maximillian basic examples

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diff --git a/.gitignore b/.gitignore
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+*.pbxuser
+*.mode1v3
+*.mode2v3
+
diff --git a/README b/README
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+                      .__        .__.__  .__               
+  _____ _____  ___  __|__| _____ |__|  | |__|____    ____  
+ /     \\__  \ \  \/  /  |/     \|  |  | |  \__  \  /    \ 
+|  Y Y  \/ __ \_>    <|  |  Y Y  \  |  |_|  |/ __ \|   |  \
+|__|_|  (____  /__/\_ \__|__|_|  /__|____/__(____  /___|  /
+      \/     \/      \/        \/                \/     \/ 
+
+
+:::::::::::::::::::::::::::BASIC EXAMPLES
+
+You can choose between using RTAudio and PortAudio drivers in player.h by uncommenting the appropriate line.  To use PortAudio, you will need to compile the portAudio library from http://http://www.portaudio.com/ and link it with your executable.
+
+Examples demonstrating different features can be found in the maximilian_examples folder.  To try them, replace the contents of main.cpp with the contents of a tutorial file and compile.
+
+
+:::MAC OSX XCODE PROJECT
+
+You can run the examples using the 'maximilianTest' XCode 3 project provided.
+
+
+:::COMMAND LINE COMPILATION IN MACOSX/LINUX etc
+
+> g++ -Wall -D__MACOSX_CORE__ -o maximilian main.cpp RtAudio.cpp player.cpp maximilian.cpp -framework CoreAudio -lpthread
+
+> ./maximilian
+
+
+
diff --git a/RtAudio.cpp b/RtAudio.cpp
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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) 2001-2011 Gary P. Scavone

+

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

+    obtaining a copy of this software and associated documentation files

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

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

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

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

+    subject to the following conditions:

+

+    The above copyright notice and this permission notice shall be

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

+

+    Any person wishing to distribute modifications to the Software is

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

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

+    however, not a binding provision of this license.

+

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

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

+    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.

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

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

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

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

+*/

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

+

+// RtAudio: Version 4.0.9

+

+#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.

+

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

+// implementation.

+struct CoreHandle {

+  AudioDeviceID id[2];    // device ids

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

+  AudioDeviceIOProcID procId[2];

+#endif

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

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

+  bool xrun[2];

+  char *deviceBuffer;

+  pthread_cond_t condition;

+  int drainCounter;       // Tracks callback counts when draining

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

+

+  CoreHandle()

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

+};

+

+RtApiCore:: RtApiCore()

+{

+#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )

+  // This is a largely undocumented but absolutely necessary

+  // requirement starting with OS-X 10.6.  If not called, queries and

+  // updates to various audio device properties are not handled

+  // correctly.

+  CFRunLoopRef theRunLoop = NULL;

+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,

+                                          kAudioObjectPropertyScopeGlobal,

+                                          kAudioObjectPropertyElementMaster };

+  OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);

+  if ( result != noErr ) {

+    errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";

+    error( RtError::WARNING );

+  }

+#endif

+}

+

+RtApiCore :: ~RtApiCore()

+{

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

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

+  // apiDeviceId memory.

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

+}

+

+unsigned int RtApiCore :: getDeviceCount( void )

+{

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

+  UInt32 dataSize;

+  AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };

+  OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );

+  if ( result != noErr ) {

+    errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";

+    error( RtError::WARNING );

+    return 0;

+  }

+

+  return dataSize / sizeof( AudioDeviceID );

+}

+

+unsigned int RtApiCore :: getDefaultInputDevice( void )

+{

+  unsigned int nDevices = getDeviceCount();

+  if ( nDevices <= 1 ) return 0;

+

+  AudioDeviceID id;

+  UInt32 dataSize = sizeof( AudioDeviceID );

+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };

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

+  if ( result != noErr ) {

+    errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";

+    error( RtError::WARNING );

+    return 0;

+  }

+

+  dataSize *= nDevices;

+  AudioDeviceID deviceList[ nDevices ];

+  property.mSelector = kAudioHardwarePropertyDevices;

+  result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );

+  if ( result != noErr ) {

+    errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";

+    error( RtError::WARNING );

+    return 0;

+  }

+

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

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

+

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

+  error( RtError::WARNING );

+  return 0;

+}

+

+unsigned int RtApiCore :: getDefaultOutputDevice( void )

+{

+  unsigned int nDevices = getDeviceCount();

+  if ( nDevices <= 1 ) return 0;

+

+  AudioDeviceID id;

+  UInt32 dataSize = sizeof( AudioDeviceID );

+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };

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

+  if ( result != noErr ) {

+    errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";

+    error( RtError::WARNING );

+    return 0;

+  }

+

+  dataSize = sizeof( AudioDeviceID ) * nDevices;

+  AudioDeviceID deviceList[ nDevices ];

+  property.mSelector = kAudioHardwarePropertyDevices;

+  result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );

+  if ( result != noErr ) {

+    errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";

+    error( RtError::WARNING );

+    return 0;

+  }

+

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

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

+

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

+  error( RtError::WARNING );

+  return 0;

+}

+

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

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

+  // Get device ID

+  unsigned int nDevices = getDeviceCount();

+  if ( nDevices == 0 ) {

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

+    error( RtError::INVALID_USE );

+  }

+

+  if ( device >= nDevices ) {

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

+    error( RtError::INVALID_USE );

+  }

+

+  AudioDeviceID deviceList[ nDevices ];

+  UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;

+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,

+                                          kAudioObjectPropertyScopeGlobal,

+                                          kAudioObjectPropertyElementMaster };

+  OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,

+                                                0, NULL, &dataSize, (void *) &deviceList );

+  if ( result != noErr ) {

+    errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";

+    error( RtError::WARNING );

+    return info;

+  }

+

+  AudioDeviceID id = deviceList[ device ];

+

+  // Get the device name.

+  info.name.erase();

+  CFStringRef cfname;

+  dataSize = sizeof( CFStringRef );

+  property.mSelector = kAudioObjectPropertyManufacturer;

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

+  if ( result != noErr ) {

+    errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

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

+  int length = CFStringGetLength(cfname);

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

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

+  info.name.append( (const char *)mname, strlen(mname) );

+  info.name.append( ": " );

+  CFRelease( cfname );

+  free(mname);

+

+  property.mSelector = kAudioObjectPropertyName;

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

+  if ( result != noErr ) {

+    errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

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

+  length = CFStringGetLength(cfname);

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

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

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

+  CFRelease( cfname );

+  free(name);

+

+  // Get the output stream "configuration".

+  AudioBufferList	*bufferList = nil;

+  property.mSelector = kAudioDevicePropertyStreamConfiguration;

+  property.mScope = kAudioDevicePropertyScopeOutput;

+  //  property.mElement = kAudioObjectPropertyElementWildcard;

+  dataSize = 0;

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

+  if ( result != noErr || dataSize == 0 ) {

+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Allocate the AudioBufferList.

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

+  if ( bufferList == NULL ) {

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

+    error( RtError::WARNING );

+    return info;

+  }

+

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

+  if ( result != noErr || dataSize == 0 ) {

+    free( bufferList );

+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Get output channel information.

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

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

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

+  free( bufferList );

+

+  // Get the input stream "configuration".

+  property.mScope = kAudioDevicePropertyScopeInput;

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

+  if ( result != noErr || dataSize == 0 ) {

+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Allocate the AudioBufferList.

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

+  if ( bufferList == NULL ) {

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

+    error( RtError::WARNING );

+    return info;

+  }

+

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

+  if (result != noErr || dataSize == 0) {

+    free( bufferList );

+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Get input channel information.

+  nStreams = bufferList->mNumberBuffers;

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

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

+  free( bufferList );

+

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

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

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

+

+  // Probe the device sample rates.

+  bool isInput = false;

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

+

+  // Determine the supported sample rates.

+  property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;

+  if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;

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

+  if ( result != kAudioHardwareNoError || dataSize == 0 ) {

+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  UInt32 nRanges = dataSize / sizeof( AudioValueRange );

+  AudioValueRange rangeList[ nRanges ];

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

+  if ( result != kAudioHardwareNoError ) {

+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  Float64 minimumRate = 100000000.0, maximumRate = 0.0;

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

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

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

+  }

+

+  info.sampleRates.clear();

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

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

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

+  }

+

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

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

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

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

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

+  // no interest to the client.

+  info.nativeFormats = RTAUDIO_FLOAT32;

+

+  if ( info.outputChannels > 0 )

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

+  if ( info.inputChannels > 0 )

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

+

+  info.probed = true;

+  return info;

+}

+

+OSStatus callbackHandler( AudioDeviceID inDevice,

+                          const AudioTimeStamp* inNow,

+                          const AudioBufferList* inInputData,

+                          const AudioTimeStamp* inInputTime,

+                          AudioBufferList* outOutputData,

+                          const AudioTimeStamp* inOutputTime, 

+                          void* infoPointer )

+{

+  CallbackInfo *info = (CallbackInfo *) infoPointer;

+

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

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

+    return kAudioHardwareUnspecifiedError;

+  else

+    return kAudioHardwareNoError;

+}

+

+OSStatus xrunListener( AudioObjectID inDevice,

+                         UInt32 nAddresses,

+                         const AudioObjectPropertyAddress properties[],

+                         void* handlePointer )

+{

+  CoreHandle *handle = (CoreHandle *) handlePointer;

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

+    if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {

+      if ( properties[i].mScope == kAudioDevicePropertyScopeInput )

+        handle->xrun[1] = true;

+      else

+        handle->xrun[0] = true;

+    }

+  }

+

+  return kAudioHardwareNoError;

+}

+

+OSStatus rateListener( AudioObjectID inDevice,

+                       UInt32 nAddresses,

+                       const AudioObjectPropertyAddress properties[],

+                       void* ratePointer )

+{

+

+  Float64 *rate = (Float64 *) ratePointer;

+  UInt32 dataSize = sizeof( Float64 );

+  AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,

+                                          kAudioObjectPropertyScopeGlobal,

+                                          kAudioObjectPropertyElementMaster };

+  AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );

+  return kAudioHardwareNoError;

+}

+

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

+                                   unsigned int firstChannel, unsigned int sampleRate,

+                                   RtAudioFormat format, unsigned int *bufferSize,

+                                   RtAudio::StreamOptions *options )

+{

+  // Get device ID

+  unsigned int nDevices = getDeviceCount();

+  if ( nDevices == 0 ) {

+    // This should not happen because a check is made before this function is called.

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

+    return FAILURE;

+  }

+

+  if ( device >= nDevices ) {

+    // This should not happen because a check is made before this function is called.

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

+    return FAILURE;

+  }

+

+  AudioDeviceID deviceList[ nDevices ];

+  UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;

+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,

+                                          kAudioObjectPropertyScopeGlobal,

+                                          kAudioObjectPropertyElementMaster };

+  OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,

+                                                0, NULL, &dataSize, (void *) &deviceList );

+  if ( result != noErr ) {

+    errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";

+    return FAILURE;

+  }

+

+  AudioDeviceID id = deviceList[ device ];

+

+  // Setup for stream mode.

+  bool isInput = false;

+  if ( mode == INPUT ) {

+    isInput = true;

+    property.mScope = kAudioDevicePropertyScopeInput;

+  }

+  else

+    property.mScope = kAudioDevicePropertyScopeOutput;

+

+  // Get the stream "configuration".

+  AudioBufferList	*bufferList = nil;

+  dataSize = 0;

+  property.mSelector = kAudioDevicePropertyStreamConfiguration;

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

+  if ( result != noErr || dataSize == 0 ) {

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Allocate the AudioBufferList.

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

+  if ( bufferList == NULL ) {

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

+    return FAILURE;

+  }

+

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

+  if (result != noErr || dataSize == 0) {

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Search for one or more streams that contain the desired number of

+  // channels. CoreAudio devices can have an arbitrary number of

+  // streams and each stream can have an arbitrary number of channels.

+  // For each stream, a single buffer of interleaved samples is

+  // provided.  RtAudio prefers the use of one stream of interleaved

+  // data or multiple consecutive single-channel streams.  However, we

+  // now support multiple consecutive multi-channel streams of

+  // interleaved data as well.

+  UInt32 iStream, offsetCounter = firstChannel;

+  UInt32 nStreams = bufferList->mNumberBuffers;

+  bool monoMode = false;

+  bool foundStream = false;

+

+  // First check that the device supports the requested number of

+  // channels.

+  UInt32 deviceChannels = 0;

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

+    deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;

+

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

+    free( bufferList );

+    errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Look for a single stream meeting our needs.

+  UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;

+  for ( iStream=0; iStream<nStreams; iStream++ ) {

+    streamChannels = bufferList->mBuffers[iStream].mNumberChannels;

+    if ( streamChannels >= channels + offsetCounter ) {

+      firstStream = iStream;

+      channelOffset = offsetCounter;

+      foundStream = true;

+      break;

+    }

+    if ( streamChannels > offsetCounter ) break;

+    offsetCounter -= streamChannels;

+  }

+

+  // If we didn't find a single stream above, then we should be able

+  // to meet the channel specification with multiple streams.

+  if ( foundStream == false ) {

+    monoMode = true;

+    offsetCounter = firstChannel;

+    for ( iStream=0; iStream<nStreams; iStream++ ) {

+      streamChannels = bufferList->mBuffers[iStream].mNumberChannels;

+      if ( streamChannels > offsetCounter ) break;

+      offsetCounter -= streamChannels;

+    }

+

+    firstStream = iStream;

+    channelOffset = offsetCounter;

+    Int32 channelCounter = channels + offsetCounter - streamChannels;

+

+    if ( streamChannels > 1 ) monoMode = false;

+    while ( channelCounter > 0 ) {

+      streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;

+      if ( streamChannels > 1 ) monoMode = false;

+      channelCounter -= streamChannels;

+      streamCount++;

+    }

+  }

+

+  free( bufferList );

+

+  // Determine the buffer size.

+  AudioValueRange	bufferRange;

+  dataSize = sizeof( AudioValueRange );

+  property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;

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

+

+  if ( result != noErr ) {

+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;

+  else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;

+  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;

+

+  // Set the buffer size.  For multiple streams, I'm assuming we only

+  // need to make this setting for the master channel.

+  UInt32 theSize = (UInt32) *bufferSize;

+  dataSize = sizeof( UInt32 );

+  property.mSelector = kAudioDevicePropertyBufferFrameSize;

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

+

+  if ( result != noErr ) {

+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // If attempting to setup a duplex stream, the bufferSize parameter

+  // MUST be the same in both directions!

+  *bufferSize = theSize;

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

+    errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  stream_.bufferSize = *bufferSize;

+  stream_.nBuffers = 1;

+

+  // Try to set "hog" mode ... it's not clear to me this is working.

+  if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {

+    pid_t hog_pid;

+    dataSize = sizeof( hog_pid );

+    property.mSelector = kAudioDevicePropertyHogMode;

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

+    if ( result != noErr ) {

+      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    if ( hog_pid != getpid() ) {

+      hog_pid = getpid();

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

+      if ( result != noErr ) {

+        errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";

+        errorText_ = errorStream_.str();

+        return FAILURE;

+      }

+    }

+  }

+

+  // Check and if necessary, change the sample rate for the device.

+  Float64 nominalRate;

+  dataSize = sizeof( Float64 );

+  property.mSelector = kAudioDevicePropertyNominalSampleRate;

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

+

+  if ( result != noErr ) {

+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Only change the sample rate if off by more than 1 Hz.

+  if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {

+

+    // Set a property listener for the sample rate change

+    Float64 reportedRate = 0.0;

+    AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };

+    result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );

+    if ( result != noErr ) {

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

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    nominalRate = (Float64) sampleRate;

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

+

+    if ( result != noErr ) {

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

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Now wait until the reported nominal rate is what we just set.

+    UInt32 microCounter = 0;

+    while ( reportedRate != nominalRate ) {

+      microCounter += 5000;

+      if ( microCounter > 5000000 ) break;

+      usleep( 5000 );

+    }

+

+    // Remove the property listener.

+    AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );

+

+    if ( microCounter > 5000000 ) {

+      errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

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

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

+  AudioStreamBasicDescription	description;

+  dataSize = sizeof( AudioStreamBasicDescription );

+  property.mSelector = kAudioStreamPropertyVirtualFormat;

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

+  if ( result != noErr ) {

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the sample rate and data format id.  However, only make the

+  // change if the sample rate is not within 1.0 of the desired

+  // rate and the format is not linear pcm.

+  bool updateFormat = false;

+  if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {

+    description.mSampleRate = (Float64) sampleRate;

+    updateFormat = true;

+  }

+

+  if ( description.mFormatID != kAudioFormatLinearPCM ) {

+    description.mFormatID = kAudioFormatLinearPCM;

+    updateFormat = true;

+  }

+

+  if ( updateFormat ) {

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

+    if ( result != noErr ) {

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

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

+  // Now check the physical format.

+  property.mSelector = kAudioStreamPropertyPhysicalFormat;

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

+  if ( result != noErr ) {

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  //std::cout << "Current physical stream format:" << std::endl;

+  //std::cout << "   mBitsPerChan = " << description.mBitsPerChannel << std::endl;

+  //std::cout << "   aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;

+  //std::cout << "   bytesPerFrame = " << description.mBytesPerFrame << std::endl;

+  //std::cout << "   sample rate = " << description.mSampleRate << std::endl;

+

+  if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {

+    description.mFormatID = kAudioFormatLinearPCM;

+    //description.mSampleRate = (Float64) sampleRate;

+    AudioStreamBasicDescription	testDescription = description;

+    UInt32 formatFlags;

+

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

+    std::vector< std::pair<UInt32, UInt32>  > physicalFormats;

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

+    physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );

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

+    physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );

+    physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) );   // 24-bit packed

+    formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );

+    physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low

+    formatFlags |= kAudioFormatFlagIsAlignedHigh;

+    physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high

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

+    physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );

+    physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );

+

+    bool setPhysicalFormat = false;

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

+      testDescription = description;

+      testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;

+      testDescription.mFormatFlags = physicalFormats[i].second;

+      if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )

+        testDescription.mBytesPerFrame =  4 * testDescription.mChannelsPerFrame;

+      else

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

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

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

+      if ( result == noErr ) {

+        setPhysicalFormat = true;

+        //std::cout << "Updated physical stream format:" << std::endl;

+        //std::cout << "   mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;

+        //std::cout << "   aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;

+        //std::cout << "   bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;

+        //std::cout << "   sample rate = " << testDescription.mSampleRate << std::endl;

+        break;

+      }

+    }

+

+    if ( !setPhysicalFormat ) {

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

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  } // done setting virtual/physical formats.

+

+  // Get the stream / device latency.

+  UInt32 latency;

+  dataSize = sizeof( UInt32 );

+  property.mSelector = kAudioDevicePropertyLatency;

+  if ( AudioObjectHasProperty( id, &property ) == true ) {

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

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

+    else {

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

+      errorText_ = errorStream_.str();

+      error( RtError::WARNING );

+    }

+  }

+

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

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

+  // need to byte swap.

+  stream_.doByteSwap[mode] = false;

+

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

+  // 32-bit floats.

+  stream_.userFormat = format;

+  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;

+

+  if ( streamCount == 1 )

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

+  else // multiple streams

+    stream_.nDeviceChannels[mode] = channels;

+  stream_.nUserChannels[mode] = channels;

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

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

+  else stream_.userInterleaved = true;

+  stream_.deviceInterleaved[mode] = true;

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

+

+  // Set flags for buffer conversion.

+  stream_.doConvertBuffer[mode] = false;

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

+    stream_.doConvertBuffer[mode] = true;

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

+    stream_.doConvertBuffer[mode] = true;

+  if ( streamCount == 1 ) {

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

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

+      stream_.doConvertBuffer[mode] = true;

+  }

+  else if ( monoMode && stream_.userInterleaved )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate our CoreHandle structure for the stream.

+  CoreHandle *handle = 0;

+  if ( stream_.apiHandle == 0 ) {

+    try {

+      handle = new CoreHandle;

+    }

+    catch ( std::bad_alloc& ) {

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

+      goto error;

+    }

+

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

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

+      goto error;

+    }

+    stream_.apiHandle = (void *) handle;

+  }

+  else

+    handle = (CoreHandle *) stream_.apiHandle;

+  handle->iStream[mode] = firstStream;

+  handle->nStreams[mode] = streamCount;

+  handle->id[mode] = id;

+

+  // Allocate necessary internal buffers.

+  unsigned long bufferBytes;

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

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

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

+  memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );

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

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

+    goto error;

+  }

+

+  // If possible, we will make use of the CoreAudio stream buffers as

+  // "device buffers".  However, we can't do this if using multiple

+  // streams.

+  if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {

+

+    bool makeBuffer = true;

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

+    if ( mode == INPUT ) {

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

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

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

+      }

+    }

+

+    if ( makeBuffer ) {

+      bufferBytes *= *bufferSize;

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

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

+      if ( stream_.deviceBuffer == NULL ) {

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

+        goto error;

+      }

+    }

+  }

+

+  stream_.sampleRate = sampleRate;

+  stream_.device[mode] = device;

+  stream_.state = STREAM_STOPPED;

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

+

+  // Setup the buffer conversion information structure.

+  if ( stream_.doConvertBuffer[mode] ) {

+    if ( streamCount > 1 ) setConvertInfo( mode, 0 );

+    else setConvertInfo( mode, channelOffset );

+  }

+

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

+    // Only one callback procedure per device.

+    stream_.mode = DUPLEX;

+  else {

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

+    result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );

+#else

+    // deprecated in favor of AudioDeviceCreateIOProcID()

+    result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );

+#endif

+    if ( result != noErr ) {

+      errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";

+      errorText_ = errorStream_.str();

+      goto error;

+    }

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

+      stream_.mode = DUPLEX;

+    else

+      stream_.mode = mode;

+  }

+

+  // Setup the device property listener for over/underload.

+  property.mSelector = kAudioDeviceProcessorOverload;

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

+

+  return SUCCESS;

+

+ error:

+  if ( handle ) {

+    pthread_cond_destroy( &handle->condition );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

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

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiCore :: closeStream( void )

+{

+  if ( stream_.state == STREAM_CLOSED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

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

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

+    if ( stream_.state == STREAM_RUNNING )

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

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

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

+#else

+    // deprecated in favor of AudioDeviceDestroyIOProcID()

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

+#endif

+  }

+

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

+    if ( stream_.state == STREAM_RUNNING )

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

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

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

+#else

+    // deprecated in favor of AudioDeviceDestroyIOProcID()

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

+#endif

+  }

+

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

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  // Destroy pthread condition variable.

+  pthread_cond_destroy( &handle->condition );

+  delete handle;

+  stream_.apiHandle = 0;

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiCore :: startStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  OSStatus result = noErr;

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

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

+

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

+    if ( result != noErr ) {

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

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  if ( stream_.mode == INPUT ||

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

+

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

+    if ( result != noErr ) {

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

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  handle->drainCounter = 0;

+  handle->internalDrain = false;

+  stream_.state = STREAM_RUNNING;

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result == noErr ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiCore :: stopStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+

+  OSStatus result = noErr;

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

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

+

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

+      handle->drainCounter = 2;

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

+    }

+

+    MUTEX_UNLOCK( &stream_.mutex );

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

+    MUTEX_LOCK( &stream_.mutex );

+    if ( result != noErr ) {

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

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

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

+

+    MUTEX_UNLOCK( &stream_.mutex );

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

+    MUTEX_LOCK( &stream_.mutex );

+    if ( result != noErr ) {

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

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  stream_.state = STREAM_STOPPED;

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result == noErr ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiCore :: abortStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

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

+  handle->drainCounter = 2;

+

+  stopStream();

+}

+

+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,

+                                 const AudioBufferList *inBufferList,

+                                 const AudioBufferList *outBufferList )

+{

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

+  if ( stream_.state == STREAM_CLOSED ) {

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

+    error( RtError::WARNING );

+    return FAILURE;

+  }

+

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

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

+

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

+  if ( handle->drainCounter > 3 ) {

+    if ( handle->internalDrain == true )

+      stopStream();

+    else // external call to stopStream()

+      pthread_cond_signal( &handle->condition );

+    return SUCCESS;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

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

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return SUCCESS;

+  }

+

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

+

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

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

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

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

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

+    double streamTime = getStreamTime();

+    RtAudioStreamStatus status = 0;

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

+      status |= RTAUDIO_OUTPUT_UNDERFLOW;

+      handle->xrun[0] = false;

+    }

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

+      status |= RTAUDIO_INPUT_OVERFLOW;

+      handle->xrun[1] = false;

+    }

+

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

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

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

+      MUTEX_UNLOCK( &stream_.mutex );

+      abortStream();

+      return SUCCESS;

+    }

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

+      handle->internalDrain = true;

+  }

+

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

+

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

+

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

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

+                0,

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

+      }

+      else { // fill multiple streams with zeros

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

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

+                  0,

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

+        }

+      }

+    }

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

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

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

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

+      }

+      else { // copy from user buffer

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

+                stream_.userBuffer[0],

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

+      }

+    }

+    else { // fill multiple streams

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

+      if ( stream_.doConvertBuffer[0] ) {

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

+        inBuffer = (Float32 *) stream_.deviceBuffer;

+      }

+

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

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

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

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

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

+        }

+      }

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

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

+        Float32 *out, *in;

+

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

+        UInt32 inChannels = stream_.nUserChannels[0];

+        if ( stream_.doConvertBuffer[0] ) {

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

+          inChannels = stream_.nDeviceChannels[0];

+        }

+

+        if ( inInterleaved ) inOffset = 1;

+        else inOffset = stream_.bufferSize;

+

+        channelsLeft = inChannels;

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

+          in = inBuffer;

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

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

+

+          outJump = 0;

+          // Account for possible channel offset in first stream

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

+            streamChannels -= stream_.channelOffset[0];

+            outJump = stream_.channelOffset[0];

+            out += outJump;

+          }

+

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

+          if ( streamChannels > channelsLeft ) {

+            outJump = streamChannels - channelsLeft;

+            streamChannels = channelsLeft;

+          }

+

+          // Determine input buffer offsets and skips

+          if ( inInterleaved ) {

+            inJump = inChannels;

+            in += inChannels - channelsLeft;

+          }

+          else {

+            inJump = 1;

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

+          }

+

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

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

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

+            }

+            out += outJump;

+            in += inJump;

+          }

+          channelsLeft -= streamChannels;

+        }

+      }

+    }

+

+    if ( handle->drainCounter ) {

+      handle->drainCounter++;

+      goto unlock;

+    }

+  }

+

+  AudioDeviceID inputDevice;

+  inputDevice = handle->id[1];

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

+

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

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

+        convertBuffer( stream_.userBuffer[1],

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

+                       stream_.convertInfo[1] );

+      }

+      else { // copy to user buffer

+        memcpy( stream_.userBuffer[1],

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

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

+      }

+    }

+    else { // read from multiple streams

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

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

+

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

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

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

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

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

+        }

+      }

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

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

+        Float32 *out, *in;

+

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

+        UInt32 outChannels = stream_.nUserChannels[1];

+        if ( stream_.doConvertBuffer[1] ) {

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

+          outChannels = stream_.nDeviceChannels[1];

+        }

+

+        if ( outInterleaved ) outOffset = 1;

+        else outOffset = stream_.bufferSize;

+

+        channelsLeft = outChannels;

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

+          out = outBuffer;

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

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

+

+          inJump = 0;

+          // Account for possible channel offset in first stream

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

+            streamChannels -= stream_.channelOffset[1];

+            inJump = stream_.channelOffset[1];

+            in += inJump;

+          }

+

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

+          if ( streamChannels > channelsLeft ) {

+            inJump = streamChannels - channelsLeft;

+            streamChannels = channelsLeft;

+          }

+

+          // Determine output buffer offsets and skips

+          if ( outInterleaved ) {

+            outJump = outChannels;

+            out += outChannels - channelsLeft;

+          }

+          else {

+            outJump = 1;

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

+          }

+

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

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

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

+            }

+            out += outJump;

+            in += inJump;

+          }

+          channelsLeft -= streamChannels;

+        }

+      }

+      

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

+        convertBuffer( stream_.userBuffer[1],

+                       stream_.deviceBuffer,

+                       stream_.convertInfo[1] );

+      }

+    }

+  }

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  RtApi::tickStreamTime();

+  return SUCCESS;

+}

+

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

+{

+  switch( code ) {

+

+  case kAudioHardwareNotRunningError:

+    return "kAudioHardwareNotRunningError";

+

+  case kAudioHardwareUnspecifiedError:

+    return "kAudioHardwareUnspecifiedError";

+

+  case kAudioHardwareUnknownPropertyError:

+    return "kAudioHardwareUnknownPropertyError";

+

+  case kAudioHardwareBadPropertySizeError:

+    return "kAudioHardwareBadPropertySizeError";

+

+  case kAudioHardwareIllegalOperationError:

+    return "kAudioHardwareIllegalOperationError";

+

+  case kAudioHardwareBadObjectError:

+    return "kAudioHardwareBadObjectError";

+

+  case kAudioHardwareBadDeviceError:

+    return "kAudioHardwareBadDeviceError";

+

+  case kAudioHardwareBadStreamError:

+    return "kAudioHardwareBadStreamError";

+

+  case kAudioHardwareUnsupportedOperationError:

+    return "kAudioHardwareUnsupportedOperationError";

+

+  case kAudioDeviceUnsupportedFormatError:

+    return "kAudioDeviceUnsupportedFormatError";

+

+  case kAudioDevicePermissionsError:

+    return "kAudioDevicePermissionsError";

+

+  default:

+    return "CoreAudio unknown error";

+  }

+}

+

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

+#endif

+

+#if defined(__UNIX_JACK__)

+

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

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

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

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

+//

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

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

+// started in a terminal as follows:

+//

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

+//

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

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

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

+// particular,

+//

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

+//

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

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

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

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

+//

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

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

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

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

+// stream cannot be opened.

+

+#include <jack/jack.h>

+#include <unistd.h>

+#include <cstdio>

+

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

+// implementation.

+struct JackHandle {

+  jack_client_t *client;

+  jack_port_t **ports[2];

+  std::string deviceName[2];

+  bool xrun[2];

+  pthread_cond_t condition;

+  int drainCounter;       // Tracks callback counts when draining

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

+

+  JackHandle()

+    :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }

+};

+

+ThreadHandle threadId;

+void jackSilentError( const char * ) {};

+

+RtApiJack :: RtApiJack()

+{

+  // Nothing to do here.

+#if !defined(__RTAUDIO_DEBUG__)

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

+  jack_set_error_function( &jackSilentError );

+#endif

+}

+

+RtApiJack :: ~RtApiJack()

+{

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

+}

+

+unsigned int RtApiJack :: getDeviceCount( void )

+{

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

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

+  jack_status_t *status = NULL;

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

+  if ( client == 0 ) return 0;

+

+  const char **ports;

+  std::string port, previousPort;

+  unsigned int nChannels = 0, nDevices = 0;

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

+  if ( ports ) {

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

+    size_t iColon = 0;

+    do {

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

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

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

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

+        if ( port != previousPort ) {

+          nDevices++;

+          previousPort = port;

+        }

+      }

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

+    free( ports );

+  }

+

+  jack_client_close( client );

+  return nDevices;

+}

+

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

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

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

+  jack_status_t *status = NULL;

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

+  if ( client == 0 ) {

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

+    error( RtError::WARNING );

+    return info;

+  }

+

+  const char **ports;

+  std::string port, previousPort;

+  unsigned int nPorts = 0, nDevices = 0;

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

+  if ( ports ) {

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

+    size_t iColon = 0;

+    do {

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

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

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

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

+        if ( port != previousPort ) {

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

+          nDevices++;

+          previousPort = port;

+        }

+      }

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

+    free( ports );

+  }

+

+  if ( device >= nDevices ) {

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

+    error( RtError::INVALID_USE );

+  }

+

+  // Get the current jack server sample rate.

+  info.sampleRates.clear();

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

+

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

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

+  unsigned int nChannels = 0;

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

+  if ( ports ) {

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

+    free( ports );

+    info.outputChannels = nChannels;

+  }

+

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

+  nChannels = 0;

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

+  if ( ports ) {

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

+    free( ports );

+    info.inputChannels = nChannels;

+  }

+

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

+    jack_client_close(client);

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

+    error( RtError::WARNING );

+    return info;

+  }

+

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

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

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

+

+  // Jack always uses 32-bit floats.

+  info.nativeFormats = RTAUDIO_FLOAT32;

+

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

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

+    info.isDefaultOutput = true;

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

+    info.isDefaultInput = true;

+

+  jack_client_close(client);

+  info.probed = true;

+  return info;

+}

+

+int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )

+{

+  CallbackInfo *info = (CallbackInfo *) infoPointer;

+

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

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

+

+  return 0;

+}

+

+// This function will be called by a spawned thread when the Jack

+// server signals that it is shutting down.  It is necessary to handle

+// it this way because the jackShutdown() function must return before

+// the jack_deactivate() function (in closeStream()) will return.

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

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

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

+

+  object->closeStream();

+

+  pthread_exit( NULL );

+}

+void jackShutdown( void *infoPointer )

+{

+  CallbackInfo *info = (CallbackInfo *) infoPointer;

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

+

+  // Check current stream state.  If stopped, then we'll assume this

+  // was called as a result of a call to RtApiJack::stopStream (the

+  // deactivation of a client handle causes this function to be called).

+  // If not, we'll assume the Jack server is shutting down or some

+  // other problem occurred and we should close the stream.

