packages feed

proteaaudio (empty) → 0.6.2

raw patch · 28 files changed

+22769/−0 lines, 28 filesdep +basesetup-changed

Dependencies added: base

Files

+ LICENSE view
@@ -0,0 +1,28 @@+Copyright (c) 2012, Csaba Hruska+All rights reserved.++Redistribution and use in source and binary forms, with or without+modification, are permitted provided that the following conditions are met:++1. Redistributions of source code must retain the above copyright notice,+   this list of conditions and the following disclaimer.++2. Redistributions in binary form must reproduce the above copyright+   notice, this list of conditions and the following disclaimer in the+   documentation and/or other materials provided with the distribution.++3. Neither the name of the author nor the names of its contributors may be+   used to endorse or promote products derived from this software without+   specific prior written permission.++THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE+POSSIBILITY OF SUCH DAMAGE.
+ Setup.hs view
@@ -0,0 +1,2 @@+import Distribution.Simple+main = defaultMain
+ Sound/ProteaAudio.chs view
@@ -0,0 +1,37 @@+{-#LANGUAGE ForeignFunctionInterface#-}+#include "proteaaudio_binding.h"+module Sound.ProteaAudio (+    initAudio,+    finishAudio,+    loaderAvailable,+    volume,+    sampleFromFile,+    soundActive,+    soundStopAll,+    soundLoop,+    soundPlay,+    soundUpdate,+    soundStop,+    Sample()+ ) where++import Foreign+import Foreign.C++newtype Sample = Sample {#type sample_t#}++toSample s = Sample s+fromSample (Sample s) = s++{#fun initAudio {`Int', `Int', `Int'} -> `Bool'#}+{#fun finishAudio {} -> `()'#}+{#fun loaderAvailable {`String'} -> `Bool'#}+{#fun sampleFromFile {`String', `Float'} -> `Sample' toSample#}+{#fun volume {`Float', `Float'} -> `()'#}+{#fun soundActive {} -> `Int'#}+{#fun soundStopAll {} -> `()'#}++{#fun soundLoop {fromSample `Sample', `Float', `Float', `Float', `Float'} -> `()'#}+{#fun soundPlay {fromSample `Sample', `Float', `Float', `Float', `Float'} -> `()'#}+{#fun soundUpdate {fromSample `Sample', `Float', `Float', `Float', `Float'} -> `Bool'#}+{#fun soundStop {fromSample `Sample'} -> `Bool'#}
+ cbits/RtAudio.cpp view
@@ -0,0 +1,8350 @@+/************************************************************************/
+/*! \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) 2001-2012 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 4.0.11
+
+#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(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+  // pthread API
+  #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
+  #define MUTEX_DESTROY(A)    pthread_mutex_destroy(A)
+  #define MUTEX_LOCK(A)       pthread_mutex_lock(A)
+  #define MUTEX_UNLOCK(A)     pthread_mutex_unlock(A)
+#else
+  #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
+  #define MUTEX_DESTROY(A)    abs(*A) // dummy definitions
+#endif
+
+// *************************************************** //
+//
+// RtAudio definitions.
+//
+// *************************************************** //
+
+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_PULSE__)
+  apis.push_back( LINUX_PULSE );
+#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 ( rtapi_ )
+    delete rtapi_;
+  rtapi_ = 0;
+
+#if defined(__UNIX_JACK__)
+  if ( api == UNIX_JACK )
+    rtapi_ = new RtApiJack();
+#endif
+#if defined(__LINUX_ALSA__)
+  if ( api == LINUX_ALSA )
+    rtapi_ = new RtApiAlsa();
+#endif
+#if defined(__LINUX_PULSE__)
+  if ( api == LINUX_PULSE )
+    rtapi_ = new RtApiPulse();
+#endif
+#if defined(__LINUX_OSS__)
+  if ( api == LINUX_OSS )
+    rtapi_ = new RtApiOss();
+#endif
+#if defined(__WINDOWS_ASIO__)
+  if ( api == WINDOWS_ASIO )
+    rtapi_ = new RtApiAsio();
+#endif
+#if defined(__WINDOWS_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; }
+};
+
+ThreadHandle threadId;
+
+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;
+  }
+
+  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:
+  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;
+  }
+
+  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
+    }
+
+    result = AudioDeviceStop( handle->id[0], callbackHandler );
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+  }
+
+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+
+    result = AudioDeviceStop( handle->id[1], callbackHandler );
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+  }
+
+  stream_.state = STREAM_STOPPED;
+
+ unlock:
+  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();
+}
+
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted.  It is better to handle it this way because the
+// callbackEvent() function probably should return before the AudioDeviceStop()
+// function is called.
+extern "C" void *coreStopStream( void *ptr )
+{
+  CallbackInfo *info = (CallbackInfo *) ptr;
+  RtApiCore *object = (RtApiCore *) info->object;
+
+  object->stopStream();
+  pthread_exit( NULL );
+}
+
+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
+                                 const AudioBufferList *inBufferList,
+                                 const AudioBufferList *outBufferList )
+{
+  if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+    error( 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 ) {
+
+    stream_.state = STREAM_STOPPING;
+    if ( handle->internalDrain == true )
+      pthread_create( &threadId, NULL, coreStopStream, info );
+    else // external call to stopStream()
+      pthread_cond_signal( &handle->condition );
+    return SUCCESS;
+  }
+
+  AudioDeviceID outputDevice = handle->id[0];
+
+  // Invoke user callback to get fresh output data UNLESS we are
+  // draining stream or duplex mode AND the input/output devices are
+  // different AND this function is called for the input device.
+  if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
+    RtAudioCallback callback = (RtAudioCallback) info->callback;
+    double streamTime = getStreamTime();
+    RtAudioStreamStatus status = 0;
+    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+      status |= RTAUDIO_OUTPUT_UNDERFLOW;
+      handle->xrun[0] = false;
+    }
+    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+      status |= RTAUDIO_INPUT_OVERFLOW;
+      handle->xrun[1] = false;
+    }
+
+    int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                  stream_.bufferSize, streamTime, status, info->userData );
+    if ( cbReturnValue == 2 ) {
+      stream_.state = STREAM_STOPPING;
+      handle->drainCounter = 2;
+      abortStream();
+      return SUCCESS;
+    }
+    else if ( cbReturnValue == 1 ) {
+      handle->drainCounter = 1;
+      handle->internalDrain = true;
+    }
+  }
+
+  if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
+
+    if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+      if ( handle->nStreams[0] == 1 ) {
+        memset( outBufferList->mBuffers[handle->iStream[0]].mData,
+                0,
+                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+      }
+      else { // fill multiple streams with zeros
+        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+          memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+                  0,
+                  outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
+        }
+      }
+    }
+    else if ( handle->nStreams[0] == 1 ) {
+      if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
+        convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
+                       stream_.userBuffer[0], stream_.convertInfo[0] );
+      }
+      else { // copy from user buffer
+        memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
+                stream_.userBuffer[0],
+                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+      }
+    }
+    else { // fill multiple streams
+      Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
+      if ( stream_.doConvertBuffer[0] ) {
+        convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+        inBuffer = (Float32 *) stream_.deviceBuffer;
+      }
+
+      if ( stream_.deviceInterleaved[0] == false ) { // mono mode
+        UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
+        for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+          memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+                  (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
+        }
+      }
+      else { // fill multiple multi-channel streams with interleaved data
+        UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
+        Float32 *out, *in;
+
+        bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
+        UInt32 inChannels = stream_.nUserChannels[0];
+        if ( stream_.doConvertBuffer[0] ) {
+          inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
+          inChannels = stream_.nDeviceChannels[0];
+        }
+
+        if ( inInterleaved ) inOffset = 1;
+        else inOffset = stream_.bufferSize;
+
+        channelsLeft = inChannels;
+        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+          in = inBuffer;
+          out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
+          streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
+
+          outJump = 0;
+          // Account for possible channel offset in first stream
+          if ( i == 0 && stream_.channelOffset[0] > 0 ) {
+            streamChannels -= stream_.channelOffset[0];
+            outJump = stream_.channelOffset[0];
+            out += outJump;
+          }
+
+          // Account for possible unfilled channels at end of the last stream
+          if ( streamChannels > channelsLeft ) {
+            outJump = streamChannels - channelsLeft;
+            streamChannels = channelsLeft;
+          }
+
+          // Determine input buffer offsets and skips
+          if ( inInterleaved ) {
+            inJump = inChannels;
+            in += inChannels - channelsLeft;
+          }
+          else {
+            inJump = 1;
+            in += (inChannels - channelsLeft) * inOffset;
+          }
+
+          for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+            for ( unsigned int j=0; j<streamChannels; j++ ) {
+              *out++ = in[j*inOffset];
+            }
+            out += outJump;
+            in += inJump;
+          }
+          channelsLeft -= streamChannels;
+        }
+      }
+    }
+
+    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 ) {
+    jack_client_close( client );
+    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;
+  }
+
+  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:
+  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;
+  }
+
+  JackHandle *handle = (JackHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+    if ( handle->drainCounter == 0 ) {
+      handle->drainCounter = 2;
+      pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+    }
+  }
+
+  jack_deactivate( handle->client );
+  stream_.state = STREAM_STOPPED;
+}
+
+void RtApiJack :: abortStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+    error( 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 || stream_.state == STREAM_STOPPING ) 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 ) {
+
+    stream_.state = STREAM_STOPPING;
+    if ( handle->internalDrain == true )
+      pthread_create( &threadId, NULL, jackStopStream, info );
+    else
+      pthread_cond_signal( &handle->condition );
+    return SUCCESS;
+  }
+
+  // Invoke user callback first, to get fresh output data.
+  if ( handle->drainCounter == 0 ) {
+    RtAudioCallback callback = (RtAudioCallback) info->callback;
+    double streamTime = getStreamTime();
+    RtAudioStreamStatus status = 0;
+    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+      status |= RTAUDIO_OUTPUT_UNDERFLOW;
+      handle->xrun[0] = false;
+    }
+    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+      status |= RTAUDIO_INPUT_OVERFLOW;
+      handle->xrun[1] = false;
+    }
+    int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                  stream_.bufferSize, streamTime, status, info->userData );
+    if ( cbReturnValue == 2 ) {
+      stream_.state = STREAM_STOPPING;
+      handle->drainCounter = 2;
+      ThreadHandle id;
+      pthread_create( &id, NULL, jackStopStream, info );
+      return SUCCESS;
+    }
+    else if ( cbReturnValue == 1 ) {
+      handle->drainCounter = 1;
+      handle->internalDrain = true;
+    }
+  }
+
+  jack_default_audio_sample_t *jackbuffer;
+  unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+    if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+      for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+        memset( jackbuffer, 0, bufferBytes );
+      }
+
+    }
+    else if ( stream_.doConvertBuffer[0] ) {
+
+      convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+
+      for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+        memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
+      }
+    }
+    else { // no buffer conversion
+      for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+        memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+      }
+    }
+
+    if ( handle->drainCounter ) {
+      handle->drainCounter++;
+      goto unlock;
+    }
+  }
+
+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+    if ( stream_.doConvertBuffer[1] ) {
+      for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+        memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
+      }
+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+    }
+    else { // no buffer conversion
+      for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+        memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
+      }
+    }
+  }
+
+ unlock:
+  RtApi::tickStreamTime();
+  return SUCCESS;
+}
+  //******************** End of __UNIX_JACK__ *********************//
+#endif
+
+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
+
+// The ASIO API is designed around a callback scheme, so this
+// implementation is similar to that used for OS-X CoreAudio and Linux
+// Jack.  The primary constraint with ASIO is that it only allows
+// access to a single driver at a time.  Thus, it is not possible to
+// have more than one simultaneous RtAudio stream.
+//
+// This implementation also requires a number of external ASIO files
+// and a few global variables.  The ASIO callback scheme does not
+// allow for the passing of user data, so we must create a global
+// pointer to our callbackInfo structure.
+//
+// On unix systems, we make use of a pthread condition variable.
+// Since there is no equivalent in Windows, I hacked something based
+// on information found in
+// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
+
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
+#include <cmath>
+
+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;
+  }
+
+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+  ASIOError result = ASIOStart();
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
+    errorText_ = errorStream_.str();
+    goto unlock;
+  }
+
+  handle->drainCounter = 0;
+  handle->internalDrain = false;
+  ResetEvent( handle->condition );
+  stream_.state = STREAM_RUNNING;
+  asioXRun = false;
+
+ unlock:
+  stopThreadCalled = false;
+
+  if ( result == ASE_OK ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiAsio :: stopStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    if ( handle->drainCounter == 0 ) {
+      handle->drainCounter = 2;
+      WaitForSingleObject( handle->condition, INFINITE );  // block until signaled
+    }
+  }
+
+  stream_.state = STREAM_STOPPED;
+
+  ASIOError result = ASIOStop();
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
+    errorText_ = errorStream_.str();
+  }
+
+  if ( result == ASE_OK ) return;
+  error( 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 || stream_.state == STREAM_STOPPING ) 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 ) {
+
+    stream_.state = STREAM_STOPPING;
+    if ( handle->internalDrain == false )
+      SetEvent( handle->condition );
+    else { // spawn a thread to stop the stream
+      unsigned threadId;
+      stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+                                                    &stream_.callbackInfo, 0, &threadId );
+    }
+    return SUCCESS;
+  }
+
+  // Invoke user callback to get fresh output data UNLESS we are
+  // draining stream.
+  if ( handle->drainCounter == 0 ) {
+    RtAudioCallback callback = (RtAudioCallback) info->callback;
+    double streamTime = getStreamTime();
+    RtAudioStreamStatus status = 0;
+    if ( stream_.mode != INPUT && asioXRun == true ) {
+      status |= RTAUDIO_OUTPUT_UNDERFLOW;
+      asioXRun = false;
+    }
+    if ( stream_.mode != OUTPUT && asioXRun == true ) {
+      status |= RTAUDIO_INPUT_OVERFLOW;
+      asioXRun = false;
+    }
+    int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                     stream_.bufferSize, streamTime, status, info->userData );
+    if ( cbReturnValue == 2 ) {
+      stream_.state = STREAM_STOPPING;
+      handle->drainCounter = 2;
+      unsigned threadId;
+      stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+                                                    &stream_.callbackInfo, 0, &threadId );
+      return SUCCESS;
+    }
+    else if ( cbReturnValue == 1 ) {
+      handle->drainCounter = 1;
+      handle->internalDrain = true;
+    }
+  }
+
+  unsigned int nChannels, bufferBytes, i, j;
+  nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+    bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
+
+    if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+      for ( i=0, j=0; i<nChannels; i++ ) {
+        if ( handle->bufferInfos[i].isInput != ASIOTrue )
+          memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
+      }
+
+    }
+    else if ( stream_.doConvertBuffer[0] ) {
+
+      convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+      if ( stream_.doByteSwap[0] )
+        byteSwapBuffer( stream_.deviceBuffer,
+                        stream_.bufferSize * stream_.nDeviceChannels[0],
+                        stream_.deviceFormat[0] );
+
+      for ( i=0, j=0; i<nChannels; i++ ) {
+        if ( handle->bufferInfos[i].isInput != ASIOTrue )
+          memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+                  &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
+      }
+
+    }
+    else {
+
+      if ( stream_.doByteSwap[0] )
+        byteSwapBuffer( stream_.userBuffer[0],
+                        stream_.bufferSize * stream_.nUserChannels[0],
+                        stream_.userFormat );
+
+      for ( i=0, j=0; i<nChannels; i++ ) {
+        if ( handle->bufferInfos[i].isInput != ASIOTrue )
+          memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+                  &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
+      }
+
+    }
+
+    if ( handle->drainCounter ) {
+      handle->drainCounter++;
+      goto unlock;
+    }
+  }
+
+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+    bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
+
+    if (stream_.doConvertBuffer[1]) {
+
+      // Always interleave ASIO input data.
+      for ( i=0, j=0; i<nChannels; i++ ) {
+        if ( handle->bufferInfos[i].isInput == ASIOTrue )
+          memcpy( &stream_.deviceBuffer[j++*bufferBytes],
+                  handle->bufferInfos[i].buffers[bufferIndex],
+                  bufferBytes );
+      }
+
+      if ( stream_.doByteSwap[1] )
+        byteSwapBuffer( stream_.deviceBuffer,
+                        stream_.bufferSize * stream_.nDeviceChannels[1],
+                        stream_.deviceFormat[1] );
+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+
+    }
+    else {
+      for ( i=0, j=0; i<nChannels; i++ ) {
+        if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
+          memcpy( &stream_.userBuffer[1][bufferBytes*j++],
+                  handle->bufferInfos[i].buffers[bufferIndex],
+                  bufferBytes );
+        }
+      }
+
+      if ( stream_.doByteSwap[1] )
+        byteSwapBuffer( stream_.userBuffer[1],
+                        stream_.bufferSize * stream_.nUserChannels[1],
+                        stream_.userFormat );
+    }
+  }
+
+ unlock:
+  // The following call was suggested by Malte Clasen.  While the API
+  // documentation indicates it should not be required, some device
+  // drivers apparently do not function correctly without it.
+  ASIOOutputReady();
+
+  RtApi::tickStreamTime();
+  return SUCCESS;
+}
+
+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 );
+  unsigned int nErased = 0;
+  for ( unsigned int i=0; i<indices.size(); i++ )
+    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;
+  }
+
+  DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+  // Increase scheduler frequency on lesser windows (a side-effect of
+  // increasing timer accuracy).  On greater windows (Win2K or later),
+  // this is already in effect.
+  timeBeginPeriod( 1 ); 
+
+  buffersRolling = false;
+  duplexPrerollBytes = 0;
+
+  if ( stream_.mode == DUPLEX ) {
+    // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
+    duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
+  }
+
+  HRESULT result = 0;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+    LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+    result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+  }
+
+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+    LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+    result = buffer->Start( DSCBSTART_LOOPING );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+  }
+
+  handle->drainCounter = 0;
+  handle->internalDrain = false;
+  ResetEvent( handle->condition );
+  stream_.state = STREAM_RUNNING;
+
+ unlock:
+  if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: stopStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  HRESULT result = 0;
+  LPVOID audioPtr;
+  DWORD dataLen;
+  DsHandle *handle = (DsHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    if ( handle->drainCounter == 0 ) {
+      handle->drainCounter = 2;
+      WaitForSingleObject( handle->condition, INFINITE );  // block until signaled
+    }
+
+    stream_.state = STREAM_STOPPED;
+
+    // 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.
+  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 || stream_.state == STREAM_STOPPING ) {
+    Sleep( 50 ); // sleep 50 milliseconds
+    return;
+  }
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+    error( 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 ) {
+
+    stream_.state = STREAM_STOPPING;
+    if ( handle->internalDrain == false )
+      SetEvent( handle->condition );
+    else
+      stopStream();
+    return;
+  }
+
+  // Invoke user callback to get fresh output data UNLESS we are
+  // draining stream.
+  if ( handle->drainCounter == 0 ) {
+    RtAudioCallback callback = (RtAudioCallback) info->callback;
+    double streamTime = getStreamTime();
+    RtAudioStreamStatus status = 0;
+    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+      status |= RTAUDIO_OUTPUT_UNDERFLOW;
+      handle->xrun[0] = false;
+    }
+    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+      status |= RTAUDIO_INPUT_OVERFLOW;
+      handle->xrun[1] = false;
+    }
+    int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                  stream_.bufferSize, streamTime, status, info->userData );
+    if ( cbReturnValue == 2 ) {
+      stream_.state = STREAM_STOPPING;
+      handle->drainCounter = 2;
+      abortStream();
+      return;
+    }
+    else if ( cbReturnValue == 1 ) {
+      handle->drainCounter = 1;
+      handle->internalDrain = true;
+    }
+  }
+
+  HRESULT result;
+  DWORD currentWritePointer, safeWritePointer;
+  DWORD currentReadPointer, safeReadPointer;
+  UINT nextWritePointer;
+
+  LPVOID buffer1 = NULL;
+  LPVOID buffer2 = NULL;
+  DWORD bufferSize1 = 0;
+  DWORD bufferSize2 = 0;
+
+  char *buffer;
+  long bufferBytes;
+
+  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 && stream_.callbackInfo.isRunning ) {
+        // See comments for playback.
+        double millis = (endRead - safeReadPointer) * 1000.0;
+        millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+        if ( millis < 1.0 ) millis = 1.0;
+        Sleep( (DWORD) millis );
+
+        // Wake up and find out where we are now.
+        result = dsBuffer->GetCurrentPosition( &currentReadPointer, &safeReadPointer );
+        if ( FAILED( result ) ) {
+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+          errorText_ = errorStream_.str();
+          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:
+  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 ) ) {
+    snd_ctl_close( chandle );
+    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.
+  snd_pcm_close( phandle );
+  errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
+  errorText_ = errorStream_.str();
+  return FAILURE;
+
+ setFormat:
+  result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
+  if ( result < 0 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Determine whether byte-swaping is necessary.
+  stream_.doByteSwap[mode] = false;
+  if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
+    result = snd_pcm_format_cpu_endian( deviceFormat );
+    if ( result == 0 )
+      stream_.doByteSwap[mode] = true;
+    else if (result < 0) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+  }
+
+  // Set the sample rate.
+  result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
+  if ( result < 0 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Determine the number of channels for this device.  We support a possible
+  // minimum device channel number > than the value requested by the user.
+  stream_.nUserChannels[mode] = channels;
+  unsigned int value;
+  result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
+  unsigned int deviceChannels = value;
+  if ( result < 0 || deviceChannels < channels + firstChannel ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
+  if ( result < 0 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+  deviceChannels = value;
+  if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
+  stream_.nDeviceChannels[mode] = deviceChannels;
+
+  // Set the device channels.
+  result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
+  if ( result < 0 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Set the buffer (or period) size.
+  int dir = 0;
+  snd_pcm_uframes_t periodSize = *bufferSize;
+  result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
+  if ( result < 0 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+  *bufferSize = periodSize;
+
+  // Set the buffer number, which in ALSA is referred to as the "period".
+  unsigned int periods = 0;
+  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
+  if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
+  if ( periods < 2 ) periods = 4; // a fairly safe default value
+  result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
+  if ( result < 0 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // If attempting to setup a duplex stream, the bufferSize parameter
+  // MUST be the same in both directions!
+  if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  stream_.bufferSize = *bufferSize;
+
+  // Install the hardware configuration
+  result = snd_pcm_hw_params( phandle, hw_params );
+  if ( result < 0 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+#if defined(__RTAUDIO_DEBUG__)
+  fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
+  snd_pcm_hw_params_dump( hw_params, out );
+#endif
+
+  // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
+  snd_pcm_sw_params_t *sw_params = NULL;
+  snd_pcm_sw_params_alloca( &sw_params );
+  snd_pcm_sw_params_current( phandle, sw_params );
+  snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
+  snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
+  snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
+
+  // The following two settings were suggested by Theo Veenker
+  //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
+  //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
+
+  // here are two options for a fix
+  //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
+  snd_pcm_uframes_t val;
+  snd_pcm_sw_params_get_boundary( sw_params, &val );
+  snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
+
+  result = snd_pcm_sw_params( phandle, sw_params );
+  if ( result < 0 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+#if defined(__RTAUDIO_DEBUG__)
+  fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
+  snd_pcm_sw_params_dump( sw_params, out );
+#endif
+
+  // Set flags for buffer conversion
+  stream_.doConvertBuffer[mode] = false;
+  if ( stream_.userFormat != stream_.deviceFormat[mode] )
+    stream_.doConvertBuffer[mode] = true;
+  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+    stream_.doConvertBuffer[mode] = true;
+  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+       stream_.nUserChannels[mode] > 1 )
+    stream_.doConvertBuffer[mode] = true;
+
+  // Allocate the ApiHandle if necessary and then save.
+  AlsaHandle *apiInfo = 0;
+  if ( stream_.apiHandle == 0 ) {
+    try {
+      apiInfo = (AlsaHandle *) new AlsaHandle;
+    }
+    catch ( std::bad_alloc& ) {
+      errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
+      goto error;
+    }
+
+    if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
+      errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
+      goto error;
+    }
+
+    stream_.apiHandle = (void *) apiInfo;
+    apiInfo->handles[0] = 0;
+    apiInfo->handles[1] = 0;
+  }
+  else {
+    apiInfo = (AlsaHandle *) stream_.apiHandle;
+  }
+  apiInfo->handles[mode] = phandle;
+  phandle = 0;
+
+  // Allocate necessary internal buffers.
+  unsigned long bufferBytes;
+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+  if ( stream_.userBuffer[mode] == NULL ) {
+    errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
+    goto error;
+  }
+
+  if ( stream_.doConvertBuffer[mode] ) {
+
+    bool makeBuffer = true;
+    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+    if ( mode == INPUT ) {
+      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+        if ( bufferBytes <= bytesOut ) makeBuffer = false;
+      }
+    }
+
+    if ( makeBuffer ) {
+      bufferBytes *= *bufferSize;
+      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+      if ( stream_.deviceBuffer == NULL ) {
+        errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
+        goto error;
+      }
+    }
+  }
+
+  stream_.sampleRate = sampleRate;
+  stream_.nBuffers = periods;
+  stream_.device[mode] = device;
+  stream_.state = STREAM_STOPPED;
+
+  // Setup the buffer conversion information structure.
+  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+  // Setup thread if necessary.
+  if ( stream_.mode == OUTPUT && mode == INPUT ) {
+    // We had already set up an output stream.
+    stream_.mode = DUPLEX;
+    // Link the streams if possible.
+    apiInfo->synchronized = false;
+    if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
+      apiInfo->synchronized = true;
+    else {
+      errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
+      error( 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;
+  }
+
+  if ( phandle) snd_pcm_close( phandle );
+
+  for ( int i=0; i<2; i++ ) {
+    if ( stream_.userBuffer[i] ) {
+      free( stream_.userBuffer[i] );
+      stream_.userBuffer[i] = 0;
+    }
+  }
+
+  if ( stream_.deviceBuffer ) {
+    free( stream_.deviceBuffer );
+    stream_.deviceBuffer = 0;
+  }
+
+  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 );
+
+  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:
+  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 );
+
+  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:
+  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_PULSE__)
+
+// Code written by Peter Meerwald, pmeerw@pmeerw.net
+// and Tristan Matthews.
+
+#include <pulse/error.h>
+#include <pulse/simple.h>
+#include <cstdio>
+
+namespace {
+const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
+                                               44100, 48000, 96000, 0}; }
+
+struct rtaudio_pa_format_mapping_t {
+  RtAudioFormat rtaudio_format;
+  pa_sample_format_t pa_format;
+};
+
+static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
+  {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
+  {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
+  {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
+  {0, PA_SAMPLE_INVALID}};
+
+struct PulseAudioHandle {
+  pa_simple *s_play;
+  pa_simple *s_rec;
+  pthread_t thread;
+  pthread_cond_t runnable_cv;
+  bool runnable;
+  PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
+};
+
+RtApiPulse::~RtApiPulse()
+{
+  if ( stream_.state != STREAM_CLOSED )
+    closeStream();
+}
+
+unsigned int RtApiPulse::getDeviceCount( void )
+{
+  return 1;
+}
+
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device )
+{
+  RtAudio::DeviceInfo info;
+  info.probed = true;
+  info.name = "PulseAudio";
+  info.outputChannels = 2;
+  info.inputChannels = 2;
+  info.duplexChannels = 2;
+  info.isDefaultOutput = true;
+  info.isDefaultInput = true;
+
+  for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+    info.sampleRates.push_back( *sr );
+
+  info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
+
+  return info;
+}
+
+extern "C" void *pulseaudio_callback( void * user )
+{
+  CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
+  RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
+  volatile bool *isRunning = &cbi->isRunning;
+
+  while ( *isRunning ) {
+    pthread_testcancel();
+    context->callbackEvent();
+  }
+
+  pthread_exit( NULL );
+}
+
+void RtApiPulse::closeStream( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  stream_.callbackInfo.isRunning = false;
+  if ( pah ) {
+    MUTEX_LOCK( &stream_.mutex );
+    if ( stream_.state == STREAM_STOPPED ) {
+      pah->runnable = true;
+      pthread_cond_signal( &pah->runnable_cv );
+    }
+    MUTEX_UNLOCK( &stream_.mutex );
+
+    pthread_join( pah->thread, 0 );
+    if ( pah->s_play ) {
+      pa_simple_flush( pah->s_play, NULL );
+      pa_simple_free( pah->s_play );
+    }
+    if ( pah->s_rec )
+      pa_simple_free( pah->s_rec );
+
+    pthread_cond_destroy( &pah->runnable_cv );
+    delete pah;
+    stream_.apiHandle = 0;
+  }
+
+  if ( stream_.userBuffer[0] ) {
+    free( stream_.userBuffer[0] );
+    stream_.userBuffer[0] = 0;
+  }
+  if ( stream_.userBuffer[1] ) {
+    free( stream_.userBuffer[1] );
+    stream_.userBuffer[1] = 0;
+  }
+
+  stream_.state = STREAM_CLOSED;
+  stream_.mode = UNINITIALIZED;
+}
+
+void RtApiPulse::callbackEvent( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_LOCK( &stream_.mutex );
+    while ( !pah->runnable )
+      pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
+
+    if ( stream_.state != STREAM_RUNNING ) {
+      MUTEX_UNLOCK( &stream_.mutex );
+      return;
+    }
+    MUTEX_UNLOCK( &stream_.mutex );
+  }
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
+      "this shouldn't happen!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+  double streamTime = getStreamTime();
+  RtAudioStreamStatus status = 0;
+  int doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                               stream_.bufferSize, streamTime, status,
+                               stream_.callbackInfo.userData );
+
+  if ( doStopStream == 2 ) {
+    abortStream();
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( stream_.state != STREAM_RUNNING )
+    goto unlock;
+
+  int pa_error;
+  size_t bytes;
+  switch ( stream_.mode ) {
+  case INPUT:
+    bytes = stream_.nUserChannels[1] * stream_.bufferSize * formatBytes( stream_.userFormat );
+    if ( pa_simple_read( pah->s_rec, stream_.userBuffer[1], bytes, &pa_error ) < 0 ) {
+      errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
+        pa_strerror( pa_error ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+    }
+    break;
+  case OUTPUT:
+    bytes = stream_.nUserChannels[0] * stream_.bufferSize * formatBytes( stream_.userFormat );
+    if ( pa_simple_write( pah->s_play, stream_.userBuffer[0], bytes, &pa_error ) < 0 ) {
+      errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
+        pa_strerror( pa_error ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+    }
+    break;
+  case DUPLEX:
+    bytes = stream_.nUserChannels[1] * stream_.bufferSize * formatBytes( stream_.userFormat );
+    if ( pa_simple_read( pah->s_rec, stream_.userBuffer[1], bytes, &pa_error ) < 0 ) {
+      errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
+        pa_strerror( pa_error ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+    }
+    bytes = stream_.nUserChannels[0] * stream_.bufferSize * formatBytes( stream_.userFormat );
+    if ( pa_simple_write( pah->s_play, stream_.userBuffer[0], bytes, &pa_error ) < 0) {
+      errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
+        pa_strerror( pa_error ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+    }
+    break;
+  default:
+    // ERROR
+    break;
+  }
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+  RtApi::tickStreamTime();
+
+  if ( doStopStream == 1 )
+    stopStream();
+}
+
+void RtApiPulse::startStream( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiPulse::startStream(): the stream is not open!";
+    error( RtError::INVALID_USE );
+    return;
+  }
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiPulse::startStream(): the stream is already running!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  stream_.state = STREAM_RUNNING;
+
+  pah->runnable = true;
+  pthread_cond_signal( &pah->runnable_cv );
+  MUTEX_UNLOCK( &stream_.mutex );
+}
+
+void RtApiPulse::stopStream( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
+    error( RtError::INVALID_USE );
+    return;
+  }
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( pah && pah->s_play ) {
+    int pa_error;
+    if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
+      errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
+        pa_strerror( pa_error ) << ".";
+      errorText_ = errorStream_.str();
+      MUTEX_UNLOCK( &stream_.mutex );
+      error( RtError::SYSTEM_ERROR );
+    }
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_UNLOCK( &stream_.mutex );
+}
+
+void RtApiPulse::abortStream( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
+    error( RtError::INVALID_USE );
+    return;
+  }
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( pah && pah->s_play ) {
+    int pa_error;
+    if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
+      errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
+        pa_strerror( pa_error ) << ".";
+      errorText_ = errorStream_.str();
+      MUTEX_UNLOCK( &stream_.mutex );
+      error( RtError::SYSTEM_ERROR );
+    }
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_UNLOCK( &stream_.mutex );
+}
+
+bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
+                                  unsigned int channels, unsigned int firstChannel,
+                                  unsigned int sampleRate, RtAudioFormat format,
+                                  unsigned int *bufferSize, RtAudio::StreamOptions *options )
+{
+  PulseAudioHandle *pah = 0;
+  unsigned long bufferBytes = 0;
+  pa_sample_spec ss;
+
+  if ( device != 0 ) return false;
+  if ( mode != INPUT && mode != OUTPUT ) return false;
+  if ( channels != 1 && channels != 2 ) {
+    errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
+    return false;
+  }
+  ss.channels = channels;
+
+  if ( firstChannel != 0 ) return false;
+
+  bool sr_found = false;
+  for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
+    if ( sampleRate == *sr ) {
+      sr_found = true;
+      stream_.sampleRate = sampleRate;
+      ss.rate = sampleRate;
+      break;
+    }
+  }
+  if ( !sr_found ) {
+    errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
+    return false;
+  }
+
+  bool sf_found = 0;
+  for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
+        sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
+    if ( format == sf->rtaudio_format ) {
+      sf_found = true;
+      stream_.userFormat = sf->rtaudio_format;
+      ss.format = sf->pa_format;
+      break;
+    }
+  }
+  if ( !sf_found ) {
+    errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample format.";
+    return false;
+  }
+
+  if ( options && ( options->flags & RTAUDIO_NONINTERLEAVED ) ) {
+    errorText_ = "RtApiPulse::probeDeviceOpen: only interleaved audio data supported.";
+    return false;
+  }
+
+  stream_.userInterleaved = true;
+  stream_.nBuffers = 1;
+
+  stream_.deviceInterleaved[mode] = true;
+  stream_.doByteSwap[mode] = false;
+  stream_.doConvertBuffer[mode] = false;
+  stream_.deviceFormat[mode] = stream_.userFormat;
+  stream_.nUserChannels[mode] = channels;
+  stream_.nDeviceChannels[mode] = channels;
+  stream_.channelOffset[mode] = 0;
+
+  // Allocate necessary internal buffers.
+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+  if ( stream_.userBuffer[mode] == NULL ) {
+    errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
+    goto error;
+  }
+  stream_.bufferSize = *bufferSize;
+
+  if ( !stream_.apiHandle ) {
+    PulseAudioHandle *pah = new PulseAudioHandle;
+    if ( !pah ) {
+      errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
+      goto error;
+    }
+
+    stream_.apiHandle = pah;
+    if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
+      errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
+      goto error;
+    }
+  }
+  pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  int error;
+  switch ( mode ) {
+  case INPUT:
+    pah->s_rec = pa_simple_new( NULL, "RtAudio", PA_STREAM_RECORD, NULL, "Record", &ss, NULL, NULL, &error );
+    if ( !pah->s_rec ) {
+      errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
+      goto error;
+    }
+    break;
+  case OUTPUT:
+    pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
+    if ( !pah->s_play ) {
+      errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
+      goto error;
+    }
+    break;
+  default:
+    goto error;
+  }
+
+  if ( stream_.mode == UNINITIALIZED )
+    stream_.mode = mode;
+  else if ( stream_.mode == mode )
+    goto error;
+  else
+    stream_.mode = DUPLEX;
+
+  stream_.state = STREAM_STOPPED;
+
+  if ( !stream_.callbackInfo.isRunning ) {
+    stream_.callbackInfo.object = this;
+    stream_.callbackInfo.isRunning = true;
+    if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
+      errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
+      goto error;
+    }
+  }
+  return true;
+ 
+ error:
+  closeStream();
+  return false;
+}
+
+//******************** End of __LINUX_PULSE__ *********************//
+#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
+
+ cbits/RtAudio.h view
@@ -0,0 +1,1014 @@+/************************************************************************/+/*! \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) 2001-2012 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 4.0.11++#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_PULSE,    /*!< The Linux PulseAudio 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(__LINUX_PULSE__) || 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, 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_STOPPING,+    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_PULSE__)++class RtApiPulse: public RtApi+{+public:+  ~RtApiPulse();+  RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; };+  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
+ cbits/RtError.h view
@@ -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
+ cbits/include/asio.cpp view
@@ -0,0 +1,257 @@+/*
+	Steinberg Audio Stream I/O API
+	(c) 1996, Steinberg Soft- und Hardware GmbH
+
+	asio.cpp
+	
+	asio functions entries which translate the
+	asio interface to the asiodrvr class methods
+*/ 
+	
+#include <string.h>
+#include "asiosys.h"		// platform definition
+#include "asio.h"
+
+#if MAC
+#include "asiodrvr.h"
+
+#pragma export on
+
+AsioDriver *theAsioDriver = 0;
+
+extern "C"
+{
+
+long main()
+{
+	return 'ASIO';
+}
+
+#elif WINDOWS
+
+#include "windows.h"
+#include "iasiodrv.h"
+#include "asiodrivers.h"
+
+IASIO *theAsioDriver = 0;
+extern AsioDrivers *asioDrivers;
+
+#elif SGI || SUN || BEOS || LINUX
+#include "asiodrvr.h"
+static AsioDriver *theAsioDriver = 0;
+#endif
+
+//-----------------------------------------------------------------------------------------------------
+ASIOError ASIOInit(ASIODriverInfo *info)
+{
+#if MAC || SGI || SUN || BEOS || LINUX
+	if(theAsioDriver)
+	{
+		delete theAsioDriver;
+		theAsioDriver = 0;
+	}		
+	info->driverVersion = 0;
+	strcpy(info->name, "No ASIO Driver");
+	theAsioDriver = getDriver();
+	if(!theAsioDriver)
+	{
+		strcpy(info->errorMessage, "Not enough memory for the ASIO driver!"); 
+		return ASE_NotPresent;
+	}
+	if(!theAsioDriver->init(info->sysRef))
+	{
+		theAsioDriver->getErrorMessage(info->errorMessage);
+		delete theAsioDriver;
+		theAsioDriver = 0;
+		return ASE_NotPresent;
+	}
+	strcpy(info->errorMessage, "No ASIO Driver Error");
+	theAsioDriver->getDriverName(info->name);
+	info->driverVersion = theAsioDriver->getDriverVersion();
+	return ASE_OK;
+
+#else
+
+	info->driverVersion = 0;
+	strcpy(info->name, "No ASIO Driver");
+	if(theAsioDriver)	// must be loaded!
+	{
+		if(!theAsioDriver->init(info->sysRef))
+		{
+			theAsioDriver->getErrorMessage(info->errorMessage);
+			theAsioDriver = 0;
+			return ASE_NotPresent;
+		}		
+
+		strcpy(info->errorMessage, "No ASIO Driver Error");
+		theAsioDriver->getDriverName(info->name);
+		info->driverVersion = theAsioDriver->getDriverVersion();
+		return ASE_OK;
+	}
+	return ASE_NotPresent;
+
+#endif	// !MAC
+}
+
+ASIOError ASIOExit(void)
+{
+	if(theAsioDriver)
+	{
+#if WINDOWS
+		asioDrivers->removeCurrentDriver();
+#else
+		delete theAsioDriver;
+#endif
+	}		
+	theAsioDriver = 0;
+	return ASE_OK;
+}
+
+ASIOError ASIOStart(void)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->start();
+}
+
+ASIOError ASIOStop(void)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->stop();
+}
+
+ASIOError ASIOGetChannels(long *numInputChannels, long *numOutputChannels)
+{
+	if(!theAsioDriver)
+	{
+		*numInputChannels = *numOutputChannels = 0;
+		return ASE_NotPresent;
+	}
+	return theAsioDriver->getChannels(numInputChannels, numOutputChannels);
+}
+
+ASIOError ASIOGetLatencies(long *inputLatency, long *outputLatency)
+{
+	if(!theAsioDriver)
+	{
+		*inputLatency = *outputLatency = 0;
+		return ASE_NotPresent;
+	}
+	return theAsioDriver->getLatencies(inputLatency, outputLatency);
+}
+
+ASIOError ASIOGetBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity)
+{
+	if(!theAsioDriver)
+	{
+		*minSize = *maxSize = *preferredSize = *granularity = 0;
+		return ASE_NotPresent;
+	}
+	return theAsioDriver->getBufferSize(minSize, maxSize, preferredSize, granularity);
+}
+
+ASIOError ASIOCanSampleRate(ASIOSampleRate sampleRate)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->canSampleRate(sampleRate);
+}
+
+ASIOError ASIOGetSampleRate(ASIOSampleRate *currentRate)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->getSampleRate(currentRate);
+}
+
+ASIOError ASIOSetSampleRate(ASIOSampleRate sampleRate)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->setSampleRate(sampleRate);
+}
+
+ASIOError ASIOGetClockSources(ASIOClockSource *clocks, long *numSources)
+{
+	if(!theAsioDriver)
+	{
+		*numSources = 0;
+		return ASE_NotPresent;
+	}
+	return theAsioDriver->getClockSources(clocks, numSources);
+}
+
+ASIOError ASIOSetClockSource(long reference)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->setClockSource(reference);
+}
+
+ASIOError ASIOGetSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->getSamplePosition(sPos, tStamp);
+}
+
+ASIOError ASIOGetChannelInfo(ASIOChannelInfo *info)
+{
+	if(!theAsioDriver)
+	{
+		info->channelGroup = -1;
+		info->type = ASIOSTInt16MSB;
+		strcpy(info->name, "None");
+		return ASE_NotPresent;
+	}
+	return theAsioDriver->getChannelInfo(info);
+}
+
+ASIOError ASIOCreateBuffers(ASIOBufferInfo *bufferInfos, long numChannels,
+	long bufferSize, ASIOCallbacks *callbacks)
+{
+	if(!theAsioDriver)
+	{
+		ASIOBufferInfo *info = bufferInfos;
+		for(long i = 0; i < numChannels; i++, info++)
+			info->buffers[0] = info->buffers[1] = 0;
+		return ASE_NotPresent;
+	}
+	return theAsioDriver->createBuffers(bufferInfos, numChannels, bufferSize, callbacks);
+}
+
+ASIOError ASIODisposeBuffers(void)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->disposeBuffers();
+}
+
+ASIOError ASIOControlPanel(void)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->controlPanel();
+}
+
+ASIOError ASIOFuture(long selector, void *opt)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->future(selector, opt);
+}
+
+ASIOError ASIOOutputReady(void)
+{
+	if(!theAsioDriver)
+		return ASE_NotPresent;
+	return theAsioDriver->outputReady();
+}
+
+#if MAC
+}	// extern "C"
+#pragma export off
+#endif
+
+
+ cbits/include/asio.h view
@@ -0,0 +1,1054 @@+//---------------------------------------------------------------------------------------------------
+//---------------------------------------------------------------------------------------------------
+
+/*
+	Steinberg Audio Stream I/O API
+	(c) 1997 - 2005, Steinberg Media Technologies GmbH
+
+	ASIO Interface Specification v 2.1
+
+	2005 - Added support for DSD sample data (in cooperation with Sony)
+
+
+	basic concept is an i/o synchronous double-buffer scheme:
+	
+	on bufferSwitch(index == 0), host will read/write:
+
+		after ASIOStart(), the
+  read  first input buffer A (index 0)
+	|   will be invalid (empty)
+	*   ------------------------
+	|------------------------|-----------------------|
+	|                        |                       |
+	|  Input Buffer A (0)    |   Input Buffer B (1)  |
+	|                        |                       |
+	|------------------------|-----------------------|
+	|                        |                       |
+	|  Output Buffer A (0)   |   Output Buffer B (1) |
+	|                        |                       |
+	|------------------------|-----------------------|
+	*                        -------------------------
+	|                        before calling ASIOStart(),
+  write                      host will have filled output
+                             buffer B (index 1) already
+
+  *please* take special care of proper statement of input
+  and output latencies (see ASIOGetLatencies()), these
+  control sequencer sync accuracy
+
+*/
+
+//---------------------------------------------------------------------------------------------------
+//---------------------------------------------------------------------------------------------------
+
+/*
+
+prototypes summary:
+
+ASIOError ASIOInit(ASIODriverInfo *info);
+ASIOError ASIOExit(void);
+ASIOError ASIOStart(void);
+ASIOError ASIOStop(void);
+ASIOError ASIOGetChannels(long *numInputChannels, long *numOutputChannels);
+ASIOError ASIOGetLatencies(long *inputLatency, long *outputLatency);
+ASIOError ASIOGetBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity);
+ASIOError ASIOCanSampleRate(ASIOSampleRate sampleRate);
+ASIOError ASIOGetSampleRate(ASIOSampleRate *currentRate);
+ASIOError ASIOSetSampleRate(ASIOSampleRate sampleRate);
+ASIOError ASIOGetClockSources(ASIOClockSource *clocks, long *numSources);
+ASIOError ASIOSetClockSource(long reference);
+ASIOError ASIOGetSamplePosition (ASIOSamples *sPos, ASIOTimeStamp *tStamp);
+ASIOError ASIOGetChannelInfo(ASIOChannelInfo *info);
+ASIOError ASIOCreateBuffers(ASIOBufferInfo *bufferInfos, long numChannels,
+	long bufferSize, ASIOCallbacks *callbacks);
+ASIOError ASIODisposeBuffers(void);
+ASIOError ASIOControlPanel(void);
+void *ASIOFuture(long selector, void *params);
+ASIOError ASIOOutputReady(void);
+
+*/
+
+//---------------------------------------------------------------------------------------------------
+//---------------------------------------------------------------------------------------------------
+
+#ifndef __ASIO_H
+#define __ASIO_H
+
+// force 4 byte alignment
+#if defined(_MSC_VER) && !defined(__MWERKS__) 
+#pragma pack(push,4)
+#elif PRAGMA_ALIGN_SUPPORTED
+#pragma options align = native
+#endif
+
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+// Type definitions
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+
+// number of samples data type is 64 bit integer
+#if NATIVE_INT64
+	typedef long long int ASIOSamples;
+#else
+	typedef struct ASIOSamples {
+		unsigned long hi;
+		unsigned long lo;
+	} ASIOSamples;
+#endif
+
+// Timestamp data type is 64 bit integer,
+// Time format is Nanoseconds.
+#if NATIVE_INT64
+	typedef long long int ASIOTimeStamp ;
+#else
+	typedef struct ASIOTimeStamp {
+		unsigned long hi;
+		unsigned long lo;
+	} ASIOTimeStamp;
+#endif
+
+// Samplerates are expressed in IEEE 754 64 bit double float,
+// native format as host computer
+#if IEEE754_64FLOAT
+	typedef double ASIOSampleRate;
+#else
+	typedef struct ASIOSampleRate {
+		char ieee[8];
+	} ASIOSampleRate;
+#endif
+
+// Boolean values are expressed as long
+typedef long ASIOBool;
+enum {
+	ASIOFalse = 0,
+	ASIOTrue = 1
+};
+
+// Sample Types are expressed as long
+typedef long ASIOSampleType;
+enum {
+	ASIOSTInt16MSB   = 0,
+	ASIOSTInt24MSB   = 1,		// used for 20 bits as well
+	ASIOSTInt32MSB   = 2,
+	ASIOSTFloat32MSB = 3,		// IEEE 754 32 bit float
+	ASIOSTFloat64MSB = 4,		// IEEE 754 64 bit double float
+
+	// these are used for 32 bit data buffer, with different alignment of the data inside
+	// 32 bit PCI bus systems can be more easily used with these
+	ASIOSTInt32MSB16 = 8,		// 32 bit data with 16 bit alignment
+	ASIOSTInt32MSB18 = 9,		// 32 bit data with 18 bit alignment
+	ASIOSTInt32MSB20 = 10,		// 32 bit data with 20 bit alignment
+	ASIOSTInt32MSB24 = 11,		// 32 bit data with 24 bit alignment
+	
+	ASIOSTInt16LSB   = 16,
+	ASIOSTInt24LSB   = 17,		// used for 20 bits as well
+	ASIOSTInt32LSB   = 18,
+	ASIOSTFloat32LSB = 19,		// IEEE 754 32 bit float, as found on Intel x86 architecture
+	ASIOSTFloat64LSB = 20, 		// IEEE 754 64 bit double float, as found on Intel x86 architecture
+
+	// these are used for 32 bit data buffer, with different alignment of the data inside
+	// 32 bit PCI bus systems can more easily used with these
+	ASIOSTInt32LSB16 = 24,		// 32 bit data with 18 bit alignment
+	ASIOSTInt32LSB18 = 25,		// 32 bit data with 18 bit alignment
+	ASIOSTInt32LSB20 = 26,		// 32 bit data with 20 bit alignment
+	ASIOSTInt32LSB24 = 27,		// 32 bit data with 24 bit alignment
+
+	//	ASIO DSD format.
+	ASIOSTDSDInt8LSB1   = 32,		// DSD 1 bit data, 8 samples per byte. First sample in Least significant bit.
+	ASIOSTDSDInt8MSB1   = 33,		// DSD 1 bit data, 8 samples per byte. First sample in Most significant bit.
+	ASIOSTDSDInt8NER8	= 40,		// DSD 8 bit data, 1 sample per byte. No Endianness required.
+
+	ASIOSTLastEntry
+};
+
+/*-----------------------------------------------------------------------------
+// DSD operation and buffer layout
+// Definition by Steinberg/Sony Oxford.
+//
+// We have tried to treat DSD as PCM and so keep a consistant structure across
+// the ASIO interface.
+//
+// DSD's sample rate is normally referenced as a multiple of 44.1Khz, so
+// the standard sample rate is refered to as 64Fs (or 2.8224Mhz). We looked
+// at making a special case for DSD and adding a field to the ASIOFuture that
+// would allow the user to select the Over Sampleing Rate (OSR) as a seperate
+// entity but decided in the end just to treat it as a simple value of
+// 2.8224Mhz and use the standard interface to set it.
+//
+// The second problem was the "word" size, in PCM the word size is always a
+// greater than or equal to 8 bits (a byte). This makes life easy as we can
+// then pack the samples into the "natural" size for the machine.
+// In DSD the "word" size is 1 bit. This is not a major problem and can easily
+// be dealt with if we ensure that we always deal with a multiple of 8 samples.
+//
+// DSD brings with it another twist to the Endianness religion. How are the
+// samples packed into the byte. It would be nice to just say the most significant
+// bit is always the first sample, however there would then be a performance hit
+// on little endian machines. Looking at how some of the processing goes...
+// Little endian machines like the first sample to be in the Least Significant Bit,
+//   this is because when you write it to memory the data is in the correct format
+//   to be shifted in and out of the words.
+// Big endian machine prefer the first sample to be in the Most Significant Bit,
+//   again for the same reasion.
+//
+// And just when things were looking really muddy there is a proposed extension to
+// DSD that uses 8 bit word sizes. It does not care what endianness you use.
+//
+// Switching the driver between DSD and PCM mode
+// ASIOFuture allows for extending the ASIO API quite transparently.
+// See kAsioSetIoFormat, kAsioGetIoFormat, kAsioCanDoIoFormat
+//
+//-----------------------------------------------------------------------------*/
+
+
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+// Error codes
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+
+typedef long ASIOError;
+enum {
+	ASE_OK = 0,             // This value will be returned whenever the call succeeded
+	ASE_SUCCESS = 0x3f4847a0,	// unique success return value for ASIOFuture calls
+	ASE_NotPresent = -1000, // hardware input or output is not present or available
+	ASE_HWMalfunction,      // hardware is malfunctioning (can be returned by any ASIO function)
+	ASE_InvalidParameter,   // input parameter invalid
+	ASE_InvalidMode,        // hardware is in a bad mode or used in a bad mode
+	ASE_SPNotAdvancing,     // hardware is not running when sample position is inquired
+	ASE_NoClock,            // sample clock or rate cannot be determined or is not present
+	ASE_NoMemory            // not enough memory for completing the request
+};
+
+//---------------------------------------------------------------------------------------------------
+//---------------------------------------------------------------------------------------------------
+
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+// Time Info support
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+
+typedef struct ASIOTimeCode
+{       
+	double          speed;                  // speed relation (fraction of nominal speed)
+	                                        // optional; set to 0. or 1. if not supported
+	ASIOSamples     timeCodeSamples;        // time in samples
+	unsigned long   flags;                  // some information flags (see below)
+	char future[64];
+} ASIOTimeCode;
+
+typedef enum ASIOTimeCodeFlags
+{
+	kTcValid                = 1,
+	kTcRunning              = 1 << 1,
+	kTcReverse              = 1 << 2,
+	kTcOnspeed              = 1 << 3,
+	kTcStill                = 1 << 4,
+	
+	kTcSpeedValid           = 1 << 8
+}  ASIOTimeCodeFlags;
+
+typedef struct AsioTimeInfo
+{
+	double          speed;                  // absolute speed (1. = nominal)
+	ASIOTimeStamp   systemTime;             // system time related to samplePosition, in nanoseconds
+	                                        // on mac, must be derived from Microseconds() (not UpTime()!)
+	                                        // on windows, must be derived from timeGetTime()
+	ASIOSamples     samplePosition;
+	ASIOSampleRate  sampleRate;             // current rate
+	unsigned long flags;                    // (see below)
+	char reserved[12];
+} AsioTimeInfo;
+
+typedef enum AsioTimeInfoFlags
+{
+	kSystemTimeValid        = 1,            // must always be valid
+	kSamplePositionValid    = 1 << 1,       // must always be valid
+	kSampleRateValid        = 1 << 2,
+	kSpeedValid             = 1 << 3,
+	
+	kSampleRateChanged      = 1 << 4,
+	kClockSourceChanged     = 1 << 5
+} AsioTimeInfoFlags;
+
+typedef struct ASIOTime                          // both input/output
+{
+	long reserved[4];                       // must be 0
+	struct AsioTimeInfo     timeInfo;       // required
+	struct ASIOTimeCode     timeCode;       // optional, evaluated if (timeCode.flags & kTcValid)
+} ASIOTime;
+
+/*
+
+using time info:
+it is recommended to use the new method with time info even if the asio
+device does not support timecode; continuous calls to ASIOGetSamplePosition
+and ASIOGetSampleRate are avoided, and there is a more defined relationship
+between callback time and the time info.
+
+see the example below.
+to initiate time info mode, after you have received the callbacks pointer in
+ASIOCreateBuffers, you will call the asioMessage callback with kAsioSupportsTimeInfo
+as the argument. if this returns 1, host has accepted time info mode.
+now host expects the new callback bufferSwitchTimeInfo to be used instead
+of the old bufferSwitch method. the ASIOTime structure is assumed to be valid
+and accessible until the callback returns.
+
+using time code:
+if the device supports reading time code, it will call host's asioMessage callback
+with kAsioSupportsTimeCode as the selector. it may then fill the according
+fields and set the kTcValid flag.
+host will call the future method with the kAsioEnableTimeCodeRead selector when
+it wants to enable or disable tc reading by the device. you should also support
+the kAsioCanTimeInfo and kAsioCanTimeCode selectors in ASIOFuture (see example).
+
+note:
+the AsioTimeInfo/ASIOTimeCode pair is supposed to work in both directions.
+as a matter of convention, the relationship between the sample
+position counter and the time code at buffer switch time is
+(ignoring offset between tc and sample pos when tc is running):
+
+on input:	sample 0 -> input  buffer sample 0 -> time code 0
+on output:	sample 0 -> output buffer sample 0 -> time code 0
+
+this means that for 'real' calculations, one has to take into account
+the according latencies.
+
+example:
+
+ASIOTime asioTime;
+
+in createBuffers()
+{
+	memset(&asioTime, 0, sizeof(ASIOTime));
+	AsioTimeInfo* ti = &asioTime.timeInfo;
+	ti->sampleRate = theSampleRate;
+	ASIOTimeCode* tc = &asioTime.timeCode;
+	tc->speed = 1.;
+	timeInfoMode = false;
+	canTimeCode = false;
+	if(callbacks->asioMessage(kAsioSupportsTimeInfo, 0, 0, 0) == 1)
+	{
+		timeInfoMode = true;
+#if kCanTimeCode
+		if(callbacks->asioMessage(kAsioSupportsTimeCode, 0, 0, 0) == 1)
+			canTimeCode = true;
+#endif
+	}
+}
+
+void switchBuffers(long doubleBufferIndex, bool processNow)
+{
+	if(timeInfoMode)
+	{
+		AsioTimeInfo* ti = &asioTime.timeInfo;
+		ti->flags =	kSystemTimeValid | kSamplePositionValid | kSampleRateValid;
+		ti->systemTime = theNanoSeconds;
+		ti->samplePosition = theSamplePosition;
+		if(ti->sampleRate != theSampleRate)
+			ti->flags |= kSampleRateChanged;
+		ti->sampleRate = theSampleRate;
+
+#if kCanTimeCode
+		if(canTimeCode && timeCodeEnabled)
+		{
+			ASIOTimeCode* tc = &asioTime.timeCode;
+			tc->timeCodeSamples = tcSamples;						// tc in samples
+			tc->flags = kTcValid | kTcRunning | kTcOnspeed;			// if so...
+		}
+		ASIOTime* bb = callbacks->bufferSwitchTimeInfo(&asioTime, doubleBufferIndex, processNow ? ASIOTrue : ASIOFalse);
+#else
+		callbacks->bufferSwitchTimeInfo(&asioTime, doubleBufferIndex, processNow ? ASIOTrue : ASIOFalse);
+#endif
+	}
+	else
+		callbacks->bufferSwitch(doubleBufferIndex, ASIOFalse);
+}
+
+ASIOError ASIOFuture(long selector, void *params)
+{
+	switch(selector)
+	{
+		case kAsioEnableTimeCodeRead:
+			timeCodeEnabled = true;
+			return ASE_SUCCESS;
+		case kAsioDisableTimeCodeRead:
+			timeCodeEnabled = false;
+			return ASE_SUCCESS;
+		case kAsioCanTimeInfo:
+			return ASE_SUCCESS;
+		#if kCanTimeCode
+		case kAsioCanTimeCode:
+			return ASE_SUCCESS;
+		#endif
+	}
+	return ASE_NotPresent;
+};
+
+*/
+
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+// application's audio stream handler callbacks
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+
+typedef struct ASIOCallbacks
+{
+	void (*bufferSwitch) (long doubleBufferIndex, ASIOBool directProcess);
+		// bufferSwitch indicates that both input and output are to be processed.
+		// the current buffer half index (0 for A, 1 for B) determines
+		// - the output buffer that the host should start to fill. the other buffer
+		//   will be passed to output hardware regardless of whether it got filled
+		//   in time or not.
+		// - the input buffer that is now filled with incoming data. Note that
+		//   because of the synchronicity of i/o, the input always has at
+		//   least one buffer latency in relation to the output.
+		// directProcess suggests to the host whether it should immedeately
+		// start processing (directProcess == ASIOTrue), or whether its process
+		// should be deferred because the call comes from a very low level
+		// (for instance, a high level priority interrupt), and direct processing
+		// would cause timing instabilities for the rest of the system. If in doubt,
+		// directProcess should be set to ASIOFalse.
+		// Note: bufferSwitch may be called at interrupt time for highest efficiency.
+
+	void (*sampleRateDidChange) (ASIOSampleRate sRate);
+		// gets called when the AudioStreamIO detects a sample rate change
+		// If sample rate is unknown, 0 is passed (for instance, clock loss
+		// when externally synchronized).
+
+	long (*asioMessage) (long selector, long value, void* message, double* opt);
+		// generic callback for various purposes, see selectors below.
+		// note this is only present if the asio version is 2 or higher
+
+	ASIOTime* (*bufferSwitchTimeInfo) (ASIOTime* params, long doubleBufferIndex, ASIOBool directProcess);
+		// new callback with time info. makes ASIOGetSamplePosition() and various
+		// calls to ASIOGetSampleRate obsolete,
+		// and allows for timecode sync etc. to be preferred; will be used if
+		// the driver calls asioMessage with selector kAsioSupportsTimeInfo.
+} ASIOCallbacks;
+
+// asioMessage selectors
+enum
+{
+	kAsioSelectorSupported = 1,	// selector in <value>, returns 1L if supported,
+								// 0 otherwise
+    kAsioEngineVersion,			// returns engine (host) asio implementation version,
+								// 2 or higher
+	kAsioResetRequest,			// request driver reset. if accepted, this
+								// will close the driver (ASIO_Exit() ) and
+								// re-open it again (ASIO_Init() etc). some
+								// drivers need to reconfigure for instance
+								// when the sample rate changes, or some basic
+								// changes have been made in ASIO_ControlPanel().
+								// returns 1L; note the request is merely passed
+								// to the application, there is no way to determine
+								// if it gets accepted at this time (but it usually
+								// will be).
+	kAsioBufferSizeChange,		// not yet supported, will currently always return 0L.
+								// for now, use kAsioResetRequest instead.
+								// once implemented, the new buffer size is expected
+								// in <value>, and on success returns 1L
+	kAsioResyncRequest,			// the driver went out of sync, such that
+								// the timestamp is no longer valid. this
+								// is a request to re-start the engine and
+								// slave devices (sequencer). returns 1 for ok,
+								// 0 if not supported.
+	kAsioLatenciesChanged, 		// the drivers latencies have changed. The engine
+								// will refetch the latencies.
+	kAsioSupportsTimeInfo,		// if host returns true here, it will expect the
+								// callback bufferSwitchTimeInfo to be called instead
+								// of bufferSwitch
+	kAsioSupportsTimeCode,		// 
+	kAsioMMCCommand,			// unused - value: number of commands, message points to mmc commands
+	kAsioSupportsInputMonitor,	// kAsioSupportsXXX return 1 if host supports this
+	kAsioSupportsInputGain,     // unused and undefined
+	kAsioSupportsInputMeter,    // unused and undefined
+	kAsioSupportsOutputGain,    // unused and undefined
+	kAsioSupportsOutputMeter,   // unused and undefined
+	kAsioOverload,              // driver detected an overload
+
+	kAsioNumMessageSelectors
+};
+
+//---------------------------------------------------------------------------------------------------
+//---------------------------------------------------------------------------------------------------
+
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+// (De-)Construction
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+
+typedef struct ASIODriverInfo
+{
+	long asioVersion;		// currently, 2
+	long driverVersion;		// driver specific
+	char name[32];
+	char errorMessage[124];
+	void *sysRef;			// on input: system reference
+							// (Windows: application main window handle, Mac & SGI: 0)
+} ASIODriverInfo;
+
+ASIOError ASIOInit(ASIODriverInfo *info);
+/* Purpose:
+	  Initialize the AudioStreamIO.
+	Parameter:
+	  info: pointer to an ASIODriver structure:
+	    - asioVersion:
+			- on input, the host version. *** Note *** this is 0 for earlier asio
+			implementations, and the asioMessage callback is implemeted
+			only if asioVersion is 2 or greater. sorry but due to a design fault
+			the driver doesn't have access to the host version in ASIOInit :-(
+			added selector for host (engine) version in the asioMessage callback
+			so we're ok from now on.
+			- on return, asio implementation version.
+			  older versions are 1
+			  if you support this version (namely, ASIO_outputReady() )
+			  this should be 2 or higher. also see the note in
+			  ASIO_getTimeStamp() !
+	    - version: on return, the driver version (format is driver specific)
+	    - name: on return, a null-terminated string containing the driver's name
+		- error message: on return, should contain a user message describing
+		  the type of error that occured during ASIOInit(), if any.
+		- sysRef: platform specific
+	Returns:
+	  If neither input nor output is present ASE_NotPresent
+	  will be returned.
+	  ASE_NoMemory, ASE_HWMalfunction are other possible error conditions
+*/
+
+ASIOError ASIOExit(void);
+/* Purpose:
+	  Terminates the AudioStreamIO.
+	Parameter:
+	  None.
+	Returns:
+	  If neither input nor output is present ASE_NotPresent
+	  will be returned.
+	Notes: this implies ASIOStop() and ASIODisposeBuffers(),
+	  meaning that no host callbacks must be accessed after ASIOExit().
+*/
+
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+// Start/Stop
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+
+ASIOError ASIOStart(void);
+/* Purpose:
+	  Start input and output processing synchronously.
+	  This will
+	  - reset the sample counter to zero
+	  - start the hardware (both input and output)
+	    The first call to the hosts' bufferSwitch(index == 0) then tells
+	    the host to read from input buffer A (index 0), and start
+	    processing to output buffer A while output buffer B (which
+	    has been filled by the host prior to calling ASIOStart())
+	    is possibly sounding (see also ASIOGetLatencies()) 
+	Parameter:
+	  None.
+	Returns:
+	  If neither input nor output is present, ASE_NotPresent
+	  will be returned.
+	  If the hardware fails to start, ASE_HWMalfunction will be returned.
+	Notes:
+	  There is no restriction on the time that ASIOStart() takes
+	  to perform (that is, it is not considered a realtime trigger).
+*/
+
+ASIOError ASIOStop(void);
+/* Purpose:
+	  Stops input and output processing altogether.
+	Parameter:
+	  None.
+	Returns:
+	  If neither input nor output is present ASE_NotPresent
+	  will be returned.
+	Notes:
+	  On return from ASIOStop(), the driver must in no
+	  case call the hosts' bufferSwitch() routine.
+*/
+
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+// Inquiry methods and sample rate
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+
+ASIOError ASIOGetChannels(long *numInputChannels, long *numOutputChannels);
+/* Purpose:
+	  Returns number of individual input/output channels.
+	Parameter:
+	  numInputChannels will hold the number of available input channels
+	  numOutputChannels will hold the number of available output channels
+	Returns:
+	  If no input/output is present ASE_NotPresent will be returned.
+	  If only inputs, or only outputs are available, the according
+	  other parameter will be zero, and ASE_OK is returned.
+*/
+
+ASIOError ASIOGetLatencies(long *inputLatency, long *outputLatency);
+/* Purpose:
+	  Returns the input and output latencies. This includes
+	  device specific delays, like FIFOs etc.
+	Parameter:
+	  inputLatency will hold the 'age' of the first sample frame
+	  in the input buffer when the hosts reads it in bufferSwitch()
+	  (this is theoretical, meaning it does not include the overhead
+	  and delay between the actual physical switch, and the time
+	  when bufferSitch() enters).
+	  This will usually be the size of one block in sample frames, plus
+	  device specific latencies.
+
+	  outputLatency will specify the time between the buffer switch,
+	  and the time when the next play buffer will start to sound.
+	  The next play buffer is defined as the one the host starts
+	  processing after (or at) bufferSwitch(), indicated by the
+	  index parameter (0 for buffer A, 1 for buffer B).
+	  It will usually be either one block, if the host writes directly
+	  to a dma buffer, or two or more blocks if the buffer is 'latched' by
+	  the driver. As an example, on ASIOStart(), the host will have filled
+	  the play buffer at index 1 already; when it gets the callback (with
+	  the parameter index == 0), this tells it to read from the input
+	  buffer 0, and start to fill the play buffer 0 (assuming that now
+	  play buffer 1 is already sounding). In this case, the output
+	  latency is one block. If the driver decides to copy buffer 1
+	  at that time, and pass it to the hardware at the next slot (which
+	  is most commonly done, but should be avoided), the output latency
+	  becomes two blocks instead, resulting in a total i/o latency of at least
+	  3 blocks. As memory access is the main bottleneck in native dsp processing,
+	  and to acheive less latency, it is highly recommended to try to avoid
+	  copying (this is also why the driver is the owner of the buffers). To
+	  summarize, the minimum i/o latency can be acheived if the input buffer
+	  is processed by the host into the output buffer which will physically
+	  start to sound on the next time slice. Also note that the host expects
+	  the bufferSwitch() callback to be accessed for each time slice in order
+	  to retain sync, possibly recursively; if it fails to process a block in
+	  time, it will suspend its operation for some time in order to recover.
+	Returns:
+	  If no input/output is present ASE_NotPresent will be returned.
+*/
+
+ASIOError ASIOGetBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity);
+/* Purpose:
+	  Returns min, max, and preferred buffer sizes for input/output
+	Parameter:
+	  minSize will hold the minimum buffer size
+	  maxSize will hold the maxium possible buffer size
+	  preferredSize will hold the preferred buffer size (a size which
+	  best fits performance and hardware requirements)
+	  granularity will hold the granularity at which buffer sizes
+	  may differ. Usually, the buffer size will be a power of 2;
+	  in this case, granularity will hold -1 on return, signalling
+	  possible buffer sizes starting from minSize, increased in
+	  powers of 2 up to maxSize.
+	Returns:
+	  If no input/output is present ASE_NotPresent will be returned.
+	Notes:
+	  When minimum and maximum buffer size are equal,
+	  the preferred buffer size has to be the same value as well; granularity
+	  should be 0 in this case.
+*/
+
+ASIOError ASIOCanSampleRate(ASIOSampleRate sampleRate);
+/* Purpose:
+	  Inquires the hardware for the available sample rates.
+	Parameter:
+	  sampleRate is the rate in question.
+	Returns:
+	  If the inquired sample rate is not supported, ASE_NoClock will be returned.
+	  If no input/output is present ASE_NotPresent will be returned.
+*/
+ASIOError ASIOGetSampleRate(ASIOSampleRate *currentRate);
+/* Purpose:
+	  Get the current sample Rate.
+	Parameter:
+	  currentRate will hold the current sample rate on return.
+	Returns:
+	  If sample rate is unknown, sampleRate will be 0 and ASE_NoClock will be returned.
+	  If no input/output is present ASE_NotPresent will be returned.
+	Notes:
+*/
+
+ASIOError ASIOSetSampleRate(ASIOSampleRate sampleRate);
+/* Purpose:
+	  Set the hardware to the requested sample Rate. If sampleRate == 0,
+	  enable external sync.
+	Parameter:
+	  sampleRate: on input, the requested rate
+	Returns:
+	  If sampleRate is unknown ASE_NoClock will be returned.
+	  If the current clock is external, and sampleRate is != 0,
+	  ASE_InvalidMode will be returned
+	  If no input/output is present ASE_NotPresent will be returned.
+	Notes:
+*/
+
+typedef struct ASIOClockSource
+{
+	long index;					// as used for ASIOSetClockSource()
+	long associatedChannel;		// for instance, S/PDIF or AES/EBU
+	long associatedGroup;		// see channel groups (ASIOGetChannelInfo())
+	ASIOBool isCurrentSource;	// ASIOTrue if this is the current clock source
+	char name[32];				// for user selection
+} ASIOClockSource;
+
+ASIOError ASIOGetClockSources(ASIOClockSource *clocks, long *numSources);
+/* Purpose:
+	  Get the available external audio clock sources
+	Parameter:
+	  clocks points to an array of ASIOClockSource structures:
+	  	- index: this is used to identify the clock source
+	  	  when ASIOSetClockSource() is accessed, should be
+	  	  an index counting from zero
+	  	- associatedInputChannel: the first channel of an associated
+	  	  input group, if any.
+	  	- associatedGroup: the group index of that channel.
+	  	  groups of channels are defined to seperate for
+	  	  instance analog, S/PDIF, AES/EBU, ADAT connectors etc,
+	  	  when present simultaniously. Note that associated channel
+	  	  is enumerated according to numInputs/numOutputs, means it
+	  	  is independant from a group (see also ASIOGetChannelInfo())
+	  	  inputs are associated to a clock if the physical connection
+	  	  transfers both data and clock (like S/PDIF, AES/EBU, or
+	  	  ADAT inputs). if there is no input channel associated with
+	  	  the clock source (like Word Clock, or internal oscillator), both
+	  	  associatedChannel and associatedGroup should be set to -1.