+  if ( object->isStreamRunning() == false ) return;

+

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

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

+}

+

+int jackXrun( void *infoPointer )

+{

+  JackHandle *handle = (JackHandle *) infoPointer;

+

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

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

+

+  return 0;

+}

+

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

+                                   unsigned int firstChannel, unsigned int sampleRate,

+                                   RtAudioFormat format, unsigned int *bufferSize,

+                                   RtAudio::StreamOptions *options )

+{

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

+

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

+  jack_client_t *client = 0;

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

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

+    jack_status_t *status = NULL;

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

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

+    else

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

+    if ( client == 0 ) {

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

+      error( RtError::WARNING );

+      return FAILURE;

+    }

+  }

+  else {

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

+    client = handle->client;

+  }

+

+  const char **ports;

+  std::string port, previousPort, deviceName;

+  unsigned int nPorts = 0, nDevices = 0;

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

+  if ( ports ) {

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

+    size_t iColon = 0;

+    do {

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

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

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

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

+        if ( port != previousPort ) {

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

+          nDevices++;

+          previousPort = port;

+        }

+      }

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

+    free( ports );

+  }

+

+  if ( device >= nDevices ) {

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

+    return FAILURE;

+  }

+

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

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

+  unsigned int nChannels = 0;

+  unsigned long flag = JackPortIsInput;

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

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

+  if ( ports ) {

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

+    free( ports );

+  }

+

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

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

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Check the jack server sample rate.

+  unsigned int jackRate = jack_get_sample_rate( client );

+  if ( sampleRate != jackRate ) {

+    jack_client_close( client );

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.sampleRate = jackRate;

+

+  // Get the latency of the JACK port.

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

+  if ( ports[ firstChannel ] )

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

+  free( ports );

+

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

+  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;

+  stream_.userFormat = format;

+

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

+  else stream_.userInterleaved = true;

+

+  // Jack always uses non-interleaved buffers.

+  stream_.deviceInterleaved[mode] = false;

+

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

+  stream_.doByteSwap[mode] = false;

+

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

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

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

+  *bufferSize = stream_.bufferSize;

+

+  stream_.nDeviceChannels[mode] = channels;

+  stream_.nUserChannels[mode] = channels;

+

+  // Set flags for buffer conversion.

+  stream_.doConvertBuffer[mode] = false;

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

+    stream_.doConvertBuffer[mode] = true;

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

+       stream_.nUserChannels[mode] > 1 )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate our JackHandle structure for the stream.

+  if ( handle == 0 ) {

+    try {

+      handle = new JackHandle;

+    }

+    catch ( std::bad_alloc& ) {

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

+      goto error;

+    }

+

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

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

+      goto error;

+    }

+    stream_.apiHandle = (void *) handle;

+    handle->client = client;

+  }

+  handle->deviceName[mode] = deviceName;

+

+  // Allocate necessary internal buffers.

+  unsigned long bufferBytes;

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

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

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

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

+    goto error;

+  }

+

+  if ( stream_.doConvertBuffer[mode] ) {

+

+    bool makeBuffer = true;

+    if ( mode == OUTPUT )

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

+    else { // mode == INPUT

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

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

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

+        if ( bufferBytes < bytesOut ) makeBuffer = false;

+      }

+    }

+

+    if ( makeBuffer ) {

+      bufferBytes *= *bufferSize;

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

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

+      if ( stream_.deviceBuffer == NULL ) {

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

+        goto error;

+      }

+    }

+  }

+

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

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

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

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

+    goto error;

+  }

+

+  stream_.device[mode] = device;

+  stream_.channelOffset[mode] = firstChannel;

+  stream_.state = STREAM_STOPPED;

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

+

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

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

+    stream_.mode = DUPLEX;

+  else {

+    stream_.mode = mode;

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

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

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

+  }

+

+  // Register our ports.

+  char label[64];

+  if ( mode == OUTPUT ) {

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

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

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

+                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );

+    }

+  }

+  else {

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

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

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

+                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );

+    }

+  }

+

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

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

+  // here.

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

+

+  return SUCCESS;

+

+ error:

+  if ( handle ) {

+    pthread_cond_destroy( &handle->condition );

+    jack_client_close( handle->client );

+

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

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

+

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

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

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiJack :: closeStream( void )

+{

+  if ( stream_.state == STREAM_CLOSED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

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

+  if ( handle ) {

+

+    if ( stream_.state == STREAM_RUNNING )

+      jack_deactivate( handle->client );

+

+    jack_client_close( handle->client );

+  }

+

+  if ( handle ) {

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

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

+    pthread_cond_destroy( &handle->condition );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

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

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiJack :: startStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK(&stream_.mutex);

+

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

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

+  if ( result ) {

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

+    goto unlock;

+  }

+

+  const char **ports;

+

+  // Get the list of available ports.

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

+    result = 1;

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

+    if ( ports == NULL) {

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

+      goto unlock;

+    }

+

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

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

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

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

+      result = 1;

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

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

+      if ( result ) {

+        free( ports );

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

+        goto unlock;

+      }

+    }

+    free(ports);

+  }

+

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

+    result = 1;

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

+    if ( ports == NULL) {

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

+      goto unlock;

+    }

+

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

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

+      result = 1;

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

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

+      if ( result ) {

+        free( ports );

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

+        goto unlock;

+      }

+    }

+    free(ports);

+  }

+

+  handle->drainCounter = 0;

+  handle->internalDrain = false;

+  stream_.state = STREAM_RUNNING;

+

+ unlock:

+  MUTEX_UNLOCK(&stream_.mutex);

+

+  if ( result == 0 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiJack :: stopStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+

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

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

+

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

+      handle->drainCounter = 2;

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

+    }

+  }

+

+  jack_deactivate( handle->client );

+  stream_.state = STREAM_STOPPED;

+

+  MUTEX_UNLOCK( &stream_.mutex );

+}

+

+void RtApiJack :: abortStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

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

+  handle->drainCounter = 2;

+

+  stopStream();

+}

+

+// This function will be called by a spawned thread when the user

+// callback function signals that the stream should be stopped or

+// aborted.  It is necessary to handle it this way because the

+// callbackEvent() function must return before the jack_deactivate()

+// function will return.

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

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

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

+

+  object->stopStream();

+

+  pthread_exit( NULL );

+}

+

+bool RtApiJack :: callbackEvent( unsigned long nframes )

+{

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

+  if ( stream_.state == STREAM_CLOSED ) {

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

+    error( RtError::WARNING );

+    return FAILURE;

+  }

+  if ( stream_.bufferSize != nframes ) {

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

+    error( RtError::WARNING );

+    return FAILURE;

+  }

+

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

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

+

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

+  if ( handle->drainCounter > 3 ) {

+    if ( handle->internalDrain == true )

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

+    else

+      pthread_cond_signal( &handle->condition );

+    return SUCCESS;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

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

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return SUCCESS;

+  }

+

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

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

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

+    double streamTime = getStreamTime();

+    RtAudioStreamStatus status = 0;

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

+      status |= RTAUDIO_OUTPUT_UNDERFLOW;

+      handle->xrun[0] = false;

+    }

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

+      status |= RTAUDIO_INPUT_OVERFLOW;

+      handle->xrun[1] = false;

+    }

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

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

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

+      MUTEX_UNLOCK( &stream_.mutex );

+      ThreadHandle id;

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

+      return SUCCESS;

+    }

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

+      handle->internalDrain = true;

+  }

+

+  jack_default_audio_sample_t *jackbuffer;

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

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

+

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

+

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

+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );

+        memset( jackbuffer, 0, bufferBytes );

+      }

+

+    }

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

+

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

+

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

+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );

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

+      }

+    }

+    else { // no buffer conversion

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

+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );

+        memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );

+      }

+    }

+

+    if ( handle->drainCounter ) {

+      handle->drainCounter++;

+      goto unlock;

+    }

+  }

+

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

+

+    if ( stream_.doConvertBuffer[1] ) {

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

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

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

+      }

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

+    }

+    else { // no buffer conversion

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

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

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

+      }

+    }

+  }

+

+ unlock:

+  MUTEX_UNLOCK(&stream_.mutex);

+

+  RtApi::tickStreamTime();

+  return SUCCESS;

+}

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

+#endif

+

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

+

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

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

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

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

+// have more than one simultaneous RtAudio stream.

+//

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

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

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

+// pointer to our callbackInfo structure.

+//

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

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

+// on information found in

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

+

+#include "asiosys.h"

+#include "asio.h"

+#include "iasiothiscallresolver.h"

+#include "asiodrivers.h"

+#include <cmath>

+

+AsioDrivers drivers;

+ASIOCallbacks asioCallbacks;

+ASIODriverInfo driverInfo;

+CallbackInfo *asioCallbackInfo;

+bool asioXRun;

+

+struct AsioHandle {

+  int drainCounter;       // Tracks callback counts when draining

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

+  ASIOBufferInfo *bufferInfos;

+  HANDLE condition;

+

+  AsioHandle()

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

+};

+

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

+static const char* getAsioErrorString( ASIOError result );

+void sampleRateChanged( ASIOSampleRate sRate );

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

+

+RtApiAsio :: RtApiAsio()

+{

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

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

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

+  coInitialized_ = false;

+  HRESULT hr = CoInitialize( NULL ); 

+  if ( FAILED(hr) ) {

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

+    error( RtError::WARNING );

+  }

+  coInitialized_ = true;

+

+  drivers.removeCurrentDriver();

+  driverInfo.asioVersion = 2;

+

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

+  driverInfo.sysRef = GetForegroundWindow();

+}

+

+RtApiAsio :: ~RtApiAsio()

+{

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

+  if ( coInitialized_ ) CoUninitialize();

+}

+

+unsigned int RtApiAsio :: getDeviceCount( void )

+{

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

+}

+

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

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

+  // Get device ID

+  unsigned int nDevices = getDeviceCount();

+  if ( nDevices == 0 ) {

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

+    error( RtError::INVALID_USE );

+  }

+

+  if ( device >= nDevices ) {

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

+    error( RtError::INVALID_USE );

+  }

+

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

+  if ( stream_.state != STREAM_CLOSED ) {

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

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

+      error( RtError::WARNING );

+      return info;

+    }

+    return devices_[ device ];

+  }

+

+  char driverName[32];

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

+  if ( result != ASE_OK ) {

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

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  info.name = driverName;

+

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

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

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  result = ASIOInit( &driverInfo );

+  if ( result != ASE_OK ) {

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

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Determine the device channel information.

+  long inputChannels, outputChannels;

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

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

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

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  info.outputChannels = outputChannels;

+  info.inputChannels = inputChannels;

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

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

+

+  // Determine the supported sample rates.

+  info.sampleRates.clear();

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

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

+    if ( result == ASE_OK )

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

+  }

+

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

+  ASIOChannelInfo channelInfo;

+  channelInfo.channel = 0;

+  channelInfo.isInput = true;

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

+  result = ASIOGetChannelInfo( &channelInfo );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

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

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  info.nativeFormats = 0;

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

+    info.nativeFormats |= RTAUDIO_SINT16;

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

+    info.nativeFormats |= RTAUDIO_SINT32;

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

+    info.nativeFormats |= RTAUDIO_FLOAT32;

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

+    info.nativeFormats |= RTAUDIO_FLOAT64;

+

+  if ( info.outputChannels > 0 )

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

+  if ( info.inputChannels > 0 )

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

+

+  info.probed = true;

+  drivers.removeCurrentDriver();

+  return info;

+}

+

+void bufferSwitch( long index, ASIOBool processNow )

+{

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

+  object->callbackEvent( index );

+}

+

+void RtApiAsio :: saveDeviceInfo( void )

+{

+  devices_.clear();

+

+  unsigned int nDevices = getDeviceCount();

+  devices_.resize( nDevices );

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

+    devices_[i] = getDeviceInfo( i );

+}

+

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

+                                   unsigned int firstChannel, unsigned int sampleRate,

+                                   RtAudioFormat format, unsigned int *bufferSize,

+                                   RtAudio::StreamOptions *options )

+{

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

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

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

+    return FAILURE;

+  }

+

+  char driverName[32];

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

+  if ( result != ASE_OK ) {

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

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

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

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

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

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

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

+    this->saveDeviceInfo();

+

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

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

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    result = ASIOInit( &driverInfo );

+    if ( result != ASE_OK ) {

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

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

+  // Check the device channel count.

+  long inputChannels, outputChannels;

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

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

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

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

+    drivers.removeCurrentDriver();

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.nDeviceChannels[mode] = channels;

+  stream_.nUserChannels[mode] = channels;

+  stream_.channelOffset[mode] = firstChannel;

+

+  // Verify the sample rate is supported.

+  result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Get the current sample rate

+  ASIOSampleRate currentRate;

+  result = ASIOGetSampleRate( &currentRate );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the sample rate only if necessary

+  if ( currentRate != sampleRate ) {

+    result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );

+    if ( result != ASE_OK ) {

+      drivers.removeCurrentDriver();

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

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

+  // Determine the driver data type.

+  ASIOChannelInfo channelInfo;

+  channelInfo.channel = 0;

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

+  else channelInfo.isInput = true;

+  result = ASIOGetChannelInfo( &channelInfo );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

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

+  stream_.doByteSwap[mode] = false;

+  stream_.userFormat = format;

+  stream_.deviceFormat[mode] = 0;

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

+    stream_.deviceFormat[mode] = RTAUDIO_SINT16;

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

+  }

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

+    stream_.deviceFormat[mode] = RTAUDIO_SINT32;

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

+  }

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

+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;

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

+  }

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

+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;

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

+  }

+

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

+    drivers.removeCurrentDriver();

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

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

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

+  // should be ok.

+  long minSize, maxSize, preferSize, granularity;

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

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

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

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

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

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

+  else if ( granularity == -1 ) {

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

+    int log2_of_min_size = 0;

+    int log2_of_max_size = 0;

+

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

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

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

+    }

+

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

+    int min_delta_num = log2_of_min_size;

+

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

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

+      if (current_delta < min_delta) {

+        min_delta = current_delta;

+        min_delta_num = i;

+      }

+    }

+

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

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

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

+  }

+  else if ( granularity != 0 ) {

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

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

+  }

+

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

+    drivers.removeCurrentDriver();

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

+    return FAILURE;

+  }

+

+  stream_.bufferSize = *bufferSize;

+  stream_.nBuffers = 2;

+

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

+  else stream_.userInterleaved = true;

+

+  // ASIO always uses non-interleaved buffers.

+  stream_.deviceInterleaved[mode] = false;

+

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

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

+  if ( handle == 0 ) {

+    try {

+      handle = new AsioHandle;

+    }

+    catch ( std::bad_alloc& ) {

+      //if ( handle == NULL ) {    

+      drivers.removeCurrentDriver();

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

+      return FAILURE;

+    }

+    handle->bufferInfos = 0;

+

+    // Create a manual-reset event.

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

+                                     TRUE,   // manual-reset

+                                     FALSE,  // non-signaled initially

+                                     NULL ); // unnamed

+    stream_.apiHandle = (void *) handle;

+  }

+

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

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

+  // created output buffers for a duplex stream.

+  long inputLatency, outputLatency;

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

+    ASIODisposeBuffers();

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

+  }

+

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

+  bool buffersAllocated = false;

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

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

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

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

+    errorText_ = errorStream_.str();

+    goto error;

+  }

+

+  ASIOBufferInfo *infos;

+  infos = handle->bufferInfos;

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

+    infos->isInput = ASIOFalse;

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

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

+  }

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

+    infos->isInput = ASIOTrue;

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

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

+  }

+

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

+  asioCallbacks.bufferSwitch = &bufferSwitch;

+  asioCallbacks.sampleRateDidChange = &sampleRateChanged;

+  asioCallbacks.asioMessage = &asioMessages;

+  asioCallbacks.bufferSwitchTimeInfo = NULL;

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

+  if ( result != ASE_OK ) {

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

+    errorText_ = errorStream_.str();

+    goto error;

+  }

+  buffersAllocated = true;

+

+  // Set flags for buffer conversion.

+  stream_.doConvertBuffer[mode] = false;

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

+    stream_.doConvertBuffer[mode] = true;

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

+       stream_.nUserChannels[mode] > 1 )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate necessary internal buffers

+  unsigned long bufferBytes;

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

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

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

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

+    goto error;

+  }

+

+  if ( stream_.doConvertBuffer[mode] ) {

+

+    bool makeBuffer = true;

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

+    if ( mode == INPUT ) {

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

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

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

+      }

+    }

+

+    if ( makeBuffer ) {

+      bufferBytes *= *bufferSize;

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

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

+      if ( stream_.deviceBuffer == NULL ) {

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

+        goto error;

+      }

+    }

+  }

+

+  stream_.sampleRate = sampleRate;

+  stream_.device[mode] = device;

+  stream_.state = STREAM_STOPPED;

+  asioCallbackInfo = &stream_.callbackInfo;

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

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

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

+    stream_.mode = DUPLEX;

+  else

+    stream_.mode = mode;

+

+  // Determine device latencies

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

+  if ( result != ASE_OK ) {

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

+    errorText_ = errorStream_.str();

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

+  }

+  else {

+    stream_.latency[0] = outputLatency;

+    stream_.latency[1] = inputLatency;

+  }

+

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

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

+  // here.

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

+

+  return SUCCESS;

+

+ error:

+  if ( buffersAllocated )

+    ASIODisposeBuffers();

+  drivers.removeCurrentDriver();

+

+  if ( handle ) {

+    CloseHandle( handle->condition );

+    if ( handle->bufferInfos )

+      free( handle->bufferInfos );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

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

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiAsio :: closeStream()

+{

+  if ( stream_.state == STREAM_CLOSED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  if ( stream_.state == STREAM_RUNNING ) {

+    stream_.state = STREAM_STOPPED;

+    ASIOStop();

+  }

+  ASIODisposeBuffers();

+  drivers.removeCurrentDriver();

+

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

+  if ( handle ) {

+    CloseHandle( handle->condition );

+    if ( handle->bufferInfos )

+      free( handle->bufferInfos );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

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

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+bool stopThreadCalled = false;

+

+void RtApiAsio :: startStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  //MUTEX_LOCK( &stream_.mutex );

+

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

+  ASIOError result = ASIOStart();

+  if ( result != ASE_OK ) {

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

+    errorText_ = errorStream_.str();

+    goto unlock;

+  }

+

+  handle->drainCounter = 0;

+  handle->internalDrain = false;

+  ResetEvent( handle->condition );

+  stream_.state = STREAM_RUNNING;

+  asioXRun = false;

+

+ unlock:

+  //MUTEX_UNLOCK( &stream_.mutex );

+

+  stopThreadCalled = false;

+

+  if ( result == ASE_OK ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAsio :: stopStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  /*

+  MUTEX_LOCK( &stream_.mutex );

+

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+  */

+

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

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

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

+      handle->drainCounter = 2;

+      //      MUTEX_UNLOCK( &stream_.mutex );

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

+      //ResetEvent( handle->condition );

+      //      MUTEX_LOCK( &stream_.mutex );

+    }

+  }

+

+  stream_.state = STREAM_STOPPED;

+

+  ASIOError result = ASIOStop();

+  if ( result != ASE_OK ) {

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

+    errorText_ = errorStream_.str();

+  }

+

+  //  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result == ASE_OK ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAsio :: abortStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

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

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

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

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

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

+  // handle->drainCounter = 2;

+  stopStream();

+}

+

+// This function will be called by a spawned thread when the user

+// callback function signals that the stream should be stopped or

+// aborted.  It is necessary to handle it this way because the

+// callbackEvent() function must return before the ASIOStop()

+// function will return.

+extern "C" unsigned __stdcall asioStopStream( void *ptr )

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

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

+

+  object->stopStream();

+

+  _endthreadex( 0 );

+  return 0;

+}

+

+bool RtApiAsio :: callbackEvent( long bufferIndex )

+{

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

+  if ( stopThreadCalled ) return SUCCESS;

+  if ( stream_.state == STREAM_CLOSED ) {

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

+    error( RtError::WARNING );

+    return FAILURE;

+  }

+

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

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

+

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

+  if ( handle->drainCounter > 3 ) {

+    if ( handle->internalDrain == false )

+      SetEvent( handle->condition );

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

+      unsigned threadId;

+      stopThreadCalled = true;

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

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

+    }

+    return SUCCESS;

+  }

+

+  /*MUTEX_LOCK( &stream_.mutex );

+

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

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

+

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

+  // draining stream.

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

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

+    double streamTime = getStreamTime();

+    RtAudioStreamStatus status = 0;

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

+      status |= RTAUDIO_OUTPUT_UNDERFLOW;

+      asioXRun = false;

+    }

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

+      status |= RTAUDIO_INPUT_OVERFLOW;

+      asioXRun = false;

+    }

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

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

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

+      //      MUTEX_UNLOCK( &stream_.mutex );

+      //      abortStream();

+      unsigned threadId;

+      stopThreadCalled = true;

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

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

+      return SUCCESS;

+    }

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

+      handle->internalDrain = true;

+  }

+

+  unsigned int nChannels, bufferBytes, i, j;

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

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

+

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

+

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

+

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

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

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

+      }

+

+    }

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

+

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

+      if ( stream_.doByteSwap[0] )

+        byteSwapBuffer( stream_.deviceBuffer,

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

+                        stream_.deviceFormat[0] );

+

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

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

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

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

+      }

+

+    }

+    else {

+

+      if ( stream_.doByteSwap[0] )

+        byteSwapBuffer( stream_.userBuffer[0],

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

+                        stream_.userFormat );

+

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

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

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

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

+      }

+

+    }

+

+    if ( handle->drainCounter ) {

+      handle->drainCounter++;

+      goto unlock;

+    }

+  }

+

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

+

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

+

+    if (stream_.doConvertBuffer[1]) {

+

+      // Always interleave ASIO input data.

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

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

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

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

+                  bufferBytes );

+      }

+

+      if ( stream_.doByteSwap[1] )

+        byteSwapBuffer( stream_.deviceBuffer,

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

+                        stream_.deviceFormat[1] );

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

+

+    }

+    else {

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

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

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

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

+                  bufferBytes );

+        }

+      }

+

+      if ( stream_.doByteSwap[1] )

+        byteSwapBuffer( stream_.userBuffer[1],

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

+                        stream_.userFormat );

+    }

+  }

+

+ unlock:

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

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

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

+  ASIOOutputReady();

+

+  //  MUTEX_UNLOCK( &stream_.mutex );

+

+  RtApi::tickStreamTime();

+  return SUCCESS;

+}

+

+void sampleRateChanged( ASIOSampleRate sRate )

+{

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

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

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

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

+  // audio device.

+

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

+  try {

+    object->stopStream();

+  }

+  catch ( RtError &exception ) {

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

+    return;

+  }

+

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

+}

+

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

+{

+  long ret = 0;

+

+  switch( selector ) {

+  case kAsioSelectorSupported:

+    if ( value == kAsioResetRequest

+         || value == kAsioEngineVersion

+         || value == kAsioResyncRequest

+         || value == kAsioLatenciesChanged

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

+         // necessarily have to support them.

+         || value == kAsioSupportsTimeInfo

+         || value == kAsioSupportsTimeCode

+         || value == kAsioSupportsInputMonitor)

+      ret = 1L;

+    break;

+  case kAsioResetRequest:

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

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

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

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

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

+    // driver again.

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

+    ret = 1L;

+    break;

+  case kAsioResyncRequest:

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

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

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

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

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

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

+    // situations, too.

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

+    asioXRun = true;

+    ret = 1L;

+    break;

+  case kAsioLatenciesChanged:

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

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

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

+    // delay data.

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

+    ret = 1L;

+    break;

+  case kAsioEngineVersion:

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

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

+    // is assumed by the driver.

+    ret = 2L;

+    break;

+  case kAsioSupportsTimeInfo:

+    // Informs the driver whether the

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

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

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

+    ret = 0;

+    break;

+  case kAsioSupportsTimeCode:

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

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

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

+    ret = 0;

+    break;

+  }

+  return ret;

+}

+

+static const char* getAsioErrorString( ASIOError result )

+{

+  struct Messages 

+  {

+    ASIOError value;

+    const char*message;

+  };

+

+  static Messages m[] = 

+    {

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

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

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

+      {   ASE_InvalidMode,      "Invalid mode." },

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

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

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

+    };

+

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

+    if ( m[i].value == result ) return m[i].message;

+

+  return "Unknown error.";

+}

+//******************** End of __WINDOWS_ASIO__ *********************//

+#endif

+

+

+#if defined(__WINDOWS_DS__) // Windows DirectSound API

+

+// Modified by Robin Davies, October 2005

+// - Improvements to DirectX pointer chasing. 

+// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.

+// - Auto-call CoInitialize for DSOUND and ASIO platforms.

+// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007

+// Changed device query structure for RtAudio 4.0.7, January 2010

+

+#include <dsound.h>

+#include <assert.h>

+#include <algorithm>

+

+#if defined(__MINGW32__)

+  // missing from latest mingw winapi

+#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */

+#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */

+#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */

+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */

+#endif

+

+#define MINIMUM_DEVICE_BUFFER_SIZE 32768

+

+#ifdef _MSC_VER // if Microsoft Visual C++

+#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.

+#endif

+

+static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )

+{

+  if ( pointer > bufferSize ) pointer -= bufferSize;

+  if ( laterPointer < earlierPointer ) laterPointer += bufferSize;

+  if ( pointer < earlierPointer ) pointer += bufferSize;

+  return pointer >= earlierPointer && pointer < laterPointer;

+}

+

+// A structure to hold various information related to the DirectSound

+// API implementation.

+struct DsHandle {

+  unsigned int drainCounter; // Tracks callback counts when draining

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

+  void *id[2];

+  void *buffer[2];

+  bool xrun[2];

+  UINT bufferPointer[2];  

+  DWORD dsBufferSize[2];

+  DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.

+  HANDLE condition;

+

+  DsHandle()

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

+};

+

+// Declarations for utility functions, callbacks, and structures

+// specific to the DirectSound implementation.

+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,

+                                          LPCTSTR description,

+                                          LPCTSTR module,

+                                          LPVOID lpContext );

+

+static const char* getErrorString( int code );

+

+extern "C" unsigned __stdcall callbackHandler( void *ptr );

+

+struct DsDevice {

+  LPGUID id[2];

+  bool validId[2];

+  bool found;

+  std::string name;

+

+  DsDevice()

+  : found(false) { validId[0] = false; validId[1] = false; }

+};

+

+std::vector< DsDevice > dsDevices;

+

+RtApiDs :: RtApiDs()

+{

+  // Dsound will run both-threaded. If CoInitialize fails, then just

+  // accept whatever the mainline chose for a threading model.

+  coInitialized_ = false;

+  HRESULT hr = CoInitialize( NULL );

+  if ( !FAILED( hr ) ) coInitialized_ = true;

+}

+

+RtApiDs :: ~RtApiDs()

+{

+  if ( coInitialized_ ) CoUninitialize(); // balanced call.

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

+}

+

+// The DirectSound default output is always the first device.

+unsigned int RtApiDs :: getDefaultOutputDevice( void )

+{

+  return 0;

+}

+

+// The DirectSound default input is always the first input device,

+// which is the first capture device enumerated.

+unsigned int RtApiDs :: getDefaultInputDevice( void )

+{

+  return 0;

+}

+

+unsigned int RtApiDs :: getDeviceCount( void )

+{

+  // Set query flag for previously found devices to false, so that we

+  // can check for any devices that have disappeared.

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

+    dsDevices[i].found = false;

+

+  // Query DirectSound devices.

+  bool isInput = false;

+  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );

+  if ( FAILED( result ) ) {

+    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+  }

+

+  // Query DirectSoundCapture devices.

+  isInput = true;

+  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );

+  if ( FAILED( result ) ) {

+    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+  }

+

+  // Clean out any devices that may have disappeared.

+  std::vector< int > indices;

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

+    if ( dsDevices[i].found == false ) indices.push_back( i );

+  for ( unsigned int nErased=0, unsigned int i=0; i<indices.size(); i++, nErased++ ) {

+    dsDevices.erase( dsDevices.begin()-nErased );

+  }

+

+  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] );

+  }

+  std::sort( info.sampleRates.begin(), info.sampleRates.end() );

+

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

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

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

+

+  if ( device == 0 ) info.isDefaultInput = true;

+

+  // Copy name and return.

+  info.name = dsDevices[ device ].name;

+  info.probed = true;

+  return info;

+}

+

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

+                                 unsigned int firstChannel, unsigned int sampleRate,

+                                 RtAudioFormat format, unsigned int *bufferSize,

+                                 RtAudio::StreamOptions *options )

+{

+  if ( channels + firstChannel > 2 ) {

+    errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";

+    return FAILURE;

+  }

+

+  unsigned int nDevices = dsDevices.size();

+  if ( nDevices == 0 ) {

+    // This should not happen because a check is made before this function is called.

+    errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";

+    return FAILURE;

+  }

+

+  if ( device >= nDevices ) {

+    // This should not happen because a check is made before this function is called.

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

+    return FAILURE;

+  }

+

+  if ( mode == OUTPUT ) {

+    if ( dsDevices[ device ].validId[0] == false ) {

+      errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+  else { // mode == INPUT

+    if ( dsDevices[ device ].validId[1] == false ) {

+      errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

+  // According to a note in PortAudio, using GetDesktopWindow()

+  // instead of GetForegroundWindow() is supposed to avoid problems

+  // that occur when the application's window is not the foreground

+  // window.  Also, if the application window closes before the

+  // DirectSound buffer, DirectSound can crash.  In the past, I had

+  // problems when using GetDesktopWindow() but it seems fine now

+  // (January 2010).  I'll leave it commented here.

+  // HWND hWnd = GetForegroundWindow();

+  HWND hWnd = GetDesktopWindow();

+

+  // Check the numberOfBuffers parameter and limit the lowest value to

+  // two.  This is a judgement call and a value of two is probably too

+  // low for capture, but it should work for playback.

+  int nBuffers = 0;

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

+  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;

+  if ( nBuffers < 2 ) nBuffers = 3;

+

+  // Check the lower range of the user-specified buffer size and set

+  // (arbitrarily) to a lower bound of 32.

+  if ( *bufferSize < 32 ) *bufferSize = 32;

+

+  // Create the wave format structure.  The data format setting will

+  // be determined later.

+  WAVEFORMATEX waveFormat;

+  ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );

+  waveFormat.wFormatTag = WAVE_FORMAT_PCM;

+  waveFormat.nChannels = channels + firstChannel;

+  waveFormat.nSamplesPerSec = (unsigned long) sampleRate;

+

+  // Determine the device buffer size. By default, we'll use the value

+  // defined above (32K), but we will grow it to make allowances for

+  // very large software buffer sizes.

+  DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;;

+  DWORD dsPointerLeadTime = 0;

+

+  void *ohandle = 0, *bhandle = 0;

+  HRESULT result;

+  if ( mode == OUTPUT ) {

+

+    LPDIRECTSOUND output;

+    result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    DSCAPS outCaps;

+    outCaps.dwSize = sizeof( outCaps );

+    result = output->GetCaps( &outCaps );

+    if ( FAILED( result ) ) {

+      output->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Check channel information.

+    if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {

+      errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Check format information.  Use 16-bit format unless not

+    // supported or user requests 8-bit.

+    if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&

+         !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {

+      waveFormat.wBitsPerSample = 16;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+    }

+    else {

+      waveFormat.wBitsPerSample = 8;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT8;

+    }

+    stream_.userFormat = format;

+

+    // Update wave format structure and buffer information.

+    waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;

+    waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;

+    dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;

+

+    // If the user wants an even bigger buffer, increase the device buffer size accordingly.

+    while ( dsPointerLeadTime * 2U > dsBufferSize )

+      dsBufferSize *= 2;

+

+    // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.

+    // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );

+    // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.

+    result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );

+    if ( FAILED( result ) ) {

+      output->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Even though we will write to the secondary buffer, we need to

+    // access the primary buffer to set the correct output format

+    // (since the default is 8-bit, 22 kHz!).  Setup the DS primary

+    // buffer description.

+    DSBUFFERDESC bufferDescription;

+    ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );

+    bufferDescription.dwSize = sizeof( DSBUFFERDESC );

+    bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;

+

+    // Obtain the primary buffer

+    LPDIRECTSOUNDBUFFER buffer;

+    result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );

+    if ( FAILED( result ) ) {

+      output->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Set the primary DS buffer sound format.

+    result = buffer->SetFormat( &waveFormat );

+    if ( FAILED( result ) ) {

+      output->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Setup the secondary DS buffer description.

+    ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );

+    bufferDescription.dwSize = sizeof( DSBUFFERDESC );

+    bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |

+                                  DSBCAPS_GLOBALFOCUS |

+                                  DSBCAPS_GETCURRENTPOSITION2 |

+                                  DSBCAPS_LOCHARDWARE );  // Force hardware mixing

+    bufferDescription.dwBufferBytes = dsBufferSize;

+    bufferDescription.lpwfxFormat = &waveFormat;

+

+    // Try to create the secondary DS buffer.  If that doesn't work,

+    // try to use software mixing.  Otherwise, there's a problem.

+    result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );

+    if ( FAILED( result ) ) {

+      bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |

+                                    DSBCAPS_GLOBALFOCUS |

+                                    DSBCAPS_GETCURRENTPOSITION2 |

+                                    DSBCAPS_LOCSOFTWARE );  // Force software mixing

+      result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );

+      if ( FAILED( result ) ) {

+        output->Release();

+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";

+        errorText_ = errorStream_.str();

+        return FAILURE;

+      }

+    }

+

+    // Get the buffer size ... might be different from what we specified.

+    DSBCAPS dsbcaps;

+    dsbcaps.dwSize = sizeof( DSBCAPS );

+    result = buffer->GetCaps( &dsbcaps );

+    if ( FAILED( result ) ) {

+      output->Release();

+      buffer->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    dsBufferSize = dsbcaps.dwBufferBytes;

+

+    // Lock the DS buffer

+    LPVOID audioPtr;

+    DWORD dataLen;

+    result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );

+    if ( FAILED( result ) ) {

+      output->Release();

+      buffer->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Zero the DS buffer

+    ZeroMemory( audioPtr, dataLen );

+

+    // Unlock the DS buffer

+    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );

+    if ( FAILED( result ) ) {

+      output->Release();

+      buffer->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    ohandle = (void *) output;

+    bhandle = (void *) buffer;

+  }

+

+  if ( mode == INPUT ) {

+

+    LPDIRECTSOUNDCAPTURE input;

+    result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    DSCCAPS inCaps;

+    inCaps.dwSize = sizeof( inCaps );

+    result = input->GetCaps( &inCaps );

+    if ( FAILED( result ) ) {

+      input->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Check channel information.

+    if ( inCaps.dwChannels < channels + firstChannel ) {

+      errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";

+      return FAILURE;

+    }

+

+    // Check format information.  Use 16-bit format unless user

+    // requests 8-bit.