+	  	- isCurrentSource: on exit, ASIOTrue if this is the current clock
+	  	  source, ASIOFalse else
+		- name: a null-terminated string for user selection of the available sources.
+	  numSources:
+	      on input: the number of allocated array members
+	      on output: the number of available clock sources, at least
+	      1 (internal clock generator).
+	Returns:
+	  If no input/output is present ASE_NotPresent will be returned.
+	Notes:
+*/
+
+ASIOError ASIOSetClockSource(long index);
+/* Purpose:
+	  Set the audio clock source
+	Parameter:
+	  index as obtained from an inquiry to ASIOGetClockSources()
+	Returns:
+	  If no input/output is present ASE_NotPresent will be returned.
+	  If the clock can not be selected because an input channel which
+	  carries the current clock source is active, ASE_InvalidMode
+	  *may* be returned (this depends on the properties of the driver
+	  and/or hardware).
+	Notes:
+	  Should *not* return ASE_NoClock if there is no clock signal present
+	  at the selected source; this will be inquired via ASIOGetSampleRate().
+	  It should call the host callback procedure sampleRateHasChanged(),
+	  if the switch causes a sample rate change, or if no external clock
+	  is present at the selected source.
+*/
+
+ASIOError ASIOGetSamplePosition (ASIOSamples *sPos, ASIOTimeStamp *tStamp);
+/* Purpose:
+	  Inquires the sample position/time stamp pair.
+	Parameter:
+	  sPos will hold the sample position on return. The sample
+	  position is reset to zero when ASIOStart() gets called.
+	  tStamp will hold the system time when the sample position
+	  was latched.
+	Returns:
+	  If no input/output is present, ASE_NotPresent will be returned.
+	  If there is no clock, ASE_SPNotAdvancing will be returned.
+	Notes:
+
+	  in order to be able to synchronise properly,
+	  the sample position / time stamp pair must refer to the current block,
+	  that is, the engine will call ASIOGetSamplePosition() in its bufferSwitch()
+	  callback and expect the time for the current block. thus, when requested
+	  in the very first bufferSwitch after ASIO_Start(), the sample position
+	  should be zero, and the time stamp should refer to the very time where
+	  the stream was started. it also means that the sample position must be
+	  block aligned. the driver must ensure proper interpolation if the system
+	  time can not be determined for the block position. the driver is responsible
+	  for precise time stamps as it usually has most direct access to lower
+	  level resources. proper behaviour of ASIO_GetSamplePosition() and ASIO_GetLatencies()
+	  are essential for precise media synchronization!
+*/
+
+typedef struct ASIOChannelInfo
+{
+	long channel;			// on input, channel index
+	ASIOBool isInput;		// on input
+	ASIOBool isActive;		// on exit
+	long channelGroup;		// dto
+	ASIOSampleType type;	// dto
+	char name[32];			// dto
+} ASIOChannelInfo;
+
+ASIOError ASIOGetChannelInfo(ASIOChannelInfo *info);
+/* Purpose:
+	  retreive information about the nature of a channel
+	Parameter:
+	  info: pointer to a ASIOChannelInfo structure with
+	  	- channel: on input, the channel index of the channel in question.
+	  	- isInput: on input, ASIOTrue if info for an input channel is
+	  	  requested, else output
+		- channelGroup: on return, the channel group that the channel
+		  belongs to. For drivers which support different types of
+		  channels, like analog, S/PDIF, AES/EBU, ADAT etc interfaces,
+		  there should be a reasonable grouping of these types. Groups
+		  are always independant form a channel index, that is, a channel
+		  index always counts from 0 to numInputs/numOutputs regardless
+		  of the group it may belong to.
+		  There will always be at least one group (group 0). Please
+		  also note that by default, the host may decide to activate
+		  channels 0 and 1; thus, these should belong to the most
+		  useful type (analog i/o, if present).
+	  	- type: on return, contains the sample type of the channel
+	  	- isActive: on return, ASIOTrue if channel is active as it was
+	  	  installed by ASIOCreateBuffers(), ASIOFalse else
+	  	- name:  describing the type of channel in question. Used to allow
+	  	  for user selection, and enabling of specific channels. examples:
+	      "Analog In", "SPDIF Out" etc
+	Returns:
+	  If no input/output is present ASE_NotPresent will be returned.
+	Notes:
+	  If possible, the string should be organised such that the first
+	  characters are most significantly describing the nature of the
+	  port, to allow for identification even if the view showing the
+	  port name is too small to display more than 8 characters, for
+	  instance.
+*/
+
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+// Buffer preparation
+//- - - - - - - - - - - - - - - - - - - - - - - - -
+
+typedef struct ASIOBufferInfo
+{
+	ASIOBool isInput;			// on input:  ASIOTrue: input, else output
+	long channelNum;			// on input:  channel index
+	void *buffers[2];			// on output: double buffer addresses
+} ASIOBufferInfo;
+
+ASIOError ASIOCreateBuffers(ASIOBufferInfo *bufferInfos, long numChannels,
+	long bufferSize, ASIOCallbacks *callbacks);
+
+/* Purpose:
+	  Allocates input/output buffers for all input and output channels to be activated.
+	Parameter:
+	  bufferInfos is a pointer to an array of ASIOBufferInfo structures:
+	    - isInput: on input, ASIOTrue if the buffer is to be allocated
+	      for an input, output buffer else
+	    - channelNum: on input, the index of the channel in question
+	      (counting from 0)
+	    - buffers: on exit, 2 pointers to the halves of the channels' double-buffer.
+	      the size of the buffer(s) of course depend on both the ASIOSampleType
+	      as obtained from ASIOGetChannelInfo(), and bufferSize
+	  numChannels is the sum of all input and output channels to be created;
+	  thus bufferInfos is a pointer to an array of numChannels ASIOBufferInfo
+	  structures.
+	  bufferSize selects one of the possible buffer sizes as obtained from
+	  ASIOGetBufferSizes().
+	  callbacks is a pointer to an ASIOCallbacks structure.
+	Returns:
+	  If not enough memory is available ASE_NoMemory will be returned.
+	  If no input/output is present ASE_NotPresent will be returned.
+	  If bufferSize is not supported, or one or more of the bufferInfos elements
+	  contain invalid settings, ASE_InvalidMode will be returned.
+	Notes:
+	  If individual channel selection is not possible but requested,
+	  the driver has to handle this. namely, bufferSwitch() will only
+	  have filled buffers of enabled outputs. If possible, processing
+	  and buss activities overhead should be avoided for channels which
+	  were not enabled here.
+*/
+
+ASIOError ASIODisposeBuffers(void);
+/* Purpose:
+	  Releases all buffers for the device.
+	Parameter:
+	  None.
+	Returns:
+	  If no buffer were ever prepared, ASE_InvalidMode will be returned.
+	  If no input/output is present ASE_NotPresent will be returned.
+	Notes:
+	  This implies ASIOStop().
+*/
+
+ASIOError ASIOControlPanel(void);
+/* Purpose:
+	  request the driver to start a control panel component
+	  for device specific user settings. This will not be
+	  accessed on some platforms (where the component is accessed
+	  instead).
+	Parameter:
+	  None.
+	Returns:
+	  If no panel is available ASE_NotPresent will be returned.
+	  Actually, the return code is ignored.
+	Notes:
+	  if the user applied settings which require a re-configuration
+	  of parts or all of the enigine and/or driver (such as a change of
+	  the block size), the asioMessage callback can be used (see
+	  ASIO_Callbacks).
+*/
+
+ASIOError ASIOFuture(long selector, void *params);
+/* Purpose:
+	  various
+	Parameter:
+	  selector: operation Code as to be defined. zero is reserved for
+	  testing purposes.
+	  params: depends on the selector; usually pointer to a structure
+	  for passing and retreiving any type and amount of parameters.
+	Returns:
+	  the return value is also selector dependant. if the selector
+	  is unknown, ASE_InvalidParameter should be returned to prevent
+	  further calls with this selector. on success, ASE_SUCCESS
+	  must be returned (note: ASE_OK is *not* sufficient!)
+	Notes:
+	  see selectors defined below.	  
+*/
+
+enum
+{
+	kAsioEnableTimeCodeRead = 1,	// no arguments
+	kAsioDisableTimeCodeRead,		// no arguments
+	kAsioSetInputMonitor,			// ASIOInputMonitor* in params
+	kAsioTransport,					// ASIOTransportParameters* in params
+	kAsioSetInputGain,				// ASIOChannelControls* in params, apply gain
+	kAsioGetInputMeter,				// ASIOChannelControls* in params, fill meter
+	kAsioSetOutputGain,				// ASIOChannelControls* in params, apply gain
+	kAsioGetOutputMeter,			// ASIOChannelControls* in params, fill meter
+	kAsioCanInputMonitor,			// no arguments for kAsioCanXXX selectors
+	kAsioCanTimeInfo,
+	kAsioCanTimeCode,
+	kAsioCanTransport,
+	kAsioCanInputGain,
+	kAsioCanInputMeter,
+	kAsioCanOutputGain,
+	kAsioCanOutputMeter,
+
+	//	DSD support
+	//	The following extensions are required to allow switching
+	//	and control of the DSD subsystem.
+	kAsioSetIoFormat			= 0x23111961,		/* ASIOIoFormat * in params.			*/
+	kAsioGetIoFormat			= 0x23111983,		/* ASIOIoFormat * in params.			*/
+	kAsioCanDoIoFormat			= 0x23112004,		/* ASIOIoFormat * in params.			*/
+};
+
+typedef struct ASIOInputMonitor
+{
+	long input;		// this input was set to monitor (or off), -1: all
+	long output;	// suggested output for monitoring the input (if so)
+	long gain;		// suggested gain, ranging 0 - 0x7fffffffL (-inf to +12 dB)
+	ASIOBool state;	// ASIOTrue => on, ASIOFalse => off
+	long pan;		// suggested pan, 0 => all left, 0x7fffffff => right
+} ASIOInputMonitor;
+
+typedef struct ASIOChannelControls
+{
+	long channel;			// on input, channel index
+	ASIOBool isInput;		// on input
+	long gain;				// on input,  ranges 0 thru 0x7fffffff
+	long meter;				// on return, ranges 0 thru 0x7fffffff
+	char future[32];
+} ASIOChannelControls;
+
+typedef struct ASIOTransportParameters
+{
+	long command;		// see enum below
+	ASIOSamples samplePosition;
+	long track;
+	long trackSwitches[16];		// 512 tracks on/off
+	char future[64];
+} ASIOTransportParameters;
+
+enum
+{
+	kTransStart = 1,
+	kTransStop,
+	kTransLocate,		// to samplePosition
+	kTransPunchIn,
+	kTransPunchOut,
+	kTransArmOn,		// track
+	kTransArmOff,		// track
+	kTransMonitorOn,	// track
+	kTransMonitorOff,	// track
+	kTransArm,			// trackSwitches
+	kTransMonitor		// trackSwitches
+};
+
+/*
+// DSD support
+//	Some notes on how to use ASIOIoFormatType.
+//
+//	The caller will fill the format with the request types.
+//	If the board can do the request then it will leave the
+//	values unchanged. If the board does not support the
+//	request then it will change that entry to Invalid (-1)
+//
+//	So to request DSD then
+//
+//	ASIOIoFormat NeedThis={kASIODSDFormat};
+//
+//	if(ASE_SUCCESS != ASIOFuture(kAsioSetIoFormat,&NeedThis) ){
+//		// If the board did not accept one of the parameters then the
+//		// whole call will fail and the failing parameter will
+//		// have had its value changes to -1.
+//	}
+//
+// Note: Switching between the formats need to be done before the "prepared"
+// state (see ASIO 2 documentation) is entered.
+*/
+typedef long int ASIOIoFormatType;
+enum ASIOIoFormatType_e
+{
+	kASIOFormatInvalid = -1,
+	kASIOPCMFormat = 0,
+	kASIODSDFormat = 1,
+};
+
+typedef struct ASIOIoFormat_s
+{
+	ASIOIoFormatType	FormatType;
+	char				future[512-sizeof(ASIOIoFormatType)];
+} ASIOIoFormat;
+
+
+ASIOError ASIOOutputReady(void);
+/* Purpose:
+	  this tells the driver that the host has completed processing
+	  the output buffers. if the data format required by the hardware
+	  differs from the supported asio formats, but the hardware
+	  buffers are DMA buffers, the driver will have to convert
+	  the audio stream data; as the bufferSwitch callback is
+	  usually issued at dma block switch time, the driver will
+	  have to convert the *previous* host buffer, which increases
+	  the output latency by one block.
+	  when the host finds out that ASIOOutputReady() returns
+	  true, it will issue this call whenever it completed
+	  output processing. then the driver can convert the
+	  host data directly to the dma buffer to be played next,
+	  reducing output latency by one block.
+	  another way to look at it is, that the buffer switch is called
+	  in order to pass the *input* stream to the host, so that it can
+	  process the input into the output, and the output stream is passed
+	  to the driver when the host has completed its process.
+	Parameter:
+		None
+	Returns:
+	  only if the above mentioned scenario is given, and a reduction
+	  of output latency can be acheived by this mechanism, should
+	  ASE_OK be returned. otherwise (and usually), ASE_NotPresent
+	  should be returned in order to prevent further calls to this
+	  function. note that the host may want to determine if it is
+	  to use this when the system is not yet fully initialized, so
+	  ASE_OK should always be returned if the mechanism makes sense.	  
+	Notes:
+	  please remeber to adjust ASIOGetLatencies() according to
+	  whether ASIOOutputReady() was ever called or not, if your
+	  driver supports this scenario.
+	  also note that the engine may fail to call ASIO_OutputReady()
+	  in time in overload cases. as already mentioned, bufferSwitch
+      should be called for every block regardless of whether a block
+      could be processed in time.
+*/
+
+// restore old alignment
+#if defined(_MSC_VER) && !defined(__MWERKS__) 
+#pragma pack(pop)
+#elif PRAGMA_ALIGN_SUPPORTED
+#pragma options align = reset
+#endif
+
+#endif
+
+ cbits/include/asiodrivers.cpp view
@@ -0,0 +1,186 @@+#include <string.h>
+#include "asiodrivers.h"
+
+AsioDrivers* asioDrivers = 0;
+
+bool loadAsioDriver(char *name);
+
+bool loadAsioDriver(char *name)
+{
+	if(!asioDrivers)
+		asioDrivers = new AsioDrivers();
+	if(asioDrivers)
+		return asioDrivers->loadDriver(name);
+	return false;
+}
+
+//------------------------------------------------------------------------------------
+
+#if MAC
+
+bool resolveASIO(unsigned long aconnID);
+
+AsioDrivers::AsioDrivers() : CodeFragments("ASIO Drivers", 'AsDr', 'Asio')
+{
+	connID = -1;
+	curIndex = -1;
+}
+
+AsioDrivers::~AsioDrivers()
+{
+	removeCurrentDriver();
+}
+
+bool AsioDrivers::getCurrentDriverName(char *name)
+{
+	if(curIndex >= 0)
+		return getName(curIndex, name);
+	return false;
+}
+
+long AsioDrivers::getDriverNames(char **names, long maxDrivers)
+{
+	for(long i = 0; i < getNumFragments() && i < maxDrivers; i++)
+		getName(i, names[i]);
+	return getNumFragments() < maxDrivers ? getNumFragments() : maxDrivers;
+}
+
+bool AsioDrivers::loadDriver(char *name)
+{
+	char dname[64];
+	unsigned long newID;
+
+	for(long i = 0; i < getNumFragments(); i++)
+	{
+		if(getName(i, dname) && !strcmp(name, dname))
+		{
+			if(newInstance(i, &newID))
+			{
+				if(resolveASIO(newID))
+				{
+					if(connID != -1)
+						removeInstance(curIndex, connID);
+					curIndex = i;
+					connID = newID;
+					return true;
+				}
+			}
+			break;
+		}
+	}
+	return false;
+}
+
+void AsioDrivers::removeCurrentDriver()
+{
+	if(connID != -1)
+		removeInstance(curIndex, connID);
+	connID = -1;
+	curIndex = -1;
+}
+
+//------------------------------------------------------------------------------------
+
+#elif WINDOWS
+
+#include "iasiodrv.h"
+
+extern IASIO* theAsioDriver;
+
+AsioDrivers::AsioDrivers() : AsioDriverList()
+{
+	curIndex = -1;
+}
+
+AsioDrivers::~AsioDrivers()
+{
+}
+
+bool AsioDrivers::getCurrentDriverName(char *name)
+{
+	if(curIndex >= 0)
+		return asioGetDriverName(curIndex, name, 32) == 0 ? true : false;
+	name[0] = 0;
+	return false;
+}
+
+long AsioDrivers::getDriverNames(char **names, long maxDrivers)
+{
+	for(long i = 0; i < asioGetNumDev() && i < maxDrivers; i++)
+		asioGetDriverName(i, names[i], 32);
+	return asioGetNumDev() < maxDrivers ? asioGetNumDev() : maxDrivers;
+}
+
+bool AsioDrivers::loadDriver(char *name)
+{
+	char dname[64];
+	char curName[64];
+
+	for(long i = 0; i < asioGetNumDev(); i++)
+	{
+		if(!asioGetDriverName(i, dname, 32) && !strcmp(name, dname))
+		{
+			curName[0] = 0;
+			getCurrentDriverName(curName);	// in case we fail...
+			removeCurrentDriver();
+
+			if(!asioOpenDriver(i, (void **)&theAsioDriver))
+			{
+				curIndex = i;
+				return true;
+			}
+			else
+			{
+				theAsioDriver = 0;
+				if(curName[0] && strcmp(dname, curName))
+					loadDriver(curName);	// try restore
+			}
+			break;
+		}
+	}
+	return false;
+}
+
+void AsioDrivers::removeCurrentDriver()
+{
+	if(curIndex != -1)
+		asioCloseDriver(curIndex);
+	curIndex = -1;
+}
+
+#elif SGI || BEOS
+
+#include "asiolist.h"
+
+AsioDrivers::AsioDrivers() 
+	: AsioDriverList()
+{
+	curIndex = -1;
+}
+
+AsioDrivers::~AsioDrivers()
+{
+}
+
+bool AsioDrivers::getCurrentDriverName(char *name)
+{
+	return false;
+}
+
+long AsioDrivers::getDriverNames(char **names, long maxDrivers)
+{
+	return 0;
+}
+
+bool AsioDrivers::loadDriver(char *name)
+{
+	return false;
+}
+
+void AsioDrivers::removeCurrentDriver()
+{
+}
+
+#else
+#error implement me
+#endif
+ cbits/include/asiodrivers.h view
@@ -0,0 +1,41 @@+#ifndef __AsioDrivers__
+#define __AsioDrivers__
+
+#include "ginclude.h"
+
+#if MAC
+#include "CodeFragments.hpp"
+
+class AsioDrivers : public CodeFragments
+
+#elif WINDOWS
+#include <windows.h>
+#include "asiolist.h"
+
+class AsioDrivers : public AsioDriverList
+
+#elif SGI || BEOS
+#include "asiolist.h"
+
+class AsioDrivers : public AsioDriverList
+
+#else
+#error implement me
+#endif
+
+{
+public:
+	AsioDrivers();
+	~AsioDrivers();
+	
+	bool getCurrentDriverName(char *name);
+	long getDriverNames(char **names, long maxDrivers);
+	bool loadDriver(char *name);
+	void removeCurrentDriver();
+	long getCurrentDriverIndex() {return curIndex;}
+protected:
+	unsigned long connID;
+	long curIndex;
+};
+
+#endif
+ cbits/include/asiodrvr.h view
@@ -0,0 +1,76 @@+/*
+	Steinberg Audio Stream I/O API
+	(c) 1996, Steinberg Soft- und Hardware GmbH
+	charlie (May 1996)
+
+	asiodrvr.h
+	c++ superclass to implement asio functionality. from this,
+	you can derive whatever required
+*/
+
+#ifndef _asiodrvr_
+#define _asiodrvr_
+
+// cpu and os system we are running on
+#include "asiosys.h"
+// basic "C" interface
+#include "asio.h"
+
+class AsioDriver;
+extern AsioDriver *getDriver();		// for generic constructor 
+
+#if WINDOWS
+#include <windows.h>
+#include "combase.h"
+#include "iasiodrv.h"
+class AsioDriver : public IASIO ,public CUnknown
+{
+public:
+	AsioDriver(LPUNKNOWN pUnk, HRESULT *phr);
+
+	DECLARE_IUNKNOWN
+	// Factory method
+	static CUnknown *CreateInstance(LPUNKNOWN pUnk, HRESULT *phr);
+	// IUnknown
+	virtual HRESULT STDMETHODCALLTYPE NonDelegatingQueryInterface(REFIID riid,void **ppvObject);
+
+#else
+
+class AsioDriver
+{
+public:
+	AsioDriver();
+#endif
+	virtual ~AsioDriver();
+
+	virtual ASIOBool init(void* sysRef);
+	virtual void getDriverName(char *name);	// max 32 bytes incl. terminating zero
+	virtual long getDriverVersion();
+	virtual void getErrorMessage(char *string);	// max 124 bytes incl.
+
+	virtual ASIOError start();
+	virtual ASIOError stop();
+
+	virtual ASIOError getChannels(long *numInputChannels, long *numOutputChannels);
+	virtual ASIOError getLatencies(long *inputLatency, long *outputLatency);
+	virtual ASIOError getBufferSize(long *minSize, long *maxSize,
+		long *preferredSize, long *granularity);
+
+	virtual ASIOError canSampleRate(ASIOSampleRate sampleRate);
+	virtual ASIOError getSampleRate(ASIOSampleRate *sampleRate);
+	virtual ASIOError setSampleRate(ASIOSampleRate sampleRate);
+	virtual ASIOError getClockSources(ASIOClockSource *clocks, long *numSources);
+	virtual ASIOError setClockSource(long reference);
+
+	virtual ASIOError getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp);
+	virtual ASIOError getChannelInfo(ASIOChannelInfo *info);
+
+	virtual ASIOError createBuffers(ASIOBufferInfo *bufferInfos, long numChannels,
+		long bufferSize, ASIOCallbacks *callbacks);
+	virtual ASIOError disposeBuffers();
+
+	virtual ASIOError controlPanel();
+	virtual ASIOError future(long selector, void *opt);
+	virtual ASIOError outputReady();
+};
+#endif
+ cbits/include/asiolist.cpp view
@@ -0,0 +1,268 @@+#include <windows.h>
+#include "iasiodrv.h"
+#include "asiolist.h"
+
+#define ASIODRV_DESC		"description"
+#define INPROC_SERVER		"InprocServer32"
+#define ASIO_PATH			"software\\asio"
+#define COM_CLSID			"clsid"
+
+// ******************************************************************
+// Local Functions 
+// ******************************************************************
+static LONG findDrvPath (char *clsidstr,char *dllpath,int dllpathsize)
+{
+	HKEY			hkEnum,hksub,hkpath;
+	char			databuf[512];
+	LONG 			cr,rc = -1;
+	DWORD			datatype,datasize;
+	DWORD			index;
+	OFSTRUCT		ofs;
+	HFILE			hfile;
+	BOOL			found = FALSE;
+
+	CharLowerBuff(clsidstr,strlen(clsidstr));
+	if ((cr = RegOpenKey(HKEY_CLASSES_ROOT,COM_CLSID,&hkEnum)) == ERROR_SUCCESS) {
+
+		index = 0;
+		while (cr == ERROR_SUCCESS && !found) {
+			cr = RegEnumKey(hkEnum,index++,(LPTSTR)databuf,512);
+			if (cr == ERROR_SUCCESS) {
+				CharLowerBuff(databuf,strlen(databuf));
+				if (!(strcmp(databuf,clsidstr))) {
+					if ((cr = RegOpenKeyEx(hkEnum,(LPCTSTR)databuf,0,KEY_READ,&hksub)) == ERROR_SUCCESS) {
+						if ((cr = RegOpenKeyEx(hksub,(LPCTSTR)INPROC_SERVER,0,KEY_READ,&hkpath)) == ERROR_SUCCESS) {
+							datatype = REG_SZ; datasize = (DWORD)dllpathsize;
+							cr = RegQueryValueEx(hkpath,0,0,&datatype,(LPBYTE)dllpath,&datasize);
+							if (cr == ERROR_SUCCESS) {
+								memset(&ofs,0,sizeof(OFSTRUCT));
+								ofs.cBytes = sizeof(OFSTRUCT); 
+								hfile = OpenFile(dllpath,&ofs,OF_EXIST);
+								if (hfile) rc = 0; 
+							}
+							RegCloseKey(hkpath);
+						}
+						RegCloseKey(hksub);
+					}
+					found = TRUE;	// break out 
+				}
+			}
+		}				
+		RegCloseKey(hkEnum);
+	}
+	return rc;
+}
+
+
+static LPASIODRVSTRUCT newDrvStruct (HKEY hkey,char *keyname,int drvID,LPASIODRVSTRUCT lpdrv)
+{
+	HKEY	hksub;
+	char	databuf[256];
+	char	dllpath[MAXPATHLEN];
+	WORD	wData[100];
+	CLSID	clsid;
+	DWORD	datatype,datasize;
+	LONG	cr,rc;
+
+	if (!lpdrv) {
+		if ((cr = RegOpenKeyEx(hkey,(LPCTSTR)keyname,0,KEY_READ,&hksub)) == ERROR_SUCCESS) {
+
+			datatype = REG_SZ; datasize = 256;
+			cr = RegQueryValueEx(hksub,COM_CLSID,0,&datatype,(LPBYTE)databuf,&datasize);
+			if (cr == ERROR_SUCCESS) {
+				rc = findDrvPath (databuf,dllpath,MAXPATHLEN);
+				if (rc == 0) {
+					lpdrv = new ASIODRVSTRUCT[1];
+					if (lpdrv) {
+						memset(lpdrv,0,sizeof(ASIODRVSTRUCT));
+						lpdrv->drvID = drvID;
+						MultiByteToWideChar(CP_ACP,0,(LPCSTR)databuf,-1,(LPWSTR)wData,100);
+						if ((cr = CLSIDFromString((LPOLESTR)wData,(LPCLSID)&clsid)) == S_OK) {
+							memcpy(&lpdrv->clsid,&clsid,sizeof(CLSID));
+						}
+
+						datatype = REG_SZ; datasize = 256;
+						cr = RegQueryValueEx(hksub,ASIODRV_DESC,0,&datatype,(LPBYTE)databuf,&datasize);
+						if (cr == ERROR_SUCCESS) {
+							strcpy(lpdrv->drvname,databuf);
+						}
+						else strcpy(lpdrv->drvname,keyname);
+					}
+				}
+			}
+			RegCloseKey(hksub);
+		}
+	}	
+	else lpdrv->next = newDrvStruct(hkey,keyname,drvID+1,lpdrv->next);
+
+	return lpdrv;
+}
+
+static void deleteDrvStruct (LPASIODRVSTRUCT lpdrv)
+{
+	IASIO	*iasio;
+
+	if (lpdrv != 0) {
+		deleteDrvStruct(lpdrv->next);
+		if (lpdrv->asiodrv) {
+			iasio = (IASIO *)lpdrv->asiodrv;
+			iasio->Release();
+		}
+		delete lpdrv;
+	}
+}
+
+
+static LPASIODRVSTRUCT getDrvStruct (int drvID,LPASIODRVSTRUCT lpdrv)
+{
+	while (lpdrv) {
+		if (lpdrv->drvID == drvID) return lpdrv;
+		lpdrv = lpdrv->next;
+	}
+	return 0;
+}
+// ******************************************************************
+
+
+// ******************************************************************
+//	AsioDriverList
+// ******************************************************************
+AsioDriverList::AsioDriverList ()
+{
+	HKEY			hkEnum = 0;
+	char			keyname[MAXDRVNAMELEN];
+	LPASIODRVSTRUCT	pdl;
+	LONG 			cr;
+	DWORD			index = 0;
+	BOOL			fin = FALSE;
+
+	numdrv		= 0;
+	lpdrvlist	= 0;
+
+	cr = RegOpenKey(HKEY_LOCAL_MACHINE,ASIO_PATH,&hkEnum);
+	while (cr == ERROR_SUCCESS) {
+		if ((cr = RegEnumKey(hkEnum,index++,(LPTSTR)keyname,MAXDRVNAMELEN))== ERROR_SUCCESS) {
+			lpdrvlist = newDrvStruct (hkEnum,keyname,0,lpdrvlist);
+		}
+		else fin = TRUE;
+	}
+	if (hkEnum) RegCloseKey(hkEnum);
+
+	pdl = lpdrvlist;
+	while (pdl) {
+		numdrv++;
+		pdl = pdl->next;
+	}
+
+	if (numdrv) CoInitialize(0);	// initialize COM
+}
+
+AsioDriverList::~AsioDriverList ()
+{
+	if (numdrv) {
+		deleteDrvStruct(lpdrvlist);
+		CoUninitialize();
+	}
+}
+
+
+LONG AsioDriverList::asioGetNumDev (VOID)
+{
+	return (LONG)numdrv;
+}
+
+
+LONG AsioDriverList::asioOpenDriver (int drvID,LPVOID *asiodrv)
+{
+	LPASIODRVSTRUCT	lpdrv = 0;
+	long			rc;
+
+	if (!asiodrv) return DRVERR_INVALID_PARAM;
+
+	if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) {
+		if (!lpdrv->asiodrv) {
+			rc = CoCreateInstance(lpdrv->clsid,0,CLSCTX_INPROC_SERVER,lpdrv->clsid,asiodrv);
+			if (rc == S_OK) {
+				lpdrv->asiodrv = *asiodrv;
+				return 0;
+			}
+			// else if (rc == REGDB_E_CLASSNOTREG)
+			//	strcpy (info->messageText, "Driver not registered in the Registration Database!");
+		}
+		else rc = DRVERR_DEVICE_ALREADY_OPEN;
+	}
+	else rc = DRVERR_DEVICE_NOT_FOUND;
+	
+	return rc;
+}
+
+
+LONG AsioDriverList::asioCloseDriver (int drvID)
+{
+	LPASIODRVSTRUCT	lpdrv = 0;
+	IASIO			*iasio;
+
+	if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) {
+		if (lpdrv->asiodrv) {
+			iasio = (IASIO *)lpdrv->asiodrv;
+			iasio->Release();
+			lpdrv->asiodrv = 0;
+		}
+	}
+
+	return 0;
+}
+
+LONG AsioDriverList::asioGetDriverName (int drvID,char *drvname,int drvnamesize)
+{	
+	LPASIODRVSTRUCT			lpdrv = 0;
+
+	if (!drvname) return DRVERR_INVALID_PARAM;
+
+	if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) {
+		if (strlen(lpdrv->drvname) < (unsigned int)drvnamesize) {
+			strcpy(drvname,lpdrv->drvname);
+		}
+		else {
+			memcpy(drvname,lpdrv->drvname,drvnamesize-4);
+			drvname[drvnamesize-4] = '.';
+			drvname[drvnamesize-3] = '.';
+			drvname[drvnamesize-2] = '.';
+			drvname[drvnamesize-1] = 0;
+		}
+		return 0;
+	}
+	return DRVERR_DEVICE_NOT_FOUND;
+}
+
+LONG AsioDriverList::asioGetDriverPath (int drvID,char *dllpath,int dllpathsize)
+{
+	LPASIODRVSTRUCT			lpdrv = 0;
+
+	if (!dllpath) return DRVERR_INVALID_PARAM;
+
+	if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) {
+		if (strlen(lpdrv->dllpath) < (unsigned int)dllpathsize) {
+			strcpy(dllpath,lpdrv->dllpath);
+			return 0;
+		}
+		dllpath[0] = 0;
+		return DRVERR_INVALID_PARAM;
+	}
+	return DRVERR_DEVICE_NOT_FOUND;
+}
+
+LONG AsioDriverList::asioGetDriverCLSID (int drvID,CLSID *clsid)
+{
+	LPASIODRVSTRUCT			lpdrv = 0;
+
+	if (!clsid) return DRVERR_INVALID_PARAM;
+
+	if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) {
+		memcpy(clsid,&lpdrv->clsid,sizeof(CLSID));
+		return 0;
+	}
+	return DRVERR_DEVICE_NOT_FOUND;
+}
+
+
+ cbits/include/asiolist.h view
@@ -0,0 +1,46 @@+#ifndef __asiolist__
+#define __asiolist__
+
+#define DRVERR			-5000
+#define DRVERR_INVALID_PARAM		DRVERR-1
+#define DRVERR_DEVICE_ALREADY_OPEN	DRVERR-2
+#define DRVERR_DEVICE_NOT_FOUND		DRVERR-3
+
+#define MAXPATHLEN			512
+#define MAXDRVNAMELEN		128
+
+struct asiodrvstruct
+{
+	int						drvID;
+	CLSID					clsid;
+	char					dllpath[MAXPATHLEN];
+	char					drvname[MAXDRVNAMELEN];
+	LPVOID					asiodrv;
+	struct asiodrvstruct	*next;
+};
+
+typedef struct asiodrvstruct ASIODRVSTRUCT;
+typedef ASIODRVSTRUCT	*LPASIODRVSTRUCT;
+
+class AsioDriverList {
+public:
+	AsioDriverList();
+	~AsioDriverList();
+	
+	LONG asioOpenDriver (int,VOID **);
+	LONG asioCloseDriver (int);
+
+	// nice to have
+	LONG asioGetNumDev (VOID);
+	LONG asioGetDriverName (int,char *,int);		
+	LONG asioGetDriverPath (int,char *,int);
+	LONG asioGetDriverCLSID (int,CLSID *);
+
+	// or use directly access
+	LPASIODRVSTRUCT	lpdrvlist;
+	int				numdrv;
+};
+
+typedef class AsioDriverList *LPASIODRIVERLIST;
+
+#endif
+ cbits/include/asiosys.h view
@@ -0,0 +1,82 @@+#ifndef __asiosys__
+	#define __asiosys__
+
+	#ifdef WIN32
+		#undef MAC 
+		#define PPC 0
+		#define WINDOWS 1
+		#define SGI 0
+		#define SUN 0
+		#define LINUX 0
+		#define BEOS 0
+
+		#define NATIVE_INT64 0
+		#define IEEE754_64FLOAT 1
+	
+	#elif BEOS
+		#define MAC 0
+		#define PPC 0
+		#define WINDOWS 0
+		#define PC 0
+		#define SGI 0
+		#define SUN 0
+		#define LINUX 0
+		
+		#define NATIVE_INT64 0
+		#define IEEE754_64FLOAT 1
+		
+		#ifndef DEBUG
+			#define DEBUG 0
+		 	#if DEBUG
+		 		void DEBUGGERMESSAGE(char *string);
+		 	#else
+		  		#define DEBUGGERMESSAGE(a)
+			#endif
+		#endif
+
+	#elif SGI
+		#define MAC 0
+		#define PPC 0
+		#define WINDOWS 0
+		#define PC 0
+		#define SUN 0
+		#define LINUX 0
+		#define BEOS 0
+		
+		#define NATIVE_INT64 0
+		#define IEEE754_64FLOAT 1
+		
+		#ifndef DEBUG
+			#define DEBUG 0
+		 	#if DEBUG
+		 		void DEBUGGERMESSAGE(char *string);
+		 	#else
+		  		#define DEBUGGERMESSAGE(a)
+			#endif
+		#endif
+
+	#else	// MAC
+
+		#define MAC 1
+		#define PPC 1
+		#define WINDOWS 0
+		#define PC 0
+		#define SGI 0
+		#define SUN 0
+		#define LINUX 0
+		#define BEOS 0
+
+		#define NATIVE_INT64 0
+		#define IEEE754_64FLOAT 1
+
+		#ifndef DEBUG
+			#define DEBUG 0
+			#if DEBUG
+				void DEBUGGERMESSAGE(char *string);
+			#else
+				#define DEBUGGERMESSAGE(a)
+			#endif
+		#endif
+	#endif
+
+#endif
+ cbits/include/dsound.h view
@@ -0,0 +1,2369 @@+/*==========================================================================;+ *+ *  Copyright (c) Microsoft Corporation.  All rights reserved.+ *+ *  File:       dsound.h+ *  Content:    DirectSound include file+ *+ **************************************************************************/++#define COM_NO_WINDOWS_H+#include <objbase.h>+#include <float.h>++#ifndef DIRECTSOUND_VERSION+#define DIRECTSOUND_VERSION 0x0900  /* Version 9.0 */+#endif++#ifdef __cplusplus+extern "C" {+#endif // __cplusplus++#ifndef __DSOUND_INCLUDED__+#define __DSOUND_INCLUDED__++/* Type definitions shared with Direct3D */++#ifndef DX_SHARED_DEFINES++typedef float D3DVALUE, *LPD3DVALUE;++#ifndef D3DCOLOR_DEFINED+typedef DWORD D3DCOLOR;+#define D3DCOLOR_DEFINED+#endif++#ifndef LPD3DCOLOR_DEFINED+typedef DWORD *LPD3DCOLOR;+#define LPD3DCOLOR_DEFINED+#endif++#ifndef D3DVECTOR_DEFINED+typedef struct _D3DVECTOR {+    float x;+    float y;+    float z;+} D3DVECTOR;+#define D3DVECTOR_DEFINED+#endif++#ifndef LPD3DVECTOR_DEFINED+typedef D3DVECTOR *LPD3DVECTOR;+#define LPD3DVECTOR_DEFINED+#endif++#define DX_SHARED_DEFINES+#endif // DX_SHARED_DEFINES++#define _FACDS  0x878   /* DirectSound's facility code */+#define MAKE_DSHRESULT(code)  MAKE_HRESULT(1, _FACDS, code)++// DirectSound Component GUID {47D4D946-62E8-11CF-93BC-444553540000}+DEFINE_GUID(CLSID_DirectSound, 0x47d4d946, 0x62e8, 0x11cf, 0x93, 0xbc, 0x44, 0x45, 0x53, 0x54, 0x0, 0x0);++// DirectSound 8.0 Component GUID {3901CC3F-84B5-4FA4-BA35-AA8172B8A09B}+DEFINE_GUID(CLSID_DirectSound8, 0x3901cc3f, 0x84b5, 0x4fa4, 0xba, 0x35, 0xaa, 0x81, 0x72, 0xb8, 0xa0, 0x9b);++// DirectSound Capture Component GUID {B0210780-89CD-11D0-AF08-00A0C925CD16}+DEFINE_GUID(CLSID_DirectSoundCapture, 0xb0210780, 0x89cd, 0x11d0, 0xaf, 0x8, 0x0, 0xa0, 0xc9, 0x25, 0xcd, 0x16);++// DirectSound 8.0 Capture Component GUID {E4BCAC13-7F99-4908-9A8E-74E3BF24B6E1}+DEFINE_GUID(CLSID_DirectSoundCapture8, 0xe4bcac13, 0x7f99, 0x4908, 0x9a, 0x8e, 0x74, 0xe3, 0xbf, 0x24, 0xb6, 0xe1);++// DirectSound Full Duplex Component GUID {FEA4300C-7959-4147-B26A-2377B9E7A91D}+DEFINE_GUID(CLSID_DirectSoundFullDuplex, 0xfea4300c, 0x7959, 0x4147, 0xb2, 0x6a, 0x23, 0x77, 0xb9, 0xe7, 0xa9, 0x1d);+++// DirectSound default playback device GUID {DEF00000-9C6D-47ED-AAF1-4DDA8F2B5C03}+DEFINE_GUID(DSDEVID_DefaultPlayback, 0xdef00000, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03);++// DirectSound default capture device GUID {DEF00001-9C6D-47ED-AAF1-4DDA8F2B5C03}+DEFINE_GUID(DSDEVID_DefaultCapture, 0xdef00001, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03);++// DirectSound default device for voice playback {DEF00002-9C6D-47ED-AAF1-4DDA8F2B5C03}+DEFINE_GUID(DSDEVID_DefaultVoicePlayback, 0xdef00002, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03);++// DirectSound default device for voice capture {DEF00003-9C6D-47ED-AAF1-4DDA8F2B5C03}+DEFINE_GUID(DSDEVID_DefaultVoiceCapture, 0xdef00003, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03);+++//+// Forward declarations for interfaces.+// 'struct' not 'class' per the way DECLARE_INTERFACE_ is defined+//++#ifdef __cplusplus+struct IDirectSound;+struct IDirectSoundBuffer;+struct IDirectSound3DListener;+struct IDirectSound3DBuffer;+struct IDirectSoundCapture;+struct IDirectSoundCaptureBuffer;+struct IDirectSoundNotify;+#endif // __cplusplus+++//+// DirectSound 8.0 interfaces.+//++#if DIRECTSOUND_VERSION >= 0x0800++#ifdef __cplusplus+struct IDirectSound8;+struct IDirectSoundBuffer8;+struct IDirectSoundCaptureBuffer8;+struct IDirectSoundFXGargle;+struct IDirectSoundFXChorus;+struct IDirectSoundFXFlanger;+struct IDirectSoundFXEcho;+struct IDirectSoundFXDistortion;+struct IDirectSoundFXCompressor;+struct IDirectSoundFXParamEq;+struct IDirectSoundFXWavesReverb;+struct IDirectSoundFXI3DL2Reverb;+struct IDirectSoundCaptureFXAec;+struct IDirectSoundCaptureFXNoiseSuppress;+struct IDirectSoundFullDuplex;+#endif // __cplusplus++// IDirectSound8, IDirectSoundBuffer8 and IDirectSoundCaptureBuffer8 are the+// only DirectSound 7.0 interfaces with changed functionality in version 8.0.+// The other level 8 interfaces as equivalent to their level 7 counterparts:++#define IDirectSoundCapture8            IDirectSoundCapture+#define IDirectSound3DListener8         IDirectSound3DListener+#define IDirectSound3DBuffer8           IDirectSound3DBuffer+#define IDirectSoundNotify8             IDirectSoundNotify+#define IDirectSoundFXGargle8           IDirectSoundFXGargle+#define IDirectSoundFXChorus8           IDirectSoundFXChorus+#define IDirectSoundFXFlanger8          IDirectSoundFXFlanger+#define IDirectSoundFXEcho8             IDirectSoundFXEcho+#define IDirectSoundFXDistortion8       IDirectSoundFXDistortion+#define IDirectSoundFXCompressor8       IDirectSoundFXCompressor+#define IDirectSoundFXParamEq8          IDirectSoundFXParamEq+#define IDirectSoundFXWavesReverb8      IDirectSoundFXWavesReverb+#define IDirectSoundFXI3DL2Reverb8      IDirectSoundFXI3DL2Reverb+#define IDirectSoundCaptureFXAec8       IDirectSoundCaptureFXAec+#define IDirectSoundCaptureFXNoiseSuppress8 IDirectSoundCaptureFXNoiseSuppress+#define IDirectSoundFullDuplex8         IDirectSoundFullDuplex++#endif // DIRECTSOUND_VERSION >= 0x0800++typedef struct IDirectSound                 *LPDIRECTSOUND;+typedef struct IDirectSoundBuffer           *LPDIRECTSOUNDBUFFER;+typedef struct IDirectSound3DListener       *LPDIRECTSOUND3DLISTENER;+typedef struct IDirectSound3DBuffer         *LPDIRECTSOUND3DBUFFER;+typedef struct IDirectSoundCapture          *LPDIRECTSOUNDCAPTURE;+typedef struct IDirectSoundCaptureBuffer    *LPDIRECTSOUNDCAPTUREBUFFER;+typedef struct IDirectSoundNotify           *LPDIRECTSOUNDNOTIFY;+++#if DIRECTSOUND_VERSION >= 0x0800++typedef struct IDirectSoundFXGargle         *LPDIRECTSOUNDFXGARGLE;+typedef struct IDirectSoundFXChorus         *LPDIRECTSOUNDFXCHORUS;+typedef struct IDirectSoundFXFlanger        *LPDIRECTSOUNDFXFLANGER;+typedef struct IDirectSoundFXEcho           *LPDIRECTSOUNDFXECHO;+typedef struct IDirectSoundFXDistortion     *LPDIRECTSOUNDFXDISTORTION;+typedef struct IDirectSoundFXCompressor     *LPDIRECTSOUNDFXCOMPRESSOR;+typedef struct IDirectSoundFXParamEq        *LPDIRECTSOUNDFXPARAMEQ;+typedef struct IDirectSoundFXWavesReverb    *LPDIRECTSOUNDFXWAVESREVERB;+typedef struct IDirectSoundFXI3DL2Reverb    *LPDIRECTSOUNDFXI3DL2REVERB;+typedef struct IDirectSoundCaptureFXAec     *LPDIRECTSOUNDCAPTUREFXAEC;+typedef struct IDirectSoundCaptureFXNoiseSuppress *LPDIRECTSOUNDCAPTUREFXNOISESUPPRESS;+typedef struct IDirectSoundFullDuplex       *LPDIRECTSOUNDFULLDUPLEX;++typedef struct IDirectSound8                *LPDIRECTSOUND8;+typedef struct IDirectSoundBuffer8          *LPDIRECTSOUNDBUFFER8;+typedef struct IDirectSound3DListener8      *LPDIRECTSOUND3DLISTENER8;+typedef struct IDirectSound3DBuffer8        *LPDIRECTSOUND3DBUFFER8;+typedef struct IDirectSoundCapture8         *LPDIRECTSOUNDCAPTURE8;+typedef struct IDirectSoundCaptureBuffer8   *LPDIRECTSOUNDCAPTUREBUFFER8;+typedef struct IDirectSoundNotify8          *LPDIRECTSOUNDNOTIFY8;+typedef struct IDirectSoundFXGargle8        *LPDIRECTSOUNDFXGARGLE8;+typedef struct IDirectSoundFXChorus8        *LPDIRECTSOUNDFXCHORUS8;+typedef struct IDirectSoundFXFlanger8       *LPDIRECTSOUNDFXFLANGER8;+typedef struct IDirectSoundFXEcho8          *LPDIRECTSOUNDFXECHO8;+typedef struct IDirectSoundFXDistortion8    *LPDIRECTSOUNDFXDISTORTION8;+typedef struct IDirectSoundFXCompressor8    *LPDIRECTSOUNDFXCOMPRESSOR8;+typedef struct IDirectSoundFXParamEq8       *LPDIRECTSOUNDFXPARAMEQ8;+typedef struct IDirectSoundFXWavesReverb8   *LPDIRECTSOUNDFXWAVESREVERB8;+typedef struct IDirectSoundFXI3DL2Reverb8   *LPDIRECTSOUNDFXI3DL2REVERB8;+typedef struct IDirectSoundCaptureFXAec8    *LPDIRECTSOUNDCAPTUREFXAEC8;+typedef struct IDirectSoundCaptureFXNoiseSuppress8 *LPDIRECTSOUNDCAPTUREFXNOISESUPPRESS8;+typedef struct IDirectSoundFullDuplex8      *LPDIRECTSOUNDFULLDUPLEX8;++#endif // DIRECTSOUND_VERSION >= 0x0800++//+// IID definitions for the unchanged DirectSound 8.0 interfaces+//++#if DIRECTSOUND_VERSION >= 0x0800++#define IID_IDirectSoundCapture8            IID_IDirectSoundCapture+#define IID_IDirectSound3DListener8         IID_IDirectSound3DListener+#define IID_IDirectSound3DBuffer8           IID_IDirectSound3DBuffer+#define IID_IDirectSoundNotify8             IID_IDirectSoundNotify+#define IID_IDirectSoundFXGargle8           IID_IDirectSoundFXGargle+#define IID_IDirectSoundFXChorus8           IID_IDirectSoundFXChorus+#define IID_IDirectSoundFXFlanger8          IID_IDirectSoundFXFlanger+#define IID_IDirectSoundFXEcho8             IID_IDirectSoundFXEcho+#define IID_IDirectSoundFXDistortion8       IID_IDirectSoundFXDistortion+#define IID_IDirectSoundFXCompressor8       IID_IDirectSoundFXCompressor+#define IID_IDirectSoundFXParamEq8          IID_IDirectSoundFXParamEq+#define IID_IDirectSoundFXWavesReverb8      IID_IDirectSoundFXWavesReverb+#define IID_IDirectSoundFXI3DL2Reverb8      IID_IDirectSoundFXI3DL2Reverb+#define IID_IDirectSoundCaptureFXAec8       IID_IDirectSoundCaptureFXAec+#define IID_IDirectSoundCaptureFXNoiseSuppress8 IID_IDirectSoundCaptureFXNoiseSuppress+#define IID_IDirectSoundFullDuplex8         IID_IDirectSoundFullDuplex++#endif // DIRECTSOUND_VERSION >= 0x0800++//+// Compatibility typedefs+//++#ifndef _LPCWAVEFORMATEX_DEFINED+#define _LPCWAVEFORMATEX_DEFINED+typedef const WAVEFORMATEX *LPCWAVEFORMATEX;+#endif // _LPCWAVEFORMATEX_DEFINED++#ifndef __LPCGUID_DEFINED__+#define __LPCGUID_DEFINED__+typedef const GUID *LPCGUID;+#endif // __LPCGUID_DEFINED__++typedef LPDIRECTSOUND *LPLPDIRECTSOUND;+typedef LPDIRECTSOUNDBUFFER *LPLPDIRECTSOUNDBUFFER;+typedef LPDIRECTSOUND3DLISTENER *LPLPDIRECTSOUND3DLISTENER;+typedef LPDIRECTSOUND3DBUFFER *LPLPDIRECTSOUND3DBUFFER;+typedef LPDIRECTSOUNDCAPTURE *LPLPDIRECTSOUNDCAPTURE;+typedef LPDIRECTSOUNDCAPTUREBUFFER *LPLPDIRECTSOUNDCAPTUREBUFFER;+typedef LPDIRECTSOUNDNOTIFY *LPLPDIRECTSOUNDNOTIFY;++#if DIRECTSOUND_VERSION >= 0x0800+typedef LPDIRECTSOUND8 *LPLPDIRECTSOUND8;+typedef LPDIRECTSOUNDBUFFER8 *LPLPDIRECTSOUNDBUFFER8;+typedef LPDIRECTSOUNDCAPTURE8 *LPLPDIRECTSOUNDCAPTURE8;+typedef LPDIRECTSOUNDCAPTUREBUFFER8 *LPLPDIRECTSOUNDCAPTUREBUFFER8;+#endif // DIRECTSOUND_VERSION >= 0x0800++//+// Structures+//++typedef struct _DSCAPS+{+    DWORD           dwSize;+    DWORD           dwFlags;+    DWORD           dwMinSecondarySampleRate;+    DWORD           dwMaxSecondarySampleRate;+    DWORD           dwPrimaryBuffers;+    DWORD           dwMaxHwMixingAllBuffers;+    DWORD           dwMaxHwMixingStaticBuffers;+    DWORD           dwMaxHwMixingStreamingBuffers;+    DWORD           dwFreeHwMixingAllBuffers;+    DWORD           dwFreeHwMixingStaticBuffers;+    DWORD           dwFreeHwMixingStreamingBuffers;+    DWORD           dwMaxHw3DAllBuffers;+    DWORD           dwMaxHw3DStaticBuffers;+    DWORD           dwMaxHw3DStreamingBuffers;+    DWORD           dwFreeHw3DAllBuffers;+    DWORD           dwFreeHw3DStaticBuffers;+    DWORD           dwFreeHw3DStreamingBuffers;+    DWORD           dwTotalHwMemBytes;+    DWORD           dwFreeHwMemBytes;+    DWORD           dwMaxContigFreeHwMemBytes;+    DWORD           dwUnlockTransferRateHwBuffers;+    DWORD           dwPlayCpuOverheadSwBuffers;+    DWORD           dwReserved1;+    DWORD           dwReserved2;+} DSCAPS, *LPDSCAPS;++typedef const DSCAPS *LPCDSCAPS;++typedef struct _DSBCAPS+{+    DWORD           dwSize;+    DWORD           dwFlags;+    DWORD           dwBufferBytes;+    DWORD           dwUnlockTransferRate;+    DWORD           dwPlayCpuOverhead;+} DSBCAPS, *LPDSBCAPS;++typedef const DSBCAPS *LPCDSBCAPS;++#if DIRECTSOUND_VERSION >= 0x0800++    typedef struct _DSEFFECTDESC+    {+        DWORD       dwSize;+        DWORD       dwFlags;+        GUID        guidDSFXClass;+        DWORD_PTR   dwReserved1;+        DWORD_PTR   dwReserved2;+    } DSEFFECTDESC, *LPDSEFFECTDESC;+    typedef const DSEFFECTDESC *LPCDSEFFECTDESC;++    #define DSFX_LOCHARDWARE    0x00000001+    #define DSFX_LOCSOFTWARE    0x00000002++    enum+    {+        DSFXR_PRESENT,          // 0+        DSFXR_LOCHARDWARE,      // 1+        DSFXR_LOCSOFTWARE,      // 2+        DSFXR_UNALLOCATED,      // 3+        DSFXR_FAILED,           // 4+        DSFXR_UNKNOWN,          // 5+        DSFXR_SENDLOOP          // 6+    };++    typedef struct _DSCEFFECTDESC+    {+        DWORD       dwSize;+        DWORD       dwFlags;+        GUID        guidDSCFXClass;+        GUID        guidDSCFXInstance;+        DWORD       dwReserved1;+        DWORD       dwReserved2;+    } DSCEFFECTDESC, *LPDSCEFFECTDESC;+    typedef const DSCEFFECTDESC *LPCDSCEFFECTDESC;++    #define DSCFX_LOCHARDWARE   0x00000001+    #define DSCFX_LOCSOFTWARE   0x00000002++    #define DSCFXR_LOCHARDWARE  0x00000010+    #define DSCFXR_LOCSOFTWARE  0x00000020++#endif // DIRECTSOUND_VERSION >= 0x0800++typedef struct _DSBUFFERDESC+{+    DWORD           dwSize;+    DWORD           dwFlags;+    DWORD           dwBufferBytes;+    DWORD           dwReserved;+    LPWAVEFORMATEX  lpwfxFormat;+#if DIRECTSOUND_VERSION >= 0x0700+    GUID            guid3DAlgorithm;+#endif+} DSBUFFERDESC, *LPDSBUFFERDESC;++typedef const DSBUFFERDESC *LPCDSBUFFERDESC;++// Older version of this structure:++typedef struct _DSBUFFERDESC1+{+    DWORD           dwSize;+    DWORD           dwFlags;+    DWORD           dwBufferBytes;+    DWORD           dwReserved;+    LPWAVEFORMATEX  lpwfxFormat;+} DSBUFFERDESC1, *LPDSBUFFERDESC1;++typedef const DSBUFFERDESC1 *LPCDSBUFFERDESC1;++typedef struct _DS3DBUFFER+{+    DWORD           dwSize;+    D3DVECTOR       vPosition;+    D3DVECTOR       vVelocity;+    DWORD           dwInsideConeAngle;+    DWORD           dwOutsideConeAngle;+    D3DVECTOR       vConeOrientation;+    LONG            lConeOutsideVolume;+    D3DVALUE        flMinDistance;+    D3DVALUE        flMaxDistance;+    DWORD           dwMode;+} DS3DBUFFER, *LPDS3DBUFFER;++typedef const DS3DBUFFER *LPCDS3DBUFFER;++typedef struct _DS3DLISTENER+{+    DWORD           dwSize;+    D3DVECTOR       vPosition;+    D3DVECTOR       vVelocity;+    D3DVECTOR       vOrientFront;+    D3DVECTOR       vOrientTop;+    D3DVALUE        flDistanceFactor;+    D3DVALUE        flRolloffFactor;+    D3DVALUE        flDopplerFactor;+} DS3DLISTENER, *LPDS3DLISTENER;++typedef const DS3DLISTENER *LPCDS3DLISTENER;++typedef struct _DSCCAPS+{+    DWORD           dwSize;+    DWORD           dwFlags;+    DWORD           dwFormats;+    DWORD           dwChannels;+} DSCCAPS, *LPDSCCAPS;++typedef const DSCCAPS *LPCDSCCAPS;++typedef struct _DSCBUFFERDESC1+{+    DWORD           dwSize;+    DWORD           dwFlags;+    DWORD           dwBufferBytes;+    DWORD           dwReserved;+    LPWAVEFORMATEX  lpwfxFormat;+} DSCBUFFERDESC1, *LPDSCBUFFERDESC1;++typedef struct _DSCBUFFERDESC+{+    DWORD           dwSize;+    DWORD           dwFlags;+    DWORD           dwBufferBytes;+    DWORD           dwReserved;+    LPWAVEFORMATEX  lpwfxFormat;+#if DIRECTSOUND_VERSION >= 0x0800+    DWORD           dwFXCount;+    LPDSCEFFECTDESC lpDSCFXDesc;+#endif+} DSCBUFFERDESC, *LPDSCBUFFERDESC;++typedef const DSCBUFFERDESC *LPCDSCBUFFERDESC;++typedef struct _DSCBCAPS+{+    DWORD           dwSize;+    DWORD           dwFlags;+    DWORD           dwBufferBytes;+    DWORD           dwReserved;+} DSCBCAPS, *LPDSCBCAPS;++typedef const DSCBCAPS *LPCDSCBCAPS;++typedef struct _DSBPOSITIONNOTIFY+{+    DWORD           dwOffset;+    HANDLE          hEventNotify;+} DSBPOSITIONNOTIFY, *LPDSBPOSITIONNOTIFY;++typedef const DSBPOSITIONNOTIFY *LPCDSBPOSITIONNOTIFY;++//+// DirectSound API+//++typedef BOOL (CALLBACK *LPDSENUMCALLBACKA)(LPGUID, LPCSTR, LPCSTR, LPVOID);+typedef BOOL (CALLBACK *LPDSENUMCALLBACKW)(LPGUID, LPCWSTR, LPCWSTR, LPVOID);++extern HRESULT WINAPI DirectSoundCreate(LPCGUID pcGuidDevice, LPDIRECTSOUND *ppDS, LPUNKNOWN pUnkOuter);+extern HRESULT WINAPI DirectSoundEnumerateA(LPDSENUMCALLBACKA pDSEnumCallback, LPVOID pContext);+extern HRESULT WINAPI DirectSoundEnumerateW(LPDSENUMCALLBACKW pDSEnumCallback, LPVOID pContext);++extern HRESULT WINAPI DirectSoundCaptureCreate(LPCGUID pcGuidDevice, LPDIRECTSOUNDCAPTURE *ppDSC, LPUNKNOWN pUnkOuter);+extern HRESULT WINAPI DirectSoundCaptureEnumerateA(LPDSENUMCALLBACKA pDSEnumCallback, LPVOID pContext);+extern HRESULT WINAPI DirectSoundCaptureEnumerateW(LPDSENUMCALLBACKW pDSEnumCallback, LPVOID pContext);++#if DIRECTSOUND_VERSION >= 0x0800+extern HRESULT WINAPI DirectSoundCreate8(LPCGUID pcGuidDevice, LPDIRECTSOUND8 *ppDS8, LPUNKNOWN pUnkOuter);+extern HRESULT WINAPI DirectSoundCaptureCreate8(LPCGUID pcGuidDevice, LPDIRECTSOUNDCAPTURE8 *ppDSC8, LPUNKNOWN pUnkOuter);+extern HRESULT WINAPI DirectSoundFullDuplexCreate(LPCGUID pcGuidCaptureDevice, LPCGUID pcGuidRenderDevice,+        LPCDSCBUFFERDESC pcDSCBufferDesc, LPCDSBUFFERDESC pcDSBufferDesc, HWND hWnd,+        DWORD dwLevel, LPDIRECTSOUNDFULLDUPLEX* ppDSFD, LPDIRECTSOUNDCAPTUREBUFFER8 *ppDSCBuffer8,+        LPDIRECTSOUNDBUFFER8 *ppDSBuffer8, LPUNKNOWN pUnkOuter);+#define DirectSoundFullDuplexCreate8 DirectSoundFullDuplexCreate++extern HRESULT WINAPI GetDeviceID(LPCGUID pGuidSrc, LPGUID pGuidDest);+#endif // DIRECTSOUND_VERSION >= 0x0800++#ifdef UNICODE+#define LPDSENUMCALLBACK            LPDSENUMCALLBACKW+#define DirectSoundEnumerate        DirectSoundEnumerateW+#define DirectSoundCaptureEnumerate DirectSoundCaptureEnumerateW+#else // UNICODE+#define LPDSENUMCALLBACK            LPDSENUMCALLBACKA+#define DirectSoundEnumerate        DirectSoundEnumerateA+#define DirectSoundCaptureEnumerate DirectSoundCaptureEnumerateA+#endif // UNICODE++//+// IUnknown+//++#if !defined(__cplusplus) || defined(CINTERFACE)+#ifndef IUnknown_QueryInterface+#define IUnknown_QueryInterface(p,a,b)  (p)->lpVtbl->QueryInterface(p,a,b)+#endif // IUnknown_QueryInterface+#ifndef IUnknown_AddRef+#define IUnknown_AddRef(p)              (p)->lpVtbl->AddRef(p)+#endif // IUnknown_AddRef+#ifndef IUnknown_Release+#define IUnknown_Release(p)             (p)->lpVtbl->Release(p)+#endif // IUnknown_Release+#else // !defined(__cplusplus) || defined(CINTERFACE)+#ifndef IUnknown_QueryInterface+#define IUnknown_QueryInterface(p,a,b)  (p)->QueryInterface(a,b)+#endif // IUnknown_QueryInterface+#ifndef IUnknown_AddRef+#define IUnknown_AddRef(p)              (p)->AddRef()+#endif // IUnknown_AddRef+#ifndef IUnknown_Release+#define IUnknown_Release(p)             (p)->Release()+#endif // IUnknown_Release+#endif // !defined(__cplusplus) || defined(CINTERFACE)++#ifndef __IReferenceClock_INTERFACE_DEFINED__+#define __IReferenceClock_INTERFACE_DEFINED__++typedef LONGLONG REFERENCE_TIME;+typedef REFERENCE_TIME *LPREFERENCE_TIME;++DEFINE_GUID(IID_IReferenceClock, 0x56a86897, 0x0ad4, 0x11ce, 0xb0, 0x3a, 0x00, 0x20, 0xaf, 0x0b, 0xa7, 0x70);++#undef INTERFACE+#define INTERFACE IReferenceClock++DECLARE_INTERFACE_(IReferenceClock, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IReferenceClock methods+    STDMETHOD(GetTime)              (THIS_ REFERENCE_TIME *pTime) PURE;+    STDMETHOD(AdviseTime)           (THIS_ REFERENCE_TIME rtBaseTime, REFERENCE_TIME rtStreamTime,+                                           HANDLE hEvent, LPDWORD pdwAdviseCookie) PURE;+    STDMETHOD(AdvisePeriodic)       (THIS_ REFERENCE_TIME rtStartTime, REFERENCE_TIME rtPeriodTime,+                                           HANDLE hSemaphore, LPDWORD pdwAdviseCookie) PURE;+    STDMETHOD(Unadvise)             (THIS_ DWORD dwAdviseCookie) PURE;+};++#endif // __IReferenceClock_INTERFACE_DEFINED__++#ifndef IReferenceClock_QueryInterface++#define IReferenceClock_QueryInterface(p,a,b)      IUnknown_QueryInterface(p,a,b)+#define IReferenceClock_AddRef(p)                  IUnknown_AddRef(p)+#define IReferenceClock_Release(p)                 IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IReferenceClock_GetTime(p,a)               (p)->lpVtbl->GetTime(p,a)+#define IReferenceClock_AdviseTime(p,a,b,c,d)      (p)->lpVtbl->AdviseTime(p,a,b,c,d)+#define IReferenceClock_AdvisePeriodic(p,a,b,c,d)  (p)->lpVtbl->AdvisePeriodic(p,a,b,c,d)+#define IReferenceClock_Unadvise(p,a)              (p)->lpVtbl->Unadvise(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IReferenceClock_GetTime(p,a)               (p)->GetTime(a)+#define IReferenceClock_AdviseTime(p,a,b,c,d)      (p)->AdviseTime(a,b,c,d)+#define IReferenceClock_AdvisePeriodic(p,a,b,c,d)  (p)->AdvisePeriodic(a,b,c,d)+#define IReferenceClock_Unadvise(p,a)              (p)->Unadvise(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++#endif // IReferenceClock_QueryInterface++//+// IDirectSound+//++DEFINE_GUID(IID_IDirectSound, 0x279AFA83, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60);++#undef INTERFACE+#define INTERFACE IDirectSound++DECLARE_INTERFACE_(IDirectSound, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSound methods+    STDMETHOD(CreateSoundBuffer)    (THIS_ LPCDSBUFFERDESC pcDSBufferDesc, LPDIRECTSOUNDBUFFER *ppDSBuffer, LPUNKNOWN pUnkOuter) PURE;+    STDMETHOD(GetCaps)              (THIS_ LPDSCAPS pDSCaps) PURE;+    STDMETHOD(DuplicateSoundBuffer) (THIS_ LPDIRECTSOUNDBUFFER pDSBufferOriginal, LPDIRECTSOUNDBUFFER *ppDSBufferDuplicate) PURE;+    STDMETHOD(SetCooperativeLevel)  (THIS_ HWND hwnd, DWORD dwLevel) PURE;+    STDMETHOD(Compact)              (THIS) PURE;+    STDMETHOD(GetSpeakerConfig)     (THIS_ LPDWORD pdwSpeakerConfig) PURE;+    STDMETHOD(SetSpeakerConfig)     (THIS_ DWORD dwSpeakerConfig) PURE;+    STDMETHOD(Initialize)           (THIS_ LPCGUID pcGuidDevice) PURE;+};++#define IDirectSound_QueryInterface(p,a,b)       IUnknown_QueryInterface(p,a,b)+#define IDirectSound_AddRef(p)                   IUnknown_AddRef(p)+#define IDirectSound_Release(p)                  IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSound_CreateSoundBuffer(p,a,b,c)  (p)->lpVtbl->CreateSoundBuffer(p,a,b,c)+#define IDirectSound_GetCaps(p,a)                (p)->lpVtbl->GetCaps(p,a)+#define IDirectSound_DuplicateSoundBuffer(p,a,b) (p)->lpVtbl->DuplicateSoundBuffer(p,a,b)+#define IDirectSound_SetCooperativeLevel(p,a,b)  (p)->lpVtbl->SetCooperativeLevel(p,a,b)+#define IDirectSound_Compact(p)                  (p)->lpVtbl->Compact(p)+#define IDirectSound_GetSpeakerConfig(p,a)       (p)->lpVtbl->GetSpeakerConfig(p,a)+#define IDirectSound_SetSpeakerConfig(p,b)       (p)->lpVtbl->SetSpeakerConfig(p,b)+#define IDirectSound_Initialize(p,a)             (p)->lpVtbl->Initialize(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSound_CreateSoundBuffer(p,a,b,c)  (p)->CreateSoundBuffer(a,b,c)+#define IDirectSound_GetCaps(p,a)                (p)->GetCaps(a)+#define IDirectSound_DuplicateSoundBuffer(p,a,b) (p)->DuplicateSoundBuffer(a,b)+#define IDirectSound_SetCooperativeLevel(p,a,b)  (p)->SetCooperativeLevel(a,b)+#define IDirectSound_Compact(p)                  (p)->Compact()+#define IDirectSound_GetSpeakerConfig(p,a)       (p)->GetSpeakerConfig(a)+#define IDirectSound_SetSpeakerConfig(p,b)       (p)->SetSpeakerConfig(b)+#define IDirectSound_Initialize(p,a)             (p)->Initialize(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++#if DIRECTSOUND_VERSION >= 0x0800++//+// IDirectSound8+//++DEFINE_GUID(IID_IDirectSound8, 0xC50A7E93, 0xF395, 0x4834, 0x9E, 0xF6, 0x7F, 0xA9, 0x9D, 0xE5, 0x09, 0x66);++#undef INTERFACE+#define INTERFACE IDirectSound8++DECLARE_INTERFACE_(IDirectSound8, IDirectSound)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSound methods+    STDMETHOD(CreateSoundBuffer)    (THIS_ LPCDSBUFFERDESC pcDSBufferDesc, LPDIRECTSOUNDBUFFER *ppDSBuffer, LPUNKNOWN pUnkOuter) PURE;+    STDMETHOD(GetCaps)              (THIS_ LPDSCAPS pDSCaps) PURE;+    STDMETHOD(DuplicateSoundBuffer) (THIS_ LPDIRECTSOUNDBUFFER pDSBufferOriginal, LPDIRECTSOUNDBUFFER *ppDSBufferDuplicate) PURE;+    STDMETHOD(SetCooperativeLevel)  (THIS_ HWND hwnd, DWORD dwLevel) PURE;+    STDMETHOD(Compact)              (THIS) PURE;+    STDMETHOD(GetSpeakerConfig)     (THIS_ LPDWORD pdwSpeakerConfig) PURE;+    STDMETHOD(SetSpeakerConfig)     (THIS_ DWORD dwSpeakerConfig) PURE;+    STDMETHOD(Initialize)           (THIS_ LPCGUID pcGuidDevice) PURE;++    // IDirectSound8 methods+    STDMETHOD(VerifyCertification)  (THIS_ LPDWORD pdwCertified) PURE;+};++#define IDirectSound8_QueryInterface(p,a,b)       IDirectSound_QueryInterface(p,a,b)+#define IDirectSound8_AddRef(p)                   IDirectSound_AddRef(p)+#define IDirectSound8_Release(p)                  IDirectSound_Release(p)+#define IDirectSound8_CreateSoundBuffer(p,a,b,c)  IDirectSound_CreateSoundBuffer(p,a,b,c)+#define IDirectSound8_GetCaps(p,a)                IDirectSound_GetCaps(p,a)+#define IDirectSound8_DuplicateSoundBuffer(p,a,b) IDirectSound_DuplicateSoundBuffer(p,a,b)+#define IDirectSound8_SetCooperativeLevel(p,a,b)  IDirectSound_SetCooperativeLevel(p,a,b)+#define IDirectSound8_Compact(p)                  IDirectSound_Compact(p)+#define IDirectSound8_GetSpeakerConfig(p,a)       IDirectSound_GetSpeakerConfig(p,a)+#define IDirectSound8_SetSpeakerConfig(p,a)       IDirectSound_SetSpeakerConfig(p,a)+#define IDirectSound8_Initialize(p,a)             IDirectSound_Initialize(p,a)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSound8_VerifyCertification(p,a)           (p)->lpVtbl->VerifyCertification(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSound8_VerifyCertification(p,a)           (p)->VerifyCertification(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++#endif // DIRECTSOUND_VERSION >= 0x0800++//+// IDirectSoundBuffer+//++DEFINE_GUID(IID_IDirectSoundBuffer, 0x279AFA85, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60);++#undef INTERFACE+#define INTERFACE IDirectSoundBuffer++DECLARE_INTERFACE_(IDirectSoundBuffer, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundBuffer methods+    STDMETHOD(GetCaps)              (THIS_ LPDSBCAPS pDSBufferCaps) PURE;+    STDMETHOD(GetCurrentPosition)   (THIS_ LPDWORD pdwCurrentPlayCursor, LPDWORD pdwCurrentWriteCursor) PURE;+    STDMETHOD(GetFormat)            (THIS_ LPWAVEFORMATEX pwfxFormat, DWORD dwSizeAllocated, LPDWORD pdwSizeWritten) PURE;+    STDMETHOD(GetVolume)            (THIS_ LPLONG plVolume) PURE;+    STDMETHOD(GetPan)               (THIS_ LPLONG plPan) PURE;+    STDMETHOD(GetFrequency)         (THIS_ LPDWORD pdwFrequency) PURE;+    STDMETHOD(GetStatus)            (THIS_ LPDWORD pdwStatus) PURE;+    STDMETHOD(Initialize)           (THIS_ LPDIRECTSOUND pDirectSound, LPCDSBUFFERDESC pcDSBufferDesc) PURE;+    STDMETHOD(Lock)                 (THIS_ DWORD dwOffset, DWORD dwBytes, LPVOID *ppvAudioPtr1, LPDWORD pdwAudioBytes1,+                                           LPVOID *ppvAudioPtr2, LPDWORD pdwAudioBytes2, DWORD dwFlags) PURE;+    STDMETHOD(Play)                 (THIS_ DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags) PURE;+    STDMETHOD(SetCurrentPosition)   (THIS_ DWORD dwNewPosition) PURE;+    STDMETHOD(SetFormat)            (THIS_ LPCWAVEFORMATEX pcfxFormat) PURE;+    STDMETHOD(SetVolume)            (THIS_ LONG lVolume) PURE;+    STDMETHOD(SetPan)               (THIS_ LONG lPan) PURE;+    STDMETHOD(SetFrequency)         (THIS_ DWORD dwFrequency) PURE;+    STDMETHOD(Stop)                 (THIS) PURE;+    STDMETHOD(Unlock)               (THIS_ LPVOID pvAudioPtr1, DWORD dwAudioBytes1, LPVOID pvAudioPtr2, DWORD dwAudioBytes2) PURE;+    STDMETHOD(Restore)              (THIS) PURE;+};++#define IDirectSoundBuffer_QueryInterface(p,a,b)        IUnknown_QueryInterface(p,a,b)+#define IDirectSoundBuffer_AddRef(p)                    IUnknown_AddRef(p)+#define IDirectSoundBuffer_Release(p)                   IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundBuffer_GetCaps(p,a)                 (p)->lpVtbl->GetCaps(p,a)+#define IDirectSoundBuffer_GetCurrentPosition(p,a,b)    (p)->lpVtbl->GetCurrentPosition(p,a,b)+#define IDirectSoundBuffer_GetFormat(p,a,b,c)           (p)->lpVtbl->GetFormat(p,a,b,c)+#define IDirectSoundBuffer_GetVolume(p,a)               (p)->lpVtbl->GetVolume(p,a)+#define IDirectSoundBuffer_GetPan(p,a)                  (p)->lpVtbl->GetPan(p,a)+#define IDirectSoundBuffer_GetFrequency(p,a)            (p)->lpVtbl->GetFrequency(p,a)+#define IDirectSoundBuffer_GetStatus(p,a)               (p)->lpVtbl->GetStatus(p,a)+#define IDirectSoundBuffer_Initialize(p,a,b)            (p)->lpVtbl->Initialize(p,a,b)+#define IDirectSoundBuffer_Lock(p,a,b,c,d,e,f,g)        (p)->lpVtbl->Lock(p,a,b,c,d,e,f,g)+#define IDirectSoundBuffer_Play(p,a,b,c)                (p)->lpVtbl->Play(p,a,b,c)+#define IDirectSoundBuffer_SetCurrentPosition(p,a)      (p)->lpVtbl->SetCurrentPosition(p,a)+#define IDirectSoundBuffer_SetFormat(p,a)               (p)->lpVtbl->SetFormat(p,a)+#define IDirectSoundBuffer_SetVolume(p,a)               (p)->lpVtbl->SetVolume(p,a)+#define IDirectSoundBuffer_SetPan(p,a)                  (p)->lpVtbl->SetPan(p,a)+#define IDirectSoundBuffer_SetFrequency(p,a)            (p)->lpVtbl->SetFrequency(p,a)+#define IDirectSoundBuffer_Stop(p)                      (p)->lpVtbl->Stop(p)+#define IDirectSoundBuffer_Unlock(p,a,b,c,d)            (p)->lpVtbl->Unlock(p,a,b,c,d)+#define IDirectSoundBuffer_Restore(p)                   (p)->lpVtbl->Restore(p)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundBuffer_GetCaps(p,a)                 (p)->GetCaps(a)+#define IDirectSoundBuffer_GetCurrentPosition(p,a,b)    (p)->GetCurrentPosition(a,b)+#define IDirectSoundBuffer_GetFormat(p,a,b,c)           (p)->GetFormat(a,b,c)+#define IDirectSoundBuffer_GetVolume(p,a)               (p)->GetVolume(a)+#define IDirectSoundBuffer_GetPan(p,a)                  (p)->GetPan(a)+#define IDirectSoundBuffer_GetFrequency(p,a)            (p)->GetFrequency(a)+#define IDirectSoundBuffer_GetStatus(p,a)               (p)->GetStatus(a)+#define IDirectSoundBuffer_Initialize(p,a,b)            (p)->Initialize(a,b)+#define IDirectSoundBuffer_Lock(p,a,b,c,d,e,f,g)        (p)->Lock(a,b,c,d,e,f,g)+#define IDirectSoundBuffer_Play(p,a,b,c)                (p)->Play(a,b,c)+#define IDirectSoundBuffer_SetCurrentPosition(p,a)      (p)->SetCurrentPosition(a)+#define IDirectSoundBuffer_SetFormat(p,a)               (p)->SetFormat(a)+#define IDirectSoundBuffer_SetVolume(p,a)               (p)->SetVolume(a)+#define IDirectSoundBuffer_SetPan(p,a)                  (p)->SetPan(a)+#define IDirectSoundBuffer_SetFrequency(p,a)            (p)->SetFrequency(a)+#define IDirectSoundBuffer_Stop(p)                      (p)->Stop()+#define IDirectSoundBuffer_Unlock(p,a,b,c,d)            (p)->Unlock(a,b,c,d)+#define IDirectSoundBuffer_Restore(p)                   (p)->Restore()+#endif // !defined(__cplusplus) || defined(CINTERFACE)++#if DIRECTSOUND_VERSION >= 0x0800++//+// IDirectSoundBuffer8+//++DEFINE_GUID(IID_IDirectSoundBuffer8, 