+    DWORD deviceFormats;

+    if ( channels + firstChannel == 2 ) {

+      deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;

+      if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {

+        waveFormat.wBitsPerSample = 8;

+        stream_.deviceFormat[mode] = RTAUDIO_SINT8;

+      }

+      else { // assume 16-bit is supported

+        waveFormat.wBitsPerSample = 16;

+        stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+      }

+    }

+    else { // channel == 1

+      deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;

+      if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {

+        waveFormat.wBitsPerSample = 8;

+        stream_.deviceFormat[mode] = RTAUDIO_SINT8;

+      }

+      else { // assume 16-bit is supported

+        waveFormat.wBitsPerSample = 16;

+        stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+      }

+    }

+    stream_.userFormat = format;

+

+    // Update wave format structure and buffer information.

+    waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;

+    waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;

+    dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;

+

+    // If the user wants an even bigger buffer, increase the device buffer size accordingly.

+    while ( dsPointerLeadTime * 2U > dsBufferSize )

+      dsBufferSize *= 2;

+

+    // Setup the secondary DS buffer description.

+    DSCBUFFERDESC bufferDescription;

+    ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );

+    bufferDescription.dwSize = sizeof( DSCBUFFERDESC );

+    bufferDescription.dwFlags = 0;

+    bufferDescription.dwReserved = 0;

+    bufferDescription.dwBufferBytes = dsBufferSize;

+    bufferDescription.lpwfxFormat = &waveFormat;

+

+    // Create the capture buffer.

+    LPDIRECTSOUNDCAPTUREBUFFER buffer;

+    result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );

+    if ( FAILED( result ) ) {

+      input->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Get the buffer size ... might be different from what we specified.

+    DSCBCAPS dscbcaps;

+    dscbcaps.dwSize = sizeof( DSCBCAPS );

+    result = buffer->GetCaps( &dscbcaps );

+    if ( FAILED( result ) ) {

+      input->Release();

+      buffer->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    dsBufferSize = dscbcaps.dwBufferBytes;

+

+    // NOTE: We could have a problem here if this is a duplex stream

+    // and the play and capture hardware buffer sizes are different

+    // (I'm actually not sure if that is a problem or not).

+    // Currently, we are not verifying that.

+

+    // Lock the capture buffer

+    LPVOID audioPtr;

+    DWORD dataLen;

+    result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );

+    if ( FAILED( result ) ) {

+      input->Release();

+      buffer->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    // Zero the buffer

+    ZeroMemory( audioPtr, dataLen );

+

+    // Unlock the buffer

+    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );

+    if ( FAILED( result ) ) {

+      input->Release();

+      buffer->Release();

+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    ohandle = (void *) input;

+    bhandle = (void *) buffer;

+  }

+

+  // Set various stream parameters

+  DsHandle *handle = 0;

+  stream_.nDeviceChannels[mode] = channels + firstChannel;

+  stream_.nUserChannels[mode] = channels;

+  stream_.bufferSize = *bufferSize;

+  stream_.channelOffset[mode] = firstChannel;

+  stream_.deviceInterleaved[mode] = true;

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

+  else stream_.userInterleaved = true;

+

+  // Set flag for buffer conversion

+  stream_.doConvertBuffer[mode] = false;

+  if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])

+    stream_.doConvertBuffer[mode] = true;

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

+    stream_.doConvertBuffer[mode] = true;

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

+       stream_.nUserChannels[mode] > 1 )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate necessary internal buffers

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

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

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

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

+    goto error;

+  }

+

+  if ( stream_.doConvertBuffer[mode] ) {

+

+    bool makeBuffer = true;

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

+    if ( mode == INPUT ) {

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

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

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

+      }

+    }

+

+    if ( makeBuffer ) {

+      bufferBytes *= *bufferSize;

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

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

+      if ( stream_.deviceBuffer == NULL ) {

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

+        goto error;

+      }

+    }

+  }

+

+  // Allocate our DsHandle structures for the stream.

+  if ( stream_.apiHandle == 0 ) {

+    try {

+      handle = new DsHandle;

+    }

+    catch ( std::bad_alloc& ) {

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

+      goto error;

+    }

+

+    // Create a manual-reset event.

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

+                                     TRUE,   // manual-reset

+                                     FALSE,  // non-signaled initially

+                                     NULL ); // unnamed

+    stream_.apiHandle = (void *) handle;

+  }

+  else

+    handle = (DsHandle *) stream_.apiHandle;

+  handle->id[mode] = ohandle;

+  handle->buffer[mode] = bhandle;

+  handle->dsBufferSize[mode] = dsBufferSize;

+  handle->dsPointerLeadTime[mode] = dsPointerLeadTime;

+

+  stream_.device[mode] = device;

+  stream_.state = STREAM_STOPPED;

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

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

+    stream_.mode = DUPLEX;

+  else

+    stream_.mode = mode;

+  stream_.nBuffers = nBuffers;

+  stream_.sampleRate = sampleRate;

+

+  // Setup the buffer conversion information structure.

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

+

+  // Setup the callback thread.

+  if ( stream_.callbackInfo.isRunning == false ) {

+    unsigned threadId;

+    stream_.callbackInfo.isRunning = true;

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

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

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

+    if ( stream_.callbackInfo.thread == 0 ) {

+      errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";

+      goto error;

+    }

+

+    // Boost DS thread priority

+    SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );

+  }

+  return SUCCESS;

+

+ error:

+  if ( handle ) {

+    if ( handle->buffer[0] ) { // the object pointer can be NULL and valid

+      LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];

+      LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];

+      if ( buffer ) buffer->Release();

+      object->Release();

+    }

+    if ( handle->buffer[1] ) {

+      LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];

+      LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];

+      if ( buffer ) buffer->Release();

+      object->Release();

+    }

+    CloseHandle( handle->condition );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

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

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiDs :: closeStream()

+{

+  if ( stream_.state == STREAM_CLOSED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  // Stop the callback thread.

+  stream_.callbackInfo.isRunning = false;

+  WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );

+  CloseHandle( (HANDLE) stream_.callbackInfo.thread );

+

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

+  if ( handle ) {

+    if ( handle->buffer[0] ) { // the object pointer can be NULL and valid

+      LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];

+      LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];

+      if ( buffer ) {

+        buffer->Stop();

+        buffer->Release();

+      }

+      object->Release();

+    }

+    if ( handle->buffer[1] ) {

+      LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];

+      LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];

+      if ( buffer ) {

+        buffer->Stop();

+        buffer->Release();

+      }

+      object->Release();

+    }

+    CloseHandle( handle->condition );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

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

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiDs :: startStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  //MUTEX_LOCK( &stream_.mutex );

+  

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

+

+  // Increase scheduler frequency on lesser windows (a side-effect of

+  // increasing timer accuracy).  On greater windows (Win2K or later),

+  // this is already in effect.

+  timeBeginPeriod( 1 ); 

+

+  buffersRolling = false;

+  duplexPrerollBytes = 0;

+

+  if ( stream_.mode == DUPLEX ) {

+    // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.

+    duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );

+  }

+

+  HRESULT result = 0;

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

+

+    LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];

+    result = buffer->Play( 0, 0, DSBPLAY_LOOPING );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

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

+

+    LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];

+    result = buffer->Start( DSCBSTART_LOOPING );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  handle->drainCounter = 0;

+  handle->internalDrain = false;

+  ResetEvent( handle->condition );

+  stream_.state = STREAM_RUNNING;

+

+ unlock:

+  //  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiDs :: stopStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

+  /*

+  MUTEX_LOCK( &stream_.mutex );

+

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+  */

+

+  HRESULT result = 0;

+  LPVOID audioPtr;

+  DWORD dataLen;

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

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

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

+      handle->drainCounter = 2;

+      //      MUTEX_UNLOCK( &stream_.mutex );

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

+      //ResetEvent( handle->condition );

+      //      MUTEX_LOCK( &stream_.mutex );

+    }

+

+    stream_.state = STREAM_STOPPED;

+

+    // Stop the buffer and clear memory

+    LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];

+    result = buffer->Stop();

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+

+    // Lock the buffer and clear it so that if we start to play again,

+    // we won't have old data playing.

+    result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+

+    // Zero the DS buffer

+    ZeroMemory( audioPtr, dataLen );

+

+    // Unlock the DS buffer

+    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+

+    // If we start playing again, we must begin at beginning of buffer.

+    handle->bufferPointer[0] = 0;

+  }

+

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

+    LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];

+    audioPtr = NULL;

+    dataLen = 0;

+

+    stream_.state = STREAM_STOPPED;

+

+    result = buffer->Stop();

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+

+    // Lock the buffer and clear it so that if we start to play again,

+    // we won't have old data playing.

+    result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+

+    // Zero the DS buffer

+    ZeroMemory( audioPtr, dataLen );

+

+    // Unlock the DS buffer

+    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+

+    // If we start recording again, we must begin at beginning of buffer.

+    handle->bufferPointer[1] = 0;

+  }

+

+ unlock:

+  timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.

+  //  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiDs :: abortStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

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

+    error( RtError::WARNING );

+    return;

+  }

+

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

+  handle->drainCounter = 2;

+

+  stopStream();

+}

+

+void RtApiDs :: callbackEvent()

+{

+  if ( stream_.state == STREAM_STOPPED ) {

+    Sleep( 50 ); // sleep 50 milliseconds

+    return;

+  }

+

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;

+  DsHandle *handle = (DsHandle *) stream_.apiHandle;

+

+  // Check if we were draining the stream and signal is finished.

+  if ( handle->drainCounter > stream_.nBuffers + 2 ) {

+    if ( handle->internalDrain == false )

+      SetEvent( handle->condition );

+    else

+      stopStream();

+    return;

+  }

+

+  /*

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+  */

+

+  // Invoke user callback to get fresh output data UNLESS we are

+  // draining stream.

+  if ( handle->drainCounter == 0 ) {

+    RtAudioCallback callback = (RtAudioCallback) info->callback;

+    double streamTime = getStreamTime();

+    RtAudioStreamStatus status = 0;

+    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {

+      status |= RTAUDIO_OUTPUT_UNDERFLOW;

+      handle->xrun[0] = false;

+    }

+    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {

+      status |= RTAUDIO_INPUT_OVERFLOW;

+      handle->xrun[1] = false;

+    }

+    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],

+                                     stream_.bufferSize, streamTime, status, info->userData );

+    if ( handle->drainCounter == 2 ) {

+      //      MUTEX_UNLOCK( &stream_.mutex );

+      abortStream();

+      return;

+    }

+    else if ( handle->drainCounter == 1 )

+      handle->internalDrain = true;

+  }

+

+  HRESULT result;

+  DWORD currentWritePointer, safeWritePointer;

+  DWORD currentReadPointer, safeReadPointer;

+  UINT nextWritePointer;

+

+  LPVOID buffer1 = NULL;

+  LPVOID buffer2 = NULL;

+  DWORD bufferSize1 = 0;

+  DWORD bufferSize2 = 0;

+

+  char *buffer;

+  long bufferBytes;

+

+  if ( buffersRolling == false ) {

+    if ( stream_.mode == DUPLEX ) {

+      //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );

+

+      // It takes a while for the devices to get rolling. As a result,

+      // there's no guarantee that the capture and write device pointers

+      // will move in lockstep.  Wait here for both devices to start

+      // rolling, and then set our buffer pointers accordingly.

+      // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600

+      // bytes later than the write buffer.

+

+      // Stub: a serious risk of having a pre-emptive scheduling round

+      // take place between the two GetCurrentPosition calls... but I'm

+      // really not sure how to solve the problem.  Temporarily boost to

+      // Realtime priority, maybe; but I'm not sure what priority the

+      // DirectSound service threads run at. We *should* be roughly

+      // within a ms or so of correct.

+

+      LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];

+      LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];

+

+      DWORD startSafeWritePointer, startSafeReadPointer;

+

+      result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );

+      if ( FAILED( result ) ) {

+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";

+        errorText_ = errorStream_.str();

+        error( RtError::SYSTEM_ERROR );

+      }

+      result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );

+      if ( FAILED( result ) ) {

+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";

+        errorText_ = errorStream_.str();

+        error( RtError::SYSTEM_ERROR );

+      }

+      while ( true ) {

+        result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );

+        if ( FAILED( result ) ) {

+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";

+          errorText_ = errorStream_.str();

+          error( RtError::SYSTEM_ERROR );

+        }

+        result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );

+        if ( FAILED( result ) ) {

+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";

+          errorText_ = errorStream_.str();

+          error( RtError::SYSTEM_ERROR );

+        }

+        if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;

+        Sleep( 1 );

+      }

+

+      //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );

+

+      handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];

+      if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];

+      handle->bufferPointer[1] = safeReadPointer;

+    }

+    else if ( stream_.mode == OUTPUT ) {

+

+      // Set the proper nextWritePosition after initial startup.

+      LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];

+      result = dsWriteBuffer->GetCurrentPosition( &currentWritePointer, &safeWritePointer );

+      if ( FAILED( result ) ) {

+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";

+        errorText_ = errorStream_.str();

+        error( RtError::SYSTEM_ERROR );

+      }

+      handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];

+      if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];

+    }

+

+    buffersRolling = true;

+  }

+

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    

+    LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];

+

+    if ( handle->drainCounter > 1 ) { // write zeros to the output stream

+      bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];

+      bufferBytes *= formatBytes( stream_.userFormat );

+      memset( stream_.userBuffer[0], 0, bufferBytes );

+    }

+

+    // Setup parameters and do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[0] ) {

+      buffer = stream_.deviceBuffer;

+      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );

+      bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];

+      bufferBytes *= formatBytes( stream_.deviceFormat[0] );

+    }

+    else {

+      buffer = stream_.userBuffer[0];

+      bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];

+      bufferBytes *= formatBytes( stream_.userFormat );

+    }

+

+    // No byte swapping necessary in DirectSound implementation.

+

+    // Ahhh ... windoze.  16-bit data is signed but 8-bit data is

+    // unsigned.  So, we need to convert our signed 8-bit data here to

+    // unsigned.

+    if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )

+      for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );

+

+    DWORD dsBufferSize = handle->dsBufferSize[0];

+    nextWritePointer = handle->bufferPointer[0];

+

+    DWORD endWrite, leadPointer;

+    while ( true ) {

+      // Find out where the read and "safe write" pointers are.

+      result = dsBuffer->GetCurrentPosition( &currentWritePointer, &safeWritePointer );

+      if ( FAILED( result ) ) {

+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";

+        errorText_ = errorStream_.str();

+        error( RtError::SYSTEM_ERROR );

+      }

+

+      // We will copy our output buffer into the region between

+      // safeWritePointer and leadPointer.  If leadPointer is not

+      // beyond the next endWrite position, wait until it is.

+      leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];

+      //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;

+      if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;

+      if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset

+      endWrite = nextWritePointer + bufferBytes;

+

+      // Check whether the entire write region is behind the play pointer.

+      if ( leadPointer >= endWrite ) break;

+

+      // If we are here, then we must wait until the leadPointer advances

+      // beyond the end of our next write region. We use the

+      // Sleep() function to suspend operation until that happens.

+      double millis = ( endWrite - leadPointer ) * 1000.0;

+      millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);

+      if ( millis < 1.0 ) millis = 1.0;

+      Sleep( (DWORD) millis );

+    }

+

+    if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )

+         || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { 

+      // We've strayed into the forbidden zone ... resync the read pointer.

+      handle->xrun[0] = true;

+      nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;

+      if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;

+      handle->bufferPointer[0] = nextWritePointer;

+      endWrite = nextWritePointer + bufferBytes;

+    }

+

+    // Lock free space in the buffer

+    result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,

+                             &bufferSize1, &buffer2, &bufferSize2, 0 );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";

+      errorText_ = errorStream_.str();

+      error( RtError::SYSTEM_ERROR );

+    }

+

+    // Copy our buffer into the DS buffer

+    CopyMemory( buffer1, buffer, bufferSize1 );

+    if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );

+

+    // Update our buffer offset and unlock sound buffer

+    dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";

+      errorText_ = errorStream_.str();

+      error( RtError::SYSTEM_ERROR );

+    }

+    nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;

+    handle->bufferPointer[0] = nextWritePointer;

+

+    if ( handle->drainCounter ) {

+      handle->drainCounter++;

+      goto unlock;

+    }

+  }

+

+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {

+

+    // Setup parameters.

+    if ( stream_.doConvertBuffer[1] ) {

+      buffer = stream_.deviceBuffer;

+      bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];

+      bufferBytes *= formatBytes( stream_.deviceFormat[1] );

+    }

+    else {

+      buffer = stream_.userBuffer[1];

+      bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];

+      bufferBytes *= formatBytes( stream_.userFormat );

+    }

+

+    LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];

+    long nextReadPointer = handle->bufferPointer[1];

+    DWORD dsBufferSize = handle->dsBufferSize[1];

+

+    // Find out where the write and "safe read" pointers are.

+    result = dsBuffer->GetCurrentPosition( &currentReadPointer, &safeReadPointer );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";

+      errorText_ = errorStream_.str();

+      error( RtError::SYSTEM_ERROR );

+    }

+

+    if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset

+    DWORD endRead = nextReadPointer + bufferBytes;

+

+    // Handling depends on whether we are INPUT or DUPLEX. 

+    // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,

+    // then a wait here will drag the write pointers into the forbidden zone.

+    // 

+    // In DUPLEX mode, rather than wait, we will back off the read pointer until 

+    // it's in a safe position. This causes dropouts, but it seems to be the only 

+    // practical way to sync up the read and write pointers reliably, given the 

+    // the very complex relationship between phase and increment of the read and write 

+    // pointers.

+    //

+    // In order to minimize audible dropouts in DUPLEX mode, we will

+    // provide a pre-roll period of 0.5 seconds in which we return

+    // zeros from the read buffer while the pointers sync up.

+

+    if ( stream_.mode == DUPLEX ) {

+      if ( safeReadPointer < endRead ) {

+        if ( duplexPrerollBytes <= 0 ) {

+          // Pre-roll time over. Be more agressive.

+          int adjustment = endRead-safeReadPointer;

+

+          handle->xrun[1] = true;

+          // Two cases:

+          //   - large adjustments: we've probably run out of CPU cycles, so just resync exactly,

+          //     and perform fine adjustments later.

+          //   - small adjustments: back off by twice as much.

+          if ( adjustment >= 2*bufferBytes )

+            nextReadPointer = safeReadPointer-2*bufferBytes;

+          else

+            nextReadPointer = safeReadPointer-bufferBytes-adjustment;

+

+          if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;

+

+        }

+        else {

+          // In pre=roll time. Just do it.

+          nextReadPointer = safeReadPointer - bufferBytes;

+          while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;

+        }

+        endRead = nextReadPointer + bufferBytes;

+      }

+    }

+    else { // mode == INPUT

+      while ( safeReadPointer < endRead ) {

+        // See comments for playback.

+        double millis = (endRead - safeReadPointer) * 1000.0;

+        millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);

+        if ( millis < 1.0 ) millis = 1.0;

+        Sleep( (DWORD) millis );

+

+        // Wake up and find out where we are now.

+        result = dsBuffer->GetCurrentPosition( &currentReadPointer, &safeReadPointer );

+        if ( FAILED( result ) ) {

+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";

+          errorText_ = errorStream_.str();

+          error( RtError::SYSTEM_ERROR );

+        }

+      

+        if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset

+      }

+    }

+

+    // Lock free space in the buffer

+    result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,

+                             &bufferSize1, &buffer2, &bufferSize2, 0 );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";

+      errorText_ = errorStream_.str();

+      error( RtError::SYSTEM_ERROR );

+    }

+

+    if ( duplexPrerollBytes <= 0 ) {

+      // Copy our buffer into the DS buffer

+      CopyMemory( buffer, buffer1, bufferSize1 );

+      if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );

+    }

+    else {

+      memset( buffer, 0, bufferSize1 );

+      if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );

+      duplexPrerollBytes -= bufferSize1 + bufferSize2;

+    }

+

+    // Update our buffer offset and unlock sound buffer

+    nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;

+    dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";

+      errorText_ = errorStream_.str();

+      error( RtError::SYSTEM_ERROR );

+    }

+    handle->bufferPointer[1] = nextReadPointer;

+

+    // No byte swapping necessary in DirectSound implementation.

+

+    // If necessary, convert 8-bit data from unsigned to signed.

+    if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )

+      for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );

+

+    // Do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[1] )

+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );

+  }

+

+ unlock:

+  //  MUTEX_UNLOCK( &stream_.mutex );

+

+  RtApi::tickStreamTime();

+}

+

+// Definitions for utility functions and callbacks

+// specific to the DirectSound implementation.

+

+extern "C" unsigned __stdcall callbackHandler( void *ptr )

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

+  RtApiDs *object = (RtApiDs *) info->object;

+  bool* isRunning = &info->isRunning;

+

+  while ( *isRunning == true ) {

+    object->callbackEvent();

+  }

+

+  _endthreadex( 0 );

+  return 0;

+}

+

+#include "tchar.h"

+

+std::string convertTChar( LPCTSTR name )

+{

+#if defined( UNICODE ) || defined( _UNICODE )

+  int length = WideCharToMultiByte(CP_UTF8, 0, name, -1, NULL, 0, NULL, NULL);

+  std::string s( length, 0 );

+  length = WideCharToMultiByte(CP_UTF8, 0, name, wcslen(name), &s[0], length, NULL, NULL);

+#else

+  std::string s( 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_cv;

+  bool runnable;

+

+  AlsaHandle()

+    :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }

+};

+

+extern "C" void *alsaCallbackHandler( void * ptr );

+

+RtApiAlsa :: RtApiAlsa()

+{

+  // Nothing to do here.

+}

+

+RtApiAlsa :: ~RtApiAlsa()

+{

+  if ( stream_.state != STREAM_CLOSED ) closeStream();

+}

+

+unsigned int RtApiAlsa :: getDeviceCount( void )

+{

+  unsigned nDevices = 0;

+  int result, subdevice, card;

+  char name[64];

+  snd_ctl_t *handle;

+

+  // Count cards and devices

+  card = -1;

+  snd_card_next( &card );

+  while ( card >= 0 ) {

+    sprintf( name, "hw:%d", card );

+    result = snd_ctl_open( &handle, name, 0 );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      error( RtError::WARNING );

+      goto nextcard;

+    }

+    subdevice = -1;

+    while( 1 ) {

+      result = snd_ctl_pcm_next_device( handle, &subdevice );

+      if ( result < 0 ) {

+        errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+        error( RtError::WARNING );

+        break;

+      }

+      if ( subdevice < 0 )

+        break;

+      nDevices++;

+    }

+  nextcard:

+    snd_ctl_close( handle );

+    snd_card_next( &card );

+  }

+

+  return nDevices;

+}

+

+RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

+  unsigned nDevices = 0;

+  int result, subdevice, card;

+  char name[64];

+  snd_ctl_t *chandle;

+

+  // Count cards and devices

+  card = -1;

+  snd_card_next( &card );

+  while ( card >= 0 ) {

+    sprintf( name, "hw:%d", card );

+    result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      error( RtError::WARNING );

+      goto nextcard;

+    }

+    subdevice = -1;

+    while( 1 ) {

+      result = snd_ctl_pcm_next_device( chandle, &subdevice );

+      if ( result < 0 ) {

+        errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+        error( RtError::WARNING );

+        break;

+      }

+      if ( subdevice < 0 ) break;

+      if ( nDevices == device ) {

+        sprintf( name, "hw:%d,%d", card, subdevice );

+        goto foundDevice;

+      }

+      nDevices++;

+    }

+  nextcard:

+    snd_ctl_close( chandle );

+    snd_card_next( &card );

+  }

+

+  if ( nDevices == 0 ) {

+    errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( device >= nDevices ) {

+    errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";

+    error( RtError::INVALID_USE );

+  }

+

+ foundDevice:

+

+  // If a stream is already open, we cannot probe the stream devices.

+  // Thus, use the saved results.

+  if ( stream_.state != STREAM_CLOSED &&

+       ( stream_.device[0] == device || stream_.device[1] == device ) ) {

+    if ( device >= devices_.size() ) {

+      errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";

+      error( RtError::WARNING );

+      return info;

+    }

+    return devices_[ device ];

+  }

+

+  int openMode = SND_PCM_ASYNC;

+  snd_pcm_stream_t stream;

+  snd_pcm_info_t *pcminfo;

+  snd_pcm_info_alloca( &pcminfo );

+  snd_pcm_t *phandle;

+  snd_pcm_hw_params_t *params;

+  snd_pcm_hw_params_alloca( &params );

+

+  // First try for playback

+  stream = SND_PCM_STREAM_PLAYBACK;

+  snd_pcm_info_set_device( pcminfo, subdevice );

+  snd_pcm_info_set_subdevice( pcminfo, 0 );

+  snd_pcm_info_set_stream( pcminfo, stream );

+

+  result = snd_ctl_pcm_info( chandle, pcminfo );

+  if ( result < 0 ) {

+    // Device probably doesn't support playback.

+    goto captureProbe;

+  }

+

+  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );

+  if ( result < 0 ) {

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    goto captureProbe;

+  }

+

+  // The device is open ... fill the parameter structure.

+  result = snd_pcm_hw_params_any( phandle, params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    goto captureProbe;

+  }

+

+  // Get output channel information.

+  unsigned int value;

+  result = snd_pcm_hw_params_get_channels_max( params, &value );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    goto captureProbe;

+  }

+  info.outputChannels = value;

+  snd_pcm_close( phandle );

+

+ captureProbe:

+  // Now try for capture

+  stream = SND_PCM_STREAM_CAPTURE;

+  snd_pcm_info_set_stream( pcminfo, stream );

+

+  result = snd_ctl_pcm_info( chandle, pcminfo );

+  snd_ctl_close( chandle );

+  if ( result < 0 ) {

+    // Device probably doesn't support capture.

+    if ( info.outputChannels == 0 ) return info;

+    goto probeParameters;

+  }

+

+  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);

+  if ( result < 0 ) {

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    if ( info.outputChannels == 0 ) return info;

+    goto probeParameters;

+  }

+

+  // The device is open ... fill the parameter structure.

+  result = snd_pcm_hw_params_any( phandle, params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    if ( info.outputChannels == 0 ) return info;

+    goto probeParameters;

+  }

+

+  result = snd_pcm_hw_params_get_channels_max( params, &value );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    if ( info.outputChannels == 0 ) return info;

+    goto probeParameters;

+  }

+  info.inputChannels = value;

+  snd_pcm_close( phandle );

+

+  // If device opens for both playback and capture, we determine the channels.

+  if ( info.outputChannels > 0 && info.inputChannels > 0 )

+    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;

+

+  // ALSA doesn't provide default devices so we'll use the first available one.

+  if ( device == 0 && info.outputChannels > 0 )

+    info.isDefaultOutput = true;

+  if ( device == 0 && info.inputChannels > 0 )

+    info.isDefaultInput = true;

+

+ probeParameters:

+  // At this point, we just need to figure out the supported data

+  // formats and sample rates.  We'll proceed by opening the device in

+  // the direction with the maximum number of channels, or playback if

+  // they are equal.  This might limit our sample rate options, but so

+  // be it.

+

+  if ( info.outputChannels >= info.inputChannels )

+    stream = SND_PCM_STREAM_PLAYBACK;

+  else

+    stream = SND_PCM_STREAM_CAPTURE;

+  snd_pcm_info_set_stream( pcminfo, stream );

+

+  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);

+  if ( result < 0 ) {

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // The device is open ... fill the parameter structure.

+  result = snd_pcm_hw_params_any( phandle, params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Test our discrete set of sample rate values.

+  info.sampleRates.clear();

+  for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {

+    if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )

+      info.sampleRates.push_back( SAMPLE_RATES[i] );

+  }

+  if ( info.sampleRates.size() == 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Probe the supported data formats ... we don't care about endian-ness just yet

+  snd_pcm_format_t format;

+  info.nativeFormats = 0;

+  format = SND_PCM_FORMAT_S8;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_SINT8;

+  format = SND_PCM_FORMAT_S16;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_SINT16;

+  format = SND_PCM_FORMAT_S24;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_SINT24;

+  format = SND_PCM_FORMAT_S32;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_SINT32;

+  format = SND_PCM_FORMAT_FLOAT;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_FLOAT32;

+  format = SND_PCM_FORMAT_FLOAT64;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_FLOAT64;

+

+  // Check that we have at least one supported format

+  if ( info.nativeFormats == 0 ) {

+    errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Get the device name

+  char *cardname;

+  result = snd_card_get_name( card, &cardname );

+  if ( result >= 0 )

+    sprintf( name, "hw:%s,%d", cardname, subdevice );

+  info.name = name;

+

+  // That's all ... close the device and return

+  snd_pcm_close( phandle );

+  info.probed = true;

+  return info;

+}

+

+void RtApiAlsa :: saveDeviceInfo( void )

+{

+  devices_.clear();

+

+  unsigned int nDevices = getDeviceCount();

+  devices_.resize( nDevices );

+  for ( unsigned int i=0; i<nDevices; i++ )

+    devices_[i] = getDeviceInfo( i );

+}

+

+bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,

+                                   unsigned int firstChannel, unsigned int sampleRate,

+                                   RtAudioFormat format, unsigned int *bufferSize,

+                                   RtAudio::StreamOptions *options )

+

+{

+#if defined(__RTAUDIO_DEBUG__)

+  snd_output_t *out;

+  snd_output_stdio_attach(&out, stderr, 0);

+#endif

+

+  // I'm not using the "plug" interface ... too much inconsistent behavior.

+

+  unsigned nDevices = 0;

+  int result, subdevice, card;

+  char name[64];

+  snd_ctl_t *chandle;

+

+  if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )

+    snprintf(name, sizeof(name), "%s", "default");

+  else {

+    // Count cards and devices

+    card = -1;

+    snd_card_next( &card );

+    while ( card >= 0 ) {

+      sprintf( name, "hw:%d", card );

+      result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );

+      if ( result < 0 ) {

+        errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+        return FAILURE;

+      }

+      subdevice = -1;

+      while( 1 ) {

+        result = snd_ctl_pcm_next_device( chandle, &subdevice );

+        if ( result < 0 ) break;

+        if ( subdevice < 0 ) break;

+        if ( nDevices == device ) {

+          sprintf( name, "hw:%d,%d", card, subdevice );

+          snd_ctl_close( chandle );

+          goto foundDevice;

+        }

+        nDevices++;

+      }

+      snd_ctl_close( chandle );

+      snd_card_next( &card );

+    }

+

+    if ( nDevices == 0 ) {

+      // This should not happen because a check is made before this function is called.

+      errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";

+      return FAILURE;

+    }

+

+    if ( device >= nDevices ) {

+      // This should not happen because a check is made before this function is called.

+      errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";

+      return FAILURE;

+    }

+  }

+

+ foundDevice:

+

+  // The getDeviceInfo() function will not work for a device that is

+  // already open.  Thus, we'll probe the system before opening a

+  // stream and save the results for use by getDeviceInfo().

+  if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once

+    this->saveDeviceInfo();

+

+  snd_pcm_stream_t stream;

+  if ( mode == OUTPUT )

+    stream = SND_PCM_STREAM_PLAYBACK;

+  else

+    stream = SND_PCM_STREAM_CAPTURE;

+

+  snd_pcm_t *phandle;

+  int openMode = SND_PCM_ASYNC;

+  result = snd_pcm_open( &phandle, name, stream, openMode );

+  if ( result < 0 ) {

+    if ( mode == OUTPUT )

+      errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";

+    else

+      errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Fill the parameter structure.

+  snd_pcm_hw_params_t *hw_params;

+  snd_pcm_hw_params_alloca( &hw_params );

+  result = snd_pcm_hw_params_any( phandle, hw_params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+#if defined(__RTAUDIO_DEBUG__)

+  fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );

+  snd_pcm_hw_params_dump( hw_params, out );

+#endif

+

+  // Set access ... check user preference.

+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {

+    stream_.userInterleaved = false;

+    result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );

+    if ( result < 0 ) {

+      result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );

+      stream_.deviceInterleaved[mode] =  true;

+    }

+    else

+      stream_.deviceInterleaved[mode] = false;

+  }

+  else {

+    stream_.userInterleaved = true;

+    result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );

+    if ( result < 0 ) {

+      result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );

+      stream_.deviceInterleaved[mode] =  false;

+    }

+    else

+      stream_.deviceInterleaved[mode] =  true;

+  }

+

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Determine how to set the device format.

+  stream_.userFormat = format;

+  snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;

+

+  if ( format == RTAUDIO_SINT8 )

+    deviceFormat = SND_PCM_FORMAT_S8;

+  else if ( format == RTAUDIO_SINT16 )

+    deviceFormat = SND_PCM_FORMAT_S16;

+  else if ( format == RTAUDIO_SINT24 )

+    deviceFormat = SND_PCM_FORMAT_S24;

+  else if ( format == RTAUDIO_SINT32 )

+    deviceFormat = SND_PCM_FORMAT_S32;

+  else if ( format == RTAUDIO_FLOAT32 )

+    deviceFormat = SND_PCM_FORMAT_FLOAT;

+  else if ( format == RTAUDIO_FLOAT64 )

+    deviceFormat = SND_PCM_FORMAT_FLOAT64;

+

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {

+    stream_.deviceFormat[mode] = format;

+    goto setFormat;

+  }

+

+  // The user requested format is not natively supported by the device.

+  deviceFormat = SND_PCM_FORMAT_FLOAT64;

+  if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_FLOAT;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_S32;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_S24;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_S16;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_S8;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT8;

+    goto setFormat;

+  }

+

+  // If we get here, no supported format was found.

+  errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";

+  errorText_ = errorStream_.str();

+  return FAILURE;

+

+ setFormat:

+  result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Determine whether byte-swaping is necessary.

+  stream_.doByteSwap[mode] = false;

+  if ( deviceFormat != SND_PCM_FORMAT_S8 ) {

+    result = snd_pcm_format_cpu_endian( deviceFormat );

+    if ( result == 0 )

+      stream_.doByteSwap[mode] = true;

+    else if (result < 0) {

+      snd_pcm_close( phandle );

+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

+  // Set the sample rate.

+  result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Determine the number of channels for this device.  We support a possible

+  // minimum device channel number > than the value requested by the user.

+  stream_.nUserChannels[mode] = channels;

+  unsigned int value;

+  result = snd_pcm_hw_params_get_channels_max( hw_params, &value );

+  unsigned int deviceChannels = value;

+  if ( result < 0 || deviceChannels < channels + firstChannel ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  result = snd_pcm_hw_params_get_channels_min( hw_params, &value );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  deviceChannels = value;

+  if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;

+  stream_.nDeviceChannels[mode] = deviceChannels;

+

+  // Set the device channels.

+  result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the buffer (or period) size.

+  int dir = 0;

+  snd_pcm_uframes_t periodSize = *bufferSize;

+  result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  *bufferSize = periodSize;

+

+  // Set the buffer number, which in ALSA is referred to as the "period".

+  unsigned int periods = 0;

+  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;

+  if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;

+  if ( periods < 2 ) periods = 4; // a fairly safe default value

+  result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // If attempting to setup a duplex stream, the bufferSize parameter

+  // MUST be the same in both directions!

+  if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  stream_.bufferSize = *bufferSize;

+

+  // Install the hardware configuration

+  result = snd_pcm_hw_params( phandle, hw_params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+#if defined(__RTAUDIO_DEBUG__)

+  fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");

+  snd_pcm_hw_params_dump( hw_params, out );

+#endif

+

+  // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.

+  snd_pcm_sw_params_t *sw_params = NULL;

+  snd_pcm_sw_params_alloca( &sw_params );

+  snd_pcm_sw_params_current( phandle, sw_params );

+  snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );

+  snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );

+  snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );

+

+  // The following two settings were suggested by Theo Veenker

+  //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );

+  //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );

+

+  // here are two options for a fix

+  //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );

+  snd_pcm_uframes_t val;

+  snd_pcm_sw_params_get_boundary( sw_params, &val );

+  snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );

+

+  result = snd_pcm_sw_params( phandle, sw_params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+#if defined(__RTAUDIO_DEBUG__)

+  fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");

+  snd_pcm_sw_params_dump( sw_params, out );

+#endif

+

+  // Set flags for buffer conversion

+  stream_.doConvertBuffer[mode] = false;

+  if ( stream_.userFormat != stream_.deviceFormat[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&

+       stream_.nUserChannels[mode] > 1 )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate the ApiHandle if necessary and then save.

+  AlsaHandle *apiInfo = 0;

+  if ( stream_.apiHandle == 0 ) {

+    try {

+      apiInfo = (AlsaHandle *) new AlsaHandle;

+    }

+    catch ( std::bad_alloc& ) {

+      errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";

+      goto error;

+    }

+

+    if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {

+      errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";

+      goto error;

+    }

+

+    stream_.apiHandle = (void *) apiInfo;

+    apiInfo->handles[0] = 0;

+    apiInfo->handles[1] = 0;

+  }

+  else {

+    apiInfo = (AlsaHandle *) stream_.apiHandle;

+  }

+  apiInfo->handles[mode] = phandle;

+

+  // 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_cv );

+    if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );

+    if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );

+    delete apiInfo;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  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 ) {

+    apiInfo->runnable = true;

+    pthread_cond_signal( &apiInfo->runnable_cv );

+  }

+  MUTEX_UNLOCK( &stream_.mutex );

+  pthread_join( stream_.callbackInfo.thread, NULL );

+

+  if ( stream_.state == STREAM_RUNNING ) {

+    stream_.state = STREAM_STOPPED;

+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )

+      snd_pcm_drop( apiInfo->handles[0] );

+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX )

+      snd_pcm_drop( apiInfo->handles[1] );

+  }

+

+  if ( apiInfo ) {

+    pthread_cond_destroy( &apiInfo->runnable_cv );

+    if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );

+    if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );

+    delete apiInfo;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiAlsa :: startStream()

+{

+  // This method calls snd_pcm_prepare if the device isn't already in that state.

+

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

+    errorText_ = "RtApiAlsa::startStream(): the stream is already running!";

+    error( 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:

+  apiInfo->runnable = true;

+  pthread_cond_signal( &apiInfo->runnable_cv );

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result >= 0 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAlsa :: stopStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  stream_.state = STREAM_STOPPED;

+  MUTEX_LOCK( &stream_.mutex );

+

+  //if ( stream_.state == STREAM_STOPPED ) {

+  //  MUTEX_UNLOCK( &stream_.mutex );

+  //  return;

+  //}

+

+  int result = 0;

+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;

+  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    if ( apiInfo->synchronized ) 

+      result = snd_pcm_drop( handle[0] );

+    else

+      result = snd_pcm_drain( handle[0] );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {

+    result = snd_pcm_drop( handle[1] );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+ unlock:

+  stream_.state = STREAM_STOPPED;

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result >= 0 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAlsa :: abortStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  stream_.state = STREAM_STOPPED;

+  MUTEX_LOCK( &stream_.mutex );

+

+  //if ( stream_.state == STREAM_STOPPED ) {

+  //  MUTEX_UNLOCK( &stream_.mutex );

+  //  return;

+  //}

+

+  int result = 0;

+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;

+  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    result = snd_pcm_drop( handle[0] );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {

+    result = snd_pcm_drop( handle[1] );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+ unlock:

+  stream_.state = STREAM_STOPPED;

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result >= 0 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAlsa :: callbackEvent()

+{

+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_LOCK( &stream_.mutex );

+    while ( !apiInfo->runnable )

+      pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );

+

+    if ( stream_.state != STREAM_RUNNING ) {

+      MUTEX_UNLOCK( &stream_.mutex );

+      return;

+    }

+    MUTEX_UNLOCK( &stream_.mutex );

+  }

+

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  int doStopStream = 0;

+  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;

+  double streamTime = getStreamTime();

+  RtAudioStreamStatus status = 0;

+  if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {

+    status |= RTAUDIO_OUTPUT_UNDERFLOW;

+    apiInfo->xrun[0] = false;

+  }

+  if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {

+    status |= RTAUDIO_INPUT_OVERFLOW;

+    apiInfo->xrun[1] = false;

+  }

+  doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],

+                           stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );

+

+  if ( doStopStream == 2 ) {

+    abortStream();

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) goto unlock;

+

+  int result;

+  char *buffer;

+  int channels;

+  snd_pcm_t **handle;

+  snd_pcm_sframes_t frames;

+  RtAudioFormat format;

+  handle = (snd_pcm_t **) apiInfo->handles;

+

+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {

+

+    // Setup parameters.

+    if ( stream_.doConvertBuffer[1] ) {

+      buffer = stream_.deviceBuffer;

+      channels = stream_.nDeviceChannels[1];

+      format = stream_.deviceFormat[1];

+    }

+    else {

+      buffer = stream_.userBuffer[1];

+      channels = stream_.nUserChannels[1];

+      format = stream_.userFormat;

+    }

+

+    // Read samples from device in interleaved/non-interleaved format.

+    if ( stream_.deviceInterleaved[1] )

+      result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );

+    else {

+      void *bufs[channels];

+      size_t offset = stream_.bufferSize * formatBytes( format );

+      for ( int i=0; i<channels; i++ )

+        bufs[i] = (void *) (buffer + (i * offset));

+      result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );

+    }

+

+    if ( result < (int) stream_.bufferSize ) {

+      // Either an error or overrun occured.

+      if ( result == -EPIPE ) {

+        snd_pcm_state_t state = snd_pcm_state( handle[1] );

+        if ( state == SND_PCM_STATE_XRUN ) {

+          apiInfo->xrun[1] = true;

+          result = snd_pcm_prepare( handle[1] );

+          if ( result < 0 ) {

+            errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";

+            errorText_ = errorStream_.str();

+          }

+        }

+        else {

+          errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";

+          errorText_ = errorStream_.str();

+        }

+      }

+      else {

+        errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+      }

+      error( RtError::WARNING );

+      goto tryOutput;

+    }

+

+    // Do byte swapping if necessary.

+    if ( stream_.doByteSwap[1] )

+      byteSwapBuffer( buffer, stream_.bufferSize * channels, format );

+

+    // Do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[1] )

+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );

+

+    // Check stream latency

+    result = snd_pcm_delay( handle[1], &frames );

+    if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;

+  }

+

+ tryOutput:

+

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    // Setup parameters and do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[0] ) {

+      buffer = stream_.deviceBuffer;

+      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );

+      channels = stream_.nDeviceChannels[0];

+      format = stream_.deviceFormat[0];

+    }

+    else {

+      buffer = stream_.userBuffer[0];

+      channels = stream_.nUserChannels[0];

+      format = stream_.userFormat;

+    }

+

+    // Do byte swapping if necessary.

+    if ( stream_.doByteSwap[0] )

+      byteSwapBuffer(buffer, stream_.bufferSize * channels, format);

+

+    // Write samples to device in interleaved/non-interleaved format.

+    if ( stream_.deviceInterleaved[0] )

+      result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );

+    else {

+      void *bufs[channels];

+      size_t offset = stream_.bufferSize * formatBytes( format );

+      for ( int i=0; i<channels; i++ )

+        bufs[i] = (void *) (buffer + (i * offset));

+      result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );

+    }

+

+    if ( result < (int) stream_.bufferSize ) {

+      // Either an error or underrun occured.

+      if ( result == -EPIPE ) {

+        snd_pcm_state_t state = snd_pcm_state( handle[0] );

+        if ( state == SND_PCM_STATE_XRUN ) {

+          apiInfo->xrun[0] = true;

+          result = snd_pcm_prepare( handle[0] );

+          if ( result < 0 ) {

+            errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";

+            errorText_ = errorStream_.str();

+          }

+        }

+        else {

+          errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";

+          errorText_ = errorStream_.str();

+        }

+      }

+      else {

+        errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+      }

+      error( RtError::WARNING );

+      goto unlock;

+    }

+

+    // Check stream latency

+    result = snd_pcm_delay( handle[0], &frames );

+    if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;

+  }

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  RtApi::tickStreamTime();

+  if ( doStopStream == 1 ) this->stopStream();

+}

+

+extern "C" void *alsaCallbackHandler( void *ptr )

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

+  RtApiAlsa *object = (RtApiAlsa *) info->object;

+  bool *isRunning = &info->isRunning;

+

+  while ( *isRunning == true ) {

+    pthread_testcancel();

+    object->callbackEvent();

+  }

+

+  pthread_exit( NULL );

+}

+

+//******************** End of __LINUX_ALSA__ *********************//

+#endif

+

+

+#if defined(__LINUX_OSS__)

+

+#include <unistd.h>

+#include <sys/ioctl.h>

+#include <unistd.h>

+#include <fcntl.h>

+#include "soundcard.h"

+#include <errno.h>

+#include <math.h>

+

+extern "C" void *ossCallbackHandler(void * ptr);

+

+// A structure to hold various information related to the OSS API

+// implementation.

+struct OssHandle {

+  int id[2];    // device ids

+  bool xrun[2];

+  bool triggered;

+  pthread_cond_t runnable;

+

+  OssHandle()

+    :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }

+};

+

+RtApiOss :: RtApiOss()

+{

+  // Nothing to do here.

+}

+

+RtApiOss :: ~RtApiOss()

+{

+  if ( stream_.state != STREAM_CLOSED ) closeStream();

+}

+

+unsigned int RtApiOss :: getDeviceCount( void )

+{

+  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );

+  if ( mixerfd == -1 ) {

+    errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";

+    error( RtError::WARNING );

+    return 0;

+  }

+

+  oss_sysinfo sysinfo;

+  if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";

+    error( RtError::WARNING );

+    return 0;

+  }

+

+  close( mixerfd );

+  return sysinfo.numaudios;

+}

+

+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

+  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );

+  if ( mixerfd == -1 ) {

+    errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";

+    error( RtError::WARNING );

+    return info;

+  }

+

+  oss_sysinfo sysinfo;

+  int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );

+  if ( result == -1 ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";

+    error( RtError::WARNING );

+    return info;

+  }

+

+  unsigned nDevices = sysinfo.numaudios;

+  if ( nDevices == 0 ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::getDeviceInfo: no devices found!";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( device >= nDevices ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";

+    error( RtError::INVALID_USE );

+  }

+

+  oss_audioinfo ainfo;

+  ainfo.dev = device;

+  result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );

+  close( mixerfd );

+  if ( result == -1 ) {

+    errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Probe channels

+  if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;

+  if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;

+  if ( ainfo.caps & PCM_CAP_DUPLEX ) {

+    if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )

+      info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;

+  }

+

+  // Probe data formats ... do for input

+  unsigned long mask = ainfo.iformats;

+  if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )

+    info.nativeFormats |= RTAUDIO_SINT16;

+  if ( mask & AFMT_S8 )

+    info.nativeFormats |= RTAUDIO_SINT8;

+  if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )

+    info.nativeFormats |= RTAUDIO_SINT32;

+  if ( mask & AFMT_FLOAT )

+    info.nativeFormats |= RTAUDIO_FLOAT32;

+  if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )

+    info.nativeFormats |= RTAUDIO_SINT24;

+

+  // Check that we have at least one supported format

+  if ( info.nativeFormats == 0 ) {

+    errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Probe the supported sample rates.

+  info.sampleRates.clear();

+  if ( ainfo.nrates ) {

+    for ( unsigned int i=0; i<ainfo.nrates; i++ ) {

+      for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {

+        if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {

+          info.sampleRates.push_back( SAMPLE_RATES[k] );

+          break;

+        }

+      }

+    }

+  }

+  else {

+    // Check min and max rate values;

+    for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {

+      if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )

+        info.sampleRates.push_back( SAMPLE_RATES[k] );

+    }

+  }

+

+  if ( info.sampleRates.size() == 0 ) {

+    errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+  }

+  else {

+    info.probed = true;

+    info.name = ainfo.name;

+  }

+

+  return info;

+}

+

+

+bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,

+                                  unsigned int firstChannel, unsigned int sampleRate,

+                                  RtAudioFormat format, unsigned int *bufferSize,

+                                  RtAudio::StreamOptions *options )

+{

+  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );

+  if ( mixerfd == -1 ) {

+    errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";

+    return FAILURE;

+  }

+

+  oss_sysinfo sysinfo;

+  int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );

+  if ( result == -1 ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";

+    return FAILURE;

+  }

+

+  unsigned nDevices = sysinfo.numaudios;

+  if ( nDevices == 0 ) {

+    // This should not happen because a check is made before this function is called.

+    close( mixerfd );

+    errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";

+    return FAILURE;

+  }

+

+  if ( device >= nDevices ) {

+    // This should not happen because a check is made before this function is called.

+    close( mixerfd );

+    errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";

+    return FAILURE;

+  }

+

+  oss_audioinfo ainfo;

+  ainfo.dev = device;

+  result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );

+  close( mixerfd );

+  if ( result == -1 ) {

+    errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Check if device supports input or output

+  if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||

+       ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {

+    if ( mode == OUTPUT )

+      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";

+    else

+      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  int flags = 0;

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  if ( mode == OUTPUT )

+    flags |= O_WRONLY;

+  else { // mode == INPUT

+    if (stream_.mode == OUTPUT && stream_.device[0] == device) {

+      // We just set the same device for playback ... close and reopen for duplex (OSS only).

+      close( handle->id[0] );

+      handle->id[0] = 0;

+      if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {

+        errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";

+        errorText_ = errorStream_.str();

+        return FAILURE;

+      }

+      // Check that the number previously set channels is the same.

+      if ( stream_.nUserChannels[0] != channels ) {

+        errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";

+        errorText_ = errorStream_.str();

+        return FAILURE;

+      }

+      flags |= O_RDWR;

+    }

+    else

+      flags |= O_RDONLY;

+  }

+

+  // Set exclusive access if specified.

+  if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;

+

+  // Try to open the device.

+  int fd;

+  fd = open( ainfo.devnode, flags, 0 );

+  if ( fd == -1 ) {

+    if ( errno == EBUSY )

+      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";

+    else

+      errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // For duplex operation, specifically set this mode (this doesn't seem to work).

+  /*

+    if ( flags | O_RDWR ) {

+    result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );

+    if ( result == -1) {

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+    }

+    }

+  */

+

+  // Check the device channel support.

+  stream_.nUserChannels[mode] = channels;

+  if ( ainfo.max_channels < (int)(channels + firstChannel) ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the number of channels.

+  int deviceChannels = channels + firstChannel;

+  result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );

+  if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.nDeviceChannels[mode] = deviceChannels;

+

+  // Get the data format mask

+  int mask;

+  result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );

+  if ( result == -1 ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Determine how to set the device format.

+  stream_.userFormat = format;

+  int deviceFormat = -1;

+  stream_.doByteSwap[mode] = false;

+  if ( format == RTAUDIO_SINT8 ) {

+    if ( mask & AFMT_S8 ) {

+      deviceFormat = AFMT_S8;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT8;

+    }

+  }

+  else if ( format == RTAUDIO_SINT16 ) {

+    if ( mask & AFMT_S16_NE ) {

+      deviceFormat = AFMT_S16_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+    }

+    else if ( mask & AFMT_S16_OE ) {

+      deviceFormat = AFMT_S16_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+      stream_.doByteSwap[mode] = true;

+    }

+  }

+  else if ( format == RTAUDIO_SINT24 ) {

+    if ( mask & AFMT_S24_NE ) {

+      deviceFormat = AFMT_S24_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+    }

+    else if ( mask & AFMT_S24_OE ) {

+      deviceFormat = AFMT_S24_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+      stream_.doByteSwap[mode] = true;

+    }

+  }

+  else if ( format == RTAUDIO_SINT32 ) {

+    if ( mask & AFMT_S32_NE ) {

+      deviceFormat = AFMT_S32_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+    }

+    else if ( mask & AFMT_S32_OE ) {

+      deviceFormat = AFMT_S32_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+      stream_.doByteSwap[mode] = true;

+    }

+  }

+

+  if ( deviceFormat == -1 ) {

+    // The user requested format is not natively supported by the device.

+    if ( mask & AFMT_S16_NE ) {

+      deviceFormat = AFMT_S16_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+    }

+    else if ( mask & AFMT_S32_NE ) {

+      deviceFormat = AFMT_S32_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+    }

+    else if ( mask & AFMT_S24_NE ) {

+      deviceFormat = AFMT_S24_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+    }

+    else if ( mask & AFMT_S16_OE ) {

+      deviceFormat = AFMT_S16_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+      stream_.doByteSwap[mode] = true;

+    }

+    else if ( mask & AFMT_S32_OE ) {

+      deviceFormat = AFMT_S32_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+      stream_.doByteSwap[mode] = true;

+    }

+    else if ( mask & AFMT_S24_OE ) {

+      deviceFormat = AFMT_S24_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+      stream_.doByteSwap[mode] = true;

+    }

+    else if ( mask & AFMT_S8) {

+      deviceFormat = AFMT_S8;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT8;

+    }

+  }

+

+  if ( stream_.deviceFormat[mode] == 0 ) {

+    // This really shouldn't happen ...

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the data format.

+  int temp = deviceFormat;

+  result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );

+  if ( result == -1 || deviceFormat != temp ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Attempt to set the buffer size.  According to OSS, the minimum

+  // number of buffers is two.  The supposed minimum buffer size is 16

+  // bytes, so that will be our lower bound.  The argument to this

+  // call is in the form 0xMMMMSSSS (hex), where the buffer size (in

+  // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.

+  // We'll check the actual value used near the end of the setup

+  // procedure.

+  int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;

+  if ( ossBufferBytes < 16 ) ossBufferBytes = 16;

+  int buffers = 0;

+  if ( options ) buffers = options->numberOfBuffers;

+  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;

+  if ( buffers < 2 ) buffers = 3;

+  temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );

+  result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );

+  if ( result == -1 ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.nBuffers = buffers;

+

+  // Save buffer size (in sample frames).

+  *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );

+  stream_.bufferSize = *bufferSize;

+

+  // Set the sample rate.

+  int srate = sampleRate;

+  result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );

+  if ( result == -1 ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Verify the sample rate setup worked.

+  if ( abs( srate - sampleRate ) > 100 ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.sampleRate = sampleRate;

+

+  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {

+    // We're doing duplex setup here.

+    stream_.deviceFormat[0] = stream_.deviceFormat[1];

+    stream_.nDeviceChannels[0] = deviceChannels;

+  }

+

+  // Set interleaving parameters.

+  stream_.userInterleaved = true;

+  stream_.deviceInterleaved[mode] =  true;

+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED )

+    stream_.userInterleaved = false;

+

+  // Set flags for buffer conversion

+  stream_.doConvertBuffer[mode] = false;

+  if ( stream_.userFormat != stream_.deviceFormat[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&

+       stream_.nUserChannels[mode] > 1 )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate the stream handles if necessary and then save.

+  if ( stream_.apiHandle == 0 ) {

+    try {

+      handle = new OssHandle;

+    }

+    catch ( std::bad_alloc& ) {

+      errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";

+      goto error;

+    }

+

+    if ( pthread_cond_init( &handle->runnable, NULL ) ) {

+      errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";

+      goto error;

+    }

+

+    stream_.apiHandle = (void *) handle;

+  }

+  else {

+    handle = (OssHandle *) stream_.apiHandle;

+  }

+  handle->id[mode] = fd;

+

+  // Allocate necessary internal buffers.

+  unsigned long bufferBytes;

+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );

+  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );

+  if ( stream_.userBuffer[mode] == NULL ) {

+    errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";

+    goto error;

+  }

+

+  if ( stream_.doConvertBuffer[mode] ) {

+

+    bool makeBuffer = true;

+    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );

+    if ( mode == INPUT ) {

+      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {

+        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );

+        if ( bufferBytes <= bytesOut ) makeBuffer = false;

+      }

+    }

+

+    if ( makeBuffer ) {

+      bufferBytes *= *bufferSize;

+      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );

+      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );

+      if ( stream_.deviceBuffer == NULL ) {

+        errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";

+        goto error;

+      }

+    }

+  }

+

+  stream_.device[mode] = device;

+  stream_.state = STREAM_STOPPED;

+

+  // Setup the buffer conversion information structure.

+  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );

+

+  // Setup thread if necessary.

+  if ( stream_.mode == OUTPUT && mode == INPUT ) {

+    // We had already set up an output stream.

+    stream_.mode = DUPLEX;

+    if ( stream_.device[0] == device ) handle->id[0] = fd;

+  }

+  else {

+    stream_.mode = mode;

+

+    // Setup callback thread.

+    stream_.callbackInfo.object = (void *) this;

+

+    // Set the thread attributes for joinable and realtime scheduling

+    // priority.  The higher priority will only take affect if the

+    // program is run as root or suid.

+    pthread_attr_t attr;

+    pthread_attr_init( &attr );

+    pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );

+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)

+    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {

+      struct sched_param param;

+      int priority = options->priority;

+      int min = sched_get_priority_min( SCHED_RR );

+      int max = sched_get_priority_max( SCHED_RR );

+      if ( priority < min ) priority = min;

+      else if ( priority > max ) priority = max;

+      param.sched_priority = priority;

+      pthread_attr_setschedparam( &attr, &param );

+      pthread_attr_setschedpolicy( &attr, SCHED_RR );

+    }

+    else

+      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );

+#else

+    pthread_attr_setschedpolicy( &attr, SCHED_OTHER );

+#endif

+

+    stream_.callbackInfo.isRunning = true;

+    result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );

+    pthread_attr_destroy( &attr );

+    if ( result ) {

+      stream_.callbackInfo.isRunning = false;

+      errorText_ = "RtApiOss::error creating callback thread!";

+      goto error;

+    }

+  }

+

+  return SUCCESS;

+

+ error:

+  if ( handle ) {

+    pthread_cond_destroy( &handle->runnable );

+    if ( handle->id[0] ) close( handle->id[0] );

+    if ( handle->id[1] ) close( handle->id[1] );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiOss :: closeStream()

+{

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiOss::closeStream(): no open stream to close!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  stream_.callbackInfo.isRunning = false;

+  MUTEX_LOCK( &stream_.mutex );

+  if ( stream_.state == STREAM_STOPPED )

+    pthread_cond_signal( &handle->runnable );

+  MUTEX_UNLOCK( &stream_.mutex );

+  pthread_join( stream_.callbackInfo.thread, NULL );

+

+  if ( stream_.state == STREAM_RUNNING ) {

+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )

+      ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );

+    else

+      ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );

+    stream_.state = STREAM_STOPPED;

+  }

+

+  if ( handle ) {

+    pthread_cond_destroy( &handle->runnable );

+    if ( handle->id[0] ) close( handle->id[0] );

+    if ( handle->id[1] ) close( handle->id[1] );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiOss :: startStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

+    errorText_ = "RtApiOss::startStream(): the stream is already running!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  stream_.state = STREAM_RUNNING;

+

+  // No need to do anything else here ... OSS automatically starts

+  // when fed samples.

+

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  pthread_cond_signal( &handle->runnable );

+}

+

+void RtApiOss :: stopStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+

+  int result = 0;

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    // Flush the output with zeros a few times.

+    char *buffer;

+    int samples;

+    RtAudioFormat format;

+

+    if ( stream_.doConvertBuffer[0] ) {

+      buffer = stream_.deviceBuffer;

+      samples = stream_.bufferSize * stream_.nDeviceChannels[0];

+      format = stream_.deviceFormat[0];

+    }

+    else {

+      buffer = stream_.userBuffer[0];

+      samples = stream_.bufferSize * stream_.nUserChannels[0];

+      format = stream_.userFormat;

+    }

+

+    memset( buffer, 0, samples * formatBytes(format) );

+    for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {

+      result = write( handle->id[0], buffer, samples * formatBytes(format) );

+      if ( result == -1 ) {

+        errorText_ = "RtApiOss::stopStream: audio write error.";

+        error( RtError::WARNING );

+      }

+    }

+

+    result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );

+    if ( result == -1 ) {

+      errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+    handle->triggered = false;

+  }

+

+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {

+    result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );

+    if ( result == -1 ) {

+      errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+ unlock:

+  stream_.state = STREAM_STOPPED;

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result != -1 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiOss :: abortStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+

+  int result = 0;

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );

+    if ( result == -1 ) {

+      errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+    handle->triggered = false;

+  }

+

+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {

+    result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );

+    if ( result == -1 ) {

+      errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+ unlock:

+  stream_.state = STREAM_STOPPED;

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result != -1 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiOss :: callbackEvent()

+{

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_LOCK( &stream_.mutex );

+    pthread_cond_wait( &handle->runnable, &stream_.mutex );

+    if ( stream_.state != STREAM_RUNNING ) {

+      MUTEX_UNLOCK( &stream_.mutex );

+      return;

+    }

+    MUTEX_UNLOCK( &stream_.mutex );

+  }

+

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  // Invoke user callback to get fresh output data.

+  int doStopStream = 0;

+  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;

+  double streamTime = getStreamTime();

+  RtAudioStreamStatus status = 0;

+  if ( stream_.mode != INPUT && handle->xrun[0] == true ) {

+    status |= RTAUDIO_OUTPUT_UNDERFLOW;

+    handle->xrun[0] = false;

+  }

+  if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {

+    status |= RTAUDIO_INPUT_OVERFLOW;

+    handle->xrun[1] = false;

+  }

+  doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],

+                           stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );

+  if ( doStopStream == 2 ) {

+    this->abortStream();

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) goto unlock;

+

+  int result;

+  char *buffer;

+  int samples;

+  RtAudioFormat format;

+

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    // Setup parameters and do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[0] ) {

+      buffer = stream_.deviceBuffer;

+      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );

+      samples = stream_.bufferSize * stream_.nDeviceChannels[0];

+      format = stream_.deviceFormat[0];

+    }

+    else {

+      buffer = stream_.userBuffer[0];

+      samples = stream_.bufferSize * stream_.nUserChannels[0];

+      format = stream_.userFormat;

+    }

+

+    // Do byte swapping if necessary.

+    if ( stream_.doByteSwap[0] )

+      byteSwapBuffer( buffer, samples, format );

+

+    if ( stream_.mode == DUPLEX && handle->triggered == false ) {

+      int trig = 0;

+      ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );

+      result = write( handle->id[0], buffer, samples * formatBytes(format) );

+      trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;

+      ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );

+      handle->triggered = true;

+    }

+    else

+      // Write samples to device.

+      result = write( handle->id[0], buffer, samples * formatBytes(format) );

+

+    if ( result == -1 ) {

+      // We'll assume this is an underrun, though there isn't a

+      // specific means for determining that.

+      handle->xrun[0] = true;

+      errorText_ = "RtApiOss::callbackEvent: audio write error.";

+      error( RtError::WARNING );

+      // Continue on to input section.

+    }

+  }

+

+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {

+

+    // Setup parameters.

+    if ( stream_.doConvertBuffer[1] ) {

+      buffer = stream_.deviceBuffer;

+      samples = stream_.bufferSize * stream_.nDeviceChannels[1];

+      format = stream_.deviceFormat[1];

+    }

+    else {

+      buffer = stream_.userBuffer[1];

+      samples = stream_.bufferSize * stream_.nUserChannels[1];

+      format = stream_.userFormat;

+    }

+

+    // Read samples from device.

+    result = read( handle->id[1], buffer, samples * formatBytes(format) );

+

+    if ( result == -1 ) {

+      // We'll assume this is an overrun, though there isn't a

+      // specific means for determining that.

+      handle->xrun[1] = true;

+      errorText_ = "RtApiOss::callbackEvent: audio read error.";

+      error( RtError::WARNING );

+      goto unlock;

+    }

+

+    // Do byte swapping if necessary.

+    if ( stream_.doByteSwap[1] )

+      byteSwapBuffer( buffer, samples, format );

+

+    // Do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[1] )

+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );

+  }

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  RtApi::tickStreamTime();

+  if ( doStopStream == 1 ) this->stopStream();

+}

+

+extern "C" void *ossCallbackHandler( void *ptr )

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

+  RtApiOss *object = (RtApiOss *) info->object;

+  bool *isRunning = &info->isRunning;

+

+  while ( *isRunning == true ) {

+    pthread_testcancel();

+    object->callbackEvent();

+  }

+

+  pthread_exit( NULL );

+}

+

+//******************** End of __LINUX_OSS__ *********************//

+#endif

+

+

+// *************************************************** //

+//

+// Protected common (OS-independent) RtAudio methods.

+//

+// *************************************************** //

+

+// This method can be modified to control the behavior of error

+// message printing.

+void RtApi :: error( RtError::Type type )

+{

+  errorStream_.str(""); // clear the ostringstream

+  if ( type == RtError::WARNING && showWarnings_ == true )

+    std::cerr << '\n' << errorText_ << "\n\n";

+  else if ( type != RtError::WARNING )

+    throw( RtError( errorText_, type ) );

+}

+

+void RtApi :: verifyStream()

+{

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApi:: a stream is not open!";

+    error( RtError::INVALID_USE );

+  }

+}

+

+void RtApi :: clearStreamInfo()

+{

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+  stream_.sampleRate = 0;

+  stream_.bufferSize = 0;

+  stream_.nBuffers = 0;

+  stream_.userFormat = 0;

+  stream_.userInterleaved = true;

+  stream_.streamTime = 0.0;

+  stream_.apiHandle = 0;

+  stream_.deviceBuffer = 0;

+  stream_.callbackInfo.callback = 0;

+  stream_.callbackInfo.userData = 0;

+  stream_.callbackInfo.isRunning = false;

+  for ( int i=0; i<2; i++ ) {

+    stream_.device[i] = 11111;

+    stream_.doConvertBuffer[i] = false;

+    stream_.deviceInterleaved[i] = true;

+    stream_.doByteSwap[i] = false;

+    stream_.nUserChannels[i] = 0;

+    stream_.nDeviceChannels[i] = 0;

+    stream_.channelOffset[i] = 0;

+    stream_.deviceFormat[i] = 0;

+    stream_.latency[i] = 0;

+    stream_.userBuffer[i] = 0;

+    stream_.convertInfo[i].channels = 0;

+    stream_.convertInfo[i].inJump = 0;

+    stream_.convertInfo[i].outJump = 0;

+    stream_.convertInfo[i].inFormat = 0;

+    stream_.convertInfo[i].outFormat = 0;

+    stream_.convertInfo[i].inOffset.clear();

+    stream_.convertInfo[i].outOffset.clear();

+  }

+}

+

+unsigned int RtApi :: formatBytes( RtAudioFormat format )

+{

+  if ( format == RTAUDIO_SINT16 )

+    return 2;

+  else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||

+            format == RTAUDIO_FLOAT32 )

+    return 4;

+  else if ( format == RTAUDIO_FLOAT64 )

+    return 8;

+  else if ( format == RTAUDIO_SINT8 )

+    return 1;

+

+  errorText_ = "RtApi::formatBytes: undefined format.";

+  error( RtError::WARNING );

+

+  return 0;

+}

+

+void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )

+{

+  if ( mode == INPUT ) { // convert device to user buffer

+    stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];

+    stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];

+    stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];

+    stream_.convertInfo[mode].outFormat = stream_.userFormat;

+  }

+  else { // convert user to device buffer

+    stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];

+    stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];

+    stream_.convertInfo[mode].inFormat = stream_.userFormat;

+    stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];

+  }

+

+  if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )

+    stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;

+  else

+    stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;

+

+  // Set up the interleave/deinterleave offsets.

+  if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {

+    if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||

+         ( mode == INPUT && stream_.userInterleaved ) ) {

+      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {

+        stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );

+        stream_.convertInfo[mode].outOffset.push_back( k );

+        stream_.convertInfo[mode].inJump = 1;

+      }

+    }

+    else {

+      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {

+        stream_.convertInfo[mode].inOffset.push_back( k );

+        stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );

+        stream_.convertInfo[mode].outJump = 1;

+      }

+    }

+  }

+  else { // no (de)interleaving

+    if ( stream_.userInterleaved ) {

+      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {

+        stream_.convertInfo[mode].inOffset.push_back( k );

+        stream_.convertInfo[mode].outOffset.push_back( k );

+      }

+    }

+    else {

+      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {

+        stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );

+        stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );

+        stream_.convertInfo[mode].inJump = 1;

+        stream_.convertInfo[mode].outJump = 1;

+      }

+    }

+  }

+

+  // Add channel offset.