0x6825a449, 0x7524, 0x4d82, 0x92, 0x0f, 0x50, 0xe3, 0x6a, 0xb3, 0xab, 0x1e);++#undef INTERFACE+#define INTERFACE IDirectSoundBuffer8++DECLARE_INTERFACE_(IDirectSoundBuffer8, IDirectSoundBuffer)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundBuffer methods+    STDMETHOD(GetCaps)              (THIS_ LPDSBCAPS pDSBufferCaps) PURE;+    STDMETHOD(GetCurrentPosition)   (THIS_ LPDWORD pdwCurrentPlayCursor, LPDWORD pdwCurrentWriteCursor) PURE;+    STDMETHOD(GetFormat)            (THIS_ LPWAVEFORMATEX pwfxFormat, DWORD dwSizeAllocated, LPDWORD pdwSizeWritten) PURE;+    STDMETHOD(GetVolume)            (THIS_ LPLONG plVolume) PURE;+    STDMETHOD(GetPan)               (THIS_ LPLONG plPan) PURE;+    STDMETHOD(GetFrequency)         (THIS_ LPDWORD pdwFrequency) PURE;+    STDMETHOD(GetStatus)            (THIS_ LPDWORD pdwStatus) PURE;+    STDMETHOD(Initialize)           (THIS_ LPDIRECTSOUND pDirectSound, LPCDSBUFFERDESC pcDSBufferDesc) PURE;+    STDMETHOD(Lock)                 (THIS_ DWORD dwOffset, DWORD dwBytes, LPVOID *ppvAudioPtr1, LPDWORD pdwAudioBytes1,+                                           LPVOID *ppvAudioPtr2, LPDWORD pdwAudioBytes2, DWORD dwFlags) PURE;+    STDMETHOD(Play)                 (THIS_ DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags) PURE;+    STDMETHOD(SetCurrentPosition)   (THIS_ DWORD dwNewPosition) PURE;+    STDMETHOD(SetFormat)            (THIS_ LPCWAVEFORMATEX pcfxFormat) PURE;+    STDMETHOD(SetVolume)            (THIS_ LONG lVolume) PURE;+    STDMETHOD(SetPan)               (THIS_ LONG lPan) PURE;+    STDMETHOD(SetFrequency)         (THIS_ DWORD dwFrequency) PURE;+    STDMETHOD(Stop)                 (THIS) PURE;+    STDMETHOD(Unlock)               (THIS_ LPVOID pvAudioPtr1, DWORD dwAudioBytes1, LPVOID pvAudioPtr2, DWORD dwAudioBytes2) PURE;+    STDMETHOD(Restore)              (THIS) PURE;++    // IDirectSoundBuffer8 methods+    STDMETHOD(SetFX)                (THIS_ DWORD dwEffectsCount, LPDSEFFECTDESC pDSFXDesc, LPDWORD pdwResultCodes) PURE;+    STDMETHOD(AcquireResources)     (THIS_ DWORD dwFlags, DWORD dwEffectsCount, LPDWORD pdwResultCodes) PURE;+    STDMETHOD(GetObjectInPath)      (THIS_ REFGUID rguidObject, DWORD dwIndex, REFGUID rguidInterface, LPVOID *ppObject) PURE;+};++// Special GUID meaning "select all objects" for use in GetObjectInPath()+DEFINE_GUID(GUID_All_Objects, 0xaa114de5, 0xc262, 0x4169, 0xa1, 0xc8, 0x23, 0xd6, 0x98, 0xcc, 0x73, 0xb5);++#define IDirectSoundBuffer8_QueryInterface(p,a,b)           IUnknown_QueryInterface(p,a,b)+#define IDirectSoundBuffer8_AddRef(p)                       IUnknown_AddRef(p)+#define IDirectSoundBuffer8_Release(p)                      IUnknown_Release(p)++#define IDirectSoundBuffer8_GetCaps(p,a)                    IDirectSoundBuffer_GetCaps(p,a)+#define IDirectSoundBuffer8_GetCurrentPosition(p,a,b)       IDirectSoundBuffer_GetCurrentPosition(p,a,b)+#define IDirectSoundBuffer8_GetFormat(p,a,b,c)              IDirectSoundBuffer_GetFormat(p,a,b,c)+#define IDirectSoundBuffer8_GetVolume(p,a)                  IDirectSoundBuffer_GetVolume(p,a)+#define IDirectSoundBuffer8_GetPan(p,a)                     IDirectSoundBuffer_GetPan(p,a)+#define IDirectSoundBuffer8_GetFrequency(p,a)               IDirectSoundBuffer_GetFrequency(p,a)+#define IDirectSoundBuffer8_GetStatus(p,a)                  IDirectSoundBuffer_GetStatus(p,a)+#define IDirectSoundBuffer8_Initialize(p,a,b)               IDirectSoundBuffer_Initialize(p,a,b)+#define IDirectSoundBuffer8_Lock(p,a,b,c,d,e,f,g)           IDirectSoundBuffer_Lock(p,a,b,c,d,e,f,g)+#define IDirectSoundBuffer8_Play(p,a,b,c)                   IDirectSoundBuffer_Play(p,a,b,c)+#define IDirectSoundBuffer8_SetCurrentPosition(p,a)         IDirectSoundBuffer_SetCurrentPosition(p,a)+#define IDirectSoundBuffer8_SetFormat(p,a)                  IDirectSoundBuffer_SetFormat(p,a)+#define IDirectSoundBuffer8_SetVolume(p,a)                  IDirectSoundBuffer_SetVolume(p,a)+#define IDirectSoundBuffer8_SetPan(p,a)                     IDirectSoundBuffer_SetPan(p,a)+#define IDirectSoundBuffer8_SetFrequency(p,a)               IDirectSoundBuffer_SetFrequency(p,a)+#define IDirectSoundBuffer8_Stop(p)                         IDirectSoundBuffer_Stop(p)+#define IDirectSoundBuffer8_Unlock(p,a,b,c,d)               IDirectSoundBuffer_Unlock(p,a,b,c,d)+#define IDirectSoundBuffer8_Restore(p)                      IDirectSoundBuffer_Restore(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundBuffer8_SetFX(p,a,b,c)                  (p)->lpVtbl->SetFX(p,a,b,c)+#define IDirectSoundBuffer8_AcquireResources(p,a,b,c)       (p)->lpVtbl->AcquireResources(p,a,b,c)+#define IDirectSoundBuffer8_GetObjectInPath(p,a,b,c,d)      (p)->lpVtbl->GetObjectInPath(p,a,b,c,d)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundBuffer8_SetFX(p,a,b,c)                  (p)->SetFX(a,b,c)+#define IDirectSoundBuffer8_AcquireResources(p,a,b,c)       (p)->AcquireResources(a,b,c)+#define IDirectSoundBuffer8_GetObjectInPath(p,a,b,c,d)      (p)->GetObjectInPath(a,b,c,d)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++#endif // DIRECTSOUND_VERSION >= 0x0800++//+// IDirectSound3DListener+//++DEFINE_GUID(IID_IDirectSound3DListener, 0x279AFA84, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60);++#undef INTERFACE+#define INTERFACE IDirectSound3DListener++DECLARE_INTERFACE_(IDirectSound3DListener, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)           (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)            (THIS) PURE;+    STDMETHOD_(ULONG,Release)           (THIS) PURE;++    // IDirectSound3DListener methods+    STDMETHOD(GetAllParameters)         (THIS_ LPDS3DLISTENER pListener) PURE;+    STDMETHOD(GetDistanceFactor)        (THIS_ D3DVALUE* pflDistanceFactor) PURE;+    STDMETHOD(GetDopplerFactor)         (THIS_ D3DVALUE* pflDopplerFactor) PURE;+    STDMETHOD(GetOrientation)           (THIS_ D3DVECTOR* pvOrientFront, D3DVECTOR* pvOrientTop) PURE;+    STDMETHOD(GetPosition)              (THIS_ D3DVECTOR* pvPosition) PURE;+    STDMETHOD(GetRolloffFactor)         (THIS_ D3DVALUE* pflRolloffFactor) PURE;+    STDMETHOD(GetVelocity)              (THIS_ D3DVECTOR* pvVelocity) PURE;+    STDMETHOD(SetAllParameters)         (THIS_ LPCDS3DLISTENER pcListener, DWORD dwApply) PURE;+    STDMETHOD(SetDistanceFactor)        (THIS_ D3DVALUE flDistanceFactor, DWORD dwApply) PURE;+    STDMETHOD(SetDopplerFactor)         (THIS_ D3DVALUE flDopplerFactor, DWORD dwApply) PURE;+    STDMETHOD(SetOrientation)           (THIS_ D3DVALUE xFront, D3DVALUE yFront, D3DVALUE zFront,+                                               D3DVALUE xTop, D3DVALUE yTop, D3DVALUE zTop, DWORD dwApply) PURE;+    STDMETHOD(SetPosition)              (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE;+    STDMETHOD(SetRolloffFactor)         (THIS_ D3DVALUE flRolloffFactor, DWORD dwApply) PURE;+    STDMETHOD(SetVelocity)              (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE;+    STDMETHOD(CommitDeferredSettings)   (THIS) PURE;+};++#define IDirectSound3DListener_QueryInterface(p,a,b)            IUnknown_QueryInterface(p,a,b)+#define IDirectSound3DListener_AddRef(p)                        IUnknown_AddRef(p)+#define IDirectSound3DListener_Release(p)                       IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSound3DListener_GetAllParameters(p,a)            (p)->lpVtbl->GetAllParameters(p,a)+#define IDirectSound3DListener_GetDistanceFactor(p,a)           (p)->lpVtbl->GetDistanceFactor(p,a)+#define IDirectSound3DListener_GetDopplerFactor(p,a)            (p)->lpVtbl->GetDopplerFactor(p,a)+#define IDirectSound3DListener_GetOrientation(p,a,b)            (p)->lpVtbl->GetOrientation(p,a,b)+#define IDirectSound3DListener_GetPosition(p,a)                 (p)->lpVtbl->GetPosition(p,a)+#define IDirectSound3DListener_GetRolloffFactor(p,a)            (p)->lpVtbl->GetRolloffFactor(p,a)+#define IDirectSound3DListener_GetVelocity(p,a)                 (p)->lpVtbl->GetVelocity(p,a)+#define IDirectSound3DListener_SetAllParameters(p,a,b)          (p)->lpVtbl->SetAllParameters(p,a,b)+#define IDirectSound3DListener_SetDistanceFactor(p,a,b)         (p)->lpVtbl->SetDistanceFactor(p,a,b)+#define IDirectSound3DListener_SetDopplerFactor(p,a,b)          (p)->lpVtbl->SetDopplerFactor(p,a,b)+#define IDirectSound3DListener_SetOrientation(p,a,b,c,d,e,f,g)  (p)->lpVtbl->SetOrientation(p,a,b,c,d,e,f,g)+#define IDirectSound3DListener_SetPosition(p,a,b,c,d)           (p)->lpVtbl->SetPosition(p,a,b,c,d)+#define IDirectSound3DListener_SetRolloffFactor(p,a,b)          (p)->lpVtbl->SetRolloffFactor(p,a,b)+#define IDirectSound3DListener_SetVelocity(p,a,b,c,d)           (p)->lpVtbl->SetVelocity(p,a,b,c,d)+#define IDirectSound3DListener_CommitDeferredSettings(p)        (p)->lpVtbl->CommitDeferredSettings(p)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSound3DListener_GetAllParameters(p,a)            (p)->GetAllParameters(a)+#define IDirectSound3DListener_GetDistanceFactor(p,a)           (p)->GetDistanceFactor(a)+#define IDirectSound3DListener_GetDopplerFactor(p,a)            (p)->GetDopplerFactor(a)+#define IDirectSound3DListener_GetOrientation(p,a,b)            (p)->GetOrientation(a,b)+#define IDirectSound3DListener_GetPosition(p,a)                 (p)->GetPosition(a)+#define IDirectSound3DListener_GetRolloffFactor(p,a)            (p)->GetRolloffFactor(a)+#define IDirectSound3DListener_GetVelocity(p,a)                 (p)->GetVelocity(a)+#define IDirectSound3DListener_SetAllParameters(p,a,b)          (p)->SetAllParameters(a,b)+#define IDirectSound3DListener_SetDistanceFactor(p,a,b)         (p)->SetDistanceFactor(a,b)+#define IDirectSound3DListener_SetDopplerFactor(p,a,b)          (p)->SetDopplerFactor(a,b)+#define IDirectSound3DListener_SetOrientation(p,a,b,c,d,e,f,g)  (p)->SetOrientation(a,b,c,d,e,f,g)+#define IDirectSound3DListener_SetPosition(p,a,b,c,d)           (p)->SetPosition(a,b,c,d)+#define IDirectSound3DListener_SetRolloffFactor(p,a,b)          (p)->SetRolloffFactor(a,b)+#define IDirectSound3DListener_SetVelocity(p,a,b,c,d)           (p)->SetVelocity(a,b,c,d)+#define IDirectSound3DListener_CommitDeferredSettings(p)        (p)->CommitDeferredSettings()+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSound3DBuffer+//++DEFINE_GUID(IID_IDirectSound3DBuffer, 0x279AFA86, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60);++#undef INTERFACE+#define INTERFACE IDirectSound3DBuffer++DECLARE_INTERFACE_(IDirectSound3DBuffer, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSound3DBuffer methods+    STDMETHOD(GetAllParameters)     (THIS_ LPDS3DBUFFER pDs3dBuffer) PURE;+    STDMETHOD(GetConeAngles)        (THIS_ LPDWORD pdwInsideConeAngle, LPDWORD pdwOutsideConeAngle) PURE;+    STDMETHOD(GetConeOrientation)   (THIS_ D3DVECTOR* pvOrientation) PURE;+    STDMETHOD(GetConeOutsideVolume) (THIS_ LPLONG plConeOutsideVolume) PURE;+    STDMETHOD(GetMaxDistance)       (THIS_ D3DVALUE* pflMaxDistance) PURE;+    STDMETHOD(GetMinDistance)       (THIS_ D3DVALUE* pflMinDistance) PURE;+    STDMETHOD(GetMode)              (THIS_ LPDWORD pdwMode) PURE;+    STDMETHOD(GetPosition)          (THIS_ D3DVECTOR* pvPosition) PURE;+    STDMETHOD(GetVelocity)          (THIS_ D3DVECTOR* pvVelocity) PURE;+    STDMETHOD(SetAllParameters)     (THIS_ LPCDS3DBUFFER pcDs3dBuffer, DWORD dwApply) PURE;+    STDMETHOD(SetConeAngles)        (THIS_ DWORD dwInsideConeAngle, DWORD dwOutsideConeAngle, DWORD dwApply) PURE;+    STDMETHOD(SetConeOrientation)   (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE;+    STDMETHOD(SetConeOutsideVolume) (THIS_ LONG lConeOutsideVolume, DWORD dwApply) PURE;+    STDMETHOD(SetMaxDistance)       (THIS_ D3DVALUE flMaxDistance, DWORD dwApply) PURE;+    STDMETHOD(SetMinDistance)       (THIS_ D3DVALUE flMinDistance, DWORD dwApply) PURE;+    STDMETHOD(SetMode)              (THIS_ DWORD dwMode, DWORD dwApply) PURE;+    STDMETHOD(SetPosition)          (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE;+    STDMETHOD(SetVelocity)          (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE;+};++#define IDirectSound3DBuffer_QueryInterface(p,a,b)          IUnknown_QueryInterface(p,a,b)+#define IDirectSound3DBuffer_AddRef(p)                      IUnknown_AddRef(p)+#define IDirectSound3DBuffer_Release(p)                     IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSound3DBuffer_GetAllParameters(p,a)          (p)->lpVtbl->GetAllParameters(p,a)+#define IDirectSound3DBuffer_GetConeAngles(p,a,b)           (p)->lpVtbl->GetConeAngles(p,a,b)+#define IDirectSound3DBuffer_GetConeOrientation(p,a)        (p)->lpVtbl->GetConeOrientation(p,a)+#define IDirectSound3DBuffer_GetConeOutsideVolume(p,a)      (p)->lpVtbl->GetConeOutsideVolume(p,a)+#define IDirectSound3DBuffer_GetPosition(p,a)               (p)->lpVtbl->GetPosition(p,a)+#define IDirectSound3DBuffer_GetMinDistance(p,a)            (p)->lpVtbl->GetMinDistance(p,a)+#define IDirectSound3DBuffer_GetMaxDistance(p,a)            (p)->lpVtbl->GetMaxDistance(p,a)+#define IDirectSound3DBuffer_GetMode(p,a)                   (p)->lpVtbl->GetMode(p,a)+#define IDirectSound3DBuffer_GetVelocity(p,a)               (p)->lpVtbl->GetVelocity(p,a)+#define IDirectSound3DBuffer_SetAllParameters(p,a,b)        (p)->lpVtbl->SetAllParameters(p,a,b)+#define IDirectSound3DBuffer_SetConeAngles(p,a,b,c)         (p)->lpVtbl->SetConeAngles(p,a,b,c)+#define IDirectSound3DBuffer_SetConeOrientation(p,a,b,c,d)  (p)->lpVtbl->SetConeOrientation(p,a,b,c,d)+#define IDirectSound3DBuffer_SetConeOutsideVolume(p,a,b)    (p)->lpVtbl->SetConeOutsideVolume(p,a,b)+#define IDirectSound3DBuffer_SetPosition(p,a,b,c,d)         (p)->lpVtbl->SetPosition(p,a,b,c,d)+#define IDirectSound3DBuffer_SetMinDistance(p,a,b)          (p)->lpVtbl->SetMinDistance(p,a,b)+#define IDirectSound3DBuffer_SetMaxDistance(p,a,b)          (p)->lpVtbl->SetMaxDistance(p,a,b)+#define IDirectSound3DBuffer_SetMode(p,a,b)                 (p)->lpVtbl->SetMode(p,a,b)+#define IDirectSound3DBuffer_SetVelocity(p,a,b,c,d)         (p)->lpVtbl->SetVelocity(p,a,b,c,d)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSound3DBuffer_GetAllParameters(p,a)          (p)->GetAllParameters(a)+#define IDirectSound3DBuffer_GetConeAngles(p,a,b)           (p)->GetConeAngles(a,b)+#define IDirectSound3DBuffer_GetConeOrientation(p,a)        (p)->GetConeOrientation(a)+#define IDirectSound3DBuffer_GetConeOutsideVolume(p,a)      (p)->GetConeOutsideVolume(a)+#define IDirectSound3DBuffer_GetPosition(p,a)               (p)->GetPosition(a)+#define IDirectSound3DBuffer_GetMinDistance(p,a)            (p)->GetMinDistance(a)+#define IDirectSound3DBuffer_GetMaxDistance(p,a)            (p)->GetMaxDistance(a)+#define IDirectSound3DBuffer_GetMode(p,a)                   (p)->GetMode(a)+#define IDirectSound3DBuffer_GetVelocity(p,a)               (p)->GetVelocity(a)+#define IDirectSound3DBuffer_SetAllParameters(p,a,b)        (p)->SetAllParameters(a,b)+#define IDirectSound3DBuffer_SetConeAngles(p,a,b,c)         (p)->SetConeAngles(a,b,c)+#define IDirectSound3DBuffer_SetConeOrientation(p,a,b,c,d)  (p)->SetConeOrientation(a,b,c,d)+#define IDirectSound3DBuffer_SetConeOutsideVolume(p,a,b)    (p)->SetConeOutsideVolume(a,b)+#define IDirectSound3DBuffer_SetPosition(p,a,b,c,d)         (p)->SetPosition(a,b,c,d)+#define IDirectSound3DBuffer_SetMinDistance(p,a,b)          (p)->SetMinDistance(a,b)+#define IDirectSound3DBuffer_SetMaxDistance(p,a,b)          (p)->SetMaxDistance(a,b)+#define IDirectSound3DBuffer_SetMode(p,a,b)                 (p)->SetMode(a,b)+#define IDirectSound3DBuffer_SetVelocity(p,a,b,c,d)         (p)->SetVelocity(a,b,c,d)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundCapture+//++DEFINE_GUID(IID_IDirectSoundCapture, 0xb0210781, 0x89cd, 0x11d0, 0xaf, 0x8, 0x0, 0xa0, 0xc9, 0x25, 0xcd, 0x16);++#undef INTERFACE+#define INTERFACE IDirectSoundCapture++DECLARE_INTERFACE_(IDirectSoundCapture, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundCapture methods+    STDMETHOD(CreateCaptureBuffer)  (THIS_ LPCDSCBUFFERDESC pcDSCBufferDesc, LPDIRECTSOUNDCAPTUREBUFFER *ppDSCBuffer, LPUNKNOWN pUnkOuter) PURE;+    STDMETHOD(GetCaps)              (THIS_ LPDSCCAPS pDSCCaps) PURE;+    STDMETHOD(Initialize)           (THIS_ LPCGUID pcGuidDevice) PURE;+};++#define IDirectSoundCapture_QueryInterface(p,a,b)           IUnknown_QueryInterface(p,a,b)+#define IDirectSoundCapture_AddRef(p)                       IUnknown_AddRef(p)+#define IDirectSoundCapture_Release(p)                      IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCapture_CreateCaptureBuffer(p,a,b,c)    (p)->lpVtbl->CreateCaptureBuffer(p,a,b,c)+#define IDirectSoundCapture_GetCaps(p,a)                    (p)->lpVtbl->GetCaps(p,a)+#define IDirectSoundCapture_Initialize(p,a)                 (p)->lpVtbl->Initialize(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCapture_CreateCaptureBuffer(p,a,b,c)    (p)->CreateCaptureBuffer(a,b,c)+#define IDirectSoundCapture_GetCaps(p,a)                    (p)->GetCaps(a)+#define IDirectSoundCapture_Initialize(p,a)                 (p)->Initialize(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundCaptureBuffer+//++DEFINE_GUID(IID_IDirectSoundCaptureBuffer, 0xb0210782, 0x89cd, 0x11d0, 0xaf, 0x8, 0x0, 0xa0, 0xc9, 0x25, 0xcd, 0x16);++#undef INTERFACE+#define INTERFACE IDirectSoundCaptureBuffer++DECLARE_INTERFACE_(IDirectSoundCaptureBuffer, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundCaptureBuffer methods+    STDMETHOD(GetCaps)              (THIS_ LPDSCBCAPS pDSCBCaps) PURE;+    STDMETHOD(GetCurrentPosition)   (THIS_ LPDWORD pdwCapturePosition, LPDWORD pdwReadPosition) PURE;+    STDMETHOD(GetFormat)            (THIS_ LPWAVEFORMATEX pwfxFormat, DWORD dwSizeAllocated, LPDWORD pdwSizeWritten) PURE;+    STDMETHOD(GetStatus)            (THIS_ LPDWORD pdwStatus) PURE;+    STDMETHOD(Initialize)           (THIS_ LPDIRECTSOUNDCAPTURE pDirectSoundCapture, LPCDSCBUFFERDESC pcDSCBufferDesc) PURE;+    STDMETHOD(Lock)                 (THIS_ DWORD dwOffset, DWORD dwBytes, LPVOID *ppvAudioPtr1, LPDWORD pdwAudioBytes1,+                                           LPVOID *ppvAudioPtr2, LPDWORD pdwAudioBytes2, DWORD dwFlags) PURE;+    STDMETHOD(Start)                (THIS_ DWORD dwFlags) PURE;+    STDMETHOD(Stop)                 (THIS) PURE;+    STDMETHOD(Unlock)               (THIS_ LPVOID pvAudioPtr1, DWORD dwAudioBytes1, LPVOID pvAudioPtr2, DWORD dwAudioBytes2) PURE;+};++#define IDirectSoundCaptureBuffer_QueryInterface(p,a,b)         IUnknown_QueryInterface(p,a,b)+#define IDirectSoundCaptureBuffer_AddRef(p)                     IUnknown_AddRef(p)+#define IDirectSoundCaptureBuffer_Release(p)                    IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCaptureBuffer_GetCaps(p,a)                  (p)->lpVtbl->GetCaps(p,a)+#define IDirectSoundCaptureBuffer_GetCurrentPosition(p,a,b)     (p)->lpVtbl->GetCurrentPosition(p,a,b)+#define IDirectSoundCaptureBuffer_GetFormat(p,a,b,c)            (p)->lpVtbl->GetFormat(p,a,b,c)+#define IDirectSoundCaptureBuffer_GetStatus(p,a)                (p)->lpVtbl->GetStatus(p,a)+#define IDirectSoundCaptureBuffer_Initialize(p,a,b)             (p)->lpVtbl->Initialize(p,a,b)+#define IDirectSoundCaptureBuffer_Lock(p,a,b,c,d,e,f,g)         (p)->lpVtbl->Lock(p,a,b,c,d,e,f,g)+#define IDirectSoundCaptureBuffer_Start(p,a)                    (p)->lpVtbl->Start(p,a)+#define IDirectSoundCaptureBuffer_Stop(p)                       (p)->lpVtbl->Stop(p)+#define IDirectSoundCaptureBuffer_Unlock(p,a,b,c,d)             (p)->lpVtbl->Unlock(p,a,b,c,d)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCaptureBuffer_GetCaps(p,a)                  (p)->GetCaps(a)+#define IDirectSoundCaptureBuffer_GetCurrentPosition(p,a,b)     (p)->GetCurrentPosition(a,b)+#define IDirectSoundCaptureBuffer_GetFormat(p,a,b,c)            (p)->GetFormat(a,b,c)+#define IDirectSoundCaptureBuffer_GetStatus(p,a)                (p)->GetStatus(a)+#define IDirectSoundCaptureBuffer_Initialize(p,a,b)             (p)->Initialize(a,b)+#define IDirectSoundCaptureBuffer_Lock(p,a,b,c,d,e,f,g)         (p)->Lock(a,b,c,d,e,f,g)+#define IDirectSoundCaptureBuffer_Start(p,a)                    (p)->Start(a)+#define IDirectSoundCaptureBuffer_Stop(p)                       (p)->Stop()+#define IDirectSoundCaptureBuffer_Unlock(p,a,b,c,d)             (p)->Unlock(a,b,c,d)+#endif // !defined(__cplusplus) || defined(CINTERFACE)+++#if DIRECTSOUND_VERSION >= 0x0800++//+// IDirectSoundCaptureBuffer8+//++DEFINE_GUID(IID_IDirectSoundCaptureBuffer8, 0x990df4, 0xdbb, 0x4872, 0x83, 0x3e, 0x6d, 0x30, 0x3e, 0x80, 0xae, 0xb6);++#undef INTERFACE+#define INTERFACE IDirectSoundCaptureBuffer8++DECLARE_INTERFACE_(IDirectSoundCaptureBuffer8, IDirectSoundCaptureBuffer)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundCaptureBuffer methods+    STDMETHOD(GetCaps)              (THIS_ LPDSCBCAPS pDSCBCaps) PURE;+    STDMETHOD(GetCurrentPosition)   (THIS_ LPDWORD pdwCapturePosition, LPDWORD pdwReadPosition) PURE;+    STDMETHOD(GetFormat)            (THIS_ LPWAVEFORMATEX pwfxFormat, DWORD dwSizeAllocated, LPDWORD pdwSizeWritten) PURE;+    STDMETHOD(GetStatus)            (THIS_ LPDWORD pdwStatus) PURE;+    STDMETHOD(Initialize)           (THIS_ LPDIRECTSOUNDCAPTURE pDirectSoundCapture, LPCDSCBUFFERDESC pcDSCBufferDesc) PURE;+    STDMETHOD(Lock)                 (THIS_ DWORD dwOffset, DWORD dwBytes, LPVOID *ppvAudioPtr1, LPDWORD pdwAudioBytes1,+                                           LPVOID *ppvAudioPtr2, LPDWORD pdwAudioBytes2, DWORD dwFlags) PURE;+    STDMETHOD(Start)                (THIS_ DWORD dwFlags) PURE;+    STDMETHOD(Stop)                 (THIS) PURE;+    STDMETHOD(Unlock)               (THIS_ LPVOID pvAudioPtr1, DWORD dwAudioBytes1, LPVOID pvAudioPtr2, DWORD dwAudioBytes2) PURE;++    // IDirectSoundCaptureBuffer8 methods+    STDMETHOD(GetObjectInPath)      (THIS_ REFGUID rguidObject, DWORD dwIndex, REFGUID rguidInterface, LPVOID *ppObject) PURE;+    STDMETHOD(GetFXStatus)          (DWORD dwFXCount, LPDWORD pdwFXStatus) PURE;+};++#define IDirectSoundCaptureBuffer8_QueryInterface(p,a,b)            IUnknown_QueryInterface(p,a,b)+#define IDirectSoundCaptureBuffer8_AddRef(p)                        IUnknown_AddRef(p)+#define IDirectSoundCaptureBuffer8_Release(p)                       IUnknown_Release(p)++#define IDirectSoundCaptureBuffer8_GetCaps(p,a)                     IDirectSoundCaptureBuffer_GetCaps(p,a)+#define IDirectSoundCaptureBuffer8_GetCurrentPosition(p,a,b)        IDirectSoundCaptureBuffer_GetCurrentPosition(p,a,b)+#define IDirectSoundCaptureBuffer8_GetFormat(p,a,b,c)               IDirectSoundCaptureBuffer_GetFormat(p,a,b,c)+#define IDirectSoundCaptureBuffer8_GetStatus(p,a)                   IDirectSoundCaptureBuffer_GetStatus(p,a)+#define IDirectSoundCaptureBuffer8_Initialize(p,a,b)                IDirectSoundCaptureBuffer_Initialize(p,a,b)+#define IDirectSoundCaptureBuffer8_Lock(p,a,b,c,d,e,f,g)            IDirectSoundCaptureBuffer_Lock(p,a,b,c,d,e,f,g)+#define IDirectSoundCaptureBuffer8_Start(p,a)                       IDirectSoundCaptureBuffer_Start(p,a)+#define IDirectSoundCaptureBuffer8_Stop(p)                          IDirectSoundCaptureBuffer_Stop(p))+#define IDirectSoundCaptureBuffer8_Unlock(p,a,b,c,d)                IDirectSoundCaptureBuffer_Unlock(p,a,b,c,d)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCaptureBuffer8_GetObjectInPath(p,a,b,c,d)       (p)->lpVtbl->GetObjectInPath(p,a,b,c,d)+#define IDirectSoundCaptureBuffer8_GetFXStatus(p,a,b)               (p)->lpVtbl->GetFXStatus(p,a,b)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCaptureBuffer8_GetObjectInPath(p,a,b,c,d)       (p)->GetObjectInPath(a,b,c,d)+#define IDirectSoundCaptureBuffer8_GetFXStatus(p,a,b)               (p)->GetFXStatus(a,b)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++#endif // DIRECTSOUND_VERSION >= 0x0800++//+// IDirectSoundNotify+//++DEFINE_GUID(IID_IDirectSoundNotify, 0xb0210783, 0x89cd, 0x11d0, 0xaf, 0x8, 0x0, 0xa0, 0xc9, 0x25, 0xcd, 0x16);++#undef INTERFACE+#define INTERFACE IDirectSoundNotify++DECLARE_INTERFACE_(IDirectSoundNotify, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)           (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)            (THIS) PURE;+    STDMETHOD_(ULONG,Release)           (THIS) PURE;++    // IDirectSoundNotify methods+    STDMETHOD(SetNotificationPositions) (THIS_ DWORD dwPositionNotifies, LPCDSBPOSITIONNOTIFY pcPositionNotifies) PURE;+};++#define IDirectSoundNotify_QueryInterface(p,a,b)            IUnknown_QueryInterface(p,a,b)+#define IDirectSoundNotify_AddRef(p)                        IUnknown_AddRef(p)+#define IDirectSoundNotify_Release(p)                       IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundNotify_SetNotificationPositions(p,a,b)  (p)->lpVtbl->SetNotificationPositions(p,a,b)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundNotify_SetNotificationPositions(p,a,b)  (p)->SetNotificationPositions(a,b)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IKsPropertySet+//++#ifndef _IKsPropertySet_+#define _IKsPropertySet_++#ifdef __cplusplus+// 'struct' not 'class' per the way DECLARE_INTERFACE_ is defined+struct IKsPropertySet;+#endif // __cplusplus++typedef struct IKsPropertySet *LPKSPROPERTYSET;++#define KSPROPERTY_SUPPORT_GET  0x00000001+#define KSPROPERTY_SUPPORT_SET  0x00000002++DEFINE_GUID(IID_IKsPropertySet, 0x31efac30, 0x515c, 0x11d0, 0xa9, 0xaa, 0x00, 0xaa, 0x00, 0x61, 0xbe, 0x93);++#undef INTERFACE+#define INTERFACE IKsPropertySet++DECLARE_INTERFACE_(IKsPropertySet, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)   (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)    (THIS) PURE;+    STDMETHOD_(ULONG,Release)   (THIS) PURE;++    // IKsPropertySet methods+    STDMETHOD(Get)              (THIS_ REFGUID rguidPropSet, ULONG ulId, LPVOID pInstanceData, ULONG ulInstanceLength,+                                       LPVOID pPropertyData, ULONG ulDataLength, PULONG pulBytesReturned) PURE;+    STDMETHOD(Set)              (THIS_ REFGUID rguidPropSet, ULONG ulId, LPVOID pInstanceData, ULONG ulInstanceLength,+                                       LPVOID pPropertyData, ULONG ulDataLength) PURE;+    STDMETHOD(QuerySupport)     (THIS_ REFGUID rguidPropSet, ULONG ulId, PULONG pulTypeSupport) PURE;+};++#define IKsPropertySet_QueryInterface(p,a,b)       IUnknown_QueryInterface(p,a,b)+#define IKsPropertySet_AddRef(p)                   IUnknown_AddRef(p)+#define IKsPropertySet_Release(p)                  IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IKsPropertySet_Get(p,a,b,c,d,e,f,g)        (p)->lpVtbl->Get(p,a,b,c,d,e,f,g)+#define IKsPropertySet_Set(p,a,b,c,d,e,f)          (p)->lpVtbl->Set(p,a,b,c,d,e,f)+#define IKsPropertySet_QuerySupport(p,a,b,c)       (p)->lpVtbl->QuerySupport(p,a,b,c)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IKsPropertySet_Get(p,a,b,c,d,e,f,g)        (p)->Get(a,b,c,d,e,f,g)+#define IKsPropertySet_Set(p,a,b,c,d,e,f)          (p)->Set(a,b,c,d,e,f)+#define IKsPropertySet_QuerySupport(p,a,b,c)       (p)->QuerySupport(a,b,c)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++#endif // _IKsPropertySet_++#if DIRECTSOUND_VERSION >= 0x0800++//+// IDirectSoundFXGargle+//++DEFINE_GUID(IID_IDirectSoundFXGargle, 0xd616f352, 0xd622, 0x11ce, 0xaa, 0xc5, 0x00, 0x20, 0xaf, 0x0b, 0x99, 0xa3);++typedef struct _DSFXGargle+{+    DWORD       dwRateHz;               // Rate of modulation in hz+    DWORD       dwWaveShape;            // DSFXGARGLE_WAVE_xxx+} DSFXGargle, *LPDSFXGargle;++#define DSFXGARGLE_WAVE_TRIANGLE        0+#define DSFXGARGLE_WAVE_SQUARE          1++typedef const DSFXGargle *LPCDSFXGargle;++#define DSFXGARGLE_RATEHZ_MIN           1+#define DSFXGARGLE_RATEHZ_MAX           1000++#undef INTERFACE+#define INTERFACE IDirectSoundFXGargle++DECLARE_INTERFACE_(IDirectSoundFXGargle, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundFXGargle methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSFXGargle pcDsFxGargle) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSFXGargle pDsFxGargle) PURE;+};++#define IDirectSoundFXGargle_QueryInterface(p,a,b)          IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFXGargle_AddRef(p)                      IUnknown_AddRef(p)+#define IDirectSoundFXGargle_Release(p)                     IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXGargle_SetAllParameters(p,a)          (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundFXGargle_GetAllParameters(p,a)          (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXGargle_SetAllParameters(p,a)          (p)->SetAllParameters(a)+#define IDirectSoundFXGargle_GetAllParameters(p,a)          (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundFXChorus+//++DEFINE_GUID(IID_IDirectSoundFXChorus, 0x880842e3, 0x145f, 0x43e6, 0xa9, 0x34, 0xa7, 0x18, 0x06, 0xe5, 0x05, 0x47);++typedef struct _DSFXChorus+{+    FLOAT       fWetDryMix;+    FLOAT       fDepth;+    FLOAT       fFeedback;+    FLOAT       fFrequency;+    LONG        lWaveform;          // LFO shape; DSFXCHORUS_WAVE_xxx+    FLOAT       fDelay;+    LONG        lPhase;+} DSFXChorus, *LPDSFXChorus;++typedef const DSFXChorus *LPCDSFXChorus;++#define DSFXCHORUS_WAVE_TRIANGLE        0+#define DSFXCHORUS_WAVE_SIN             1++#define DSFXCHORUS_WETDRYMIX_MIN        0.0f+#define DSFXCHORUS_WETDRYMIX_MAX        100.0f+#define DSFXCHORUS_DEPTH_MIN            0.0f+#define DSFXCHORUS_DEPTH_MAX            100.0f+#define DSFXCHORUS_FEEDBACK_MIN         -99.0f+#define DSFXCHORUS_FEEDBACK_MAX         99.0f+#define DSFXCHORUS_FREQUENCY_MIN        0.0f+#define DSFXCHORUS_FREQUENCY_MAX        10.0f+#define DSFXCHORUS_DELAY_MIN            0.0f+#define DSFXCHORUS_DELAY_MAX            20.0f+#define DSFXCHORUS_PHASE_MIN            0+#define DSFXCHORUS_PHASE_MAX            4++#define DSFXCHORUS_PHASE_NEG_180        0+#define DSFXCHORUS_PHASE_NEG_90         1+#define DSFXCHORUS_PHASE_ZERO           2+#define DSFXCHORUS_PHASE_90             3+#define DSFXCHORUS_PHASE_180            4++#undef INTERFACE+#define INTERFACE IDirectSoundFXChorus++DECLARE_INTERFACE_(IDirectSoundFXChorus, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundFXChorus methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSFXChorus pcDsFxChorus) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSFXChorus pDsFxChorus) PURE;+};++#define IDirectSoundFXChorus_QueryInterface(p,a,b)          IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFXChorus_AddRef(p)                      IUnknown_AddRef(p)+#define IDirectSoundFXChorus_Release(p)                     IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXChorus_SetAllParameters(p,a)          (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundFXChorus_GetAllParameters(p,a)          (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXChorus_SetAllParameters(p,a)          (p)->SetAllParameters(a)+#define IDirectSoundFXChorus_GetAllParameters(p,a)          (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundFXFlanger+//++DEFINE_GUID(IID_IDirectSoundFXFlanger, 0x903e9878, 0x2c92, 0x4072, 0x9b, 0x2c, 0xea, 0x68, 0xf5, 0x39, 0x67, 0x83);++typedef struct _DSFXFlanger+{+    FLOAT       fWetDryMix;+    FLOAT       fDepth;+    FLOAT       fFeedback;+    FLOAT       fFrequency;+    LONG        lWaveform;+    FLOAT       fDelay;+    LONG        lPhase;+} DSFXFlanger, *LPDSFXFlanger;++typedef const DSFXFlanger *LPCDSFXFlanger;++#define DSFXFLANGER_WAVE_TRIANGLE       0+#define DSFXFLANGER_WAVE_SIN            1++#define DSFXFLANGER_WETDRYMIX_MIN       0.0f+#define DSFXFLANGER_WETDRYMIX_MAX       100.0f+#define DSFXFLANGER_FREQUENCY_MIN       0.0f+#define DSFXFLANGER_FREQUENCY_MAX       10.0f+#define DSFXFLANGER_DEPTH_MIN           0.0f+#define DSFXFLANGER_DEPTH_MAX           100.0f+#define DSFXFLANGER_PHASE_MIN           0+#define DSFXFLANGER_PHASE_MAX           4+#define DSFXFLANGER_FEEDBACK_MIN        -99.0f+#define DSFXFLANGER_FEEDBACK_MAX        99.0f+#define DSFXFLANGER_DELAY_MIN           0.0f+#define DSFXFLANGER_DELAY_MAX           4.0f++#define DSFXFLANGER_PHASE_NEG_180       0+#define DSFXFLANGER_PHASE_NEG_90        1+#define DSFXFLANGER_PHASE_ZERO          2+#define DSFXFLANGER_PHASE_90            3+#define DSFXFLANGER_PHASE_180           4++#undef INTERFACE+#define INTERFACE IDirectSoundFXFlanger++DECLARE_INTERFACE_(IDirectSoundFXFlanger, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundFXFlanger methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSFXFlanger pcDsFxFlanger) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSFXFlanger pDsFxFlanger) PURE;+};++#define IDirectSoundFXFlanger_QueryInterface(p,a,b)         IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFXFlanger_AddRef(p)                     IUnknown_AddRef(p)+#define IDirectSoundFXFlanger_Release(p)                    IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXFlanger_SetAllParameters(p,a)         (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundFXFlanger_GetAllParameters(p,a)         (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXFlanger_SetAllParameters(p,a)         (p)->SetAllParameters(a)+#define IDirectSoundFXFlanger_GetAllParameters(p,a)         (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundFXEcho+//++DEFINE_GUID(IID_IDirectSoundFXEcho, 0x8bd28edf, 0x50db, 0x4e92, 0xa2, 0xbd, 0x44, 0x54, 0x88, 0xd1, 0xed, 0x42);++typedef struct _DSFXEcho+{+    FLOAT   fWetDryMix;+    FLOAT   fFeedback;+    FLOAT   fLeftDelay;+    FLOAT   fRightDelay;+    LONG    lPanDelay;+} DSFXEcho, *LPDSFXEcho;++typedef const DSFXEcho *LPCDSFXEcho;++#define DSFXECHO_WETDRYMIX_MIN      0.0f+#define DSFXECHO_WETDRYMIX_MAX      100.0f+#define DSFXECHO_FEEDBACK_MIN       0.0f+#define DSFXECHO_FEEDBACK_MAX       100.0f+#define DSFXECHO_LEFTDELAY_MIN      1.0f+#define DSFXECHO_LEFTDELAY_MAX      2000.0f+#define DSFXECHO_RIGHTDELAY_MIN     1.0f+#define DSFXECHO_RIGHTDELAY_MAX     2000.0f+#define DSFXECHO_PANDELAY_MIN       0+#define DSFXECHO_PANDELAY_MAX       1++#undef INTERFACE+#define INTERFACE IDirectSoundFXEcho++DECLARE_INTERFACE_(IDirectSoundFXEcho, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundFXEcho methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSFXEcho pcDsFxEcho) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSFXEcho pDsFxEcho) PURE;+};++#define IDirectSoundFXEcho_QueryInterface(p,a,b)            IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFXEcho_AddRef(p)                        IUnknown_AddRef(p)+#define IDirectSoundFXEcho_Release(p)                       IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXEcho_SetAllParameters(p,a)            (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundFXEcho_GetAllParameters(p,a)            (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXEcho_SetAllParameters(p,a)            (p)->SetAllParameters(a)+#define IDirectSoundFXEcho_GetAllParameters(p,a)            (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundFXDistortion+//++DEFINE_GUID(IID_IDirectSoundFXDistortion, 0x8ecf4326, 0x455f, 0x4d8b, 0xbd, 0xa9, 0x8d, 0x5d, 0x3e, 0x9e, 0x3e, 0x0b);++typedef struct _DSFXDistortion+{+    FLOAT   fGain;+    FLOAT   fEdge;+    FLOAT   fPostEQCenterFrequency;+    FLOAT   fPostEQBandwidth;+    FLOAT   fPreLowpassCutoff;+} DSFXDistortion, *LPDSFXDistortion;++typedef const DSFXDistortion *LPCDSFXDistortion;++#define DSFXDISTORTION_GAIN_MIN                     -60.0f+#define DSFXDISTORTION_GAIN_MAX                     0.0f+#define DSFXDISTORTION_EDGE_MIN                     0.0f+#define DSFXDISTORTION_EDGE_MAX                     100.0f+#define DSFXDISTORTION_POSTEQCENTERFREQUENCY_MIN    100.0f+#define DSFXDISTORTION_POSTEQCENTERFREQUENCY_MAX    8000.0f+#define DSFXDISTORTION_POSTEQBANDWIDTH_MIN          100.0f+#define DSFXDISTORTION_POSTEQBANDWIDTH_MAX          8000.0f+#define DSFXDISTORTION_PRELOWPASSCUTOFF_MIN         100.0f+#define DSFXDISTORTION_PRELOWPASSCUTOFF_MAX         8000.0f++#undef INTERFACE+#define INTERFACE IDirectSoundFXDistortion++DECLARE_INTERFACE_(IDirectSoundFXDistortion, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundFXDistortion methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSFXDistortion pcDsFxDistortion) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSFXDistortion pDsFxDistortion) PURE;+};++#define IDirectSoundFXDistortion_QueryInterface(p,a,b)      IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFXDistortion_AddRef(p)                  IUnknown_AddRef(p)+#define IDirectSoundFXDistortion_Release(p)                 IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXDistortion_SetAllParameters(p,a)      (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundFXDistortion_GetAllParameters(p,a)      (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXDistortion_SetAllParameters(p,a)      (p)->SetAllParameters(a)+#define IDirectSoundFXDistortion_GetAllParameters(p,a)      (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundFXCompressor+//++DEFINE_GUID(IID_IDirectSoundFXCompressor, 0x4bbd1154, 0x62f6, 0x4e2c, 0xa1, 0x5c, 0xd3, 0xb6, 0xc4, 0x17, 0xf7, 0xa0);++typedef struct _DSFXCompressor+{+    FLOAT   fGain;+    FLOAT   fAttack;+    FLOAT   fRelease;+    FLOAT   fThreshold;+    FLOAT   fRatio;+    FLOAT   fPredelay;+} DSFXCompressor, *LPDSFXCompressor;++typedef const DSFXCompressor *LPCDSFXCompressor;++#define DSFXCOMPRESSOR_GAIN_MIN             -60.0f+#define DSFXCOMPRESSOR_GAIN_MAX             60.0f+#define DSFXCOMPRESSOR_ATTACK_MIN           0.01f+#define DSFXCOMPRESSOR_ATTACK_MAX           500.0f+#define DSFXCOMPRESSOR_RELEASE_MIN          50.0f+#define DSFXCOMPRESSOR_RELEASE_MAX          3000.0f+#define DSFXCOMPRESSOR_THRESHOLD_MIN        -60.0f+#define DSFXCOMPRESSOR_THRESHOLD_MAX        0.0f+#define DSFXCOMPRESSOR_RATIO_MIN            1.0f+#define DSFXCOMPRESSOR_RATIO_MAX            100.0f+#define DSFXCOMPRESSOR_PREDELAY_MIN         0.0f+#define DSFXCOMPRESSOR_PREDELAY_MAX         4.0f++#undef INTERFACE+#define INTERFACE IDirectSoundFXCompressor++DECLARE_INTERFACE_(IDirectSoundFXCompressor, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundFXCompressor methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSFXCompressor pcDsFxCompressor) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSFXCompressor pDsFxCompressor) PURE;+};++#define IDirectSoundFXCompressor_QueryInterface(p,a,b)      IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFXCompressor_AddRef(p)                  IUnknown_AddRef(p)+#define IDirectSoundFXCompressor_Release(p)                 IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXCompressor_SetAllParameters(p,a)      (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundFXCompressor_GetAllParameters(p,a)      (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXCompressor_SetAllParameters(p,a)      (p)->SetAllParameters(a)+#define IDirectSoundFXCompressor_GetAllParameters(p,a)      (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundFXParamEq+//++DEFINE_GUID(IID_IDirectSoundFXParamEq, 0xc03ca9fe, 0xfe90, 0x4204, 0x80, 0x78, 0x82, 0x33, 0x4c, 0xd1, 0x77, 0xda);++typedef struct _DSFXParamEq+{+    FLOAT   fCenter;+    FLOAT   fBandwidth;+    FLOAT   fGain;+} DSFXParamEq, *LPDSFXParamEq;++typedef const DSFXParamEq *LPCDSFXParamEq;++#define DSFXPARAMEQ_CENTER_MIN      80.0f+#define DSFXPARAMEQ_CENTER_MAX      16000.0f+#define DSFXPARAMEQ_BANDWIDTH_MIN   1.0f+#define DSFXPARAMEQ_BANDWIDTH_MAX   36.0f+#define DSFXPARAMEQ_GAIN_MIN        -15.0f+#define DSFXPARAMEQ_GAIN_MAX        15.0f++#undef INTERFACE+#define INTERFACE IDirectSoundFXParamEq++DECLARE_INTERFACE_(IDirectSoundFXParamEq, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundFXParamEq methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSFXParamEq pcDsFxParamEq) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSFXParamEq pDsFxParamEq) PURE;+};++#define IDirectSoundFXParamEq_QueryInterface(p,a,b)      IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFXParamEq_AddRef(p)                  IUnknown_AddRef(p)+#define IDirectSoundFXParamEq_Release(p)                 IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXParamEq_SetAllParameters(p,a)      (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundFXParamEq_GetAllParameters(p,a)      (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXParamEq_SetAllParameters(p,a)      (p)->SetAllParameters(a)+#define IDirectSoundFXParamEq_GetAllParameters(p,a)      (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundFXI3DL2Reverb+//++DEFINE_GUID(IID_IDirectSoundFXI3DL2Reverb, 0x4b166a6a, 0x0d66, 0x43f3, 0x80, 0xe3, 0xee, 0x62, 0x80, 0xde, 0xe1, 0xa4);++typedef struct _DSFXI3DL2Reverb+{+    LONG    lRoom;                  // [-10000, 0]      default: -1000 mB+    LONG    lRoomHF;                // [-10000, 0]      default: 0 mB+    FLOAT   flRoomRolloffFactor;    // [0.0, 10.0]      default: 0.0+    FLOAT   flDecayTime;            // [0.1, 20.0]      default: 1.49s+    FLOAT   flDecayHFRatio;         // [0.1, 2.0]       default: 0.83+    LONG    lReflections;           // [-10000, 1000]   default: -2602 mB+    FLOAT   flReflectionsDelay;     // [0.0, 0.3]       default: 0.007 s+    LONG    lReverb;                // [-10000, 2000]   default: 200 mB+    FLOAT   flReverbDelay;          // [0.0, 0.1]       default: 0.011 s+    FLOAT   flDiffusion;            // [0.0, 100.0]     default: 100.0 %+    FLOAT   flDensity;              // [0.0, 100.0]     default: 100.0 %+    FLOAT   flHFReference;          // [20.0, 20000.0]  default: 5000.0 Hz+} DSFXI3DL2Reverb, *LPDSFXI3DL2Reverb;++typedef const DSFXI3DL2Reverb *LPCDSFXI3DL2Reverb;++#define DSFX_I3DL2REVERB_ROOM_MIN                   (-10000)+#define DSFX_I3DL2REVERB_ROOM_MAX                   0+#define DSFX_I3DL2REVERB_ROOM_DEFAULT               (-1000)++#define DSFX_I3DL2REVERB_ROOMHF_MIN                 (-10000)+#define DSFX_I3DL2REVERB_ROOMHF_MAX                 0+#define DSFX_I3DL2REVERB_ROOMHF_DEFAULT             (-100)++#define DSFX_I3DL2REVERB_ROOMROLLOFFFACTOR_MIN      0.0f+#define DSFX_I3DL2REVERB_ROOMROLLOFFFACTOR_MAX      10.0f+#define DSFX_I3DL2REVERB_ROOMROLLOFFFACTOR_DEFAULT  0.0f++#define DSFX_I3DL2REVERB_DECAYTIME_MIN              0.1f+#define DSFX_I3DL2REVERB_DECAYTIME_MAX              20.0f+#define DSFX_I3DL2REVERB_DECAYTIME_DEFAULT          1.49f++#define DSFX_I3DL2REVERB_DECAYHFRATIO_MIN           0.1f+#define DSFX_I3DL2REVERB_DECAYHFRATIO_MAX           2.0f+#define DSFX_I3DL2REVERB_DECAYHFRATIO_DEFAULT       0.83f++#define DSFX_I3DL2REVERB_REFLECTIONS_MIN            (-10000)+#define DSFX_I3DL2REVERB_REFLECTIONS_MAX            1000+#define DSFX_I3DL2REVERB_REFLECTIONS_DEFAULT        (-2602)++#define DSFX_I3DL2REVERB_REFLECTIONSDELAY_MIN       0.0f+#define DSFX_I3DL2REVERB_REFLECTIONSDELAY_MAX       0.3f+#define DSFX_I3DL2REVERB_REFLECTIONSDELAY_DEFAULT   0.007f++#define DSFX_I3DL2REVERB_REVERB_MIN                 (-10000)+#define DSFX_I3DL2REVERB_REVERB_MAX                 2000+#define DSFX_I3DL2REVERB_REVERB_DEFAULT             (200)++#define DSFX_I3DL2REVERB_REVERBDELAY_MIN            0.0f+#define DSFX_I3DL2REVERB_REVERBDELAY_MAX            0.1f+#define DSFX_I3DL2REVERB_REVERBDELAY_DEFAULT        0.011f++#define DSFX_I3DL2REVERB_DIFFUSION_MIN              0.0f+#define DSFX_I3DL2REVERB_DIFFUSION_MAX              100.0f+#define DSFX_I3DL2REVERB_DIFFUSION_DEFAULT          100.0f++#define DSFX_I3DL2REVERB_DENSITY_MIN                0.0f+#define DSFX_I3DL2REVERB_DENSITY_MAX                100.0f+#define DSFX_I3DL2REVERB_DENSITY_DEFAULT            100.0f++#define DSFX_I3DL2REVERB_HFREFERENCE_MIN            20.0f+#define DSFX_I3DL2REVERB_HFREFERENCE_MAX            20000.0f+#define DSFX_I3DL2REVERB_HFREFERENCE_DEFAULT        5000.0f++#define DSFX_I3DL2REVERB_QUALITY_MIN                0+#define DSFX_I3DL2REVERB_QUALITY_MAX                3+#define DSFX_I3DL2REVERB_QUALITY_DEFAULT            2++#undef INTERFACE+#define INTERFACE IDirectSoundFXI3DL2Reverb++DECLARE_INTERFACE_(IDirectSoundFXI3DL2Reverb, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundFXI3DL2Reverb methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSFXI3DL2Reverb pcDsFxI3DL2Reverb) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSFXI3DL2Reverb pDsFxI3DL2Reverb) PURE;+    STDMETHOD(SetPreset)            (THIS_ DWORD dwPreset) PURE;+    STDMETHOD(GetPreset)            (THIS_ LPDWORD pdwPreset) PURE;+    STDMETHOD(SetQuality)           (THIS_ LONG lQuality) PURE;+    STDMETHOD(GetQuality)           (THIS_ LONG *plQuality) PURE;+};++#define IDirectSoundFXI3DL2Reverb_QueryInterface(p,a,b)     IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFXI3DL2Reverb_AddRef(p)                 IUnknown_AddRef(p)+#define IDirectSoundFXI3DL2Reverb_Release(p)                IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXI3DL2Reverb_SetAllParameters(p,a)     (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundFXI3DL2Reverb_GetAllParameters(p,a)     (p)->lpVtbl->GetAllParameters(p,a)+#define IDirectSoundFXI3DL2Reverb_SetPreset(p,a)            (p)->lpVtbl->SetPreset(p,a)+#define IDirectSoundFXI3DL2Reverb_GetPreset(p,a)            (p)->lpVtbl->GetPreset(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXI3DL2Reverb_SetAllParameters(p,a)     (p)->SetAllParameters(a)+#define IDirectSoundFXI3DL2Reverb_GetAllParameters(p,a)     (p)->GetAllParameters(a)+#define IDirectSoundFXI3DL2Reverb_SetPreset(p,a)            (p)->SetPreset(a)+#define IDirectSoundFXI3DL2Reverb_GetPreset(p,a)            (p)->GetPreset(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundFXWavesReverb+//++DEFINE_GUID(IID_IDirectSoundFXWavesReverb,0x46858c3a,0x0dc6,0x45e3,0xb7,0x60,0xd4,0xee,0xf1,0x6c,0xb3,0x25);++typedef struct _DSFXWavesReverb+{+    FLOAT   fInGain;                // [-96.0,0.0]            default: 0.0 dB+    FLOAT   fReverbMix;             // [-96.0,0.0]            default: 0.0 db+    FLOAT   fReverbTime;            // [0.001,3000.0]         default: 1000.0 ms+    FLOAT   fHighFreqRTRatio;       // [0.001,0.999]          default: 0.001+} DSFXWavesReverb, *LPDSFXWavesReverb;++typedef const DSFXWavesReverb *LPCDSFXWavesReverb;++#define DSFX_WAVESREVERB_INGAIN_MIN                 -96.0f+#define DSFX_WAVESREVERB_INGAIN_MAX                 0.0f+#define DSFX_WAVESREVERB_INGAIN_DEFAULT             0.0f+#define DSFX_WAVESREVERB_REVERBMIX_MIN              -96.0f+#define DSFX_WAVESREVERB_REVERBMIX_MAX              0.0f+#define DSFX_WAVESREVERB_REVERBMIX_DEFAULT          0.0f+#define DSFX_WAVESREVERB_REVERBTIME_MIN             0.001f+#define DSFX_WAVESREVERB_REVERBTIME_MAX             3000.0f+#define DSFX_WAVESREVERB_REVERBTIME_DEFAULT         1000.0f+#define DSFX_WAVESREVERB_HIGHFREQRTRATIO_MIN        0.001f+#define DSFX_WAVESREVERB_HIGHFREQRTRATIO_MAX        0.999f+#define DSFX_WAVESREVERB_HIGHFREQRTRATIO_DEFAULT    0.001f++#undef INTERFACE+#define INTERFACE IDirectSoundFXWavesReverb++DECLARE_INTERFACE_(IDirectSoundFXWavesReverb, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundFXWavesReverb methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSFXWavesReverb pcDsFxWavesReverb) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSFXWavesReverb pDsFxWavesReverb) PURE;+};++#define IDirectSoundFXWavesReverb_QueryInterface(p,a,b)     IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFXWavesReverb_AddRef(p)                 IUnknown_AddRef(p)+#define IDirectSoundFXWavesReverb_Release(p)                IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXWavesReverb_SetAllParameters(p,a)     (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundFXWavesReverb_GetAllParameters(p,a)     (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFXWavesReverb_SetAllParameters(p,a)     (p)->SetAllParameters(a)+#define IDirectSoundFXWavesReverb_GetAllParameters(p,a)     (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++//+// IDirectSoundCaptureFXAec+//++DEFINE_GUID(IID_IDirectSoundCaptureFXAec, 0xad74143d, 0x903d, 0x4ab7, 0x80, 0x66, 0x28, 0xd3, 0x63, 0x03, 0x6d, 0x65);++typedef struct _DSCFXAec+{+    BOOL    fEnable;+    BOOL    fNoiseFill;+    DWORD   dwMode;+} DSCFXAec, *LPDSCFXAec;++typedef const DSCFXAec *LPCDSCFXAec;++// These match the AEC_MODE_* constants in the DDK's ksmedia.h file+#define DSCFX_AEC_MODE_PASS_THROUGH                     0x0+#define DSCFX_AEC_MODE_HALF_DUPLEX                      0x1+#define DSCFX_AEC_MODE_FULL_DUPLEX                      0x2++// These match the AEC_STATUS_* constants in ksmedia.h+#define DSCFX_AEC_STATUS_HISTORY_UNINITIALIZED          0x0+#define DSCFX_AEC_STATUS_HISTORY_CONTINUOUSLY_CONVERGED 0x1+#define DSCFX_AEC_STATUS_HISTORY_PREVIOUSLY_DIVERGED    0x2+#define DSCFX_AEC_STATUS_CURRENTLY_CONVERGED            0x8++#undef INTERFACE+#define INTERFACE IDirectSoundCaptureFXAec++DECLARE_INTERFACE_(IDirectSoundCaptureFXAec, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundCaptureFXAec methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSCFXAec pDscFxAec) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSCFXAec pDscFxAec) PURE;+    STDMETHOD(GetStatus)            (THIS_ PDWORD pdwStatus) PURE;+    STDMETHOD(Reset)                (THIS) PURE;+};++#define IDirectSoundCaptureFXAec_QueryInterface(p,a,b)     IUnknown_QueryInterface(p,a,b)+#define IDirectSoundCaptureFXAec_AddRef(p)                 IUnknown_AddRef(p)+#define IDirectSoundCaptureFXAec_Release(p)                IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCaptureFXAec_SetAllParameters(p,a)     (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundCaptureFXAec_GetAllParameters(p,a)     (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCaptureFXAec_SetAllParameters(p,a)     (p)->SetAllParameters(a)+#define IDirectSoundCaptureFXAec_GetAllParameters(p,a)     (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)+++//+// IDirectSoundCaptureFXNoiseSuppress+//++DEFINE_GUID(IID_IDirectSoundCaptureFXNoiseSuppress, 0xed311e41, 0xfbae, 0x4175, 0x96, 0x25, 0xcd, 0x8, 0x54, 0xf6, 0x93, 0xca);++typedef struct _DSCFXNoiseSuppress+{+    BOOL    fEnable;+} DSCFXNoiseSuppress, *LPDSCFXNoiseSuppress;++typedef const DSCFXNoiseSuppress *LPCDSCFXNoiseSuppress;++#undef INTERFACE+#define INTERFACE IDirectSoundCaptureFXNoiseSuppress++DECLARE_INTERFACE_(IDirectSoundCaptureFXNoiseSuppress, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)       (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)        (THIS) PURE;+    STDMETHOD_(ULONG,Release)       (THIS) PURE;++    // IDirectSoundCaptureFXNoiseSuppress methods+    STDMETHOD(SetAllParameters)     (THIS_ LPCDSCFXNoiseSuppress pcDscFxNoiseSuppress) PURE;+    STDMETHOD(GetAllParameters)     (THIS_ LPDSCFXNoiseSuppress pDscFxNoiseSuppress) PURE;+    STDMETHOD(Reset)                (THIS) PURE;+};++#define IDirectSoundCaptureFXNoiseSuppress_QueryInterface(p,a,b)     IUnknown_QueryInterface(p,a,b)+#define IDirectSoundCaptureFXNoiseSuppress_AddRef(p)                 IUnknown_AddRef(p)+#define IDirectSoundCaptureFXNoiseSuppress_Release(p)                IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCaptureFXNoiseSuppress_SetAllParameters(p,a)     (p)->lpVtbl->SetAllParameters(p,a)+#define IDirectSoundCaptureFXNoiseSuppress_GetAllParameters(p,a)     (p)->lpVtbl->GetAllParameters(p,a)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundCaptureFXNoiseSuppress_SetAllParameters(p,a)     (p)->SetAllParameters(a)+#define IDirectSoundCaptureFXNoiseSuppress_GetAllParameters(p,a)     (p)->GetAllParameters(a)+#endif // !defined(__cplusplus) || defined(CINTERFACE)+++//+// IDirectSoundFullDuplex+//++#ifndef _IDirectSoundFullDuplex_+#define _IDirectSoundFullDuplex_++#ifdef __cplusplus+// 'struct' not 'class' per the way DECLARE_INTERFACE_ is defined+struct IDirectSoundFullDuplex;+#endif // __cplusplus++typedef struct IDirectSoundFullDuplex *LPDIRECTSOUNDFULLDUPLEX;++DEFINE_GUID(IID_IDirectSoundFullDuplex, 0xedcb4c7a, 0xdaab, 0x4216, 0xa4, 0x2e, 0x6c, 0x50, 0x59, 0x6d, 0xdc, 0x1d);++#undef INTERFACE+#define INTERFACE IDirectSoundFullDuplex++DECLARE_INTERFACE_(IDirectSoundFullDuplex, IUnknown)+{+    // IUnknown methods+    STDMETHOD(QueryInterface)   (THIS_ REFIID, LPVOID *) PURE;+    STDMETHOD_(ULONG,AddRef)    (THIS) PURE;+    STDMETHOD_(ULONG,Release)   (THIS) PURE;++    // IDirectSoundFullDuplex methods+    STDMETHOD(Initialize)     (THIS_ LPCGUID pCaptureGuid, LPCGUID pRenderGuid, LPCDSCBUFFERDESC lpDscBufferDesc, LPCDSBUFFERDESC lpDsBufferDesc, HWND hWnd, DWORD dwLevel, LPLPDIRECTSOUNDCAPTUREBUFFER8 lplpDirectSoundCaptureBuffer8, LPLPDIRECTSOUNDBUFFER8 lplpDirectSoundBuffer8) PURE;+};++#define IDirectSoundFullDuplex_QueryInterface(p,a,b)    IUnknown_QueryInterface(p,a,b)+#define IDirectSoundFullDuplex_AddRef(p)                IUnknown_AddRef(p)+#define IDirectSoundFullDuplex_Release(p)               IUnknown_Release(p)++#if !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFullDuplex_Initialize(p,a,b,c,d,e,f,g,h)     (p)->lpVtbl->Initialize(p,a,b,c,d,e,f,g,h)+#else // !defined(__cplusplus) || defined(CINTERFACE)+#define IDirectSoundFullDuplex_Initialize(p,a,b,c,d,e,f,g,h)     (p)->Initialize(a,b,c,d,e,f,g,h)+#endif // !defined(__cplusplus) || defined(CINTERFACE)++#endif // _IDirectSoundFullDuplex_++#endif // DIRECTSOUND_VERSION >= 0x0800++//+// Return Codes+//++// The function completed successfully+#define DS_OK                           S_OK++// The call succeeded, but we had to substitute the 3D algorithm+#define DS_NO_VIRTUALIZATION            MAKE_HRESULT(0, _FACDS, 10)++// The call failed because resources (such as a priority level)+// were already being used by another caller+#define DSERR_ALLOCATED                 MAKE_DSHRESULT(10)++// The control (vol, pan, etc.) requested by the caller is not available+#define DSERR_CONTROLUNAVAIL            MAKE_DSHRESULT(30)++// An invalid parameter was passed to the returning function+#define DSERR_INVALIDPARAM              E_INVALIDARG++// This call is not valid for the current state of this object+#define DSERR_INVALIDCALL               MAKE_DSHRESULT(50)++// An undetermined error occurred inside the DirectSound subsystem+#define DSERR_GENERIC                   E_FAIL++// The caller does not have the priority level required for the function to+// succeed+#define DSERR_PRIOLEVELNEEDED           MAKE_DSHRESULT(70)++// Not enough free memory is available to complete the operation+#define DSERR_OUTOFMEMORY               E_OUTOFMEMORY++// The specified WAVE format is not supported+#define DSERR_BADFORMAT                 MAKE_DSHRESULT(100)++// The function called is not supported at this time+#define DSERR_UNSUPPORTED               E_NOTIMPL++// No sound driver is available for use+#define DSERR_NODRIVER                  MAKE_DSHRESULT(120)+// This object is already initialized+#define DSERR_ALREADYINITIALIZED        MAKE_DSHRESULT(130)++// This object does not support aggregation+#define DSERR_NOAGGREGATION             CLASS_E_NOAGGREGATION++// The buffer memory has been lost, and must be restored+#define DSERR_BUFFERLOST                MAKE_DSHRESULT(150)++// Another app has a higher priority level, preventing this call from+// succeeding+#define DSERR_OTHERAPPHASPRIO           MAKE_DSHRESULT(160)++// This object has not been initialized+#define DSERR_UNINITIALIZED             MAKE_DSHRESULT(170)++// The requested COM interface is not available+#define DSERR_NOINTERFACE               E_NOINTERFACE++// Access is denied+#define DSERR_ACCESSDENIED              E_ACCESSDENIED++// Tried to create a DSBCAPS_CTRLFX buffer shorter than DSBSIZE_FX_MIN milliseconds+#define DSERR_BUFFERTOOSMALL            MAKE_DSHRESULT(180)++// Attempt to use DirectSound 8 functionality on an older DirectSound object+#define DSERR_DS8_REQUIRED              MAKE_DSHRESULT(190)++// A circular loop of send effects was detected+#define DSERR_SENDLOOP                  MAKE_DSHRESULT(200)++// The GUID specified in an audiopath file does not match a valid MIXIN buffer+#define DSERR_BADSENDBUFFERGUID         MAKE_DSHRESULT(210)++// The object requested was not found (numerically equal to DMUS_E_NOT_FOUND)+#define DSERR_OBJECTNOTFOUND            MAKE_DSHRESULT(4449)++// The effects requested could not be found on the system, or they were found+// but in the wrong order, or in the wrong hardware/software locations.+#define DSERR_FXUNAVAILABLE             MAKE_DSHRESULT(220)++//+// Flags+//++#define DSCAPS_PRIMARYMONO          0x00000001+#define DSCAPS_PRIMARYSTEREO        0x00000002+#define DSCAPS_PRIMARY8BIT          0x00000004+#define DSCAPS_PRIMARY16BIT         0x00000008+#define DSCAPS_CONTINUOUSRATE       0x00000010+#define DSCAPS_EMULDRIVER           0x00000020+#define DSCAPS_CERTIFIED            0x00000040+#define DSCAPS_SECONDARYMONO        0x00000100+#define DSCAPS_SECONDARYSTEREO      0x00000200+#define DSCAPS_SECONDARY8BIT        0x00000400+#define DSCAPS_SECONDARY16BIT       0x00000800++#define DSSCL_NORMAL                0x00000001+#define DSSCL_PRIORITY              0x00000002+#define DSSCL_EXCLUSIVE             0x00000003+#define DSSCL_WRITEPRIMARY          0x00000004++#define DSSPEAKER_DIRECTOUT         0x00000000+#define DSSPEAKER_HEADPHONE         0x00000001+#define DSSPEAKER_MONO              0x00000002+#define DSSPEAKER_QUAD              0x00000003+#define DSSPEAKER_STEREO            0x00000004+#define DSSPEAKER_SURROUND          0x00000005+#define DSSPEAKER_5POINT1           0x00000006  // obsolete 5.1 setting+#define DSSPEAKER_7POINT1           0x00000007  // obsolete 7.1 setting+#define DSSPEAKER_7POINT1_SURROUND  0x00000008  // correct 7.1 Home Theater setting+#define DSSPEAKER_7POINT1_WIDE      DSSPEAKER_7POINT1+#if (DIRECTSOUND_VERSION >= 0x1000)+    #define DSSPEAKER_5POINT1_SURROUND  0x00000009  // correct 5.1 setting+    #define DSSPEAKER_5POINT1_BACK      DSSPEAKER_5POINT1+#endif++#define DSSPEAKER_GEOMETRY_MIN      0x00000005  //   5 degrees+#define DSSPEAKER_GEOMETRY_NARROW   0x0000000A  //  10 degrees+#define DSSPEAKER_GEOMETRY_WIDE     0x00000014  //  20 degrees+#define DSSPEAKER_GEOMETRY_MAX      0x000000B4  // 180 degrees++#define DSSPEAKER_COMBINED(c, g)    ((DWORD)(((BYTE)(c)) | ((DWORD)((BYTE)(g))) << 16))+#define DSSPEAKER_CONFIG(a)         ((BYTE)(a))+#define DSSPEAKER_GEOMETRY(a)       ((BYTE)(((DWORD)(a) >> 16) & 0x00FF))++#define DSBCAPS_PRIMARYBUFFER       0x00000001+#define DSBCAPS_STATIC              0x00000002+#define DSBCAPS_LOCHARDWARE         0x00000004+#define DSBCAPS_LOCSOFTWARE         0x00000008+#define DSBCAPS_CTRL3D              0x00000010+#define DSBCAPS_CTRLFREQUENCY       0x00000020+#define DSBCAPS_CTRLPAN             0x00000040+#define DSBCAPS_CTRLVOLUME          0x00000080+#define DSBCAPS_CTRLPOSITIONNOTIFY  0x00000100+#define DSBCAPS_CTRLFX              0x00000200+#define DSBCAPS_STICKYFOCUS         0x00004000+#define DSBCAPS_GLOBALFOCUS         0x00008000+#define DSBCAPS_GETCURRENTPOSITION2 0x00010000+#define DSBCAPS_MUTE3DATMAXDISTANCE 0x00020000+#define DSBCAPS_LOCDEFER            0x00040000+#if (DIRECTSOUND_VERSION >= 0x1000)+    // Force GetCurrentPosition() to return a buffer's true play position;+    // unmodified by aids to enhance backward compatibility.