+  if ( firstChannel > 0 ) {

+    if ( stream_.deviceInterleaved[mode] ) {

+      if ( mode == OUTPUT ) {

+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )

+          stream_.convertInfo[mode].outOffset[k] += firstChannel;

+      }

+      else {

+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )

+          stream_.convertInfo[mode].inOffset[k] += firstChannel;

+      }

+    }

+    else {

+      if ( mode == OUTPUT ) {

+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )

+          stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );

+      }

+      else {

+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )

+          stream_.convertInfo[mode].inOffset[k] += ( firstChannel  * stream_.bufferSize );

+      }

+    }

+  }

+}

+

+void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )

+{

+  // This function does format conversion, input/output channel compensation, and

+  // data interleaving/deinterleaving.  24-bit integers are assumed to occupy

+  // the lower three bytes of a 32-bit integer.

+

+  // Clear our device buffer when in/out duplex device channels are different

+  if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&

+       ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )

+    memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );

+

+  int j;

+  if (info.outFormat == RTAUDIO_FLOAT64) {

+    Float64 scale;

+    Float64 *out = (Float64 *)outBuffer;

+

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      scale = 1.0 / 127.5;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      scale = 1.0 / 32767.5;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      Int32 *in = (Int32 *)inBuffer;

+      scale = 1.0 / 8388607.5;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      scale = 1.0 / 2147483647.5;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      // Channel compensation and/or (de)interleaving only.

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_FLOAT32) {

+    Float32 scale;

+    Float32 *out = (Float32 *)outBuffer;

+

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      scale = (Float32) ( 1.0 / 127.5 );

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      scale = (Float32) ( 1.0 / 32767.5 );

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      Int32 *in = (Int32 *)inBuffer;

+      scale = (Float32) ( 1.0 / 8388607.5 );

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      scale = (Float32) ( 1.0 / 2147483647.5 );

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      // Channel compensation and/or (de)interleaving only.

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_SINT32) {

+    Int32 *out = (Int32 *)outBuffer;

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 24;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 16;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) { // Hmmm ... we could just leave it in the lower 3 bytes

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 8;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      // Channel compensation and/or (de)interleaving only.

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_SINT24) {

+    Int32 *out = (Int32 *)outBuffer;

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 16;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 8;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      // Channel compensation and/or (de)interleaving only.

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] >>= 8;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_SINT16) {

+    Int16 *out = (Int16 *)outBuffer;

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 8;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      // Channel compensation and/or (de)interleaving only.

+      Int16 *in = (Int16 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_SINT8) {

+    signed char *out = (signed char *)outBuffer;

+    if (info.inFormat == RTAUDIO_SINT8) {

+      // Channel compensation and/or (de)interleaving only.

+      signed char *in = (signed char *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+}

+

+  //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }

+  //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }

+  //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }

+

+void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )

+{

+  register char val;

+  register char *ptr;

+

+  ptr = buffer;

+  if ( format == RTAUDIO_SINT16 ) {

+    for ( unsigned int i=0; i<samples; i++ ) {

+      // Swap 1st and 2nd bytes.

+      val = *(ptr);

+      *(ptr) = *(ptr+1);

+      *(ptr+1) = val;

+

+      // Increment 2 bytes.

+      ptr += 2;

+    }

+  }

+  else if ( format == RTAUDIO_SINT24 ||

+            format == RTAUDIO_SINT32 ||

+            format == RTAUDIO_FLOAT32 ) {

+    for ( unsigned int i=0; i<samples; i++ ) {

+      // Swap 1st and 4th bytes.

+      val = *(ptr);

+      *(ptr) = *(ptr+3);

+      *(ptr+3) = val;

+

+      // Swap 2nd and 3rd bytes.

+      ptr += 1;

+      val = *(ptr);

+      *(ptr) = *(ptr+1);

+      *(ptr+1) = val;

+

+      // Increment 3 more bytes.

+      ptr += 3;

+    }

+  }

+  else if ( format == RTAUDIO_FLOAT64 ) {

+    for ( unsigned int i=0; i<samples; i++ ) {

+      // Swap 1st and 8th bytes

+      val = *(ptr);

+      *(ptr) = *(ptr+7);

+      *(ptr+7) = val;

+

+      // Swap 2nd and 7th bytes

+      ptr += 1;

+      val = *(ptr);

+      *(ptr) = *(ptr+5);

+      *(ptr+5) = val;

+

+      // Swap 3rd and 6th bytes

+      ptr += 1;

+      val = *(ptr);

+      *(ptr) = *(ptr+3);

+      *(ptr+3) = val;

+

+      // Swap 4th and 5th bytes

+      ptr += 1;

+      val = *(ptr);

+      *(ptr) = *(ptr+1);

+      *(ptr+1) = val;

+

+      // Increment 5 more bytes.

+      ptr += 5;

+    }

+  }

+}

+

+  // Indentation settings for Vim and Emacs

+  //

+  // Local Variables:

+  // c-basic-offset: 2

+  // indent-tabs-mode: nil

+  // End:

+  //

+  // vim: et sts=2 sw=2

+

diff --git a/RtAudio.h b/RtAudio.h
new file mode 100644
index 0000000..6032368
--- /dev/null
+++ b/RtAudio.h
@@ -0,0 +1,980 @@
+/************************************************************************/
+/*! \class RtAudio
+    \brief Realtime audio i/o C++ classes.
+
+    RtAudio provides a common API (Application Programming Interface)
+    for realtime audio input/output across Linux (native ALSA, Jack,
+    and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
+    (DirectSound and ASIO) operating systems.
+
+    RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
+
+    RtAudio: realtime audio i/o C++ classes
+    Copyright (c) 2001-2011 Gary P. Scavone
+
+    Permission is hereby granted, free of charge, to any person
+    obtaining a copy of this software and associated documentation files
+    (the "Software"), to deal in the Software without restriction,
+    including without limitation the rights to use, copy, modify, merge,
+    publish, distribute, sublicense, and/or sell copies of the Software,
+    and to permit persons to whom the Software is furnished to do so,
+    subject to the following conditions:
+
+    The above copyright notice and this permission notice shall be
+    included in all copies or substantial portions of the Software.
+
+    Any person wishing to distribute modifications to the Software is
+    asked to send the modifications to the original developer so that
+    they can be incorporated into the canonical version.  This is,
+    however, not a binding provision of this license.
+
+    THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+    EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+    IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+    ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+    CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+    WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+/************************************************************************/
+
+/*!
+  \file RtAudio.h
+ */
+
+// RtAudio: Version 4.0.9
+
+#ifndef __RTAUDIO_H
+#define __RTAUDIO_H
+
+#include <string>
+#include <vector>
+#include "RtError.h"
+
+/*! \typedef typedef unsigned long RtAudioFormat;
+    \brief RtAudio data format type.
+
+    Support for signed integers and floats.  Audio data fed to/from an
+    RtAudio stream is assumed to ALWAYS be in host byte order.  The
+    internal routines will automatically take care of any necessary
+    byte-swapping between the host format and the soundcard.  Thus,
+    endian-ness is not a concern in the following format definitions.
+    Note that 24-bit data is expected to be encapsulated in a 32-bit
+    format.
+
+    - \e RTAUDIO_SINT8:   8-bit signed integer.
+    - \e RTAUDIO_SINT16:  16-bit signed integer.
+    - \e RTAUDIO_SINT24:  Lower 3 bytes of 32-bit signed integer.
+    - \e RTAUDIO_SINT32:  32-bit signed integer.
+    - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0.
+    - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.
+*/
+typedef unsigned long RtAudioFormat;
+static const RtAudioFormat RTAUDIO_SINT8 = 0x1;    // 8-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT16 = 0x2;   // 16-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT24 = 0x4;   // Lower 3 bytes of 32-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT32 = 0x8;   // 32-bit signed integer.
+static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0.
+static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0.
+
+/*! \typedef typedef unsigned long RtAudioStreamFlags;
+    \brief RtAudio stream option flags.
+
+    The following flags can be OR'ed together to allow a client to
+    make changes to the default stream behavior:
+
+    - \e RTAUDIO_NONINTERLEAVED:   Use non-interleaved buffers (default = interleaved).
+    - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
+    - \e RTAUDIO_HOG_DEVICE:       Attempt grab device for exclusive use.
+    - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
+
+    By default, RtAudio streams pass and receive audio data from the
+    client in an interleaved format.  By passing the
+    RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
+    data will instead be presented in non-interleaved buffers.  In
+    this case, each buffer argument in the RtAudioCallback function
+    will point to a single array of data, with \c nFrames samples for
+    each channel concatenated back-to-back.  For example, the first
+    sample of data for the second channel would be located at index \c
+    nFrames (assuming the \c buffer pointer was recast to the correct
+    data type for the stream).
+
+    Certain audio APIs offer a number of parameters that influence the
+    I/O latency of a stream.  By default, RtAudio will attempt to set
+    these parameters internally for robust (glitch-free) performance
+    (though some APIs, like Windows Direct Sound, make this difficult).
+    By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
+    function, internal stream settings will be influenced in an attempt
+    to minimize stream latency, though possibly at the expense of stream
+    performance.
+
+    If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
+    open the input and/or output stream device(s) for exclusive use.
+    Note that this is not possible with all supported audio APIs.
+
+    If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt 
+    to select realtime scheduling (round-robin) for the callback thread.
+
+    If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
+    open the "default" PCM device when using the ALSA API. Note that this
+    will override any specified input or output device id.
+*/
+typedef unsigned int RtAudioStreamFlags;
+static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1;    // Use non-interleaved buffers (default = interleaved).
+static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2;  // Attempt to set stream parameters for lowest possible latency.
+static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4;        // Attempt grab device and prevent use by others.
+static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
+static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).
+
+/*! \typedef typedef unsigned long RtAudioStreamStatus;
+    \brief RtAudio stream status (over- or underflow) flags.
+
+    Notification of a stream over- or underflow is indicated by a
+    non-zero stream \c status argument in the RtAudioCallback function.
+    The stream status can be one of the following two options,
+    depending on whether the stream is open for output and/or input:
+
+    - \e RTAUDIO_INPUT_OVERFLOW:   Input data was discarded because of an overflow condition at the driver.
+    - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound.
+*/
+typedef unsigned int RtAudioStreamStatus;
+static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1;    // Input data was discarded because of an overflow condition at the driver.
+static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2;  // The output buffer ran low, likely causing a gap in the output sound.
+
+//! RtAudio callback function prototype.
+/*!
+   All RtAudio clients must create a function of type RtAudioCallback
+   to read and/or write data from/to the audio stream.  When the
+   underlying audio system is ready for new input or output data, this
+   function will be invoked.
+
+   \param outputBuffer For output (or duplex) streams, the client
+          should write \c nFrames of audio sample frames into this
+          buffer.  This argument should be recast to the datatype
+          specified when the stream was opened.  For input-only
+          streams, this argument will be NULL.
+
+   \param inputBuffer For input (or duplex) streams, this buffer will
+          hold \c nFrames of input audio sample frames.  This
+          argument should be recast to the datatype specified when the
+          stream was opened.  For output-only streams, this argument
+          will be NULL.
+
+   \param nFrames The number of sample frames of input or output
+          data in the buffers.  The actual buffer size in bytes is
+          dependent on the data type and number of channels in use.
+
+   \param streamTime The number of seconds that have elapsed since the
+          stream was started.
+
+   \param status If non-zero, this argument indicates a data overflow
+          or underflow condition for the stream.  The particular
+          condition can be determined by comparison with the
+          RtAudioStreamStatus flags.
+
+   \param userData A pointer to optional data provided by the client
+          when opening the stream (default = NULL).
+
+   To continue normal stream operation, the RtAudioCallback function
+   should return a value of zero.  To stop the stream and drain the
+   output buffer, the function should return a value of one.  To abort
+   the stream immediately, the client should return a value of two.
+ */
+typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer,
+                                unsigned int nFrames,
+                                double streamTime,
+                                RtAudioStreamStatus status,
+                                void *userData );
+
+
+// **************************************************************** //
+//
+// RtAudio class declaration.
+//
+// RtAudio is a "controller" used to select an available audio i/o
+// interface.  It presents a common API for the user to call but all
+// functionality is implemented by the class RtApi and its
+// subclasses.  RtAudio creates an instance of an RtApi subclass
+// based on the user's API choice.  If no choice is made, RtAudio
+// attempts to make a "logical" API selection.
+//
+// **************************************************************** //
+
+class RtApi;
+
+class RtAudio
+{
+ public:
+
+  //! Audio API specifier arguments.
+  enum Api {
+    UNSPECIFIED,    /*!< Search for a working compiled API. */
+    LINUX_ALSA,     /*!< The Advanced Linux Sound Architecture API. */
+    LINUX_OSS,      /*!< The Linux Open Sound System API. */
+    UNIX_JACK,      /*!< The Jack Low-Latency Audio Server API. */
+    MACOSX_CORE,    /*!< Macintosh OS-X Core Audio API. */
+    WINDOWS_ASIO,   /*!< The Steinberg Audio Stream I/O API. */
+    WINDOWS_DS,     /*!< The Microsoft Direct Sound API. */
+    RTAUDIO_DUMMY   /*!< A compilable but non-functional API. */
+  };
+
+  //! The public device information structure for returning queried values.
+  struct DeviceInfo {
+    bool probed;                  /*!< true if the device capabilities were successfully probed. */
+    std::string name;             /*!< Character string device identifier. */
+    unsigned int outputChannels;  /*!< Maximum output channels supported by device. */
+    unsigned int inputChannels;   /*!< Maximum input channels supported by device. */
+    unsigned int duplexChannels;  /*!< Maximum simultaneous input/output channels supported by device. */
+    bool isDefaultOutput;         /*!< true if this is the default output device. */
+    bool isDefaultInput;          /*!< true if this is the default input device. */
+    std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
+    RtAudioFormat nativeFormats;  /*!< Bit mask of supported data formats. */
+
+    // Default constructor.
+    DeviceInfo()
+      :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
+       isDefaultOutput(false), isDefaultInput(false), nativeFormats(0) {}
+  };
+
+  //! The structure for specifying input or ouput stream parameters.
+  struct StreamParameters {
+    unsigned int deviceId;     /*!< Device index (0 to getDeviceCount() - 1). */
+    unsigned int nChannels;    /*!< Number of channels. */
+    unsigned int firstChannel; /*!< First channel index on device (default = 0). */
+
+    // Default constructor.
+    StreamParameters()
+      : deviceId(0), nChannels(0), firstChannel(0) {}
+  };
+
+  //! The structure for specifying stream options.
+  /*!
+    The following flags can be OR'ed together to allow a client to
+    make changes to the default stream behavior:
+
+    - \e RTAUDIO_NONINTERLEAVED:    Use non-interleaved buffers (default = interleaved).
+    - \e RTAUDIO_MINIMIZE_LATENCY:  Attempt to set stream parameters for lowest possible latency.
+    - \e RTAUDIO_HOG_DEVICE:        Attempt grab device for exclusive use.
+    - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
+    - \e RTAUDIO_ALSA_USE_DEFAULT:  Use the "default" PCM device (ALSA only).
+
+    By default, RtAudio streams pass and receive audio data from the
+    client in an interleaved format.  By passing the
+    RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
+    data will instead be presented in non-interleaved buffers.  In
+    this case, each buffer argument in the RtAudioCallback function
+    will point to a single array of data, with \c nFrames samples for
+    each channel concatenated back-to-back.  For example, the first
+    sample of data for the second channel would be located at index \c
+    nFrames (assuming the \c buffer pointer was recast to the correct
+    data type for the stream).
+
+    Certain audio APIs offer a number of parameters that influence the
+    I/O latency of a stream.  By default, RtAudio will attempt to set
+    these parameters internally for robust (glitch-free) performance
+    (though some APIs, like Windows Direct Sound, make this difficult).
+    By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
+    function, internal stream settings will be influenced in an attempt
+    to minimize stream latency, though possibly at the expense of stream
+    performance.
+
+    If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
+    open the input and/or output stream device(s) for exclusive use.
+    Note that this is not possible with all supported audio APIs.
+
+    If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt 
+    to select realtime scheduling (round-robin) for the callback thread.
+    The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
+    flag is set. It defines the thread's realtime priority.
+
+    If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
+    open the "default" PCM device when using the ALSA API. Note that this
+    will override any specified input or output device id.
+
+    The \c numberOfBuffers parameter can be used to control stream
+    latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
+    only.  A value of two is usually the smallest allowed.  Larger
+    numbers can potentially result in more robust stream performance,
+    though likely at the cost of stream latency.  The value set by the
+    user is replaced during execution of the RtAudio::openStream()
+    function by the value actually used by the system.
+
+    The \c streamName parameter can be used to set the client name
+    when using the Jack API.  By default, the client name is set to
+    RtApiJack.  However, if you wish to create multiple instances of
+    RtAudio with Jack, each instance must have a unique client name.
+  */
+  struct StreamOptions {
+    RtAudioStreamFlags flags;      /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
+    unsigned int numberOfBuffers;  /*!< Number of stream buffers. */
+    std::string streamName;        /*!< A stream name (currently used only in Jack). */
+    int priority;                  /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
+
+    // Default constructor.
+    StreamOptions()
+    : flags(0), numberOfBuffers(0), priority(0) {}
+  };
+
+  //! A static function to determine the available compiled audio APIs.
+  /*!
+    The values returned in the std::vector can be compared against
+    the enumerated list values.  Note that there can be more than one
+    API compiled for certain operating systems.
+  */
+  static void getCompiledApi( std::vector<RtAudio::Api> &apis ) throw();
+
+  //! The class constructor.
+  /*!
+    The constructor performs minor initialization tasks.  No exceptions
+    can be thrown.
+
+    If no API argument is specified and multiple API support has been
+    compiled, the default order of use is JACK, ALSA, OSS (Linux
+    systems) and ASIO, DS (Windows systems).
+  */
+  RtAudio( RtAudio::Api api=UNSPECIFIED ) throw();
+
+  //! The destructor.
+  /*!
+    If a stream is running or open, it will be stopped and closed
+    automatically.
+  */
+  ~RtAudio() throw();
+
+  //! Returns the audio API specifier for the current instance of RtAudio.
+  RtAudio::Api getCurrentApi( void ) throw();
+
+  //! A public function that queries for the number of audio devices available.
+  /*!
+    This function performs a system query of available devices each time it
+    is called, thus supporting devices connected \e after instantiation. If
+    a system error occurs during processing, a warning will be issued. 
+  */
+  unsigned int getDeviceCount( void ) throw();
+
+  //! Return an RtAudio::DeviceInfo structure for a specified device number.
+  /*!
+
+    Any device integer between 0 and getDeviceCount() - 1 is valid.
+    If an invalid argument is provided, an RtError (type = INVALID_USE)
+    will be thrown.  If a device is busy or otherwise unavailable, the
+    structure member "probed" will have a value of "false" and all
+    other members are undefined.  If the specified device is the
+    current default input or output device, the corresponding
+    "isDefault" member will have a value of "true".
+  */
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+
+  //! A function that returns the index of the default output device.
+  /*!
+    If the underlying audio API does not provide a "default
+    device", or if no devices are available, the return value will be
+    0.  Note that this is a valid device identifier and it is the
+    client's responsibility to verify that a device is available
+    before attempting to open a stream.
+  */
+  unsigned int getDefaultOutputDevice( void ) throw();
+
+  //! A function that returns the index of the default input device.
+  /*!
+    If the underlying audio API does not provide a "default
+    device", or if no devices are available, the return value will be
+    0.  Note that this is a valid device identifier and it is the
+    client's responsibility to verify that a device is available
+    before attempting to open a stream.
+  */
+  unsigned int getDefaultInputDevice( void ) throw();
+
+  //! A public function for opening a stream with the specified parameters.
+  /*!
+    An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be
+    opened with the specified parameters or an error occurs during
+    processing.  An RtError (type = INVALID_USE) is thrown if any
+    invalid device ID or channel number parameters are specified.
+
+    \param outputParameters Specifies output stream parameters to use
+           when opening a stream, including a device ID, number of channels,
+           and starting channel number.  For input-only streams, this
+           argument should be NULL.  The device ID is an index value between
+           0 and getDeviceCount() - 1.
+    \param inputParameters Specifies input stream parameters to use
+           when opening a stream, including a device ID, number of channels,
+           and starting channel number.  For output-only streams, this
+           argument should be NULL.  The device ID is an index value between
+           0 and getDeviceCount() - 1.
+    \param format An RtAudioFormat specifying the desired sample data format.
+    \param sampleRate The desired sample rate (sample frames per second).
+    \param *bufferFrames A pointer to a value indicating the desired
+           internal buffer size in sample frames.  The actual value
+           used by the device is returned via the same pointer.  A
+           value of zero can be specified, in which case the lowest
+           allowable value is determined.
+    \param callback A client-defined function that will be invoked
+           when input data is available and/or output data is needed.
+    \param userData An optional pointer to data that can be accessed
+           from within the callback function.
+    \param options An optional pointer to a structure containing various
+           global stream options, including a list of OR'ed RtAudioStreamFlags
+           and a suggested number of stream buffers that can be used to 
+           control stream latency.  More buffers typically result in more
+           robust performance, though at a cost of greater latency.  If a
+           value of zero is specified, a system-specific median value is
+           chosen.  If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the
+           lowest allowable value is used.  The actual value used is
+           returned via the structure argument.  The parameter is API dependent.
+  */
+  void openStream( RtAudio::StreamParameters *outputParameters,
+                   RtAudio::StreamParameters *inputParameters,
+                   RtAudioFormat format, unsigned int sampleRate,
+                   unsigned int *bufferFrames, RtAudioCallback callback,
+                   void *userData = NULL, RtAudio::StreamOptions *options = NULL );
+
+  //! A function that closes a stream and frees any associated stream memory.
+  /*!
+    If a stream is not open, this function issues a warning and
+    returns (no exception is thrown).
+  */
+  void closeStream( void ) throw();
+
+  //! A function that starts a stream.
+  /*!
+    An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
+    during processing.  An RtError (type = INVALID_USE) is thrown if a
+    stream is not open.  A warning is issued if the stream is already
+    running.
+  */
+  void startStream( void );
+
+  //! Stop a stream, allowing any samples remaining in the output queue to be played.
+  /*!
+    An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
+    during processing.  An RtError (type = INVALID_USE) is thrown if a
+    stream is not open.  A warning is issued if the stream is already
+    stopped.
+  */
+  void stopStream( void );
+
+  //! Stop a stream, discarding any samples remaining in the input/output queue.
+  /*!
+    An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
+    during processing.  An RtError (type = INVALID_USE) is thrown if a
+    stream is not open.  A warning is issued if the stream is already
+    stopped.
+  */
+  void abortStream( void );
+
+  //! Returns true if a stream is open and false if not.
+  bool isStreamOpen( void ) const throw();
+
+  //! Returns true if the stream is running and false if it is stopped or not open.
+  bool isStreamRunning( void ) const throw();
+
+  //! Returns the number of elapsed seconds since the stream was started.
+  /*!
+    If a stream is not open, an RtError (type = INVALID_USE) will be thrown.
+  */
+  double getStreamTime( void );
+
+  //! Returns the internal stream latency in sample frames.
+  /*!
+    The stream latency refers to delay in audio input and/or output
+    caused by internal buffering by the audio system and/or hardware.
+    For duplex streams, the returned value will represent the sum of
+    the input and output latencies.  If a stream is not open, an
+    RtError (type = INVALID_USE) will be thrown.  If the API does not
+    report latency, the return value will be zero.
+  */
+  long getStreamLatency( void );
+
+ //! Returns actual sample rate in use by the stream.
+ /*!
+   On some systems, the sample rate used may be slightly different
+   than that specified in the stream parameters.  If a stream is not
+   open, an RtError (type = INVALID_USE) will be thrown.
+ */
+  unsigned int getStreamSampleRate( void );
+
+  //! Specify whether warning messages should be printed to stderr.
+  void showWarnings( bool value = true ) throw();
+
+ protected:
+
+  void openRtApi( RtAudio::Api api );
+  RtApi *rtapi_;
+};
+
+// Operating system dependent thread functionality.
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
+  #include <windows.h>
+  #include <process.h>
+
+  typedef unsigned long ThreadHandle;
+  typedef CRITICAL_SECTION StreamMutex;
+
+#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+  // Using pthread library for various flavors of unix.
+  #include <pthread.h>
+
+  typedef pthread_t ThreadHandle;
+  typedef pthread_mutex_t StreamMutex;
+
+#else // Setup for "dummy" behavior
+
+  #define __RTAUDIO_DUMMY__
+  typedef int ThreadHandle;
+  typedef int StreamMutex;
+
+#endif
+
+// This global structure type is used to pass callback information
+// between the private RtAudio stream structure and global callback
+// handling functions.
+struct CallbackInfo {
+  void *object;    // Used as a "this" pointer.
+  ThreadHandle thread;
+  void *callback;
+  void *userData;
+  void *apiInfo;   // void pointer for API specific callback information
+  bool isRunning;
+
+  // Default constructor.
+  CallbackInfo()
+    :object(0), callback(0), userData(0), apiInfo(0), isRunning(false) {}
+};
+
+// **************************************************************** //
+//
+// RtApi class declaration.
+//
+// Subclasses of RtApi contain all API- and OS-specific code necessary
+// to fully implement the RtAudio API.
+//
+// Note that RtApi is an abstract base class and cannot be
+// explicitly instantiated.  The class RtAudio will create an
+// instance of an RtApi subclass (RtApiOss, RtApiAlsa,
+// RtApiJack, RtApiCore, RtApiAl, RtApiDs, or RtApiAsio).
+//
+// **************************************************************** //
+
+#if defined( HAVE_GETTIMEOFDAY )
+  #include <sys/time.h>
+#endif
+
+#include <sstream>
+
+class RtApi
+{
+public:
+
+  RtApi();
+  virtual ~RtApi();
+  virtual RtAudio::Api getCurrentApi( void ) = 0;
+  virtual unsigned int getDeviceCount( void ) = 0;
+  virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0;
+  virtual unsigned int getDefaultInputDevice( void );
+  virtual unsigned int getDefaultOutputDevice( void );
+  void openStream( RtAudio::StreamParameters *outputParameters,
+                   RtAudio::StreamParameters *inputParameters,
+                   RtAudioFormat format, unsigned int sampleRate,
+                   unsigned int *bufferFrames, RtAudioCallback callback,
+                   void *userData, RtAudio::StreamOptions *options );
+  virtual void closeStream( void );
+  virtual void startStream( void ) = 0;
+  virtual void stopStream( void ) = 0;
+  virtual void abortStream( void ) = 0;
+  long getStreamLatency( void );
+  unsigned int getStreamSampleRate( void );
+  virtual double getStreamTime( void );
+  bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; };
+  bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; };
+  void showWarnings( bool value ) { showWarnings_ = value; };
+
+
+protected:
+
+  static const unsigned int MAX_SAMPLE_RATES;
+  static const unsigned int SAMPLE_RATES[];
+
+  enum { FAILURE, SUCCESS };
+
+  enum StreamState {
+    STREAM_STOPPED,
+    STREAM_RUNNING,
+    STREAM_CLOSED = -50
+  };
+
+  enum StreamMode {
+    OUTPUT,
+    INPUT,
+    DUPLEX,
+    UNINITIALIZED = -75
+  };
+
+  // A protected structure used for buffer conversion.
+  struct ConvertInfo {
+    int channels;
+    int inJump, outJump;
+    RtAudioFormat inFormat, outFormat;
+    std::vector<int> inOffset;
+    std::vector<int> outOffset;
+  };
+
+  // A protected structure for audio streams.
+  struct RtApiStream {
+    unsigned int device[2];    // Playback and record, respectively.
+    void *apiHandle;           // void pointer for API specific stream handle information
+    StreamMode mode;           // OUTPUT, INPUT, or DUPLEX.
+    StreamState state;         // STOPPED, RUNNING, or CLOSED
+    char *userBuffer[2];       // Playback and record, respectively.
+    char *deviceBuffer;
+    bool doConvertBuffer[2];   // Playback and record, respectively.
+    bool userInterleaved;
+    bool deviceInterleaved[2]; // Playback and record, respectively.
+    bool doByteSwap[2];        // Playback and record, respectively.
+    unsigned int sampleRate;
+    unsigned int bufferSize;
+    unsigned int nBuffers;
+    unsigned int nUserChannels[2];    // Playback and record, respectively.
+    unsigned int nDeviceChannels[2];  // Playback and record channels, respectively.
+    unsigned int channelOffset[2];    // Playback and record, respectively.
+    unsigned long latency[2];         // Playback and record, respectively.
+    RtAudioFormat userFormat;
+    RtAudioFormat deviceFormat[2];    // Playback and record, respectively.
+    StreamMutex mutex;
+    CallbackInfo callbackInfo;
+    ConvertInfo convertInfo[2];
+    double streamTime;         // Number of elapsed seconds since the stream started.
+
+#if defined(HAVE_GETTIMEOFDAY)
+    struct timeval lastTickTimestamp;
+#endif
+
+    RtApiStream()
+      :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; }
+  };
+
+  typedef signed short Int16;
+  typedef signed int Int32;
+  typedef float Float32;
+  typedef double Float64;
+
+  std::ostringstream errorStream_;
+  std::string errorText_;
+  bool showWarnings_;
+  RtApiStream stream_;
+
+  /*!
+    Protected, api-specific method that attempts to open a device
+    with the given parameters.  This function MUST be implemented by
+    all subclasses.  If an error is encountered during the probe, a
+    "warning" message is reported and FAILURE is returned. A
+    successful probe is indicated by a return value of SUCCESS.
+  */
+  virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                                unsigned int firstChannel, unsigned int sampleRate,
+                                RtAudioFormat format, unsigned int *bufferSize,
+                                RtAudio::StreamOptions *options );
+
+  //! A protected function used to increment the stream time.
+  void tickStreamTime( void );
+
+  //! Protected common method to clear an RtApiStream structure.
+  void clearStreamInfo();
+
+  /*!
+    Protected common method that throws an RtError (type =
+    INVALID_USE) if a stream is not open.
+  */
+  void verifyStream( void );
+
+  //! Protected common error method to allow global control over error handling.
+  void error( RtError::Type type );
+
+  /*!
+    Protected method used to perform format, channel number, and/or interleaving
+    conversions between the user and device buffers.
+  */
+  void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info );
+
+  //! Protected common method used to perform byte-swapping on buffers.
+  void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format );
+
+  //! Protected common method that returns the number of bytes for a given format.
+  unsigned int formatBytes( RtAudioFormat format );
+
+  //! Protected common method that sets up the parameters for buffer conversion.
+  void setConvertInfo( StreamMode mode, unsigned int firstChannel );
+};
+
+// **************************************************************** //
+//
+// Inline RtAudio definitions.
+//
+// **************************************************************** //
+
+inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); }
+inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); }
+inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }
+inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); }
+inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); }
+inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); }
+inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); }
+inline void RtAudio :: stopStream( void )  { return rtapi_->stopStream(); }
+inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }
+inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); }
+inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); }
+inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
+inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); };
+inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
+inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
+
+// RtApi Subclass prototypes.
+
+#if defined(__MACOSX_CORE__)
+
+#include <CoreAudio/AudioHardware.h>
+
+class RtApiCore: public RtApi
+{
+public:
+
+  RtApiCore();
+  ~RtApiCore();
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  unsigned int getDefaultOutputDevice( void );
+  unsigned int getDefaultInputDevice( void );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+  long getStreamLatency( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  bool callbackEvent( AudioDeviceID deviceId,
+                      const AudioBufferList *inBufferList,
+                      const AudioBufferList *outBufferList );
+
+  private:
+
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+  static const char* getErrorCode( OSStatus code );
+};
+
+#endif
+
+#if defined(__UNIX_JACK__)
+
+class RtApiJack: public RtApi
+{
+public:
+
+  RtApiJack();
+  ~RtApiJack();
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+  long getStreamLatency( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  bool callbackEvent( unsigned long nframes );
+
+  private:
+
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__WINDOWS_ASIO__)
+
+class RtApiAsio: public RtApi
+{
+public:
+
+  RtApiAsio();
+  ~RtApiAsio();
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+  long getStreamLatency( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  bool callbackEvent( long bufferIndex );
+
+  private:
+
+  std::vector<RtAudio::DeviceInfo> devices_;
+  void saveDeviceInfo( void );
+  bool coInitialized_;
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__WINDOWS_DS__)
+
+class RtApiDs: public RtApi
+{
+public:
+
+  RtApiDs();
+  ~RtApiDs();
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; };
+  unsigned int getDeviceCount( void );
+  unsigned int getDefaultOutputDevice( void );
+  unsigned int getDefaultInputDevice( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+  long getStreamLatency( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  void callbackEvent( void );
+
+  private:
+
+  bool coInitialized_;
+  bool buffersRolling;
+  long duplexPrerollBytes;
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__LINUX_ALSA__)
+
+class RtApiAlsa: public RtApi
+{
+public:
+
+  RtApiAlsa();
+  ~RtApiAlsa();
+  RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  void callbackEvent( void );
+
+  private:
+
+  std::vector<RtAudio::DeviceInfo> devices_;
+  void saveDeviceInfo( void );
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__LINUX_OSS__)
+
+class RtApiOss: public RtApi
+{
+public:
+
+  RtApiOss();
+  ~RtApiOss();
+  RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  void callbackEvent( void );
+
+  private:
+
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__RTAUDIO_DUMMY__)
+
+class RtApiDummy: public RtApi
+{
+public:
+
+  RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::WARNING ); };
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; };
+  unsigned int getDeviceCount( void ) { return 0; };
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; return info; };
+  void closeStream( void ) {};
+  void startStream( void ) {};
+  void stopStream( void ) {};
+  void abortStream( void ) {};
+
+  private:
+
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options ) { return false; };
+};
+
+#endif
+
+#endif
+
+// Indentation settings for Vim and Emacs
+//
+// Local Variables:
+// c-basic-offset: 2
+// indent-tabs-mode: nil
+// End:
+//
+// vim: et sts=2 sw=2
diff --git a/RtError.h b/RtError.h
new file mode 100644
index 0000000..a64f434
--- /dev/null
+++ b/RtError.h
@@ -0,0 +1,60 @@
+/************************************************************************/
+/*! \class RtError
+    \brief Exception handling class for RtAudio & RtMidi.
+
+    The RtError class is quite simple but it does allow errors to be
+    "caught" by RtError::Type. See the RtAudio and RtMidi
+    documentation to know which methods can throw an RtError.
+
+*/
+/************************************************************************/
+
+#ifndef RTERROR_H
+#define RTERROR_H
+
+#include <exception>
+#include <iostream>
+#include <string>
+
+class RtError : public std::exception
+{
+ public:
+  //! Defined RtError types.
+  enum Type {
+    WARNING,           /*!< A non-critical error. */
+    DEBUG_WARNING,     /*!< A non-critical error which might be useful for debugging. */
+    UNSPECIFIED,       /*!< The default, unspecified error type. */
+    NO_DEVICES_FOUND,  /*!< No devices found on system. */
+    INVALID_DEVICE,    /*!< An invalid device ID was specified. */
+    MEMORY_ERROR,      /*!< An error occured during memory allocation. */
+    INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
+    INVALID_USE,       /*!< The function was called incorrectly. */
+    DRIVER_ERROR,      /*!< A system driver error occured. */
+    SYSTEM_ERROR,      /*!< A system error occured. */
+    THREAD_ERROR       /*!< A thread error occured. */
+  };
+
+  //! The constructor.
+  RtError( const std::string& message, Type type = RtError::UNSPECIFIED ) throw() : message_(message), type_(type) {}
+ 
+  //! The destructor.
+  virtual ~RtError( void ) throw() {}
+
+  //! Prints thrown error message to stderr.
+  virtual void printMessage( void ) const throw() { std::cerr << '\n' << message_ << "\n\n"; }
+
+  //! Returns the thrown error message type.
+  virtual const Type& getType(void) const throw() { return type_; }
+
+  //! Returns the thrown error message string.
+  virtual const std::string& getMessage(void) const throw() { return message_; }
+
+  //! Returns the thrown error message as a c-style string.
+  virtual const char* what( void ) const throw() { return message_.c_str(); }
+
+ protected:
+  std::string message_;
+  Type type_;
+};
+
+#endif
diff --git a/beat2.wav b/beat2.wav
new file mode 100644
index 0000000..c71cee6
--- /dev/null
+++ b/beat2.wav
diff --git a/main.cpp b/main.cpp
new file mode 100755
index 0000000..b80b512
--- /dev/null
+++ b/main.cpp
@@ -0,0 +1,33 @@
+#include "maximilian.h"
+
+maxiOsc myCounter,mySwitchableOsc;//
+int CurrentCount;//
+double myOscOutput,myFilteredOutput;//
+double myEnvelopeData[6] = {500,0,1000,500,0,500};//this data will be used to make an envelope. Value and time to value in ms.
+maxiEnvelope myEnvelope;
+maxiFilter myFilter;
+
+void setup() {//some inits
+	myEnvelope.amplitude=myEnvelopeData[0]; //initialise the envelope
+}
+
+void play(double *output) {
+	
+	CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value.
+	
+	if (CurrentCount<5)//simple if statement
+		
+		myOscOutput=mySwitchableOsc.square(CurrentCount*100);
+	
+	else if (CurrentCount>=5)//and the 'else' bit.
+		
+		myOscOutput=mySwitchableOsc.saw(CurrentCount*50);//one osc object can produce whichever waveform you want. 
+	
+	if (CurrentCount==1) 
+		
+		myEnvelope.trigger(0,myEnvelopeData[0]); //trigger the envelope
+	
+	myFilteredOutput=myFilter.lores(myOscOutput,(myEnvelope.line(6, myEnvelopeData)),10);//lores takes an audio input, a frequency and a resonance factor (1-100)
+	
+	*output=myFilteredOutput;//point me at your speakers and fire.
+}
diff --git a/maximilian.cpp b/maximilian.cpp
new file mode 100644
index 0000000..ac2ccba
--- /dev/null
+++ b/maximilian.cpp
@@ -0,0 +1,803 @@
+
+/*
+ *  maximilian.cpp
+ *  platform independent synthesis library using portaudio or rtaudio
+ *
+ *  Created by Mick Grierson on 29/12/2009.
+ *  Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved.
+ *	Thanks to the Goldsmiths Creative Computing Team.
+ *	Special thanks to Arturo Castro for the PortAudio implementation.
+ *
+ *	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.
+ *
+ *	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.
+ *
+ */
+
+#include "maximilian.h"
+#include "math.h"
+
+
+//int channels=2;
+//int samplerate=44100; 
+//int buffersize=1024;
+float chandiv= 1;
+
+int maxiSettings::sampleRate = 44100;
+int maxiSettings::channels = 2;
+int maxiSettings::bufferSize = 1024;
+
+double sineBuffer[514]={0,0.012268,0.024536,0.036804,0.049042,0.06131,0.073547,0.085785,0.097992,0.1102,0.12241,0.13455,0.1467,0.15884,0.17093,0.18301,0.19507,0.20709,0.21909,0.23105,0.24295,0.25485,0.26669,0.2785,0.29025,0.30197,0.31366,0.32529,0.33685,0.34839,0.35986,0.37128,0.38266,0.39395,0.40521,0.41641,0.42752,0.4386,0.44958,0.46051,0.47137,0.48215,0.49286,0.50351,0.51407,0.52457,0.53497,0.54529,0.55554,0.5657,0.57578,0.58575,0.59567,0.60547,0.6152,0.62482,0.63437,0.6438,0.65314,0.66238,0.67151,0.68057,0.68951,0.69833,0.70706,0.7157,0.72421,0.7326,0.74091,0.74908,0.75717,0.76514,0.77298,0.7807,0.7883,0.79581,0.80316,0.81042,0.81754,0.82455,0.83142,0.8382,0.84482,0.85132,0.8577,0.86392,0.87006,0.87604,0.88187,0.8876,0.89319,0.89862,0.90396,0.90912,0.91415,0.91907,0.92383,0.92847,0.93295,0.93729,0.9415,0.94556,0.94949,0.95325,0.95691,0.96039,0.96375,0.96692,0.97,0.9729,0.97565,0.97827,0.98074,0.98306,0.98523,0.98724,0.98914,0.99084,0.99243,0.99387,0.99515,0.99628,0.99725,0.99808,0.99875,0.99927,0.99966,0.99988,0.99997,0.99988,0.99966,0.99927,0.99875,0.99808,0.99725,0.99628,0.99515,0.99387,0.99243,0.99084,0.98914,0.98724,0.98523,0.98306,0.98074,0.97827,0.97565,0.9729,0.97,0.96692,0.96375,0.96039,0.95691,0.95325,0.94949,0.94556,0.9415,0.93729,0.93295,0.92847,0.92383,0.91907,0.91415,0.90912,0.90396,0.89862,0.89319,0.8876,0.88187,0.87604,0.87006,0.86392,0.8577,0.85132,0.84482,0.8382,0.83142,0.82455,0.81754,0.81042,0.80316,0.79581,0.7883,0.7807,0.77298,0.76514,0.75717,0.74908,0.74091,0.7326,0.72421,0.7157,0.70706,0.69833,0.68951,0.68057,0.67151,0.66238,0.65314,0.6438,0.63437,0.62482,0.6152,0.60547,0.59567,0.58575,0.57578,0.5657,0.55554,0.54529,0.53497,0.52457,0.51407,0.50351,0.49286,0.48215,0.47137,0.46051,0.44958,0.4386,0.42752,0.41641,0.40521,0.39395,0.38266,0.37128,0.35986,0.34839,0.33685,0.32529,0.31366,0.30197,0.29025,0.2785,0.26669,0.25485,0.24295,0.23105,0.21909,0.20709,0.19507,0.18301,0.17093,0.15884,0.1467,0.13455,0.12241,0.1102,0.097992,0.085785,0.073547,0.06131,0.049042,0.036804,0.024536,0.012268,0,-0.012268,-0.024536,-0.036804,-0.049042,-0.06131,-0.073547,-0.085785,-0.097992,-0.1102,-0.12241,-0.13455,-0.1467,-0.15884,-0.17093,-0.18301,-0.19507,-0.20709,-0.21909,-0.23105,-0.24295,-0.25485,-0.26669,-0.2785,-0.29025,-0.30197,-0.31366,-0.32529,-0.33685,-0.34839,-0.35986,-0.37128,-0.38266,-0.39395,-0.40521,-0.41641,-0.42752,-0.4386,-0.44958,-0.46051,-0.47137,-0.48215,-0.49286,-0.50351,-0.51407,-0.52457,-0.53497,-0.54529,-0.55554,-0.5657,-0.57578,-0.58575,-0.59567,-0.60547,-0.6152,-0.62482,-0.63437,-0.6438,-0.65314,-0.66238,-0.67151,-0.68057,-0.68951,-0.69833,-0.70706,-0.7157,-0.72421,-0.7326,-0.74091,-0.74908,-0.75717,-0.76514,-0.77298,-0.7807,-0.7883,-0.79581,-0.80316,-0.81042,-0.81754,-0.82455,-0.83142,-0.8382,-0.84482,-0.85132,-0.8577,-0.86392,-0.87006,-0.87604,-0.88187,-0.8876,-0.89319,-0.89862,-0.90396,-0.90912,-0.91415,-0.91907,-0.92383,-0.92847,-0.93295,-0.93729,-0.9415,-0.94556,-0.94949,-0.95325,-0.95691,-0.96039,-0.96375,-0.96692,-0.97,-0.9729,-0.97565,-0.97827,-0.98074,-0.98306,-0.98523,-0.98724,-0.98914,-0.99084,-0.99243,-0.99387,-0.99515,-0.99628,-0.99725,-0.99808,-0.99875,-0.99927,-0.99966,-0.99988,-0.99997,-0.99988,-0.99966,-0.99927,-0.99875,-0.99808,-0.99725,-0.99628,-0.99515,-0.99387,-0.99243,-0.99084,-0.98914,-0.98724,-0.98523,-0.98306,-0.98074,-0.97827,-0.97565,-0.9729,-0.97,-0.96692,-0.96375,-0.96039,-0.95691,-0.95325,-0.94949,-0.94556,-0.9415,-0.93729,-0.93295,-0.92847,-0.92383,-0.91907,-0.91415,-0.90912,-0.90396,-0.89862,-0.89319,-0.8876,-0.88187,-0.87604,-0.87006,-0.86392,-0.8577,-0.85132,-0.84482,-0.8382,-0.83142,-0.82455,-0.81754,-0.81042,-0.80316,-0.79581,-0.7883,-0.7807,-0.77298,-0.76514,-0.75717,-0.74908,-0.74091,-0.7326,-0.72421,-0.7157,-0.70706,-0.69833,-0.68951,-0.68057,-0.67151,-0.66238,-0.65314,-0.6438,-0.63437,-0.62482,-0.6152,-0.60547,-0.59567,-0.58575,-0.57578,-0.5657,-0.55554,-0.54529,-0.53497,-0.52457,-0.51407,-0.50351,-0.49286,-0.48215,-0.47137,-0.46051,-0.44958,-0.4386,-0.42752,-0.41641,-0.40521,-0.39395,-0.38266,-0.37128,-0.35986,-0.34839,-0.33685,-0.32529,-0.31366,-0.30197,-0.29025,-0.2785,-0.26669,-0.25485,-0.24295,-0.23105,-0.21909,-0.20709,-0.19507,-0.18301,-0.17093,-0.15884,-0.1467,-0.13455,-0.12241,-0.1102,-0.097992,-0.085785,-0.073547,-0.06131,-0.049042,-0.036804,-0.024536,-0.012268,0,0.012268