+    #define DSBCAPS_TRUEPLAYPOSITION    0x00080000+#endif++#define DSBPLAY_LOOPING             0x00000001+#define DSBPLAY_LOCHARDWARE         0x00000002+#define DSBPLAY_LOCSOFTWARE         0x00000004+#define DSBPLAY_TERMINATEBY_TIME    0x00000008+#define DSBPLAY_TERMINATEBY_DISTANCE    0x000000010+#define DSBPLAY_TERMINATEBY_PRIORITY    0x000000020++#define DSBSTATUS_PLAYING           0x00000001+#define DSBSTATUS_BUFFERLOST        0x00000002+#define DSBSTATUS_LOOPING           0x00000004+#define DSBSTATUS_LOCHARDWARE       0x00000008+#define DSBSTATUS_LOCSOFTWARE       0x00000010+#define DSBSTATUS_TERMINATED        0x00000020++#define DSBLOCK_FROMWRITECURSOR     0x00000001+#define DSBLOCK_ENTIREBUFFER        0x00000002++#define DSBFREQUENCY_ORIGINAL       0+#define DSBFREQUENCY_MIN            100+#if DIRECTSOUND_VERSION >= 0x0900+#define DSBFREQUENCY_MAX            200000+#else+#define DSBFREQUENCY_MAX            100000+#endif++#define DSBPAN_LEFT                 -10000+#define DSBPAN_CENTER               0+#define DSBPAN_RIGHT                10000++#define DSBVOLUME_MIN               -10000+#define DSBVOLUME_MAX               0++#define DSBSIZE_MIN                 4+#define DSBSIZE_MAX                 0x0FFFFFFF+#define DSBSIZE_FX_MIN              150  // NOTE: Milliseconds, not bytes++#define DSBNOTIFICATIONS_MAX        100000UL++#define DS3DMODE_NORMAL             0x00000000+#define DS3DMODE_HEADRELATIVE       0x00000001+#define DS3DMODE_DISABLE            0x00000002++#define DS3D_IMMEDIATE              0x00000000+#define DS3D_DEFERRED               0x00000001++#define DS3D_MINDISTANCEFACTOR      FLT_MIN+#define DS3D_MAXDISTANCEFACTOR      FLT_MAX+#define DS3D_DEFAULTDISTANCEFACTOR  1.0f++#define DS3D_MINROLLOFFFACTOR       0.0f+#define DS3D_MAXROLLOFFFACTOR       10.0f+#define DS3D_DEFAULTROLLOFFFACTOR   1.0f++#define DS3D_MINDOPPLERFACTOR       0.0f+#define DS3D_MAXDOPPLERFACTOR       10.0f+#define DS3D_DEFAULTDOPPLERFACTOR   1.0f++#define DS3D_DEFAULTMINDISTANCE     1.0f+#define DS3D_DEFAULTMAXDISTANCE     1000000000.0f++#define DS3D_MINCONEANGLE           0+#define DS3D_MAXCONEANGLE           360+#define DS3D_DEFAULTCONEANGLE       360++#define DS3D_DEFAULTCONEOUTSIDEVOLUME DSBVOLUME_MAX++// IDirectSoundCapture attributes++#define DSCCAPS_EMULDRIVER          DSCAPS_EMULDRIVER+#define DSCCAPS_CERTIFIED           DSCAPS_CERTIFIED+#define DSCCAPS_MULTIPLECAPTURE     0x00000001++// IDirectSoundCaptureBuffer attributes++#define DSCBCAPS_WAVEMAPPED         0x80000000++#if DIRECTSOUND_VERSION >= 0x0800+#define DSCBCAPS_CTRLFX             0x00000200+#endif+++#define DSCBLOCK_ENTIREBUFFER       0x00000001++#define DSCBSTATUS_CAPTURING        0x00000001+#define DSCBSTATUS_LOOPING          0x00000002++#define DSCBSTART_LOOPING           0x00000001++#define DSBPN_OFFSETSTOP            0xFFFFFFFF++#define DS_CERTIFIED                0x00000000+#define DS_UNCERTIFIED              0x00000001+++//+// Flags for the I3DL2 effects+//++//+// I3DL2 Material Presets+//++enum+{+    DSFX_I3DL2_MATERIAL_PRESET_SINGLEWINDOW,+    DSFX_I3DL2_MATERIAL_PRESET_DOUBLEWINDOW,+    DSFX_I3DL2_MATERIAL_PRESET_THINDOOR,+    DSFX_I3DL2_MATERIAL_PRESET_THICKDOOR,+    DSFX_I3DL2_MATERIAL_PRESET_WOODWALL,+    DSFX_I3DL2_MATERIAL_PRESET_BRICKWALL,+    DSFX_I3DL2_MATERIAL_PRESET_STONEWALL,+    DSFX_I3DL2_MATERIAL_PRESET_CURTAIN+};++#define I3DL2_MATERIAL_PRESET_SINGLEWINDOW    -2800,0.71f+#define I3DL2_MATERIAL_PRESET_DOUBLEWINDOW    -5000,0.40f+#define I3DL2_MATERIAL_PRESET_THINDOOR        -1800,0.66f+#define I3DL2_MATERIAL_PRESET_THICKDOOR       -4400,0.64f+#define I3DL2_MATERIAL_PRESET_WOODWALL        -4000,0.50f+#define I3DL2_MATERIAL_PRESET_BRICKWALL       -5000,0.60f+#define I3DL2_MATERIAL_PRESET_STONEWALL       -6000,0.68f+#define I3DL2_MATERIAL_PRESET_CURTAIN         -1200,0.15f++enum+{+    DSFX_I3DL2_ENVIRONMENT_PRESET_DEFAULT,+    DSFX_I3DL2_ENVIRONMENT_PRESET_GENERIC,+    DSFX_I3DL2_ENVIRONMENT_PRESET_PADDEDCELL,+    DSFX_I3DL2_ENVIRONMENT_PRESET_ROOM,+    DSFX_I3DL2_ENVIRONMENT_PRESET_BATHROOM,+    DSFX_I3DL2_ENVIRONMENT_PRESET_LIVINGROOM,+    DSFX_I3DL2_ENVIRONMENT_PRESET_STONEROOM,+    DSFX_I3DL2_ENVIRONMENT_PRESET_AUDITORIUM,+    DSFX_I3DL2_ENVIRONMENT_PRESET_CONCERTHALL,+    DSFX_I3DL2_ENVIRONMENT_PRESET_CAVE,+    DSFX_I3DL2_ENVIRONMENT_PRESET_ARENA,+    DSFX_I3DL2_ENVIRONMENT_PRESET_HANGAR,+    DSFX_I3DL2_ENVIRONMENT_PRESET_CARPETEDHALLWAY,+    DSFX_I3DL2_ENVIRONMENT_PRESET_HALLWAY,+    DSFX_I3DL2_ENVIRONMENT_PRESET_STONECORRIDOR,+    DSFX_I3DL2_ENVIRONMENT_PRESET_ALLEY,+    DSFX_I3DL2_ENVIRONMENT_PRESET_FOREST,+    DSFX_I3DL2_ENVIRONMENT_PRESET_CITY,+    DSFX_I3DL2_ENVIRONMENT_PRESET_MOUNTAINS,+    DSFX_I3DL2_ENVIRONMENT_PRESET_QUARRY,+    DSFX_I3DL2_ENVIRONMENT_PRESET_PLAIN,+    DSFX_I3DL2_ENVIRONMENT_PRESET_PARKINGLOT,+    DSFX_I3DL2_ENVIRONMENT_PRESET_SEWERPIPE,+    DSFX_I3DL2_ENVIRONMENT_PRESET_UNDERWATER,+    DSFX_I3DL2_ENVIRONMENT_PRESET_SMALLROOM,+    DSFX_I3DL2_ENVIRONMENT_PRESET_MEDIUMROOM,+    DSFX_I3DL2_ENVIRONMENT_PRESET_LARGEROOM,+    DSFX_I3DL2_ENVIRONMENT_PRESET_MEDIUMHALL,+    DSFX_I3DL2_ENVIRONMENT_PRESET_LARGEHALL,+    DSFX_I3DL2_ENVIRONMENT_PRESET_PLATE+};++//+// I3DL2 Reverberation Presets Values+//++#define I3DL2_ENVIRONMENT_PRESET_DEFAULT         -1000, -100, 0.0f, 1.49f, 0.83f, -2602, 0.007f,   200, 0.011f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_GENERIC         -1000, -100, 0.0f, 1.49f, 0.83f, -2602, 0.007f,   200, 0.011f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_PADDEDCELL      -1000,-6000, 0.0f, 0.17f, 0.10f, -1204, 0.001f,   207, 0.002f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_ROOM            -1000, -454, 0.0f, 0.40f, 0.83f, -1646, 0.002f,    53, 0.003f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_BATHROOM        -1000,-1200, 0.0f, 1.49f, 0.54f,  -370, 0.007f,  1030, 0.011f, 100.0f,  60.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_LIVINGROOM      -1000,-6000, 0.0f, 0.50f, 0.10f, -1376, 0.003f, -1104, 0.004f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_STONEROOM       -1000, -300, 0.0f, 2.31f, 0.64f,  -711, 0.012f,    83, 0.017f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_AUDITORIUM      -1000, -476, 0.0f, 4.32f, 0.59f,  -789, 0.020f,  -289, 0.030f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_CONCERTHALL     -1000, -500, 0.0f, 3.92f, 0.70f, -1230, 0.020f,    -2, 0.029f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_CAVE            -1000,    0, 0.0f, 2.91f, 1.30f,  -602, 0.015f,  -302, 0.022f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_ARENA           -1000, -698, 0.0f, 7.24f, 0.33f, -1166, 0.020f,    16, 0.030f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_HANGAR          -1000,-1000, 0.0f,10.05f, 0.23f,  -602, 0.020f,   198, 0.030f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_CARPETEDHALLWAY -1000,-4000, 0.0f, 0.30f, 0.10f, -1831, 0.002f, -1630, 0.030f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_HALLWAY         -1000, -300, 0.0f, 1.49f, 0.59f, -1219, 0.007f,   441, 0.011f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_STONECORRIDOR   -1000, -237, 0.0f, 2.70f, 0.79f, -1214, 0.013f,   395, 0.020f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_ALLEY           -1000, -270, 0.0f, 1.49f, 0.86f, -1204, 0.007f,    -4, 0.011f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_FOREST          -1000,-3300, 0.0f, 1.49f, 0.54f, -2560, 0.162f,  -613, 0.088f,  79.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_CITY            -1000, -800, 0.0f, 1.49f, 0.67f, -2273, 0.007f, -2217, 0.011f,  50.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_MOUNTAINS       -1000,-2500, 0.0f, 1.49f, 0.21f, -2780, 0.300f, -2014, 0.100f,  27.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_QUARRY          -1000,-1000, 0.0f, 1.49f, 0.83f,-10000, 0.061f,   500, 0.025f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_PLAIN           -1000,-2000, 0.0f, 1.49f, 0.50f, -2466, 0.179f, -2514, 0.100f,  21.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_PARKINGLOT      -1000,    0, 0.0f, 1.65f, 1.50f, -1363, 0.008f, -1153, 0.012f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_SEWERPIPE       -1000,-1000, 0.0f, 2.81f, 0.14f,   429, 0.014f,   648, 0.021f,  80.0f,  60.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_UNDERWATER      -1000,-4000, 0.0f, 1.49f, 0.10f,  -449, 0.007f,  1700, 0.011f, 100.0f, 100.0f, 5000.0f++//+// Examples simulating 'musical' reverb presets+//+// Name       Decay time   Description+// Small Room    1.1s      A small size room with a length of 5m or so.+// Medium Room   1.3s      A medium size room with a length of 10m or so.+// Large Room    1.5s      A large size room suitable for live performances.+// Medium Hall   1.8s      A medium size concert hall.+// Large Hall    1.8s      A large size concert hall suitable for a full orchestra.+// Plate         1.3s      A plate reverb simulation.+//++#define I3DL2_ENVIRONMENT_PRESET_SMALLROOM       -1000, -600, 0.0f, 1.10f, 0.83f,  -400, 0.005f,   500, 0.010f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_MEDIUMROOM      -1000, -600, 0.0f, 1.30f, 0.83f, -1000, 0.010f,  -200, 0.020f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_LARGEROOM       -1000, -600, 0.0f, 1.50f, 0.83f, -1600, 0.020f, -1000, 0.040f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_MEDIUMHALL      -1000, -600, 0.0f, 1.80f, 0.70f, -1300, 0.015f,  -800, 0.030f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_LARGEHALL       -1000, -600, 0.0f, 1.80f, 0.70f, -2000, 0.030f, -1400, 0.060f, 100.0f, 100.0f, 5000.0f+#define I3DL2_ENVIRONMENT_PRESET_PLATE           -1000, -200, 0.0f, 1.30f, 0.90f,     0, 0.002f,     0, 0.010f, 100.0f,  75.0f, 5000.0f++//+// DirectSound3D Algorithms+//++// Default DirectSound3D algorithm {00000000-0000-0000-0000-000000000000}+#define DS3DALG_DEFAULT GUID_NULL++// No virtualization (Pan3D) {C241333F-1C1B-11d2-94F5-00C04FC28ACA}+DEFINE_GUID(DS3DALG_NO_VIRTUALIZATION, 0xc241333f, 0x1c1b, 0x11d2, 0x94, 0xf5, 0x0, 0xc0, 0x4f, 0xc2, 0x8a, 0xca);++// High-quality HRTF algorithm {C2413340-1C1B-11d2-94F5-00C04FC28ACA}+DEFINE_GUID(DS3DALG_HRTF_FULL, 0xc2413340, 0x1c1b, 0x11d2, 0x94, 0xf5, 0x0, 0xc0, 0x4f, 0xc2, 0x8a, 0xca);++// Lower-quality HRTF algorithm {C2413342-1C1B-11d2-94F5-00C04FC28ACA}+DEFINE_GUID(DS3DALG_HRTF_LIGHT, 0xc2413342, 0x1c1b, 0x11d2, 0x94, 0xf5, 0x0, 0xc0, 0x4f, 0xc2, 0x8a, 0xca);+++#if DIRECTSOUND_VERSION >= 0x0800++//+// DirectSound Internal Effect Algorithms+//+++// Gargle {DAFD8210-5711-4B91-9FE3-F75B7AE279BF}+DEFINE_GUID(GUID_DSFX_STANDARD_GARGLE, 0xdafd8210, 0x5711, 0x4b91, 0x9f, 0xe3, 0xf7, 0x5b, 0x7a, 0xe2, 0x79, 0xbf);++// Chorus {EFE6629C-81F7-4281-BD91-C9D604A95AF6}+DEFINE_GUID(GUID_DSFX_STANDARD_CHORUS, 0xefe6629c, 0x81f7, 0x4281, 0xbd, 0x91, 0xc9, 0xd6, 0x04, 0xa9, 0x5a, 0xf6);++// Flanger {EFCA3D92-DFD8-4672-A603-7420894BAD98}+DEFINE_GUID(GUID_DSFX_STANDARD_FLANGER, 0xefca3d92, 0xdfd8, 0x4672, 0xa6, 0x03, 0x74, 0x20, 0x89, 0x4b, 0xad, 0x98);++// Echo/Delay {EF3E932C-D40B-4F51-8CCF-3F98F1B29D5D}+DEFINE_GUID(GUID_DSFX_STANDARD_ECHO, 0xef3e932c, 0xd40b, 0x4f51, 0x8c, 0xcf, 0x3f, 0x98, 0xf1, 0xb2, 0x9d, 0x5d);++// Distortion {EF114C90-CD1D-484E-96E5-09CFAF912A21}+DEFINE_GUID(GUID_DSFX_STANDARD_DISTORTION, 0xef114c90, 0xcd1d, 0x484e, 0x96, 0xe5, 0x09, 0xcf, 0xaf, 0x91, 0x2a, 0x21);++// Compressor/Limiter {EF011F79-4000-406D-87AF-BFFB3FC39D57}+DEFINE_GUID(GUID_DSFX_STANDARD_COMPRESSOR, 0xef011f79, 0x4000, 0x406d, 0x87, 0xaf, 0xbf, 0xfb, 0x3f, 0xc3, 0x9d, 0x57);++// Parametric Equalization {120CED89-3BF4-4173-A132-3CB406CF3231}+DEFINE_GUID(GUID_DSFX_STANDARD_PARAMEQ, 0x120ced89, 0x3bf4, 0x4173, 0xa1, 0x32, 0x3c, 0xb4, 0x06, 0xcf, 0x32, 0x31);++// I3DL2 Environmental Reverberation: Reverb (Listener) Effect {EF985E71-D5C7-42D4-BA4D-2D073E2E96F4}+DEFINE_GUID(GUID_DSFX_STANDARD_I3DL2REVERB, 0xef985e71, 0xd5c7, 0x42d4, 0xba, 0x4d, 0x2d, 0x07, 0x3e, 0x2e, 0x96, 0xf4);++// Waves Reverberation {87FC0268-9A55-4360-95AA-004A1D9DE26C}+DEFINE_GUID(GUID_DSFX_WAVES_REVERB, 0x87fc0268, 0x9a55, 0x4360, 0x95, 0xaa, 0x00, 0x4a, 0x1d, 0x9d, 0xe2, 0x6c);++//+// DirectSound Capture Effect Algorithms+//+++// Acoustic Echo Canceller {BF963D80-C559-11D0-8A2B-00A0C9255AC1}+// Matches KSNODETYPE_ACOUSTIC_ECHO_CANCEL in ksmedia.h+DEFINE_GUID(GUID_DSCFX_CLASS_AEC, 0xBF963D80L, 0xC559, 0x11D0, 0x8A, 0x2B, 0x00, 0xA0, 0xC9, 0x25, 0x5A, 0xC1);++// Microsoft AEC {CDEBB919-379A-488a-8765-F53CFD36DE40}+DEFINE_GUID(GUID_DSCFX_MS_AEC, 0xcdebb919, 0x379a, 0x488a, 0x87, 0x65, 0xf5, 0x3c, 0xfd, 0x36, 0xde, 0x40);++// System AEC {1C22C56D-9879-4f5b-A389-27996DDC2810}+DEFINE_GUID(GUID_DSCFX_SYSTEM_AEC, 0x1c22c56d, 0x9879, 0x4f5b, 0xa3, 0x89, 0x27, 0x99, 0x6d, 0xdc, 0x28, 0x10);++// Noise Supression {E07F903F-62FD-4e60-8CDD-DEA7236665B5}+// Matches KSNODETYPE_NOISE_SUPPRESS in post Windows ME DDK's ksmedia.h+DEFINE_GUID(GUID_DSCFX_CLASS_NS, 0xe07f903f, 0x62fd, 0x4e60, 0x8c, 0xdd, 0xde, 0xa7, 0x23, 0x66, 0x65, 0xb5);++// Microsoft Noise Suppresion {11C5C73B-66E9-4ba1-A0BA-E814C6EED92D}+DEFINE_GUID(GUID_DSCFX_MS_NS, 0x11c5c73b, 0x66e9, 0x4ba1, 0xa0, 0xba, 0xe8, 0x14, 0xc6, 0xee, 0xd9, 0x2d);++// System Noise Suppresion {5AB0882E-7274-4516-877D-4EEE99BA4FD0}+DEFINE_GUID(GUID_DSCFX_SYSTEM_NS, 0x5ab0882e, 0x7274, 0x4516, 0x87, 0x7d, 0x4e, 0xee, 0x99, 0xba, 0x4f, 0xd0);++#endif // DIRECTSOUND_VERSION >= 0x0800++#endif // __DSOUND_INCLUDED__++++#ifdef __cplusplus+};+#endif // __cplusplus+
+ cbits/include/ginclude.h view
@@ -0,0 +1,38 @@+#ifndef __gInclude__
+#define __gInclude__
+
+#if SGI 
+	#undef BEOS 
+	#undef MAC 
+	#undef WINDOWS
+	//
+	#define ASIO_BIG_ENDIAN 1
+	#define ASIO_CPU_MIPS 1
+#elif defined WIN32
+	#undef BEOS 
+	#undef MAC 
+	#undef SGI
+	#define WINDOWS 1
+	#define ASIO_LITTLE_ENDIAN 1
+	#define ASIO_CPU_X86 1
+#elif BEOS
+	#undef MAC 
+	#undef SGI
+	#undef WINDOWS
+	#define ASIO_LITTLE_ENDIAN 1
+	#define ASIO_CPU_X86 1
+	//
+#else
+	#define MAC 1
+	#undef BEOS 
+	#undef WINDOWS
+	#undef SGI
+	#define ASIO_BIG_ENDIAN 1
+	#define ASIO_CPU_PPC 1
+#endif
+
+// always
+#define NATIVE_INT64 0
+#define IEEE754_64FLOAT 1
+
+#endif	// __gInclude__
+ cbits/include/iasiodrv.h view
@@ -0,0 +1,37 @@+#include "asiosys.h"
+#include "asio.h"
+
+/* Forward Declarations */ 
+
+#ifndef __ASIODRIVER_FWD_DEFINED__
+#define __ASIODRIVER_FWD_DEFINED__
+typedef interface IASIO IASIO;
+#endif 	/* __ASIODRIVER_FWD_DEFINED__ */
+
+interface IASIO : public IUnknown
+{
+
+	virtual ASIOBool init(void *sysHandle) = 0;
+	virtual void getDriverName(char *name) = 0;	
+	virtual long getDriverVersion() = 0;
+	virtual void getErrorMessage(char *string) = 0;	
+	virtual ASIOError start() = 0;
+	virtual ASIOError stop() = 0;
+	virtual ASIOError getChannels(long *numInputChannels, long *numOutputChannels) = 0;
+	virtual ASIOError getLatencies(long *inputLatency, long *outputLatency) = 0;
+	virtual ASIOError getBufferSize(long *minSize, long *maxSize,
+		long *preferredSize, long *granularity) = 0;
+	virtual ASIOError canSampleRate(ASIOSampleRate sampleRate) = 0;
+	virtual ASIOError getSampleRate(ASIOSampleRate *sampleRate) = 0;
+	virtual ASIOError setSampleRate(ASIOSampleRate sampleRate) = 0;
+	virtual ASIOError getClockSources(ASIOClockSource *clocks, long *numSources) = 0;
+	virtual ASIOError setClockSource(long reference) = 0;
+	virtual ASIOError getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp) = 0;
+	virtual ASIOError getChannelInfo(ASIOChannelInfo *info) = 0;
+	virtual ASIOError createBuffers(ASIOBufferInfo *bufferInfos, long numChannels,
+		long bufferSize, ASIOCallbacks *callbacks) = 0;
+	virtual ASIOError disposeBuffers() = 0;
+	virtual ASIOError controlPanel() = 0;
+	virtual ASIOError future(long selector,void *opt) = 0;
+	virtual ASIOError outputReady() = 0;
+};
+ cbits/include/iasiothiscallresolver.cpp view
@@ -0,0 +1,572 @@+/*+	IASIOThiscallResolver.cpp see the comments in iasiothiscallresolver.h for+    the top level description - this comment describes the technical details of+    the implementation.++    The latest version of this file is available from:+    http://www.audiomulch.com/~rossb/code/calliasio++    please email comments to Ross Bencina <rossb@audiomulch.com>++    BACKGROUND++    The IASIO interface declared in the Steinberg ASIO 2 SDK declares+    functions with no explicit calling convention. This causes MSVC++ to default+    to using the thiscall convention, which is a proprietary convention not+    implemented by some non-microsoft compilers - notably borland BCC,+    C++Builder, and gcc. MSVC++ is the defacto standard compiler used by+    Steinberg. As a result of this situation, the ASIO sdk will compile with+    any compiler, however attempting to execute the compiled code will cause a+    crash due to different default calling conventions on non-Microsoft+    compilers.++    IASIOThiscallResolver solves the problem by providing an adapter class that+    delegates to the IASIO interface using the correct calling convention+    (thiscall). Due to the lack of support for thiscall in the Borland and GCC+    compilers, the calls have been implemented in assembly language.++    A number of macros are defined for thiscall function calls with different+    numbers of parameters, with and without return values - it may be possible+    to modify the format of these macros to make them work with other inline+    assemblers.+++    THISCALL DEFINITION++    A number of definitions of the thiscall calling convention are floating+    around the internet. The following definition has been validated against+    output from the MSVC++ compiler:++    For non-vararg functions, thiscall works as follows: the object (this)+    pointer is passed in ECX. All arguments are passed on the stack in+    right to left order. The return value is placed in EAX. The callee+    clears the passed arguments from the stack.+++    FINDING FUNCTION POINTERS FROM AN IASIO POINTER++    The first field of a COM object is a pointer to its vtble. Thus a pointer+    to an object implementing the IASIO interface also points to a pointer to+    that object's vtbl. The vtble is a table of function pointers for all of+    the virtual functions exposed by the implemented interfaces.++    If we consider a variable declared as a pointer to IASO:++    IASIO *theAsioDriver++    theAsioDriver points to:++    object implementing IASIO+    {+        IASIOvtbl *vtbl+        other data+    }++    in other words, theAsioDriver points to a pointer to an IASIOvtbl++    vtbl points to a table of function pointers:++    IASIOvtbl ( interface IASIO : public IUnknown )+    {+    (IUnknown functions)+    0   virtual HRESULT STDMETHODCALLTYPE (*QueryInterface)(REFIID riid, void **ppv) = 0;+    4   virtual ULONG STDMETHODCALLTYPE (*AddRef)() = 0;+    8   virtual ULONG STDMETHODCALLTYPE (*Release)() = 0;      ++    (IASIO functions)+    12	virtual ASIOBool (*init)(void *sysHandle) = 0;+    16	virtual void (*getDriverName)(char *name) = 0;+    20	virtual long (*getDriverVersion)() = 0;+    24	virtual void (*getErrorMessage)(char *string) = 0;+    28	virtual ASIOError (*start)() = 0;+    32	virtual ASIOError (*stop)() = 0;+    36	virtual ASIOError (*getChannels)(long *numInputChannels, long *numOutputChannels) = 0;+    40	virtual ASIOError (*getLatencies)(long *inputLatency, long *outputLatency) = 0;+    44	virtual ASIOError (*getBufferSize)(long *minSize, long *maxSize,+            long *preferredSize, long *granularity) = 0;+    48	virtual ASIOError (*canSampleRate)(ASIOSampleRate sampleRate) = 0;+    52	virtual ASIOError (*getSampleRate)(ASIOSampleRate *sampleRate) = 0;+    56	virtual ASIOError (*setSampleRate)(ASIOSampleRate sampleRate) = 0;+    60	virtual ASIOError (*getClockSources)(ASIOClockSource *clocks, long *numSources) = 0;+    64	virtual ASIOError (*setClockSource)(long reference) = 0;+    68	virtual ASIOError (*getSamplePosition)(ASIOSamples *sPos, ASIOTimeStamp *tStamp) = 0;+    72	virtual ASIOError (*getChannelInfo)(ASIOChannelInfo *info) = 0;+    76	virtual ASIOError (*createBuffers)(ASIOBufferInfo *bufferInfos, long numChannels,+            long bufferSize, ASIOCallbacks *callbacks) = 0;+    80	virtual ASIOError (*disposeBuffers)() = 0;+    84	virtual ASIOError (*controlPanel)() = 0;+    88	virtual ASIOError (*future)(long selector,void *opt) = 0;+    92	virtual ASIOError (*outputReady)() = 0;+    };++    The numbers in the left column show the byte offset of each function ptr+    from the beginning of the vtbl. These numbers are used in the code below+    to select different functions.++    In order to find the address of a particular function, theAsioDriver+    must first be dereferenced to find the value of the vtbl pointer:++    mov     eax, theAsioDriver+    mov     edx, [theAsioDriver]  // edx now points to vtbl[0]++    Then an offset must be added to the vtbl pointer to select a+    particular function, for example vtbl+44 points to the slot containing+    a pointer to the getBufferSize function.++    Finally vtbl+x must be dereferenced to obtain the value of the function+    pointer stored in that address:++    call    [edx+44]    // call the function pointed to by+                        // the value in the getBufferSize field of the vtbl+++    SEE ALSO++    Martin Fay's OpenASIO DLL at http://www.martinfay.com solves the same+    problem by providing a new COM interface which wraps IASIO with an+    interface that uses portable calling conventions. OpenASIO must be compiled+    with MSVC, and requires that you ship the OpenASIO DLL with your+    application.++    +    ACKNOWLEDGEMENTS++    Ross Bencina: worked out the thiscall details above, wrote the original+    Borland asm macros, and a patch for asio.cpp (which is no longer needed).+    Thanks to Martin Fay for introducing me to the issues discussed here,+    and to Rene G. Ceballos for assisting with asm dumps from MSVC++.++    Antti Silvast: converted the original calliasio to work with gcc and NASM+    by implementing the asm code in a separate file.++	Fraser Adams: modified the original calliasio containing the Borland inline+    asm to add inline asm for gcc i.e. Intel syntax for Borland and AT&T syntax+    for gcc. This seems a neater approach for gcc than to have a separate .asm+    file and it means that we only need one version of the thiscall patch.++    Fraser Adams: rewrote the original calliasio patch in the form of the+    IASIOThiscallResolver class in order to avoid modifications to files from+    the Steinberg SDK, which may have had potential licence issues.++    Andrew Baldwin: contributed fixes for compatibility problems with more+    recent versions of the gcc assembler.+*/+++// We only need IASIOThiscallResolver at all if we are on Win32. For other+// platforms we simply bypass the IASIOThiscallResolver definition to allow us+// to be safely #include'd whatever the platform to keep client code portable+#if (defined(WIN32) || defined(_WIN32) || defined(__WIN32__)) && !defined(_WIN64)+++// If microsoft compiler we can call IASIO directly so IASIOThiscallResolver+// is not used.+#if !defined(_MSC_VER)+++#include <new>+#include <assert.h>++// We have a mechanism in iasiothiscallresolver.h to ensure that asio.h is+// #include'd before it in client code, we do NOT want to do this test here.+#define iasiothiscallresolver_sourcefile 1+#include "iasiothiscallresolver.h"+#undef iasiothiscallresolver_sourcefile++// iasiothiscallresolver.h redefines ASIOInit for clients, but we don't want+// this macro defined in this translation unit.+#undef ASIOInit+++// theAsioDriver is a global pointer to the current IASIO instance which the+// ASIO SDK uses to perform all actions on the IASIO interface. We substitute+// our own forwarding interface into this pointer.+extern IASIO* theAsioDriver;+++// The following macros define the inline assembler for BORLAND first then gcc++#if defined(__BCPLUSPLUS__) || defined(__BORLANDC__)          +++#define CALL_THISCALL_0( resultName, thisPtr, funcOffset )\+    void *this_ = (thisPtr);                                                \+    __asm {                                                                 \+        mov     ecx, this_            ;                                     \+        mov     eax, [ecx]            ;                                     \+        call    [eax+funcOffset]      ;                                     \+        mov     resultName, eax       ;                                     \+    }+++#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 )\+    void *this_ = (thisPtr);                                                \+    __asm {                                                                 \+        mov     eax, param1           ;                                     \+        push    eax                   ;                                     \+        mov     ecx, this_            ;                                     \+        mov     eax, [ecx]            ;                                     \+        call    [eax+funcOffset]      ;                                     \+    }+++#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 )\+    void *this_ = (thisPtr);                                                \+    __asm {                                                                 \+        mov     eax, param1           ;                                     \+        push    eax                   ;                                     \+        mov     ecx, this_            ;                                     \+        mov     eax, [ecx]            ;                                     \+        call    [eax+funcOffset]      ;                                     \+        mov     resultName, eax       ;                                     \+    }+++#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 )\+    void *this_ = (thisPtr);                                                \+    void *doubleParamPtr_ (&param1);                                        \+    __asm {                                                                 \+        mov     eax, doubleParamPtr_  ;                                     \+        push    [eax+4]               ;                                     \+        push    [eax]                 ;                                     \+        mov     ecx, this_            ;                                     \+        mov     eax, [ecx]            ;                                     \+        call    [eax+funcOffset]      ;                                     \+        mov     resultName, eax       ;                                     \+    }+++#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 )\+    void *this_ = (thisPtr);                                                \+    __asm {                                                                 \+        mov     eax, param2           ;                                     \+        push    eax                   ;                                     \+        mov     eax, param1           ;                                     \+        push    eax                   ;                                     \+        mov     ecx, this_            ;                                     \+        mov     eax, [ecx]            ;                                     \+        call    [eax+funcOffset]      ;                                     \+        mov     resultName, eax       ;                                     \+    }+++#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\+    void *this_ = (thisPtr);                                                \+    __asm {                                                                 \+        mov     eax, param4           ;                                     \+        push    eax                   ;                                     \+        mov     eax, param3           ;                                     \+        push    eax                   ;                                     \+        mov     eax, param2           ;                                     \+        push    eax                   ;                                     \+        mov     eax, param1           ;                                     \+        push    eax                   ;                                     \+        mov     ecx, this_            ;                                     \+        mov     eax, [ecx]            ;                                     \+        call    [eax+funcOffset]      ;                                     \+        mov     resultName, eax       ;                                     \+    }+++#elif defined(__GNUC__)+++#define CALL_THISCALL_0( resultName, thisPtr, funcOffset )                  \+    __asm__ __volatile__ ("movl (%1), %%edx\n\t"                            \+                          "call *"#funcOffset"(%%edx)\n\t"                  \+                          :"=a"(resultName) /* Output Operands */           \+                          :"c"(thisPtr)     /* Input Operands */            \+                          : "%edx" /* Clobbered Registers */                \+                         );                                                 \+++#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 )                 \+    __asm__ __volatile__ ("pushl %0\n\t"                                    \+                          "movl (%1), %%edx\n\t"                            \+                          "call *"#funcOffset"(%%edx)\n\t"                  \+                          :                 /* Output Operands */           \+                          :"r"(param1),     /* Input Operands */            \+                           "c"(thisPtr)                                     \+                          : "%edx" /* Clobbered Registers */                \+                         );                                                 \+++#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 )          \+    __asm__ __volatile__ ("pushl %1\n\t"                                    \+                          "movl (%2), %%edx\n\t"                            \+                          "call *"#funcOffset"(%%edx)\n\t"                  \+                          :"=a"(resultName) /* Output Operands */           \+                          :"r"(param1),     /* Input Operands */            \+                           "c"(thisPtr)                                     \+                          : "%edx" /* Clobbered Registers */                \+                          );                                                \+++#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 )   \+    do {                                                                    \+    double param1f64 = param1; /* Cast explicitly to double */              \+    double *param1f64Ptr = &param1f64; /* Make pointer to address */        \+     __asm__ __volatile__ ("pushl 4(%1)\n\t"                                \+                           "pushl (%1)\n\t"                                 \+                           "movl (%2), %%edx\n\t"                           \+                           "call *"#funcOffset"(%%edx);\n\t"                \+                           : "=a"(resultName) /* Output Operands */         \+                           : "r"(param1f64Ptr),  /* Input Operands */       \+                           "c"(thisPtr),                                    \+                           "m"(*param1f64Ptr) /* Using address */           \+                           : "%edx" /* Clobbered Registers */               \+                           );                                               \+    } while (0);                                                            \+++#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 )  \+    __asm__ __volatile__ ("pushl %1\n\t"                                    \+                          "pushl %2\n\t"                                    \+                          "movl (%3), %%edx\n\t"                            \+                          "call *"#funcOffset"(%%edx)\n\t"                  \+                          :"=a"(resultName) /* Output Operands */           \+                          :"r"(param2),     /* Input Operands */            \+                           "r"(param1),                                     \+                           "c"(thisPtr)                                     \+                          : "%edx" /* Clobbered Registers */                \+                          );                                                \+++#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\+    __asm__ __volatile__ ("pushl %1\n\t"                                    \+                          "pushl %2\n\t"                                    \+                          "pushl %3\n\t"                                    \+                          "pushl %4\n\t"                                    \+                          "movl (%5), %%edx\n\t"                            \+                          "call *"#funcOffset"(%%edx)\n\t"                  \+                          :"=a"(resultName) /* Output Operands */           \+                          :"r"(param4),     /* Input Operands  */           \+                           "r"(param3),                                     \+                           "r"(param2),                                     \+                           "r"(param1),                                     \+                           "c"(thisPtr)                                     \+                          : "%edx" /* Clobbered Registers */                \+                          );                                                \++#endif++++// Our static singleton instance.