+};
+
+void setup();//use this to do any initialisation if you want.
+
+void play(double *channels);//run dac! 
+
+maxiOsc::maxiOsc(){
+	phase = 0.0;
+//	memset(phases,0,500);
+//	memset(freqs,0,500);
+}
+
+double maxiOsc::noise() {
+	//always the same unless you seed it.
+	float r = rand()/(float)RAND_MAX;
+	output=r*2-1;
+	return(output);
+}
+
+double maxiOsc::sinewave(double frequency) {
+	output=sin (phase*(TWOPI));
+	if ( phase >= 1.0 ) phase -= 1.0;
+	phase += (1./(maxiSettings::sampleRate/(frequency)));
+	return(output);
+	
+}
+
+double maxiOsc::sinebuf4(double frequency) {
+	double remainder;
+	double a,b,c,d,a1,a2,a3;
+	phase += 512./(maxiSettings::sampleRate/(frequency));
+	if ( phase >= 511 ) phase -=512;
+	remainder = phase - floor(phase);
+	
+	if (phase==0) {
+		a=sineBuffer[(long) 512];
+		b=sineBuffer[(long) phase];
+		c=sineBuffer[(long) phase+1];
+		d=sineBuffer[(long) phase+2];
+		
+	} else {
+		a=sineBuffer[(long) phase-1];
+		b=sineBuffer[(long) phase];
+		c=sineBuffer[(long) phase+1];
+		d=sineBuffer[(long) phase+2];
+		
+	}
+	
+	a1 = 0.5f * (c - a);
+	a2 = a - 2.5 * b + 2.f * c - 0.5f * d;
+	a3 = 0.5f * (d - a) + 1.5f * (b - c);
+	output = double (((a3 * remainder + a2) * remainder + a1) * remainder + b);
+	return(output);
+}
+
+double maxiOsc::sinebuf(double frequency) {
+	double remainder;
+ 	phase += 512./(maxiSettings::sampleRate/(frequency*chandiv));
+	if ( phase >= 511 ) phase -=512;
+	remainder = phase - floor(phase);
+	output = (double) ((1-remainder) * sineBuffer[1+ (long) phase] + remainder * sineBuffer[2+(long) phase]);
+	return(output);
+}
+
+double maxiOsc::coswave(double frequency) {
+	output=cos (phase*(TWOPI));
+	if ( phase >= 1.0 ) phase -= 1.0;
+	phase += (1./(maxiSettings::sampleRate/(frequency)));
+	return(output);
+	
+}
+
+double maxiOsc::phasor(double frequency) {
+	output=phase;
+	if ( phase >= 1.0 ) phase -= 1.0;
+	phase += (1./(maxiSettings::sampleRate/(frequency)));
+	return(output);
+} 
+
+double maxiOsc::square(double frequency) {
+	if (phase<0.5) output=-1;
+	if (phase>0.5) output=1;
+	if ( phase >= 1.0 ) phase -= 1.0;
+	phase += (1./(maxiSettings::sampleRate/(frequency)));
+	return(output);
+}
+
+double maxiOsc::pulse(double frequency, double duty) {
+	if (duty<0.) duty=0;
+	if (duty>1.) duty=1;
+	if ( phase >= 1.0 ) phase -= 1.0;
+	phase += (1./(maxiSettings::sampleRate/(frequency)));
+	if (phase<duty) output=-1.;
+	if (phase>duty) output=1.;
+	return(output);
+}
+
+double maxiOsc::phasor(double frequency, double startphase, double endphase) {
+	output=phase;
+	if (phase<startphase) {
+		phase=startphase;
+	}
+	if ( phase >= endphase ) phase = startphase;
+	phase += ((endphase-startphase)/(maxiSettings::sampleRate/(frequency)));
+	return(output);
+}
+
+
+double maxiOsc::saw(double frequency) {
+	
+	output=phase;
+	if ( phase >= 1.0 ) phase -= 2.0;
+	phase += (1./(maxiSettings::sampleRate/(frequency)));
+	return(output);
+	
+} 
+
+double maxiOsc::triangle(double frequency, double phase) {
+	output=tri*2;
+	if ( phase >= 1.0 ) phase -= 1.0;
+	phase += (1./(maxiSettings::sampleRate/(frequency*chandiv)));
+	if (phase <= 0.5 ) {
+		tri = phase;
+	} else {	
+		tri =(1-phase);
+	}
+	return(output);
+	
+} 
+
+//I like this.
+double maxiEnvelope::line(int numberofsegments,double segments[1000]) {
+	period=2./(segments[valindex+1]*0.004);
+	nextval=segments[valindex+2];
+	currentval=segments[valindex];
+	if (currentval-amplitude > 0.0000001 && valindex < numberofsegments) {
+		amplitude += ((currentval-startval)/(maxiSettings::sampleRate/period));
+	} else if (currentval-amplitude < -0.0000001 && valindex < numberofsegments) {
+		amplitude -= (((currentval-startval)*(-1))/(maxiSettings::sampleRate/period));
+	} else if (valindex >numberofsegments-1) {
+		valindex=numberofsegments-2;
+	} else {
+		valindex=valindex+2;
+		startval=currentval;
+	}
+	output=amplitude;
+	return(output);	
+}
+
+//and this
+void maxiEnvelope::trigger(int index, double amp) {
+	valindex=index;
+	amplitude=amp;
+	
+}
+
+//and this
+double maxiDelayline::dl(double input, int size, double feedback)  {
+	if ( phase >=size ) phase = 0;
+	memory[phase]=(memory[phase]*feedback)+(input*feedback)*0.5;
+	phase+=1;
+	output=memory[phase+1];
+	return(output);
+	
+}
+
+double maxiDelayline::dl(double input, int size, double feedback, int position)  {
+	if ( phase >=size ) phase = 0;
+	if ( position >=size ) position = 0;
+	memory[phase]=(memory[phase]*feedback)+(input*feedback)*chandiv;
+	phase+=1;
+	output=memory[position];
+	return(output);
+	
+}
+
+//I particularly like these. cutoff between 0 and 1
+double maxiFilter::lopass(double input, double cutoff) {
+	output=outputs[0] + cutoff*(input-outputs[0]);
+	outputs[0]=output;
+	return(output);
+}
+//as above
+double maxiFilter::hipass(double input, double cutoff) {
+	output=input-(outputs[0] + cutoff*(input-outputs[0]));
+	outputs[0]=output;
+	return(output);
+}
+//awesome. cuttof is freq in hz. res is between 1 and whatever. Watch out!
+double maxiFilter::lores(double input,double cutoff1, double resonance) {
+	cutoff=cutoff1*0.5;
+	if (cutoff<10) cutoff=10;
+	if (cutoff>(maxiSettings::sampleRate*0.25)) cutoff=(maxiSettings::sampleRate*0.25);
+	if (resonance<1.) resonance = 1.;
+	z=cos(TWOPI*cutoff/maxiSettings::sampleRate);
+	c=2-2*z;
+	double r=(sqrt(2.0)*sqrt(-pow((z-1.0),3.0))+resonance*(z-1))/(resonance*(z-1));
+	x=x+(input-y)*c;
+	y=y+x;
+	x=x*r;
+	output=y;
+	return(output);
+}
+
+//working hires filter
+double maxiFilter::hires(double input,double cutoff1, double resonance) {
+	cutoff=cutoff1*0.5;
+	if (cutoff<10) cutoff=10;
+	if (cutoff>(maxiSettings::sampleRate*0.25)) cutoff=(maxiSettings::sampleRate*0.25);
+	if (resonance<1.) resonance = 1.;
+	z=cos(TWOPI*cutoff/maxiSettings::sampleRate);
+	c=2-2*z;
+	double r=(sqrt(2.0)*sqrt(-pow((z-1.0),3.0))+resonance*(z-1))/(resonance*(z-1));
+	x=x+(input-y)*c;
+	y=y+x;
+	x=x*r;
+	output=input-y;
+	return(output);
+}
+
+//This works a bit. Needs attention.
+double maxiFilter::bandpass(double input,double cutoff1, double resonance) {
+	cutoff=cutoff1;
+	if (cutoff>(maxiSettings::sampleRate*0.5)) cutoff=(maxiSettings::sampleRate*0.5);
+	if (resonance>=1.) resonance=0.999999;
+	z=cos(TWOPI*cutoff/maxiSettings::sampleRate);
+	inputs[0] = (1-resonance)*(sqrt(resonance*(resonance-4.0*pow(z,2.0)+2.0)+1));
+	inputs[1] = 2*z*resonance;
+	inputs[2] = pow((resonance*-1),2);
+	
+	output=inputs[0]*input+inputs[1]*outputs[1]+inputs[2]*outputs[2];
+	outputs[2]=outputs[1];
+	outputs[1]=output;
+	return(output);
+}
+
+//stereo bus
+double *maxiMix::stereo(double input,double two[2],double x) {
+	if (x>1) x=1;
+	if (x<0) x=0;
+	two[0]=input*sqrt(1.0-x);
+	two[1]=input*sqrt(x);
+	return(two);
+} 
+
+//quad bus
+double *maxiMix::quad(double input,double four[4],double x,double y) {
+	if (x>1) x=1;
+	if (x<0) x=0;
+	if (y>1) y=1;
+	if (y<0) y=0;
+	four[0]=input*sqrt((1.0-x)*y);
+	four[1]=input*sqrt((1.0-x)*(1.0-y));
+	four[2]=input*sqrt(x*y);
+	four[3]=input*sqrt(x*(1.0-y));
+	return(four);
+}
+
+//ambisonic bus
+double *maxiMix::ambisonic(double input,double eight[8],double x,double y,double z) {
+	if (x>1) x=1;
+	if (x<0) x=0;
+	if (y>1) y=1;
+	if (y<0) y=0;
+	if (z>1) y=1;
+	if (z<0) y=0;
+	eight[0]=input*(sqrt((1.0-x)*y)*1.0-z);
+	eight[1]=input*(sqrt((1.0-x)*(1.0-y))*1.0-z);
+	eight[2]=input*(sqrt(x*y)*1.0-z);
+	eight[3]=input*(sqrt(x*(1.0-y))*1.0-z);
+	eight[4]=input*(sqrt((1.0-x)*y)*z);
+	eight[5]=input*(sqrt((1.0-x)*(1.0-y))*z);
+	eight[6]=input*sqrt((x*y)*z);
+	eight[7]=input*sqrt((x*(1.0-y))*z);
+	return(eight);
+}
+
+
+bool maxiSample::load(string fileName) {
+	myPath = fileName;
+	return read();
+}
+
+void maxiSample::trigger() {
+	position = 0;
+}
+
+bool maxiSample::read()
+{
+	bool result;
+	ifstream inFile( myPath.c_str(), ios::in | ios::binary);
+	result = inFile;
+	if (inFile) {
+		bool datafound = false;
+		inFile.seekg(4, ios::beg);
+		inFile.read( (char*) &myChunkSize, 4 ); // read the ChunkSize
+		
+		inFile.seekg(16, ios::beg);
+		inFile.read( (char*) &mySubChunk1Size, 4 ); // read the SubChunk1Size
+		
+		//inFile.seekg(20, ios::beg);
+		inFile.read( (char*) &myFormat, sizeof(short) ); // read the file format.  This should be 1 for PCM
+		
+		//inFile.seekg(22, ios::beg);
+		inFile.read( (char*) &myChannels, sizeof(short) ); // read the # of channels (1 or 2)
+		
+		//inFile.seekg(24, ios::beg);
+		inFile.read( (char*) &mySampleRate, sizeof(int) ); // read the samplerate
+		
+		//inFile.seekg(28, ios::beg);
+		inFile.read( (char*) &myByteRate, sizeof(int) ); // read the byterate
+		
+		//inFile.seekg(32, ios::beg);
+		inFile.read( (char*) &myBlockAlign, sizeof(short) ); // read the blockalign
+		
+		//inFile.seekg(34, ios::beg);
+		inFile.read( (char*) &myBitsPerSample, sizeof(short) ); // read the bitspersample
+		
+		//ignore any extra chunks
+		char chunkID[5]="";
+		chunkID[4] = 0;
+		int filePos = 36;
+		while(!datafound && !inFile.eof()) {
+			inFile.seekg(filePos, ios::beg);
+			inFile.read((char*) &chunkID, sizeof(char) * 4);
+			inFile.seekg(filePos + 4, ios::beg);
+			inFile.read( (char*) &myDataSize, sizeof(int) ); // read the size of the data
+			filePos += 8;
+			if (strcmp(chunkID,"data") == 0) {
+				datafound = true;
+			}else{
+				filePos += myDataSize;
+			}
+		}
+		
+		// read the data chunk
+		myData = (char*) malloc(myDataSize * sizeof(char));
+		inFile.seekg(filePos, ios::beg);
+		inFile.read(myData, myDataSize);
+		length=myDataSize*(0.5/myChannels);
+		inFile.close(); // close the input file
+		
+	}else {
+		cout << "ERROR: Could not load sample: " <<myPath << endl;
+	}
+	
+	
+	return result; // this should probably be something more descriptive
+}
+
+
+double maxiSample::play() {
+//	long length=myDataSize*(1./myChannels);
+	double remainder;
+	short* buffer = (short *)myData;
+	position=(position+1);
+	remainder = position - (long) position;
+	if ((long) position>length) position=0;
+	output = (double) ((1-remainder) * buffer[1+ (long) position] + remainder * buffer[2+(long) position])/32767;//linear interpolation
+	return(output);
+}
+
+double maxiSample::playOnce() {
+//	long length=myDataSize*(0.5/myChannels);
+	double remainder;
+	short* buffer = (short *)myData;
+	position=(position+1);
+	remainder = position - (long) position;
+	if ((long) position<length)
+		output = (double) ((1-remainder) * buffer[1+ (long) position] + remainder * buffer[2+(long) position])/32767;//linear interpolation
+	else 
+		output=0;
+
+	return(output);
+}
+
+double maxiSample::play(double speed) {
+	double remainder;
+	long a,b;
+//	long length=myDataSize*0.5;	
+	short* buffer = (short *)myData;
+	position=position+((speed*chandiv*myChannels)/(maxiSettings::sampleRate/mySampleRate));
+	if (speed >=0) {
+		
+		if ((long) position>=length-1) position=1;
+		remainder = position - floor(position);
+		if (position+1<length) {
+			a=position+1;
+			
+		}
+		else {
+			a=length-1;
+		}
+		if (position+2<length)
+		{
+		b=position+2;
+		}
+		else {
+		b=length-1;
+		}
+		
+		output = (double) ((1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation
+} else {
+		if ((long) position<0) position=length;
+		remainder = position - floor(position);
+		if (position-1>=0) {
+			a=position-1;
+				
+			}
+			else {
+				a=0;
+			}
+	if (position-2>=0) {
+		b=position-2;
+			}
+			else {
+				b=0;
+			}
+		output = (double) ((-1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation
+	}	
+	return(output);
+}
+
+double maxiSample::play(double frequency, double start, double end) {
+	return play(frequency, start, end, position);
+}
+
+double maxiSample::play(double frequency, double start, double end, double &pos) {
+	double remainder;
+	//	long length=myDataSize;
+
+	if (end>=length) end=length-1;
+	long a,b;
+	short* buffer = (short *)myData;
+	if (frequency >0.) {
+		if (pos<start) {
+			pos=start;
+		}
+		
+		if ( pos >= end ) pos = start;
+		pos += ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv)));
+		remainder = pos - floor(pos);
+		long posl = floor(pos);
+		if (posl+1<length) {
+			a=posl+1;
+			
+		}
+		else {
+			a=posl-1;
+		}
+		if (posl+2<length) {
+			b=posl+2;
+		}
+		else {
+			b=length-1;
+		}
+
+		output = (double) ((1-remainder) * buffer[a] +
+						   remainder * buffer[b])/32767;//linear interpolation
+	} else {
+		frequency=frequency-(frequency+frequency);
+		if ( pos <= start ) pos = end;
+		pos -= ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv)));
+		remainder = pos - floor(pos);
+		long posl = floor(pos);
+		if (posl-1>=0) {
+			a=posl-1;
+		}
+		else {
+			a=0;
+		}
+		if (posl-2>=0) {
+			b=posl-2;
+		}
+		else {
+			b=0;
+		}		
+		output = (double) ((-1-remainder) * buffer[a] +
+						   remainder * buffer[b])/32767;//linear interpolation
+		
+	}
+	
+	return(output);
+}
+
+
+//better cubic inerpolation. Cobbled together from various (pd externals, yehar, other places).
+double maxiSample::play4(double frequency, double start, double end) {
+	double remainder;
+	double a,b,c,d,a1,a2,a3;
+	short* buffer = (short*)myData;
+	if (frequency >0.) {
+		if (position<start) {
+			position=start;
+		}
+		if ( position >= end ) position = start;
+		position += ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv)));
+		remainder = position - floor(position);
+		if (position>0) {
+			a=buffer[(int)(floor(position))-1];
+
+		} else {
+			a=buffer[0];
+			
+		}
+		
+		b=buffer[(long) position];
+		if (position<end-2) {
+			c=buffer[(long) position+1];
+
+		} else {
+			c=buffer[0];
+
+		}
+		if (position<end-3) {
+			d=buffer[(long) position+2];
+
+		} else {
+			d=buffer[0];
+		}
+		a1 = 0.5f * (c - a);
+		a2 = a - 2.5 * b + 2.f * c - 0.5f * d;
+		a3 = 0.5f * (d - a) + 1.5f * (b - c);
+		output = (double) (((a3 * remainder + a2) * remainder + a1) * remainder + b) / 32767;
+		
+	} else {
+		frequency=frequency-(frequency+frequency);
+		if ( position <= start ) position = end;
+		position -= ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv)));
+		remainder = position - floor(position);
+		if (position>start && position < end-1) {
+			a=buffer[(long) position+1];
+			
+		} else {
+			a=buffer[0];
+			
+		}
+		
+		b=buffer[(long) position];
+		if (position>start) {
+			c=buffer[(long) position-1];
+			
+		} else {
+			c=buffer[0];
+			
+		}
+		if (position>start+1) {
+			d=buffer[(long) position-2];
+			
+		} else {
+			d=buffer[0];
+		}
+		a1 = 0.5f * (c - a);
+		a2 = a - 2.5 * b + 2.f * c - 0.5f * d;
+		a3 = 0.5f * (d - a) + 1.5f * (b - c);
+		output = (double) (((a3 * remainder + a2) * -remainder + a1) * -remainder + b) / 32767;
+		
+	}
+	
+	return(output);
+}
+
+double maxiSample::bufferPlay(unsigned char &bufferin,long length) {
+	double remainder;
+	short* buffer = (short *)bufferin;
+	position=(position+1);
+	remainder = position - (long) position;
+	if ((long) position>length) position=0;
+	output = (double) ((1-remainder) * buffer[1+ (long) position] + remainder * buffer[2+(long) position])/32767;//linear interpolation
+	return(output);
+}
+
+double maxiSample::bufferPlay(unsigned char &bufferin,double speed,long length) {
+	double remainder;
+	long a,b;
+	short* buffer = (short *)bufferin;
+	position=position+((speed*chandiv*myChannels)/(maxiSettings::sampleRate/mySampleRate));
+	if (speed >=0) {
+		
+		if ((long) position>=length-1) position=1;
+		remainder = position - floor(position);
+		if (position+1<length) {
+			a=position+1;
+			
+		}
+		else {
+			a=length-1;
+		}
+		if (position+2<length)
+		{
+			b=position+2;
+		}
+		else {
+			b=length-1;
+		}
+		
+		output = (double) ((1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation
+	} else {
+		if ((long) position<0) position=length;
+		remainder = position - floor(position);
+		if (position-1>=0) {
+			a=position-1;
+			
+		}
+		else {
+			a=0;
+		}
+		if (position-2>=0) {
+			b=position-2;
+		}
+		else {
+			b=0;
+		}
+		output = (double) ((-1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation
+	}	
+	return(output);
+}
+
+double maxiSample::bufferPlay(unsigned char &bufferin,double frequency, double start, double end) {
+	double remainder;
+	length=end;
+	long a,b;
+	short* buffer = (short *)bufferin;
+	if (frequency >0.) {
+		if (position<start) {
+			position=start;
+		}
+		
+		if ( position >= end ) position = start;
+		position += ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv)));
+		remainder = position - floor(position);
+		long pos = floor(position);
+		if (pos+1<length) {
+			a=pos+1;
+			
+		}
+		else {
+			a=pos-1;
+		}
+		if (pos+2<length) {
+			b=pos+2;
+		}
+		else {
+			b=length-1;
+		}
+		
+		output = (double) ((1-remainder) * buffer[a] +
+						   remainder * buffer[b])/32767;//linear interpolation
+	} else {
+		frequency=frequency-(frequency+frequency);
+		if ( position <= start ) position = end;
+		position -= ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv)));
+		remainder = position - floor(position);
+		long pos = floor(position);
+		if (pos-1>=0) {
+			a=pos-1;
+		}
+		else {
+			a=0;
+		}
+		if (pos-2>=0) {
+			b=pos-2;
+		}
+		else {
+			b=0;
+		}		
+		output = (double) ((-1-remainder) * buffer[a] +
+						   remainder * buffer[b])/32767;//linear interpolation
+		
+	}
+	
+	return(output);
+}
+
+//better cubic inerpolation. Cobbled together from various (pd externals, yehar, other places).
+double maxiSample::bufferPlay4(unsigned char &bufferin,double frequency, double start, double end) {
+	double remainder;
+	double a,b,c,d,a1,a2,a3;
+	short* buffer = (short*)bufferin;
+	if (frequency >0.) {
+		if (position<start) {
+			position=start;
+		}
+		if ( position >= end ) position = start;
+		position += ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv)));
+		remainder = position - floor(position);
+		if (position>0) {
+			a=buffer[(int)(floor(position))-1];
+			
+		} else {
+			a=buffer[0];
+			
+		}
+		
+		b=buffer[(long) position];
+		if (position<end-2) {
+			c=buffer[(long) position+1];
+			
+		} else {
+			c=buffer[0];
+			
+		}
+		if (position<end-3) {
+			d=buffer[(long) position+2];
+			
+		} else {
+			d=buffer[0];
+		}
+		a1 = 0.5f * (c - a);
+		a2 = a - 2.5 * b + 2.f * c - 0.5f * d;
+		a3 = 0.5f * (d - a) + 1.5f * (b - c);
+		output = (double) (((a3 * remainder + a2) * remainder + a1) * remainder + b) / 32767;
+		
+	} else {
+		frequency=frequency-(frequency+frequency);
+		if ( position <= start ) position = end;
+		position -= ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv)));
+		remainder = position - floor(position);
+		if (position>start && position < end-1) {
+			a=buffer[(long) position+1];
+			
+		} else {
+			a=buffer[0];
+			
+		}
+		
+		b=buffer[(long) position];
+		if (position>start) {
+			c=buffer[(long) position-1];
+			
+		} else {
+			c=buffer[0];
+			
+		}
+		if (position>start+1) {
+			d=buffer[(long) position-2];
+			
+		} else {
+			d=buffer[0];
+		}
+		a1 = 0.5f * (c - a);
+		a2 = a - 2.5 * b + 2.f * c - 0.5f * d;
+		a3 = 0.5f * (d - a) + 1.5f * (b - c);
+		output = (double) (((a3 * remainder + a2) * -remainder + a1) * -remainder + b) / 32767;
+		
+	}
+	
+	return(output);
+}
+
+
+void maxiSample::getLength() {
+	length=myDataSize*0.5;	
+}
+
+
diff --git a/maximilian.h b/maximilian.h
new file mode 100755
index 0000000..3304bac
--- /dev/null
+++ b/maximilian.h
@@ -0,0 +1,316 @@
+/*
+ *  maximilian.h
+ *  platform independent synthesis library using portaudio or rtaudio
+ *
+ *  Created by Mick Grierson on 29/12/2009.
+ *  Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved.
+ *	Thanks to the Goldsmiths Creative Computing Team.
+ *	Special thanks to Arturo Castro for the PortAudio implementation.
+ * 
+ *	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.
+ *
+ *	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.
+ *
+ */
+
+#ifndef MAXIMILIAN_H
+#define MAXIMILIAN_H
+
+//#define MAXIMILIAN_PORTAUDIO
+#define MAXIMILIAN_RT_AUDIO
+
+//#include "ofMain.h"
+//#include "ofUtils.h"
+
+
+#include <iostream>
+#include <fstream>
+#include <string>
+#include <cstdlib>
+#include "math.h"
+
+using namespace std;
+#define TWOPI (3.14159*2)
+
+class maxiSettings {
+public:
+	static int sampleRate;
+	static int channels;
+	static int bufferSize;
+	static void setup(int initSampleRate, int initChannels, int initBufferSize) {
+		maxiSettings::sampleRate = initSampleRate;
+		maxiSettings::channels = initChannels;
+		maxiSettings::bufferSize = initBufferSize;
+	}
+};
+
+
+class maxiOsc {
+	
+	double frequency;
+	double phase;
+	double startphase;
+	double endphase;
+	double output;
+	double tri;
+
+	
+public:
+	maxiOsc();
+	double sinewave(double frequency);
+	double coswave(double frequency);
+	double phasor(double frequency);
+	double phasor(double frequency, double startphase, double endphase);
+	double saw(double frequency);
+	double triangle(double frequency,double phase);
+	double square(double frequency);
+	double pulse(double frequency, double duty);
+	double noise();
+	double sinebuf(double frequency);
+	double sinebuf4(double frequency);
+	
+};
+
+
+class maxiEnvelope {
+	
+	double period;
+	double output;
+	double startval;
+	double currentval;
+	double nextval;
+	
+public:	
+	double line(int numberofsegments,double segments[100]);
+	void trigger(int index,double amp);
+	int valindex;
+	double amplitude;
+	
+};
+
+
+class maxiDelayline {
+	double frequency;
+	int phase;
+	double startphase;
+	double endphase;
+	double output;
+	double memory[88200];
+	
+public:
+	double dl(double input, int size, double feedback);
+	double dl(double input, int size, double feedback, int position);
+
+	
+};
+
+
+class maxiFilter { 	
+	double gain;
+	double input;
+	double output;
+	double inputs[10];
+	double outputs[10];
+	double cutoff1;
+	double x;//speed
+	double y;//pos
+	double z;//pole
+	double c;//filter coefficient
+public:
+	double cutoff;
+	double resonance;
+	double lores(double input,double cutoff1, double resonance);
+	double hires(double input,double cutoff1, double resonance);
+	double bandpass(double input,double cutoff1, double resonance);
+	double lopass(double input,double cutoff);
+	double hipass(double input,double cutoff);
+	
+};
+
+class maxiMix  {
+	double input;
+	double two[2];
+	double four[4];
+	double eight[8];
+public:
+	double x;
+	double y;
+	double z;
+	double *stereo(double input,double two[2],double x);
+	double *quad(double input,double four[4], double x,double y);
+	double *ambisonic(double input,double eight[8],double x,double y, double z);
+	
+};
+
+
+class maxiSample  {
+	
+private:
+	string 	myPath;
+	int 	myChunkSize;
+	int	mySubChunk1Size;
+	short 	myFormat;
+	int   	myByteRate;
+	short 	myBlockAlign;
+	short 	myBitsPerSample;
+	int	myDataSize;
+	double position;
+	double speed;
+	double output;
+	
+public:
+	short 	myChannels;
+	int   	mySampleRate;
+	long length;
+	void getLength();
+
+	char* 	myData;
+	
+	// get/set for the Path property
+
+	~maxiSample()
+	{
+		delete myData;
+		myChunkSize = NULL;
+		mySubChunk1Size = NULL;
+		myFormat = NULL;
+		myChannels = NULL;
+		mySampleRate = NULL;
+		myByteRate = NULL;
+		myBlockAlign = NULL;
+		myBitsPerSample = NULL;
+		myDataSize = NULL;
+	}
+	
+//	maxiSample();
+
+	bool load(string fileName);
+	
+	void trigger();
+	
+	// read a wav file into this class
+	bool read();
+	
+	double play();
+	
+	double playOnce();
+	
+	double play(double speed);
+	
+	double play(double frequency, double start, double end, double &pos);
+	
+	double play(double frequency, double start, double end);
+		
+	double play4(double frequency, double start, double end);
+	
+	double bufferPlay(unsigned char &bufferin,long length);
+	
+	double bufferPlay(unsigned char &bufferin,double speed,long length);
+	
+	double bufferPlay(unsigned char &bufferin,double frequency, double start, double end);
+	
+	double bufferPlay4(unsigned char &bufferin,double frequency, double start, double end);
+	
+	//int open(char *filename) {
+	//		// open the wav file
+	//		char *path = new char[50];
+	//		strcpy(path, filename);
+	//		sample *myWav = new sample(path);
+	//		return 0;
+	//	}	
+	//		// print a summary of the wav file
+	//	int info() {
+	//		char *summary = myWav->getSummary();
+	//		printf("Summary:\n%s", summary);	
+	//		return 0;
+	//	}
+	//	int savefile() {	
+	//		// write the summary back out
+	//		strcpy(path, "testout.wav");
+	//		myWav->setPath(path);
+	//		myWav->save();
+	//		return 0:
+	//	}
+	//	int close() {
+	//		// collect the garbage
+	//		delete summary;
+	//		delete path;
+	//		delete myWav;
+	//		
+	//		return 0;
+	//	}		
+	// write out the wav file
+	bool save()
+	{
+		fstream myFile (myPath.c_str(), ios::out | ios::binary);
+		
+		// write the wav file per the wav file format
+		myFile.seekp (0, ios::beg); 
+		myFile.write ("RIFF", 4);
+		myFile.write ((char*) &myChunkSize, 4);
+		myFile.write ("WAVE", 4);
+		myFile.write ("fmt ", 4);
+		myFile.write ((char*) &mySubChunk1Size, 4);
+		myFile.write ((char*) &myFormat, 2);
+		myFile.write ((char*) &myChannels, 2);
+		myFile.write ((char*) &mySampleRate, 4);
+		myFile.write ((char*) &myByteRate, 4);
+		myFile.write ((char*) &myBlockAlign, 2);
+		myFile.write ((char*) &myBitsPerSample, 2);
+		myFile.write ("data", 4);
+		myFile.write ((char*) &myDataSize, 4);
+		myFile.write (myData, myDataSize);
+		
+		return true;
+	}
+	
+	// return a printable summary of the wav file
+	char *getSummary()
+	{
+		char *summary = new char[250];
+		sprintf(summary, " Format: %d\n Channels: %d\n SampleRate: %d\n ByteRate: %d\n BlockAlign: %d\n BitsPerSample: %d\n DataSize: %d\n", myFormat, myChannels, mySampleRate, myByteRate, myBlockAlign, myBitsPerSample, myDataSize);
+		std::cout << myDataSize;
+		return summary;
+	}
+};
+
+
+class maxiMap {
+public:
+	static double inline linlin(double val, double inMin, double inMax, double outMin, double outMax) {
+		return ((val - inMin) / (inMax - inMin) * (outMax - outMin)) + outMin;
+	}
+	
+	static double inline linexp(double val, double inMin, double inMax, double outMin, double outMax) {
+		//clipping
+		val = max(min(val, inMax), inMin);
+		return pow((outMax / outMin), (val - inMin) / (inMax - inMin)) * outMin;
+	}
+
+	static double inline explin(double val, double inMin, double inMax, double outMin, double outMax) {
+		//clipping
+		val = max(min(val, inMax), inMin);
+		return (log(val/inMin) / log(inMax/inMin) * (outMax - outMin)) + outMin;
+	}
+	
+};
+
+
+#endif
diff --git a/maximilianTest.xcodeproj/project.pbxproj b/maximilianTest.xcodeproj/project.pbxproj
new file mode 100755
index 0000000..7aa34fd
--- /dev/null
+++ b/maximilianTest.xcodeproj/project.pbxproj
@@ -0,0 +1,286 @@
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+}
diff --git a/maximilian_examples/.DS_Store b/maximilian_examples/.DS_Store
new file mode 100644
index 0000000..5008ddf
--- /dev/null
+++ b/maximilian_examples/.DS_Store
diff --git a/maximilian_examples/1.TestTone.cpp b/maximilian_examples/1.TestTone.cpp
new file mode 100755
index 0000000..77783b4
--- /dev/null
+++ b/maximilian_examples/1.TestTone.cpp
@@ -0,0 +1,14 @@
+#include "maximilian.h"
+
+maxiOsc mySine;//let's create an oscillator and give it a name.
+
+void setup() {//some inits
+	//nothing to go here this time	
+}
+
+void play(double *output) {//this is where the magic happens. Very slow magic.
+	
+	*output=mySine.sinewave(440);//simple as that! 
+	
+}
+
diff --git a/maximilian_examples/10.Filters.cpp b/maximilian_examples/10.Filters.cpp
new file mode 100755
index 0000000..b80b512
--- /dev/null
+++ b/maximilian_examples/10.Filters.cpp
@@ -0,0 +1,33 @@
+#include "maximilian.h"
+
+maxiOsc myCounter,mySwitchableOsc;//
+int CurrentCount;//
+double myOscOutput,myFilteredOutput;//
+double myEnvelopeData[6] = {500,0,1000,500,0,500};//this data will be used to make an envelope. Value and time to value in ms.
+maxiEnvelope myEnvelope;
+maxiFilter myFilter;
+
+void setup() {//some inits
+	myEnvelope.amplitude=myEnvelopeData[0]; //initialise the envelope
+}
+
+void play(double *output) {
+	
+	CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value.
+	
+	if (CurrentCount<5)//simple if statement
+		
+		myOscOutput=mySwitchableOsc.square(CurrentCount*100);
+	
+	else if (CurrentCount>=5)//and the 'else' bit.
+		
+		myOscOutput=mySwitchableOsc.saw(CurrentCount*50);//one osc object can produce whichever waveform you want. 
+	
+	if (CurrentCount==1) 
+		
+		myEnvelope.trigger(0,myEnvelopeData[0]); //trigger the envelope
+	
+	myFilteredOutput=myFilter.lores(myOscOutput,(myEnvelope.line(6, myEnvelopeData)),10);//lores takes an audio input, a frequency and a resonance factor (1-100)
+	
+	*output=myFilteredOutput;//point me at your speakers and fire.
+}
diff --git a/maximilian_examples/11.Mixing.cpp b/maximilian_examples/11.Mixing.cpp
new file mode 100755
index 0000000..6fd7a34
--- /dev/null
+++ b/maximilian_examples/11.Mixing.cpp
@@ -0,0 +1,40 @@
+#include "maximilian.h"
+
+maxiOsc myCounter,mySwitchableOsc,myAutoPanner;//
+int CurrentCount;//
+double myOscOutput,myFilteredOutput,myPanPosition;//
+double myStereoOutput[2];// we need an output for each channel. 
+double myEnvelopeData[6] = {500,0,1000,500,0,500};//this data will be used to make an envelope. Value and time to value in ms.
+maxiEnvelope myEnvelope;
+maxiFilter myFilter;
+maxiMix myOutputs;//this is the stereo mixer channel.
+
+void setup() {//some inits
+	myEnvelope.amplitude=myEnvelopeData[0]; //initialise the envelope
+}
+
+void play(double *output) {
+	
+	CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value.
+	
+	if (CurrentCount<5)//simple if statement
+		
+		myOscOutput=mySwitchableOsc.square(CurrentCount*100);
+	
+	else if (CurrentCount>=5)//and the 'else' bit.
+		
+		myOscOutput=mySwitchableOsc.saw(CurrentCount*50);//one osc object can produce whichever waveform you want. 
+	
+	if (CurrentCount==1) 
+		
+		myEnvelope.trigger(0,myEnvelopeData[0]); //trigger the envelope
+	
+	myFilteredOutput=myFilter.lores(myOscOutput,(myEnvelope.line(6, myEnvelopeData)),10);//lores takes an audio input, a frequency and a resonance factor (1-100)
+	
+	myPanPosition=myAutoPanner.sinewave(1);
+	
+	myOutputs.stereo(myFilteredOutput,myStereoOutput,myPanPosition);//Stereo, Quad or 8 Channel. Specify the input to be mixed, the output[numberofchannels], and the pan (0-1,equal power).	
+	output[0]=myStereoOutput[0];//When working with mixing, you need to specify the outputs explicityly
+	output[1]=myStereoOutput[1];//
+	
+}
diff --git a/maximilian_examples/12.SamplePlayer.cpp b/maximilian_examples/12.SamplePlayer.cpp
new file mode 100755
index 0000000..2706f0b
--- /dev/null
+++ b/maximilian_examples/12.SamplePlayer.cpp
@@ -0,0 +1,21 @@
+#include "maximilian.h"
+
+maxiSample beats; //We give our sample a name. It's called beats this time. We could have loads of them, but they have to have different names.
+
+void setup() {//some inits
+	
+	beats.load("/Users/username/somewhere/Maximilian/beat2.wav");//load in your samples. Provide the full path to a wav file.
+	printf("Summary:\n%s", beats.getSummary());//get info on samples if you like.
+	
+}
+
+void play(double *output) {//this is where the magic happens. Very slow magic.
+	
+	*output=beats.play();//just play the file. Looping is default for all play functions.
+	//	*output=beats.play(0.69);//play the file with a speed setting. 1. is normal speed.
+	//	*output=beats.play(0.5,0,44100);//linear interpolationplay with a frequency input, start point and end point. Useful for syncing.
+	//	*output=beats.play4(0.5,0,44100);//cubic interpolation play with a frequency input, start point and end point. Useful for syncing.
+	
+	
+}
+
diff --git a/maximilian_examples/2.TwoTones.cpp b/maximilian_examples/2.TwoTones.cpp
new file mode 100755
index 0000000..4e22994
--- /dev/null
+++ b/maximilian_examples/2.TwoTones.cpp
@@ -0,0 +1,14 @@
+#include "maximilian.h"
+
+maxiOsc mySine,myOtherSine;//Two oscillators with names.
+
+void setup() {//some inits
+	//nothing to go here this time	
+}
+
+void play(double *output) {//this is where the magic happens. Very slow magic.
+	
+	*output=mySine.sinewave(440)+myOtherSine.sinewave(441);//these two sines will beat together. They're now a bit too loud though..
+	
+}
+
diff --git a/maximilian_examples/3.AM1.cpp b/maximilian_examples/3.AM1.cpp
new file mode 100755
index 0000000..ec96c7f
--- /dev/null
+++ b/maximilian_examples/3.AM1.cpp
@@ -0,0 +1,15 @@
+#include "maximilian.h"
+
+//This shows how to use maximilian to do basic amplitude modulation
+
+maxiOsc mySine,myOtherSine;//Two oscillators. They can be called anything. They can be any of the available waveforms. These ones will be sinewaves
+
+void setup() {//some inits
+	//nothing to go here this time	
+}
+
+void play(double *output) {
+	
+	*output=mySine.sinewave(440)*myOtherSine.sinewave(10);
+	
+}
diff --git a/maximilian_examples/4.AM2.cpp b/maximilian_examples/4.AM2.cpp
new file mode 100755
index 0000000..4e783d3
--- /dev/null
+++ b/maximilian_examples/4.AM2.cpp
@@ -0,0 +1,16 @@
+#include "maximilian.h"
+
+//This shows how to use maximilian to do basic amplitude modulation
+
+maxiOsc mySine,myOtherSine,myPhasor;//Three oscillators. They can be called anything. They can be any of the available waveforms.
+
+
+void setup() {//some inits
+	//nothing to go here this time	
+}
+
+void play(double *output) {
+	
+	*output=mySine.sinewave(440)*myOtherSine.sinewave(myPhasor.phasor(0.1,0,440));
+	
+}
diff --git a/maximilian_examples/5.FM1.cpp b/maximilian_examples/5.FM1.cpp
new file mode 100755
index 0000000..c734dcf
--- /dev/null
+++ b/maximilian_examples/5.FM1.cpp
@@ -0,0 +1,14 @@
+#include "maximilian.h"
+
+maxiOsc mySine,myOtherSine;//Two oscillators
+
+
+void setup() {//some inits
+	//nothing to go here this time	
+}
+
+void play(double *output) {
+	
+	*output=mySine.sinewave(myOtherSine.sinewave(1)*440);
+	
+}
diff --git a/maximilian_examples/6.FM2.cpp b/maximilian_examples/6.FM2.cpp
new file mode 100755
index 0000000..d9acd72
--- /dev/null
+++ b/maximilian_examples/6.FM2.cpp
@@ -0,0 +1,14 @@
+#include "maximilian.h"
+
+maxiOsc mySine,myOtherSine,myLastSine,myPhasor;//Three oscillators
+
+
+void setup() {//some inits
+	//nothing to go here this time	
+}
+
+void play(double *output) {
+	
+	*output=mySine.sinewave(myOtherSine.sinewave(myLastSine.sinewave(0.1)*30)*440);//awesome bassline
+	
+}
diff --git a/maximilian_examples/7.Counting.cpp b/maximilian_examples/7.Counting.cpp
new file mode 100755
index 0000000..7de2f0e
--- /dev/null
+++ b/maximilian_examples/7.Counting.cpp
@@ -0,0 +1,15 @@
+#include "maximilian.h"
+
+maxiOsc myCounter,mySquare;//these oscillators will help us count and play sound
+int CurrentCount;//we're going to put the current count in this variable so that we can use it more easily.
+
+
+void setup() {//some inits
+	//nothing to go here this time	
+}
+
+void play(double *output) {
+	
+	CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value.
+	*output=mySquare.square(CurrentCount*100);
+}
diff --git a/maximilian_examples/8.Counting2.cpp b/maximilian_examples/8.Counting2.cpp
new file mode 100755
index 0000000..ce016d0
--- /dev/null
+++ b/maximilian_examples/8.Counting2.cpp
@@ -0,0 +1,24 @@
+#include "maximilian.h"
+
+maxiOsc myCounter,mySwitchableOsc;//these oscillators will help us count and make sound.
+int CurrentCount;//we're going to put the current count in this variable so that we can use it more easily.
+double myOscOutput;//we're going to stick the output here to make it easier to mess with stuff.
+
+void setup() {//some inits
+	//nothing to go here this time	
+}
+
+void play(double *output) {
+	
+	CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value.
+	
+	if (CurrentCount<5)//simple if statement
+		
+		myOscOutput=mySwitchableOsc.square(CurrentCount*100);
+	
+	else if (CurrentCount>=5)//and the 'else' bit.
+		
+		myOscOutput=mySwitchableOsc.sinewave(CurrentCount*50);//one osc object can produce whichever waveform you want. 
+	
+	*output=myOscOutput;//point me at your speakers and fire.
+}
diff --git a/maximilian_examples/9.Envelopes.cpp b/maximilian_examples/9.Envelopes.cpp
new file mode 100755
index 0000000..806e14f
--- /dev/null
+++ b/maximilian_examples/9.Envelopes.cpp
@@ -0,0 +1,33 @@
+#include "maximilian.h"
+
+maxiOsc myCounter,mySwitchableOsc;//
+int CurrentCount;//
+double myOscOutput,myCurrentVolume;//
+double myEnvelopeData[4] = {1,0,0,500};//this data will be used to make an envelope. Value and time to value in ms.
+maxiEnvelope myEnvelope;
+
+
+void setup() {//some inits
+	myEnvelope.amplitude=myEnvelopeData[0]; //initialise the envelope
+}
+
+void play(double *output) {
+	
+	myCurrentVolume=myEnvelope.line(4,myEnvelopeData);
+	
+	CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value.
+	
+	if (CurrentCount<5)//simple if statement
+		
+		myOscOutput=mySwitchableOsc.square(CurrentCount*100);
+	
+	else if (CurrentCount>=5)//and the 'else' bit.
+		
+		myOscOutput=mySwitchableOsc.sinewave(CurrentCount*50);//one osc object can produce whichever waveform you want. 
+	
+	if (CurrentCount==1) 
+		
+		myEnvelope.trigger(0,myEnvelopeData[0]); //trigger the envelope
+	
+	*output=myOscOutput*myCurrentVolume;//point me at your speakers and fire.
+}
diff --git a/player.cpp b/player.cpp
new file mode 100644
index 0000000..37edf80
--- /dev/null
+++ b/player.cpp
@@ -0,0 +1,153 @@
+/*
+ *  player.cpp
+ *  rtaudiotest
+ *
+ *  Created by Chris on 23/08/2011.
+ *  Copyright 2011 Goldsmiths Creative Computing. All rights reserved.
+ *
+ */
+
+#include "player.h"
+#include "maximilian.h"
+#include <iostream>
+
+
+#ifdef MAXIMILIAN_PORTAUDIO
+#include "portaudio.h"
+//#include "pa_mac_core.h"
+#elif defined(MAXIMILIAN_RT_AUDIO)
+#include "RtAudio.h"
+#endif
+
+
+void setup();//use this to do any initialisation if you want.
+
+void play(double *channels);//run dac! Very very often. Too often in fact. er...
+
+
+
+#ifdef MAXIMILIAN_PORTAUDIO
+int routing(const void *inputBuffer,
+		void *outputBuffer,
+		unsigned long nBufferFrames,
+		const PaStreamCallbackTimeInfo* timeInfo,
+		PaStreamCallbackFlags status,
+		void *userData ){
+#elif defined(MAXIMILIAN_RT_AUDIO)
+int routing	(void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
+			 double streamTime, RtAudioStreamStatus status, void *userData ) {
+#endif
+	
+	unsigned int i, j;
+	
+#ifdef MAXIMILIAN_PORTAUDIO
+	float *buffer = (float *) outputBuffer;
+#elif defined(MAXIMILIAN_RT_AUDIO)
+	double *buffer = (double *) outputBuffer;
+#endif
+	double *lastValues = (double *) userData;
+	//	double currentTime = (double) streamTime; Might come in handy for control
+	if ( status )
+		std::cout << "Stream underflow detected!" << std::endl;
+	for ( i=0; i<nBufferFrames; i++ ) {	
+	}
+	// Write interleaved audio data.
+	for ( i=0; i<nBufferFrames; i++ ) {
+		play(lastValues);			
+		for ( j=0; j<maxiSettings::channels; j++ ) {
+			*buffer++=lastValues[0];
+		}
+	}
+	return 0;
+}
+
+//This is main()
+int main()
+{
+	setup();
+	
+#ifdef MAXIMILIAN_PORTAUDIO
+	PaStream *stream;
+	PaError err;
+	err = Pa_Initialize();
+	if( err != paNoError )
+		std::cout <<   "PortAudio error: " << Pa_GetErrorText( err ) << std::endl;
+	
+	double data[maxiSettings::channels];
+	
+	err = Pa_OpenDefaultStream( &stream,
+							   0,          /* no input channels */
+							   maxiSettings::channels,          /* stereo output */
+							   paFloat32,  /* 64 bit floating point output */
+							   maxiSettings::sampleRate,
+							   maxiSettings::bufferSize,        /* frames per buffer, i.e. the number
+												   of sample frames that PortAudio will
+												   request from the callback. Many apps
+												   may want to use
+												   paFramesPerBufferUnspecified, which
+												   tells PortAudio to pick the best,
+												   possibly changing, buffer size.*/
+							   &routing, /* this is your callback function */
+							   &data ); /*This is a pointer that will be passed to
+										 your callback*/
+	
+	//PaAlsa_EnableRealtimeScheduling(stream,true);
+	
+	err = Pa_StartStream( stream );
+	if( err != paNoError )
+		std::cout <<   "PortAudio error: " << Pa_GetErrorText( err ) << std::endl;
+	
+	
+	char input;
+	std::cout << "\nMaximilian is playing ... press <enter> to quit.\n";
+	std::cin.get( input );
+	
+	
+	
+	err = Pa_Terminate();
+	if( err != paNoError )
+		std::cout <<  "PortAudio error: "<< Pa_GetErrorText( err ) << std::endl;
+	
+#elif defined(MAXIMILIAN_RT_AUDIO)
+	RtAudio dac;
+	if ( dac.getDeviceCount() < 1 ) {
+		std::cout << "\nNo audio devices found!\n";
+		exit( 0 );
+	}
+	
+	RtAudio::StreamParameters parameters;
+	parameters.deviceId = dac.getDefaultOutputDevice();
+	parameters.nChannels = maxiSettings::channels;
+	parameters.firstChannel = 0;
+	unsigned int sampleRate = maxiSettings::sampleRate;
+	unsigned int bufferFrames = maxiSettings::bufferSize; 
+	double data[parameters.nChannels];
+	
+	try {
+		dac.openStream( &parameters, NULL, RTAUDIO_FLOAT64,
+					   sampleRate, &bufferFrames, &routing, (void *)&data);
+		
+		dac.startStream();
+	}
+	catch ( RtError& e ) {
+		e.printMessage();
+		exit( 0 );
+	}
+	
+	char input;
+	std::cout << "\nMaximilian is playing ... press <enter> to quit.\n";
+	std::cin.get( input );
+	
+	try {
+		// Stop the stream
+		dac.stopStream();
+	}
+	catch (RtError& e) {
+		e.printMessage();
+	}
+	
+	if ( dac.isStreamOpen() ) dac.closeStream();
+#endif
+	
+	return 0;
+}
diff --git a/player.h b/player.h
new file mode 100644
index 0000000..3fea524
--- /dev/null
+++ b/player.h
@@ -0,0 +1,11 @@
+/*
+ *  player.h
+ *  rtaudiotest
+ *
+ *  Created by Chris on 23/08/2011.
+ *  Copyright 2011 Goldsmiths Creative Computing. All rights reserved.
+ *
+ */
+
+//#define MAXIMILIAN_PORTAUDIO
+#define MAXIMILIAN_RT_AUDIO
diff --git a/portaudio.h b/portaudio.h
new file mode 100644
index 0000000..89964ab
--- /dev/null
+++ b/portaudio.h
@@ -0,0 +1,1134 @@
+#ifndef PORTAUDIO_H
+#define PORTAUDIO_H
+/*
+ * $Id: portaudio.h 1337 2008-02-15 07:32:09Z rossb $
+ * PortAudio Portable Real-Time Audio Library
+ * PortAudio API Header File
+ * Latest version available at: http://www.portaudio.com/
+ *
+ * Copyright (c) 1999-2002 Ross Bencina and Phil Burk
+ *
+ * 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.
+ *
+ * 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.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
+ */
+
+/** @file
+ @ingroup public_header
+ @brief The portable PortAudio API.
+*/
+
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+ 
+/** Retrieve the release number of the currently running PortAudio build,
+ eg 1900.
+*/
+int Pa_GetVersion( void );
+
+
+/** Retrieve a textual description of the current PortAudio build,
+ eg "PortAudio V19-devel 13 October 2002".
+*/
+const char* Pa_GetVersionText( void );
+
+
+/** Error codes returned by PortAudio functions.
+ Note that with the exception of paNoError, all PaErrorCodes are negative.
+*/
+
+typedef int PaError;
+typedef enum PaErrorCode
+{
+    paNoError = 0,
+
+    paNotInitialized = -10000,
+    paUnanticipatedHostError,
+    paInvalidChannelCount,
+    paInvalidSampleRate,
+    paInvalidDevice,
+    paInvalidFlag,
+    paSampleFormatNotSupported,
+    paBadIODeviceCombination,
+    paInsufficientMemory,
+    paBufferTooBig,
+    paBufferTooSmall,
+    paNullCallback,
+    paBadStreamPtr,
+    paTimedOut,
+    paInternalError,
+    paDeviceUnavailable,
+    paIncompatibleHostApiSpecificStreamInfo,
+    paStreamIsStopped,
+    paStreamIsNotStopped,
+    paInputOverflowed,
+    paOutputUnderflowed,
+    paHostApiNotFound,
+    paInvalidHostApi,
+    paCanNotReadFromACallbackStream,      /**< @todo review error code name */
+    paCanNotWriteToACallbackStream,       /**< @todo review error code name */
+    paCanNotReadFromAnOutputOnlyStream,   /**< @todo review error code name */
+    paCanNotWriteToAnInputOnlyStream,     /**< @todo review error code name */
+    paIncompatibleStreamHostApi,
+    paBadBufferPtr
+} PaErrorCode;
+
+
+/** Translate the supplied PortAudio error code into a human readable
+ message.
+*/
+const char *Pa_GetErrorText( PaError errorCode );
+
+
+/** Library initialization function - call this before using PortAudio.
+ This function initialises internal data structures and prepares underlying
+ host APIs for use.  With the exception of Pa_GetVersion(), Pa_GetVersionText(),
+ and Pa_GetErrorText(), this function MUST be called before using any other
+ PortAudio API functions.
+
+ If Pa_Initialize() is called multiple times, each successful 
+ call must be matched with a corresponding call to Pa_Terminate(). 
+ Pairs of calls to Pa_Initialize()/Pa_Terminate() may overlap, and are not 
+ required to be fully nested.
+
+ Note that if Pa_Initialize() returns an error code, Pa_Terminate() should
+ NOT be called.
+
+ @return paNoError if successful, otherwise an error code indicating the cause
+ of failure.
+
+ @see Pa_Terminate
+*/
+PaError Pa_Initialize( void );
+
+
+/** Library termination function - call this when finished using PortAudio.
+ This function deallocates all resources allocated by PortAudio since it was
+ initializied by a call to Pa_Initialize(). In cases where Pa_Initialise() has
+ been called multiple times, each call must be matched with a corresponding call
+ to Pa_Terminate(). The final matching call to Pa_Terminate() will automatically
+ close any PortAudio streams that are still open.
+
+ Pa_Terminate() MUST be called before exiting a program which uses PortAudio.
+ Failure to do so may result in serious resource leaks, such as audio devices
+ not being available until the next reboot.
+
+ @return paNoError if successful, otherwise an error code indicating the cause
+ of failure.
+ 
+ @see Pa_Initialize
+*/
+PaError Pa_Terminate( void );
+
+
+
+/** The type used to refer to audio devices. Values of this type usually
+ range from 0 to (Pa_GetDeviceCount()-1), and may also take on the PaNoDevice
+ and paUseHostApiSpecificDeviceSpecification values.
+
+ @see Pa_GetDeviceCount, paNoDevice, paUseHostApiSpecificDeviceSpecification
+*/
+typedef int PaDeviceIndex;
+
+
+/** A special PaDeviceIndex value indicating that no device is available,
+ or should be used.
+
+ @see PaDeviceIndex
+*/
+#define paNoDevice ((PaDeviceIndex)-1)
+
+
+/** A special PaDeviceIndex value indicating that the device(s) to be used
+ are specified in the host api specific stream info structure.
+
+ @see PaDeviceIndex
+*/
+#define paUseHostApiSpecificDeviceSpecification ((PaDeviceIndex)-2)
+
+
+/* Host API enumeration mechanism */
+
+/** The type used to enumerate to host APIs at runtime. Values of this type
+ range from 0 to (Pa_GetHostApiCount()-1).
+
+ @see Pa_GetHostApiCount
+*/
+typedef int PaHostApiIndex;
+
+
+/** Retrieve the number of available host APIs. Even if a host API is
+ available it may have no devices available.
+
+ @return A non-negative value indicating the number of available host APIs
+ or, a PaErrorCode (which are always negative) if PortAudio is not initialized
+ or an error is encountered.
+
+ @see PaHostApiIndex
+*/
+PaHostApiIndex Pa_GetHostApiCount( void );
+
+
+/** Retrieve the index of the default host API. The default host API will be
+ the lowest common denominator host API on the current platform and is
+ unlikely to provide the best performance.
+
+ @return A non-negative value ranging from 0 to (Pa_GetHostApiCount()-1)
+ indicating the default host API index or, a PaErrorCode (which are always
+ negative) if PortAudio is not initialized or an error is encountered.
+*/
+PaHostApiIndex Pa_GetDefaultHostApi( void );
+
+
+/** Unchanging unique identifiers for each supported host API. This type
+ is used in the PaHostApiInfo structure. The values are guaranteed to be
+ unique and to never change, thus allowing code to be written that
+ conditionally uses host API specific extensions.
+
+ New type ids will be allocated when support for a host API reaches
+ "public alpha" status, prior to that developers should use the
+ paInDevelopment type id.
+
+ @see PaHostApiInfo
+*/
+typedef enum PaHostApiTypeId
+{
+    paInDevelopment=0, /* use while developing support for a new host API */
+    paDirectSound=1,
+    paMME=2,
+    paASIO=3,
+    paSoundManager=4,
+    paCoreAudio=5,
+    paOSS=7,
+    paALSA=8,
+    paAL=9,
+    paBeOS=10,
+    paWDMKS=11,
+    paJACK=12,
+    paWASAPI=13,
+    paAudioScienceHPI=14
+} PaHostApiTypeId;
+
+
+/** A structure containing information about a particular host API. */
+
+typedef struct PaHostApiInfo
+{
+    /** this is struct version 1 */
+    int structVersion;
+    /** The well known unique identifier of this host API @see PaHostApiTypeId */
+    PaHostApiTypeId type;
+    /** A textual description of the host API for display on user interfaces. */
+    const char *name;
+
+    /**  The number of devices belonging to this host API. This field may be
+     used in conjunction with Pa_HostApiDeviceIndexToDeviceIndex() to enumerate
+     all devices for this host API.
+     @see Pa_HostApiDeviceIndexToDeviceIndex
+    */
+    int deviceCount;
+
+    /** The default input device for this host API. The value will be a
+     device index ranging from 0 to (Pa_GetDeviceCount()-1), or paNoDevice
+     if no default input device is available.
+    */
+    PaDeviceIndex defaultInputDevice;
+
+    /** The default output device for this host API. The value will be a
+     device index ranging from 0 to (Pa_GetDeviceCount()-1), or paNoDevice
+     if no default output device is available.
+    */
+    PaDeviceIndex defaultOutputDevice;
+    
+} PaHostApiInfo;
+
+
+/** Retrieve a pointer to a structure containing information about a specific
+ host Api.
+
+ @param hostApi A valid host API index ranging from 0 to (Pa_GetHostApiCount()-1)
+
+ @return A pointer to an immutable PaHostApiInfo structure describing
+ a specific host API. If the hostApi parameter is out of range or an error
+ is encountered, the function returns NULL.
+
+ The returned structure is owned by the PortAudio implementation and must not
+ be manipulated or freed. The pointer is only guaranteed to be valid between
+ calls to Pa_Initialize() and Pa_Terminate().
+*/
+const PaHostApiInfo * Pa_GetHostApiInfo( PaHostApiIndex hostApi );
+
+
+/** Convert a static host API unique identifier, into a runtime
+ host API index.
+
+ @param type A unique host API identifier belonging to the PaHostApiTypeId
+ enumeration.
+
+ @return A valid PaHostApiIndex ranging from 0 to (Pa_GetHostApiCount()-1) or,
+ a PaErrorCode (which are always negative) if PortAudio is not initialized
+ or an error is encountered.
+ 
+ The paHostApiNotFound error code indicates that the host API specified by the
+ type parameter is not available.
+
+ @see PaHostApiTypeId
+*/
+PaHostApiIndex Pa_HostApiTypeIdToHostApiIndex( PaHostApiTypeId type );
+
+
+/** Convert a host-API-specific device index to standard PortAudio device index.
+ This function may be used in conjunction with the deviceCount field of
+ PaHostApiInfo to enumerate all devices for the specified host API.
+
+ @param hostApi A valid host API index ranging from 0 to (Pa_GetHostApiCount()-1)
+
+ @param hostApiDeviceIndex A valid per-host device index in the range
+ 0 to (Pa_GetHostApiInfo(hostApi)->deviceCount-1)
+
+ @return A non-negative PaDeviceIndex ranging from 0 to (Pa_GetDeviceCount()-1)
+ or, a PaErrorCode (which are always negative) if PortAudio is not initialized
+ or an error is encountered.
+
+ A paInvalidHostApi error code indicates that the host API index specified by
+ the hostApi parameter is out of range.
+
+ A paInvalidDevice error code indicates that the hostApiDeviceIndex parameter
+ is out of range.
+ 
+ @see PaHostApiInfo
+*/
+PaDeviceIndex Pa_HostApiDeviceIndexToDeviceIndex( PaHostApiIndex hostApi,
+        int hostApiDeviceIndex );
+
+
+
+/** Structure used to return information about a host error condition.
+*/
+typedef struct PaHostErrorInfo{
+    PaHostApiTypeId hostApiType;    /**< the host API which returned the error code */
+    long errorCode;                 /**< the error code returned */
+    const char *errorText;          /**< a textual description of the error if available, otherwise a zero-length string */
+}PaHostErrorInfo;
+
+
+/** Return information about the last host error encountered. The error
+ information returned by Pa_GetLastHostErrorInfo() will never be modified
+ asyncronously by errors occurring in other PortAudio owned threads
+ (such as the thread that manages the stream callback.)
+
+ This function is provided as a last resort, primarily to enhance debugging
+ by providing clients with access to all available error information.
+
+ @return A pointer to an immutable structure constaining information about
+ the host error. The values in this structure will only be valid if a
+ PortAudio function has previously returned the paUnanticipatedHostError
+ error code.
+*/
+const PaHostErrorInfo* Pa_GetLastHostErrorInfo( void );
+
+
+
+/* Device enumeration and capabilities */
+
+/** Retrieve the number of available devices. The number of available devices
+ may be zero.
+
+ @return A non-negative value indicating the number of available devices or,
+ a PaErrorCode (which are always negative) if PortAudio is not initialized
+ or an error is encountered.
+*/
+PaDeviceIndex Pa_GetDeviceCount( void );
+
+
+/** Retrieve the index of the default input device. The result can be
+ used in the inputDevice parameter to Pa_OpenStream().
+
+ @return The default input device index for the default host API, or paNoDevice
+ if no default input device is available or an error was encountered.
+*/
+PaDeviceIndex Pa_GetDefaultInputDevice( void );
+
+
+/** Retrieve the index of the default output device. The result can be
+ used in the outputDevice parameter to Pa_OpenStream().
+
+ @return The default output device index for the defualt host API, or paNoDevice
+ if no default output device is available or an error was encountered.
+
+ @note
+ On the PC, the user can specify a default device by
+ setting an environment variable. For example, to use device #1.
+<pre>
+ set PA_RECOMMENDED_OUTPUT_DEVICE=1
+</pre>
+ The user should first determine the available device ids by using
+ the supplied application "pa_devs".
+*/
+PaDeviceIndex Pa_GetDefaultOutputDevice( void );
+
+
+/** The type used to represent monotonic time in seconds that can be used
+ for syncronisation. The type is used for the outTime argument to the
+ PaStreamCallback and as the result of Pa_GetStreamTime().
+     
+ @see PaStreamCallback, Pa_GetStreamTime
+*/
+typedef double PaTime;
+
+
+/** A type used to specify one or more sample formats. Each value indicates
+ a possible format for sound data passed to and from the stream callback,
+ Pa_ReadStream and Pa_WriteStream.
+
+ The standard formats paFloat32, paInt16, paInt32, paInt24, paInt8
+ and aUInt8 are usually implemented by all implementations.
+
+ The floating point representation (paFloat32) uses +1.0 and -1.0 as the
+ maximum and minimum respectively.
+
+ paUInt8 is an unsigned 8 bit format where 128 is considered "ground"
+
+ The paNonInterleaved flag indicates that a multichannel buffer is passed
+ as a set of non-interleaved pointers.
+
+ @see Pa_OpenStream, Pa_OpenDefaultStream, PaDeviceInfo
+ @see paFloat32, paInt16, paInt32, paInt24, paInt8
+ @see paUInt8, paCustomFormat, paNonInterleaved
+*/
+typedef unsigned long PaSampleFormat;
+
+
+#define paFloat32        ((PaSampleFormat) 0x00000001) /**< @see PaSampleFormat */
+#define paInt32          ((PaSampleFormat) 0x00000002) /**< @see PaSampleFormat */
+#define paInt24          ((PaSampleFormat) 0x00000004) /**< Packed 24 bit format. @see PaSampleFormat */
+#define paInt16          ((PaSampleFormat) 0x00000008) /**< @see PaSampleFormat */
+#define paInt8           ((PaSampleFormat) 0x00000010) /**< @see PaSampleFormat */
+#define paUInt8          ((PaSampleFormat) 0x00000020) /**< @see PaSampleFormat */
+#define paCustomFormat   ((PaSampleFormat) 0x00010000)/**< @see PaSampleFormat */
+
+#define paNonInterleaved ((PaSampleFormat) 0x80000000)
+
+/** A structure providing information and capabilities of PortAudio devices.
+ Devices may support input, output or both input and output.
+*/
+typedef struct PaDeviceInfo
+{
+    int structVersion;  /* this is struct version 2 */
+    const char *name;
+    PaHostApiIndex hostApi; /* note this is a host API index, not a type id*/
+    
+    int maxInputChannels;
+    int maxOutputChannels;
+
+    /* Default latency values for interactive performance. */
+    PaTime defaultLowInputLatency;
+    PaTime defaultLowOutputLatency;
+    /* Default latency values for robust non-interactive applications (eg. playing sound files). */
+    PaTime defaultHighInputLatency;
+    PaTime defaultHighOutputLatency;
+
+    double defaultSampleRate;
+} PaDeviceInfo;
+
+
+/** Retrieve a pointer to a PaDeviceInfo structure containing information
+ about the specified device.
+ @return A pointer to an immutable PaDeviceInfo structure. If the device
+ parameter is out of range the function returns NULL.
+
+ @param device A valid device index in the range 0 to (Pa_GetDeviceCount()-1)
+
+ @note PortAudio manages the memory referenced by the returned pointer,
+ the client must not manipulate or free the memory. The pointer is only
+ guaranteed to be valid between calls to Pa_Initialize() and Pa_Terminate().
+
+ @see PaDeviceInfo, PaDeviceIndex
+*/
+const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceIndex device );
+
+
+/** Parameters for one direction (input or output) of a stream.
+*/
+typedef struct PaStreamParameters
+{
+    /** A valid device index in the range 0 to (Pa_GetDeviceCount()-1)
+     specifying the device to be used or the special constant
+     paUseHostApiSpecificDeviceSpecification which indicates that the actual
+     device(s) to use are specified in hostApiSpecificStreamInfo.
+     This field must not be set to paNoDevice.
+    */
+    PaDeviceIndex device;
+    
+    /** The number of channels of sound to be delivered to the
+     stream callback or accessed by Pa_ReadStream() or Pa_WriteStream().
+     It can range from 1 to the value of maxInputChannels in the
+     PaDeviceInfo record for the device specified by the device parameter.
+    */
+    int channelCount;
+
+    /** The sample format of the buffer provided to the stream callback,
+     a_ReadStream() or Pa_WriteStream(). It may be any of the formats described
+     by the PaSampleFormat enumeration.
+    */
+    PaSampleFormat sampleFormat;
+
+    /** The desired latency in seconds. Where practical, implementations should
+     configure their latency based on these parameters, otherwise they may
+     choose the closest viable latency instead. Unless the suggested latency
+     is greater than the absolute upper limit for the device implementations
+     should round the suggestedLatency up to the next practial value - ie to
+     provide an equal or higher latency than suggestedLatency wherever possibe.
+     Actual latency values for an open stream may be retrieved using the
+     inputLatency and outputLatency fields of the PaStreamInfo structure
+     returned by Pa_GetStreamInfo().
+     @see default*Latency in PaDeviceInfo, *Latency in PaStreamInfo
+    */
+    PaTime suggestedLatency;
+
+    /** An optional pointer to a host api specific data structure
+     containing additional information for device setup and/or stream processing.
+     hostApiSpecificStreamInfo is never required for correct operation,
+     if not used it should be set to NULL.
+    */
+    void *hostApiSpecificStreamInfo;
+
+} PaStreamParameters;
+
+
+/** Return code for Pa_IsFormatSupported indicating success. */
+#define paFormatIsSupported (0)
+
+/** Determine whether it would be possible to open a stream with the specified
+ parameters.
+
+ @param inputParameters A structure that describes the input parameters used to
+ open a stream. The suggestedLatency field is ignored. See PaStreamParameters
+ for a description of these parameters. inputParameters must be NULL for
+ output-only streams.
+
+ @param outputParameters A structure that describes the output parameters used
+ to open a stream. The suggestedLatency field is ignored. See PaStreamParameters
+ for a description of these parameters. outputParameters must be NULL for
+ input-only streams.
+
+ @param sampleRate The required sampleRate. For full-duplex streams it is the
+ sample rate for both input and output
+
+ @return Returns 0 if the format is supported, and an error code indicating why
+ the format is not supported otherwise. The constant paFormatIsSupported is
+ provided to compare with the return value for success.
+
+ @see paFormatIsSupported, PaStreamParameters
+*/
+PaError Pa_IsFormatSupported( const PaStreamParameters *inputParameters,
+                              const PaStreamParameters *outputParameters,
+                              double sampleRate );
+
+
+
+/* Streaming types and functions */
+
+
+/**
+ A single PaStream can provide multiple channels of real-time
+ streaming audio input and output to a client application. A stream
+ provides access to audio hardware represented by one or more
+ PaDevices. Depending on the underlying Host API, it may be possible 
+ to open multiple streams using the same device, however this behavior 
+ is implementation defined. Portable applications should assume that 
+ a PaDevice may be simultaneously used by at most one PaStream.
+
+ Pointers to PaStream objects are passed between PortAudio functions that
+ operate on streams.
+
+ @see Pa_OpenStream, Pa_OpenDefaultStream, Pa_OpenDefaultStream, Pa_CloseStream,
+ Pa_StartStream, Pa_StopStream, Pa_AbortStream, Pa_IsStreamActive,
+ Pa_GetStreamTime, Pa_GetStreamCpuLoad
+
+*/
+typedef void PaStream;
+
+
+/** Can be passed as the framesPerBuffer parameter to Pa_OpenStream()
+ or Pa_OpenDefaultStream() to indicate that the stream callback will
+ accept buffers of any size.
+*/
+#define paFramesPerBufferUnspecified  (0)
+
+
+/** Flags used to control the behavior of a stream. They are passed as
+ parameters to Pa_OpenStream or Pa_OpenDefaultStream. Multiple flags may be
+ ORed together.
+
+ @see Pa_OpenStream, Pa_OpenDefaultStream
+ @see paNoFlag, paClipOff, paDitherOff, paNeverDropInput,
+  paPrimeOutputBuffersUsingStreamCallback, paPlatformSpecificFlags
+*/
+typedef unsigned long PaStreamFlags;
+
+/** @see PaStreamFlags */
+#define   paNoFlag          ((PaStreamFlags) 0)
+
+/** Disable default clipping of out of range samples.
+ @see PaStreamFlags
+*/
+#define   paClipOff         ((PaStreamFlags) 0x00000001)
+
+/** Disable default dithering.
+ @see PaStreamFlags
+*/
+#define   paDitherOff       ((PaStreamFlags) 0x00000002)
+
+/** Flag requests that where possible a full duplex stream will not discard
+ overflowed input samples without calling the stream callback. This flag is
+ only valid for full duplex callback streams and only when used in combination
+ with the paFramesPerBufferUnspecified (0) framesPerBuffer parameter. Using
+ this flag incorrectly results in a paInvalidFlag error being returned from
+ Pa_OpenStream and Pa_OpenDefaultStream.
+
+ @see PaStreamFlags, paFramesPerBufferUnspecified
+*/
+#define   paNeverDropInput  ((PaStreamFlags) 0x00000004)
+
+/** Call the stream callback to fill initial output buffers, rather than the
+ default behavior of priming the buffers with zeros (silence). This flag has
+ no effect for input-only and blocking read/write streams.
+ 
+ @see PaStreamFlags
+*/
+#define   paPrimeOutputBuffersUsingStreamCallback ((PaStreamFlags) 0x00000008)
+
+/** A mask specifying the platform specific bits.
+ @see PaStreamFlags
+*/
+#define   paPlatformSpecificFlags ((PaStreamFlags)0xFFFF0000)
+
+/**
+ Timing information for the buffers passed to the stream callback.
+*/
+typedef struct PaStreamCallbackTimeInfo{
+    PaTime inputBufferAdcTime;
+    PaTime currentTime;
+    PaTime outputBufferDacTime;
+} PaStreamCallbackTimeInfo;
+
+
+/**
+ Flag bit constants for the statusFlags to PaStreamCallback.
+
+ @see paInputUnderflow, paInputOverflow, paOutputUnderflow, paOutputOverflow,
+ paPrimingOutput
+*/
+typedef unsigned long PaStreamCallbackFlags;
+
+/** In a stream opened with paFramesPerBufferUnspecified, indicates that
+ input data is all silence (zeros) because no real data is available. In a
+ stream opened without paFramesPerBufferUnspecified, it indicates that one or
+ more zero samples have been inserted into the input buffer to compensate
+ for an input underflow.
+ @see PaStreamCallbackFlags
+*/
+#define paInputUnderflow   ((PaStreamCallbackFlags) 0x00000001)
+
+/** In a stream opened with paFramesPerBufferUnspecified, indicates that data
+ prior to the first sample of the input buffer was discarded due to an
+ overflow, possibly because the stream callback is using too much CPU time.
+ Otherwise indicates that data prior to one or more samples in the
+ input buffer was discarded.
+ @see PaStreamCallbackFlags
+*/
+#define paInputOverflow    ((PaStreamCallbackFlags) 0x00000002)
+
+/** Indicates that output data (or a gap) was inserted, possibly because the
+ stream callback is using too much CPU time.
+ @see PaStreamCallbackFlags
+*/
+#define paOutputUnderflow  ((PaStreamCallbackFlags) 0x00000004)
+
+/** Indicates that output data will be discarded because no room is available.
+ @see PaStreamCallbackFlags
+*/
+#define paOutputOverflow   ((PaStreamCallbackFlags) 0x00000008)
+
+/** Some of all of the output data will be used to prime the stream, input
+ data may be zero.
+ @see PaStreamCallbackFlags
+*/
+#define paPrimingOutput    ((PaStreamCallbackFlags) 0x00000010)
+
+/**
+ Allowable return values for the PaStreamCallback.
+ @see PaStreamCallback
+*/
+typedef enum PaStreamCallbackResult
+{
+    paContinue=0,
+    paComplete=1,
+    paAbort=2
+} PaStreamCallbackResult;
+
+
+/**
+ Functions of type PaStreamCallback are implemented by PortAudio clients.
+ They consume, process or generate audio in response to requests from an
+ active PortAudio stream.
+     
+ @param input and @param output are arrays of interleaved samples,
+ the format, packing and number of channels used by the buffers are
+ determined by parameters to Pa_OpenStream().
+     
+ @param frameCount The number of sample frames to be processed by
+ the stream callback.
+
+ @param timeInfo The time in seconds when the first sample of the input
+ buffer was received at the audio input, the time in seconds when the first
+ sample of the output buffer will begin being played at the audio output, and
+ the time in seconds when the stream callback was called.
+ See also Pa_GetStreamTime()
+
+ @param statusFlags Flags indicating whether input and/or output buffers
+ have been inserted or will be dropped to overcome underflow or overflow
+ conditions.
+
+ @param userData The value of a user supplied pointer passed to
+ Pa_OpenStream() intended for storing synthesis data etc.
+
+ @return
+ The stream callback should return one of the values in the
+ PaStreamCallbackResult enumeration. To ensure that the callback continues
+ to be called, it should return paContinue (0). Either paComplete or paAbort
+ can be returned to finish stream processing, after either of these values is
+ returned the callback will not be called again. If paAbort is returned the
+ stream will finish as soon as possible. If paComplete is returned, the stream
+ will continue until all buffers generated by the callback have been played.
+ This may be useful in applications such as soundfile players where a specific
+ duration of output is required. However, it is not necessary to utilise this
+ mechanism as Pa_StopStream(), Pa_AbortStream() or Pa_CloseStream() can also
+ be used to stop the stream. The callback must always fill the entire output
+ buffer irrespective of its return value.
+
+ @see Pa_OpenStream, Pa_OpenDefaultStream
+
+ @note With the exception of Pa_GetStreamCpuLoad() it is not permissable to call
+ PortAudio API functions from within the stream callback.
+*/
+typedef int PaStreamCallback(
+    const void *input, void *output,
+    unsigned long frameCount,
+    const PaStreamCallbackTimeInfo* timeInfo,
+    PaStreamCallbackFlags statusFlags,
+    void *userData );
+
+
+/** Opens a stream for either input, output or both.
+     
+ @param stream The address of a PaStream pointer which will receive
+ a pointer to the newly opened stream.
+     
+ @param inputParameters A structure that describes the input parameters used by
+ the opened stream. See PaStreamParameters for a description of these parameters.
+ inputParameters must be NULL for output-only streams.
+
+ @param outputParameters A structure that describes the output parameters used by
+ the opened stream. See PaStreamParameters for a description of these parameters.
+ outputParameters must be NULL for input-only streams.
+ 
+ @param sampleRate The desired sampleRate. For full-duplex streams it is the
+ sample rate for both input and output
+     
+ @param framesPerBuffer The number of frames passed to the stream callback
+ function, or the preferred block granularity for a blocking read/write stream.
+ The special value paFramesPerBufferUnspecified (0) may be used to request that
+ the stream callback will recieve an optimal (and possibly varying) number of
+ frames based on host requirements and the requested latency settings.
+ Note: With some host APIs, the use of non-zero framesPerBuffer for a callback
+ stream may introduce an additional layer of buffering which could introduce
+ additional latency. PortAudio guarantees that the additional latency
+ will be kept to the theoretical minimum however, it is strongly recommended
+ that a non-zero framesPerBuffer value only be used when your algorithm
+ requires a fixed number of frames per stream callback.
+ 
+ @param streamFlags Flags which modify the behaviour of the streaming process.
+ This parameter may contain a combination of flags ORed together. Some flags may
+ only be relevant to certain buffer formats.
+     
+ @param streamCallback A pointer to a client supplied function that is responsible
+ for processing and filling input and output buffers. If this parameter is NULL
+ the stream will be opened in 'blocking read/write' mode. In blocking mode,
+ the client can receive sample data using Pa_ReadStream and write sample data
+ using Pa_WriteStream, the number of samples that may be read or written
+ without blocking is returned by Pa_GetStreamReadAvailable and
+ Pa_GetStreamWriteAvailable respectively.
+
+ @param userData A client supplied pointer which is passed to the stream callback
+ function. It could for example, contain a pointer to instance data necessary
+ for processing the audio buffers. This parameter is ignored if streamCallback
+ is NULL.
+     
+ @return
+ Upon success Pa_OpenStream() returns paNoError and places a pointer to a
+ valid PaStream in the stream argument. The stream is inactive (stopped).
+ If a call to Pa_OpenStream() fails, a non-zero error code is returned (see
+ PaError for possible error codes) and the value of stream is invalid.
+
+ @see PaStreamParameters, PaStreamCallback, Pa_ReadStream, Pa_WriteStream,
+ Pa_GetStreamReadAvailable, Pa_GetStreamWriteAvailable
+*/
+PaError Pa_OpenStream( PaStream** stream,
+                       const PaStreamParameters *inputParameters,
+                       const PaStreamParameters *outputParameters,
+                       double sampleRate,
+                       unsigned long framesPerBuffer,
+                       PaStreamFlags streamFlags,
+                       PaStreamCallback *streamCallback,
+                       void *userData );
+
+
+/** A simplified version of Pa_OpenStream() that opens the default input
+ and/or output devices.
+
+ @param stream The address of a PaStream pointer which will receive
+ a pointer to the newly opened stream.
+ 
+ @param numInputChannels  The number of channels of sound that will be supplied
+ to the stream callback or returned by Pa_ReadStream. It can range from 1 to
+ the value of maxInputChannels in the PaDeviceInfo record for the default input
+ device. If 0 the stream is opened as an output-only stream.
+
+ @param numOutputChannels The number of channels of sound to be delivered to the
+ stream callback or passed to Pa_WriteStream. It can range from 1 to the value
+ of maxOutputChannels in the PaDeviceInfo record for the default output dvice.
+ If 0 the stream is opened as an output-only stream.
+
+ @param sampleFormat The sample format of both the input and output buffers
+ provided to the callback or passed to and from Pa_ReadStream and Pa_WriteStream.
+ sampleFormat may be any of the formats described by the PaSampleFormat
+ enumeration.
+ 
+ @param sampleRate Same as Pa_OpenStream parameter of the same name.
+ @param framesPerBuffer Same as Pa_OpenStream parameter of the same name.
+ @param streamCallback Same as Pa_OpenStream parameter of the same name.
+ @param userData Same as Pa_OpenStream parameter of the same name.
+
+ @return As for Pa_OpenStream
+
+ @see Pa_OpenStream, PaStreamCallback
+*/
+PaError Pa_OpenDefaultStream( PaStream** stream,
+                              int numInputChannels,
+                              int numOutputChannels,
+                              PaSampleFormat sampleFormat,
+                              double sampleRate,
+                              unsigned long framesPerBuffer,
+                              PaStreamCallback *streamCallback,
+                              void *userData );
+
+
+/** Closes an audio stream. If the audio stream is active it
+ discards any pending buffers as if Pa_AbortStream() had been called.
+*/
+PaError Pa_CloseStream( PaStream *stream );
+
+
+/** Functions of type PaStreamFinishedCallback are implemented by PortAudio 
+ clients. They can be registered with a stream using the Pa_SetStreamFinishedCallback
+ function. Once registered they are called when the stream becomes inactive
+ (ie once a call to Pa_StopStream() will not block).
+ A stream will become inactive after the stream callback returns non-zero,
+ or when Pa_StopStream or Pa_AbortStream is called. For a stream providing audio
+ output, if the stream callback returns paComplete, or Pa_StopStream is called,
+ the stream finished callback will not be called until all generated sample data
+ has been played.
+ 
+ @param userData The userData parameter supplied to Pa_OpenStream()
+
+ @see Pa_SetStreamFinishedCallback
+*/
+typedef void PaStreamFinishedCallback( void *userData );
+
+
+/** Register a stream finished callback function which will be called when the 
+ stream becomes inactive. See the description of PaStreamFinishedCallback for 
+ further details about when the callback will be called.
+
+ @param stream a pointer to a PaStream that is in the stopped state - if the
+ stream is not stopped, the stream's finished callback will remain unchanged 
+ and an error code will be returned.
+
+ @param streamFinishedCallback a pointer to a function with the same signature
+ as PaStreamFinishedCallback, that will be called when the stream becomes
+ inactive. Passing NULL for this parameter will un-register a previously
+ registered stream finished callback function.
+
+ @return on success returns paNoError, otherwise an error code indicating the cause
+ of the error.
+
+ @see PaStreamFinishedCallback
+*/
+PaError Pa_SetStreamFinishedCallback( PaStream *stream, PaStreamFinishedCallback* streamFinishedCallback ); 
+
+
+/** Commences audio processing.
+*/
+PaError Pa_StartStream( PaStream *stream );
+
+
+/** Terminates audio processing. It waits until all pending
+ audio buffers have been played before it returns.
+*/
+PaError Pa_StopStream( PaStream *stream );
+
+
+/** Terminates audio processing immediately without waiting for pending
+ buffers to complete.
+*/
+PaError Pa_AbortStream( PaStream *stream );
+
+
+/** Determine whether the stream is stopped.
+ A stream is considered to be stopped prior to a successful call to
+ Pa_StartStream and after a successful call to Pa_StopStream or Pa_AbortStream.
+ If a stream callback returns a value other than paContinue the stream is NOT
+ considered to be stopped.
+
+ @return Returns one (1) when the stream is stopped, zero (0) when
+ the stream is running or, a PaErrorCode (which are always negative) if
+ PortAudio is not initialized or an error is encountered.
+
+ @see Pa_StopStream, Pa_AbortStream, Pa_IsStreamActive
+*/
+PaError Pa_IsStreamStopped( PaStream *stream );
+
+
+/** Determine whether the stream is active.
+ A stream is active after a successful call to Pa_StartStream(), until it
+ becomes inactive either as a result of a call to Pa_StopStream() or
+ Pa_AbortStream(), or as a result of a return value other than paContinue from
+ the stream callback. In the latter case, the stream is considered inactive
+ after the last buffer has finished playing.
+
+ @return Returns one (1) when the stream is active (ie playing or recording
+ audio), zero (0) when not playing or, a PaErrorCode (which are always negative)
+ if PortAudio is not initialized or an error is encountered.
+
+ @see Pa_StopStream, Pa_AbortStream, Pa_IsStreamStopped
+*/
+PaError Pa_IsStreamActive( PaStream *stream );
+
+
+
+/** A structure containing unchanging information about an open stream.
+ @see Pa_GetStreamInfo
+*/
+
+typedef struct PaStreamInfo
+{
+    /** this is struct version 1 */
+    int structVersion;
+
+    /** The input latency of the stream in seconds. This value provides the most
+     accurate estimate of input latency available to the implementation. It may
+     differ significantly from the suggestedLatency value passed to Pa_OpenStream().
+     The value of this field will be zero (0.) for output-only streams.
+     @see PaTime
+    */
+    PaTime inputLatency;
+
+    /** The output latency of the stream in seconds. This value provides the most
+     accurate estimate of output latency available to the implementation. It may
+     differ significantly from the suggestedLatency value passed to Pa_OpenStream().
+     The value of this field will be zero (0.) for input-only streams.
+     @see PaTime
+    */
+    PaTime outputLatency;
+
+    /** The sample rate of the stream in Hertz (samples per second). In cases
+     where the hardware sample rate is inaccurate and PortAudio is aware of it,
+     the value of this field may be different from the sampleRate parameter
+     passed to Pa_OpenStream(). If information about the actual hardware sample
+     rate is not available, this field will have the same value as the sampleRate
+     parameter passed to Pa_OpenStream().
+    */
+    double sampleRate;
+    
+} PaStreamInfo;
+
+
+/** Retrieve a pointer to a PaStreamInfo structure containing information
+ about the specified stream.
+ @return A pointer to an immutable PaStreamInfo structure. If the stream
+ parameter invalid, or an error is encountered, the function returns NULL.
+
+ @param stream A pointer to an open stream previously created with Pa_OpenStream.
+
+ @note PortAudio manages the memory referenced by the returned pointer,
+ the client must not manipulate or free the memory. The pointer is only
+ guaranteed to be valid until the specified stream is closed.
+
+ @see PaStreamInfo
+*/
+const PaStreamInfo* Pa_GetStreamInfo( PaStream *stream );
+
+
+/** Determine the current time for the stream according to the same clock used
+ to generate buffer timestamps. This time may be used for syncronising other
+ events to the audio stream, for example synchronizing audio to MIDI.
+                                        
+ @return The stream's current time in seconds, or 0 if an error occurred.
+
+ @see PaTime, PaStreamCallback
+*/
+PaTime Pa_GetStreamTime( PaStream *stream );
+
+
+/** Retrieve CPU usage information for the specified stream.
+ The "CPU Load" is a fraction of total CPU time consumed by a callback stream's
+ audio processing routines including, but not limited to the client supplied
+ stream callback. This function does not work with blocking read/write streams.
+
+ This function may be called from the stream callback function or the
+ application.
+     
+ @return
+ A floating point value, typically between 0.0 and 1.0, where 1.0 indicates
+ that the stream callback is consuming the maximum number of CPU cycles possible
+ to maintain real-time operation. A value of 0.5 would imply that PortAudio and
+ the stream callback was consuming roughly 50% of the available CPU time. The
+ return value may exceed 1.0. A value of 0.0 will always be returned for a
+ blocking read/write stream, or if an error occurrs.
+*/
+double Pa_GetStreamCpuLoad( PaStream* stream );
+
+
+/** Read samples from an input stream. The function doesn't return until
+ the entire buffer has been filled - this may involve waiting for the operating
+ system to supply the data.
+
+ @param stream A pointer to an open stream previously created with Pa_OpenStream.
+ 
+ @param buffer A pointer to a buffer of sample frames. The buffer contains
+ samples in the format specified by the inputParameters->sampleFormat field
+ used to open the stream, and the number of channels specified by
+ inputParameters->numChannels. If non-interleaved samples were requested,
+ buffer is a pointer to the first element of an array of non-interleaved
+ buffer pointers, one for each channel.
+
+ @param frames The number of frames to be read into buffer. This parameter
+ is not constrained to a specific range, however high performance applications
+ will want to match this parameter to the framesPerBuffer parameter used
+ when opening the stream.
+
+ @return On success PaNoError will be returned, or PaInputOverflowed if input
+ data was discarded by PortAudio after the previous call and before this call.
+*/
+PaError Pa_ReadStream( PaStream* stream,
+                       void *buffer,
+                       unsigned long frames );
+
+
+/** Write samples to an output stream. This function doesn't return until the
+ entire buffer has been consumed - this may involve waiting for the operating
+ system to consume the data.
+
+ @param stream A pointer to an open stream previously created with Pa_OpenStream.
+
+ @param buffer A pointer to a buffer of sample frames. The buffer contains
+ samples in the format specified by the outputParameters->sampleFormat field
+ used to open the stream, and the number of channels specified by
+ outputParameters->numChannels. If non-interleaved samples were requested,
+ buffer is a pointer to the first element of an array of non-interleaved
+ buffer pointers, one for each channel.
+
+ @param frames The number of frames to be written from buffer. This parameter
+ is not constrained to a specific range, however high performance applications
+ will want to match this parameter to the framesPerBuffer parameter used
+ when opening the stream.
+
+ @return On success PaNoError will be returned, or paOutputUnderflowed if
+ additional output data was inserted after the previous call and before this
+ call.
+*/
+PaError Pa_WriteStream( PaStream* stream,
+                        const void *buffer,
+                        unsigned long frames );
+
+
+/** Retrieve the number of frames that can be read from the stream without
+ waiting.
+
+ @return Returns a non-negative value representing the maximum number of frames
+ that can be read from the stream without blocking or busy waiting or, a
+ PaErrorCode (which are always negative) if PortAudio is not initialized or an
+ error is encountered.
+*/
+signed long Pa_GetStreamReadAvailable( PaStream* stream );
+
+
+/** Retrieve the number of frames that can be written to the stream without
+ waiting.
+
+ @return Returns a non-negative value representing the maximum number of frames
+ that can be written to the stream without blocking or busy waiting or, a
+ PaErrorCode (which are always negative) if PortAudio is not initialized or an
+ error is encountered.
+*/
+signed long Pa_GetStreamWriteAvailable( PaStream* stream );
+
+
+/* Miscellaneous utilities */
+
+
+/** Retrieve the size of a given sample format in bytes.
+
+ @return The size in bytes of a single sample in the specified format,
+ or paSampleFormatNotSupported if the format is not supported.
+*/
+PaError Pa_GetSampleSize( PaSampleFormat format );
+
+
+/** Put the caller to sleep for at least 'msec' milliseconds. This function is
+ provided only as a convenience for authors of portable code (such as the tests
+ and examples in the PortAudio distribution.)
+
+ The function may sleep longer than requested so don't rely on this for accurate
+ musical timing.
+*/
+void Pa_Sleep( long msec );
+
+
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif /* PORTAUDIO_H */