+IASIOThiscallResolver IASIOThiscallResolver::instance;++// Constructor called to initialize static Singleton instance above. Note that+// it is important not to clear that_ incase it has already been set by the call+// to placement new in ASIOInit().+IASIOThiscallResolver::IASIOThiscallResolver()+{+}++// Constructor called from ASIOInit() below+IASIOThiscallResolver::IASIOThiscallResolver(IASIO* that)+: that_( that )+{+}++// Implement IUnknown methods as assert(false). IASIOThiscallResolver is not+// really a COM object, just a wrapper which will work with the ASIO SDK.+// If you wanted to use ASIO without the SDK you might want to implement COM+// aggregation in these methods.+HRESULT STDMETHODCALLTYPE IASIOThiscallResolver::QueryInterface(REFIID riid, void **ppv)+{+    (void)riid;     // suppress unused variable warning++    assert( false ); // this function should never be called by the ASIO SDK.++    *ppv = NULL;+    return E_NOINTERFACE;+}++ULONG STDMETHODCALLTYPE IASIOThiscallResolver::AddRef()+{+    assert( false ); // this function should never be called by the ASIO SDK.++    return 1;+}++ULONG STDMETHODCALLTYPE IASIOThiscallResolver::Release()+{+    assert( false ); // this function should never be called by the ASIO SDK.+    +    return 1;+}+++// Implement the IASIO interface methods by performing the vptr manipulation+// described above then delegating to the real implementation.+ASIOBool IASIOThiscallResolver::init(void *sysHandle)+{+    ASIOBool result;+    CALL_THISCALL_1( result, that_, 12, sysHandle );+    return result;+}++void IASIOThiscallResolver::getDriverName(char *name)+{+    CALL_VOID_THISCALL_1( that_, 16, name );+}++long IASIOThiscallResolver::getDriverVersion()+{+    ASIOBool result;+    CALL_THISCALL_0( result, that_, 20 );+    return result;+}++void IASIOThiscallResolver::getErrorMessage(char *string)+{+     CALL_VOID_THISCALL_1( that_, 24, string );+}++ASIOError IASIOThiscallResolver::start()+{+    ASIOBool result;+    CALL_THISCALL_0( result, that_, 28 );+    return result;+}++ASIOError IASIOThiscallResolver::stop()+{+    ASIOBool result;+    CALL_THISCALL_0( result, that_, 32 );+    return result;+}++ASIOError IASIOThiscallResolver::getChannels(long *numInputChannels, long *numOutputChannels)+{+    ASIOBool result;+    CALL_THISCALL_2( result, that_, 36, numInputChannels, numOutputChannels );+    return result;+}++ASIOError IASIOThiscallResolver::getLatencies(long *inputLatency, long *outputLatency)+{+    ASIOBool result;+    CALL_THISCALL_2( result, that_, 40, inputLatency, outputLatency );+    return result;+}++ASIOError IASIOThiscallResolver::getBufferSize(long *minSize, long *maxSize,+        long *preferredSize, long *granularity)+{+    ASIOBool result;+    CALL_THISCALL_4( result, that_, 44, minSize, maxSize, preferredSize, granularity );+    return result;+}++ASIOError IASIOThiscallResolver::canSampleRate(ASIOSampleRate sampleRate)+{+    ASIOBool result;+    CALL_THISCALL_1_DOUBLE( result, that_, 48, sampleRate );+    return result;+}++ASIOError IASIOThiscallResolver::getSampleRate(ASIOSampleRate *sampleRate)+{+    ASIOBool result;+    CALL_THISCALL_1( result, that_, 52, sampleRate );+    return result;+}++ASIOError IASIOThiscallResolver::setSampleRate(ASIOSampleRate sampleRate)+{    +    ASIOBool result;+    CALL_THISCALL_1_DOUBLE( result, that_, 56, sampleRate );+    return result;+}++ASIOError IASIOThiscallResolver::getClockSources(ASIOClockSource *clocks, long *numSources)+{+    ASIOBool result;+    CALL_THISCALL_2( result, that_, 60, clocks, numSources );+    return result;+}++ASIOError IASIOThiscallResolver::setClockSource(long reference)+{+    ASIOBool result;+    CALL_THISCALL_1( result, that_, 64, reference );+    return result;+}++ASIOError IASIOThiscallResolver::getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp)+{+    ASIOBool result;+    CALL_THISCALL_2( result, that_, 68, sPos, tStamp );+    return result;+}++ASIOError IASIOThiscallResolver::getChannelInfo(ASIOChannelInfo *info)+{+    ASIOBool result;+    CALL_THISCALL_1( result, that_, 72, info );+    return result;+}++ASIOError IASIOThiscallResolver::createBuffers(ASIOBufferInfo *bufferInfos,+        long numChannels, long bufferSize, ASIOCallbacks *callbacks)+{+    ASIOBool result;+    CALL_THISCALL_4( result, that_, 76, bufferInfos, numChannels, bufferSize, callbacks );+    return result;+}++ASIOError IASIOThiscallResolver::disposeBuffers()+{+    ASIOBool result;+    CALL_THISCALL_0( result, that_, 80 );+    return result;+}++ASIOError IASIOThiscallResolver::controlPanel()+{+    ASIOBool result;+    CALL_THISCALL_0( result, that_, 84 );+    return result;+}++ASIOError IASIOThiscallResolver::future(long selector,void *opt)+{+    ASIOBool result;+    CALL_THISCALL_2( result, that_, 88, selector, opt );+    return result;+}++ASIOError IASIOThiscallResolver::outputReady()+{+    ASIOBool result;+    CALL_THISCALL_0( result, that_, 92 );+    return result;+}+++// Implement our substitute ASIOInit() method+ASIOError IASIOThiscallResolver::ASIOInit(ASIODriverInfo *info)+{+    // To ensure that our instance's vptr is correctly constructed, even if+    // ASIOInit is called prior to main(), we explicitly call its constructor+    // (potentially over the top of an existing instance). Note that this is+    // pretty ugly, and is only safe because IASIOThiscallResolver has no+    // destructor and contains no objects with destructors.+    new((void*)&instance) IASIOThiscallResolver( theAsioDriver );++    // Interpose between ASIO client code and the real driver.+    theAsioDriver = &instance;++    // Note that we never need to switch theAsioDriver back to point to the+    // real driver because theAsioDriver is reset to zero in ASIOExit().++    // Delegate to the real ASIOInit+	return ::ASIOInit(info);+}+++#endif /* !defined(_MSC_VER) */++#endif /* Win32 */+
+ cbits/include/iasiothiscallresolver.h view
@@ -0,0 +1,202 @@+// ****************************************************************************
+//
+// Changed:         I have modified this file slightly (includes) to work  with
+//                  RtAudio. RtAudio.cpp must include this file after asio.h.                                                    
+//
+// File:			IASIOThiscallResolver.h
+// Description:     The IASIOThiscallResolver class implements the IASIO
+//					interface and acts as a proxy to the real IASIO interface by
+//                  calling through its vptr table using the thiscall calling
+//                  convention. To put it another way, we interpose
+//                  IASIOThiscallResolver between ASIO SDK code and the driver.
+//                  This is necessary because most non-Microsoft compilers don't
+//                  implement the thiscall calling convention used by IASIO.
+//
+//					iasiothiscallresolver.cpp contains the background of this
+//					problem plus a technical description of the vptr
+//                  manipulations.
+//
+//					In order to use this mechanism one simply has to add
+//					iasiothiscallresolver.cpp to the list of files to compile
+//                  and #include <iasiothiscallresolver.h>
+//
+//					Note that this #include must come after the other ASIO SDK
+//                  #includes, for example:
+//
+//					#include <windows.h>
+//					#include <asiosys.h>
+//					#include <asio.h>
+//					#include <asiodrivers.h>
+//					#include <iasiothiscallresolver.h>
+//
+//					Actually the important thing is to #include
+//                  <iasiothiscallresolver.h> after <asio.h>. We have
+//                  incorporated a test to enforce this ordering.
+//
+//					The code transparently takes care of the interposition by
+//                  using macro substitution to intercept calls to ASIOInit()
+//                  and ASIOExit(). We save the original ASIO global
+//                  "theAsioDriver" in our "that" variable, and then set
+//                  "theAsioDriver" to equal our IASIOThiscallResolver instance.
+//
+// 					Whilst this method of resolving the thiscall problem requires
+//					the addition of #include <iasiothiscallresolver.h> to client
+//                  code it has the advantage that it does not break the terms
+//                  of the ASIO licence by publishing it. We are NOT modifying
+//                  any Steinberg code here, we are merely implementing the IASIO
+//					interface in the same way that we would need to do if we
+//					wished to provide an open source ASIO driver.
+//
+//					For compilation with MinGW -lole32 needs to be added to the
+//                  linker options. For BORLAND, linking with Import32.lib is
+//                  sufficient.
+//
+//					The dependencies are with: CoInitialize, CoUninitialize,
+//					CoCreateInstance, CLSIDFromString - used by asiolist.cpp
+//					and are required on Windows whether ThiscallResolver is used
+//					or not.
+//
+//					Searching for the above strings in the root library path
+//					of your compiler should enable the correct libraries to be
+//					identified if they aren't immediately obvious.
+//
+//                  Note that the current implementation of IASIOThiscallResolver
+//                  is not COM compliant - it does not correctly implement the
+//                  IUnknown interface. Implementing it is not necessary because
+//                  it is not called by parts of the ASIO SDK which call through
+//                  theAsioDriver ptr. The IUnknown methods are implemented as
+//                  assert(false) to ensure that the code fails if they are
+//                  ever called.
+// Restrictions:	None. Public Domain & Open Source distribute freely
+//					You may use IASIOThiscallResolver commercially as well as
+//                  privately.
+//					You the user assume the responsibility for the use of the
+//					files, binary or text, and there is no guarantee or warranty,
+//					expressed or implied, including but not limited to the
+//					implied warranties of merchantability and fitness for a
+//					particular purpose. You assume all responsibility and agree
+//					to hold no entity, copyright holder or distributors liable
+//					for any loss of data or inaccurate representations of data
+//					as a result of using IASIOThiscallResolver.
+// Version:         1.4 Added separate macro CALL_THISCALL_1_DOUBLE from
+//                  Andrew Baldwin, and volatile for whole gcc asm blocks,
+//                  both for compatibility with newer gcc versions. Cleaned up
+//                  Borland asm to use one less register.
+//                  1.3 Switched to including assert.h for better compatibility.
+//                  Wrapped entire .h and .cpp contents with a check for
+//                  _MSC_VER to provide better compatibility with MS compilers.
+//                  Changed Singleton implementation to use static instance
+//                  instead of freestore allocated instance. Removed ASIOExit
+//                  macro as it is no longer needed.
+//                  1.2 Removed semicolons from ASIOInit and ASIOExit macros to
+//                  allow them to be embedded in expressions (if statements).
+//                  Cleaned up some comments. Removed combase.c dependency (it
+//                  doesn't compile with BCB anyway) by stubbing IUnknown.
+//                  1.1 Incorporated comments from Ross Bencina including things
+//					such as changing name from ThiscallResolver to
+//					IASIOThiscallResolver, tidying up the constructor, fixing
+//					a bug in IASIOThiscallResolver::ASIOExit() and improving
+//					portability through the use of conditional compilation
+//					1.0 Initial working version.
+// Created:			6/09/2003
+// Authors:         Fraser Adams
+//                  Ross Bencina
+//                  Rene G. Ceballos
+//                  Martin Fay
+//                  Antti Silvast
+//                  Andrew Baldwin
+//
+// ****************************************************************************
+
+
+#ifndef included_iasiothiscallresolver_h
+#define included_iasiothiscallresolver_h
+
+// We only need IASIOThiscallResolver at all if we are on Win32. For other
+// platforms we simply bypass the IASIOThiscallResolver definition to allow us
+// to be safely #include'd whatever the platform to keep client code portable
+//#if defined(WIN32) || defined(_WIN32) || defined(__WIN32__)
+#if (defined(WIN32) || defined(_WIN32) || defined(__WIN32__)) && !defined(_WIN64)
+
+
+// If microsoft compiler we can call IASIO directly so IASIOThiscallResolver
+// is not used.
+#if !defined(_MSC_VER)
+
+
+// The following is in order to ensure that this header is only included after
+// the other ASIO headers (except for the case of iasiothiscallresolver.cpp).
+// We need to do this because IASIOThiscallResolver works by eclipsing the
+// original definition of ASIOInit() with a macro (see below).
+#if !defined(iasiothiscallresolver_sourcefile)
+	#if !defined(__ASIO_H)
+	#error iasiothiscallresolver.h must be included AFTER asio.h
+	#endif
+#endif
+
+#include <windows.h>
+#include "iasiodrv.h" /* From ASIO SDK */
+
+
+class IASIOThiscallResolver : public IASIO {
+private:
+	IASIO* that_; // Points to the real IASIO
+
+	static IASIOThiscallResolver instance; // Singleton instance
+
+	// Constructors - declared private so construction is limited to
+    // our Singleton instance
+    IASIOThiscallResolver();
+	IASIOThiscallResolver(IASIO* that);
+public:
+
+    // Methods from the IUnknown interface. We don't fully implement IUnknown
+    // because the ASIO SDK never calls these methods through theAsioDriver ptr.
+    // These methods are implemented as assert(false).
+    virtual HRESULT STDMETHODCALLTYPE QueryInterface(REFIID riid, void **ppv);
+    virtual ULONG STDMETHODCALLTYPE AddRef();
+    virtual ULONG STDMETHODCALLTYPE Release();
+
+    // Methods from the IASIO interface, implemented as forwarning calls to that.
+	virtual ASIOBool init(void *sysHandle);
+	virtual void getDriverName(char *name);
+	virtual long getDriverVersion();
+	virtual void getErrorMessage(char *string);
+	virtual ASIOError start();
+	virtual ASIOError stop();
+	virtual ASIOError getChannels(long *numInputChannels, long *numOutputChannels);
+	virtual ASIOError getLatencies(long *inputLatency, long *outputLatency);
+	virtual ASIOError getBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity);
+	virtual ASIOError canSampleRate(ASIOSampleRate sampleRate);
+	virtual ASIOError getSampleRate(ASIOSampleRate *sampleRate);
+	virtual ASIOError setSampleRate(ASIOSampleRate sampleRate);
+	virtual ASIOError getClockSources(ASIOClockSource *clocks, long *numSources);
+	virtual ASIOError setClockSource(long reference);
+	virtual ASIOError getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp);
+	virtual ASIOError getChannelInfo(ASIOChannelInfo *info);
+	virtual ASIOError createBuffers(ASIOBufferInfo *bufferInfos, long numChannels, long bufferSize, ASIOCallbacks *callbacks);
+	virtual ASIOError disposeBuffers();
+	virtual ASIOError controlPanel();
+	virtual ASIOError future(long selector,void *opt);
+	virtual ASIOError outputReady();
+
+    // Class method, see ASIOInit() macro below.
+    static ASIOError ASIOInit(ASIODriverInfo *info); // Delegates to ::ASIOInit
+};
+
+
+// Replace calls to ASIOInit with our interposing version.
+// This macro enables us to perform thiscall resolution simply by #including
+// <iasiothiscallresolver.h> after the asio #includes (this file _must_ be
+// included _after_ the asio #includes)
+
+#define ASIOInit(name) IASIOThiscallResolver::ASIOInit((name))
+
+
+#endif /* !defined(_MSC_VER) */
+
+#endif /* Win32 */
+
+#endif /* included_iasiothiscallresolver_h */
+
+
+ cbits/include/soundcard.h view
@@ -0,0 +1,1878 @@+/*+ * soundcard.h+ */++/*-+ * Copyright by Hannu Savolainen 1993 / 4Front Technologies 1993-2006+ * Modified for the new FreeBSD sound driver by Luigi Rizzo, 1997+ *+ * Redistribution and use in source and binary forms, with or without+ * modification, are permitted provided that the following conditions+ * are met:+ * 1. Redistributions of source code must retain the above copyright+ *    notice, this list of conditions and the following disclaimer.+ * 2. Redistributions in binary form must reproduce the above+ *    copyright notice, this list of conditions and the following+ *    disclaimer in the documentation and/or other materials provided+ *    with the distribution.+ *+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS''+ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED+ * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A+ * PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR+ * OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF+ * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED+ * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN+ * ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE+ * POSSIBILITY OF SUCH DAMAGE.+ *+ * $FreeBSD: src/sys/sys/soundcard.h,v 1.48 2006/11/26 11:55:48 netchild Exp $+ */++/*+ * Unless coordinating changes with 4Front Technologies, do NOT make any+ * modifications to ioctl commands, types, etc. that would break+ * compatibility with the OSS API.+ */++#ifndef _SYS_SOUNDCARD_H_+#define _SYS_SOUNDCARD_H_+ /*+  * If you make modifications to this file, please contact me before+  * distributing the modified version. There is already enough+  * diversity in the world.+  *+  * Regards,+  * Hannu Savolainen+  * hannu@voxware.pp.fi+  *+  **********************************************************************+  * PS.	The Hacker's Guide to VoxWare available from+  *     nic.funet.fi:pub/Linux/ALPHA/sound. The file is+  *	snd-sdk-doc-0.1.ps.gz (gzipped postscript). It contains+  *	some useful information about programming with VoxWare.+  *	(NOTE! The pub/Linux/ALPHA/ directories are hidden. You have+  *	to cd inside them before the files are accessible.)+  **********************************************************************+  */++/*+ * SOUND_VERSION is only used by the voxware driver. Hopefully apps+ * should not depend on it, but rather look at the capabilities+ * of the driver in the kernel!+ */+#define SOUND_VERSION  301+#define VOXWARE		/* does this have any use ? */++/*+ * Supported card ID numbers (Should be somewhere else? We keep+ * them here just for compativility with the old driver, but these+ * constants are of little or no use).+ */++#define SNDCARD_ADLIB          1+#define SNDCARD_SB             2+#define SNDCARD_PAS            3+#define SNDCARD_GUS            4+#define SNDCARD_MPU401         5+#define SNDCARD_SB16           6+#define SNDCARD_SB16MIDI       7+#define SNDCARD_UART6850       8+#define SNDCARD_GUS16          9+#define SNDCARD_MSS            10+#define SNDCARD_PSS            11+#define SNDCARD_SSCAPE         12+#define SNDCARD_PSS_MPU        13+#define SNDCARD_PSS_MSS        14+#define SNDCARD_SSCAPE_MSS     15+#define SNDCARD_TRXPRO         16+#define SNDCARD_TRXPRO_SB      17+#define SNDCARD_TRXPRO_MPU     18+#define SNDCARD_MAD16          19+#define SNDCARD_MAD16_MPU      20+#define SNDCARD_CS4232         21+#define SNDCARD_CS4232_MPU     22+#define SNDCARD_MAUI           23+#define SNDCARD_PSEUDO_MSS     24+#define SNDCARD_AWE32          25+#define SNDCARD_NSS            26+#define SNDCARD_UART16550      27+#define SNDCARD_OPL            28++#include <sys/types.h>+#include <machine/endian.h>+#ifndef _IOWR+#include <sys/ioccom.h>+#endif  /* !_IOWR */++/*+ * The first part of this file contains the new FreeBSD sound ioctl+ * interface. Tries to minimize the number of different ioctls, and+ * to be reasonably general.+ *+ * 970821: some of the new calls have not been implemented yet.+ */++/*+ * the following three calls extend the generic file descriptor+ * interface. AIONWRITE is the dual of FIONREAD, i.e. returns the max+ * number of bytes for a write operation to be non-blocking.+ *+ * AIOGSIZE/AIOSSIZE are used to change the behaviour of the device,+ * from a character device (default) to a block device. In block mode,+ * (not to be confused with blocking mode) the main difference for the+ * application is that select() will return only when a complete+ * block can be read/written to the device, whereas in character mode+ * select will return true when one byte can be exchanged. For audio+ * devices, character mode makes select almost useless since one byte+ * will always be ready by the next sample time (which is often only a+ * handful of microseconds away).+ * Use a size of 0 or 1 to return to character mode.+ */+#define	AIONWRITE   _IOR('A', 10, int)   /* get # bytes to write */+struct snd_size {+    int play_size;+    int rec_size;+};+#define	AIOGSIZE    _IOR('A', 11, struct snd_size)/* read current blocksize */+#define	AIOSSIZE    _IOWR('A', 11, struct snd_size)  /* sets blocksize */++/*+ * The following constants define supported audio formats. The+ * encoding follows voxware conventions, i.e. 1 bit for each supported+ * format. We extend it by using bit 31 (RO) to indicate full-duplex+ * capability, and bit 29 (RO) to indicate that the card supports/+ * needs different formats on capture & playback channels.+ * Bit 29 (RW) is used to indicate/ask stereo.+ *+ * The number of bits required to store the sample is:+ *  o  4 bits for the IDA ADPCM format,+ *  o  8 bits for 8-bit formats, mu-law and A-law,+ *  o  16 bits for the 16-bit formats, and+ *  o  32 bits for the 24/32-bit formats.+ *  o  undefined for the MPEG audio format.+ */++#define AFMT_QUERY	0x00000000	/* Return current format */+#define AFMT_MU_LAW	0x00000001	/* Logarithmic mu-law */+#define AFMT_A_LAW	0x00000002	/* Logarithmic A-law */+#define AFMT_IMA_ADPCM	0x00000004	/* A 4:1 compressed format where 16-bit+					 * squence represented using the+					 * the average 4 bits per sample */+#define AFMT_U8		0x00000008	/* Unsigned 8-bit */+#define AFMT_S16_LE	0x00000010	/* Little endian signed 16-bit */+#define AFMT_S16_BE	0x00000020	/* Big endian signed 16-bit */+#define AFMT_S8		0x00000040	/* Signed 8-bit */+#define AFMT_U16_LE	0x00000080	/* Little endian unsigned 16-bit */+#define AFMT_U16_BE	0x00000100	/* Big endian unsigned 16-bit */+#define AFMT_MPEG	0x00000200	/* MPEG MP2/MP3 audio */+#define AFMT_AC3	0x00000400	/* Dolby Digital AC3 */++#if _BYTE_ORDER == _LITTLE_ENDIAN+#define AFMT_S16_NE	AFMT_S16_LE	/* native endian signed 16 */+#else+#define AFMT_S16_NE	AFMT_S16_BE+#endif++/*+ * 32-bit formats below used for 24-bit audio data where the data is stored+ * in the 24 most significant bits and the least significant bits are not used+ * (should be set to 0).+ */+#define AFMT_S32_LE	0x00001000	/* Little endian signed 32-bit */+#define AFMT_S32_BE	0x00002000	/* Big endian signed 32-bit */+#define AFMT_U32_LE	0x00004000	/* Little endian unsigned 32-bit */+#define AFMT_U32_BE	0x00008000	/* Big endian unsigned 32-bit */+#define AFMT_S24_LE	0x00010000	/* Little endian signed 24-bit */+#define AFMT_S24_BE	0x00020000	/* Big endian signed 24-bit */+#define AFMT_U24_LE	0x00040000	/* Little endian unsigned 24-bit */+#define AFMT_U24_BE	0x00080000	/* Big endian unsigned 24-bit */++#define AFMT_STEREO	0x10000000	/* can do/want stereo	*/++/*+ * the following are really capabilities+ */+#define AFMT_WEIRD	0x20000000	/* weird hardware...	*/+    /*+     * AFMT_WEIRD reports that the hardware might need to operate+     * with different formats in the playback and capture+     * channels when operating in full duplex.+     * As an example, SoundBlaster16 cards only support U8 in one+     * direction and S16 in the other one, and applications should+     * be aware of this limitation.+     */+#define AFMT_FULLDUPLEX	0x80000000	/* can do full duplex	*/++/*+ * The following structure is used to get/set format and sampling rate.+ * While it would be better to have things such as stereo, bits per+ * sample, endiannes, etc split in different variables, it turns out+ * that formats are not that many, and not all combinations are possible.+ * So we followed the Voxware approach of associating one bit to each+ * format.+ */++typedef struct _snd_chan_param {+    u_long	play_rate;	/* sampling rate			*/+    u_long	rec_rate;	/* sampling rate			*/+    u_long	play_format;	/* everything describing the format	*/+    u_long	rec_format;	/* everything describing the format	*/+} snd_chan_param;+#define	AIOGFMT    _IOR('f', 12, snd_chan_param)   /* get format */+#define	AIOSFMT    _IOWR('f', 12, snd_chan_param)  /* sets format */++/*+ * The following structure is used to get/set the mixer setting.+ * Up to 32 mixers are supported, each one with up to 32 channels.+ */+typedef struct _snd_mix_param {+    u_char	subdev;	/* which output				*/+    u_char	line;	/* which input				*/+    u_char	left,right; /* volumes, 0..255, 0 = mute	*/+} snd_mix_param ;++/* XXX AIOGMIX, AIOSMIX not implemented yet */+#define AIOGMIX	_IOWR('A', 13, snd_mix_param)	/* return mixer status */+#define AIOSMIX	_IOWR('A', 14, snd_mix_param)	/* sets mixer status   */++/*+ * channel specifiers used in AIOSTOP and AIOSYNC+ */+#define	AIOSYNC_PLAY	0x1	/* play chan */+#define	AIOSYNC_CAPTURE	0x2	/* capture chan */+/* AIOSTOP stop & flush a channel, returns the residual count */+#define	AIOSTOP	_IOWR ('A', 15, int)++/* alternate method used to notify the sync condition */+#define	AIOSYNC_SIGNAL	0x100+#define	AIOSYNC_SELECT	0x200++/* what the 'pos' field refers to */+#define AIOSYNC_READY	0x400+#define AIOSYNC_FREE	0x800++typedef struct _snd_sync_parm {+    long chan ; /* play or capture channel, plus modifier */+    long pos;+} snd_sync_parm;+#define	AIOSYNC	_IOWR ('A', 15, snd_sync_parm)	/* misc. synchronization */++/*+ * The following is used to return device capabilities. If the structure+ * passed to the ioctl is zeroed, default values are returned for rate+ * and formats, a bitmap of available mixers is returned, and values+ * (inputs, different levels) for the first one are returned.+ *+ * If  formats, mixers, inputs are instantiated, then detailed info+ * are returned depending on the call.+ */+typedef struct _snd_capabilities {+    u_long	rate_min, rate_max;	/* min-max sampling rate */+    u_long	formats;+    u_long	bufsize; /* DMA buffer size */+    u_long	mixers; /* bitmap of available mixers */+    u_long	inputs; /* bitmap of available inputs (per mixer) */+    u_short	left, right;	/* how many levels are supported */+} snd_capabilities;+#define AIOGCAP	_IOWR('A', 15, snd_capabilities)	/* get capabilities */++/*+ * here is the old (Voxware) ioctl interface+ */++/*+ * IOCTL Commands for /dev/sequencer+ */++#define SNDCTL_SEQ_RESET	_IO  ('Q', 0)+#define SNDCTL_SEQ_SYNC		_IO  ('Q', 1)+#define SNDCTL_SYNTH_INFO	_IOWR('Q', 2, struct synth_info)+#define SNDCTL_SEQ_CTRLRATE	_IOWR('Q', 3, int) /* Set/get timer res.(hz) */+#define SNDCTL_SEQ_GETOUTCOUNT	_IOR ('Q', 4, int)+#define SNDCTL_SEQ_GETINCOUNT	_IOR ('Q', 5, int)+#define SNDCTL_SEQ_PERCMODE	_IOW ('Q', 6, int)+#define SNDCTL_FM_LOAD_INSTR	_IOW ('Q', 7, struct sbi_instrument)	/* Valid for FM only */+#define SNDCTL_SEQ_TESTMIDI	_IOW ('Q', 8, int)+#define SNDCTL_SEQ_RESETSAMPLES	_IOW ('Q', 9, int)+#define SNDCTL_SEQ_NRSYNTHS	_IOR ('Q',10, int)+#define SNDCTL_SEQ_NRMIDIS	_IOR ('Q',11, int)+#define SNDCTL_MIDI_INFO	_IOWR('Q',12, struct midi_info)+#define SNDCTL_SEQ_THRESHOLD	_IOW ('Q',13, int)+#define SNDCTL_SEQ_TRESHOLD	SNDCTL_SEQ_THRESHOLD	/* there was once a typo */+#define SNDCTL_SYNTH_MEMAVL	_IOWR('Q',14, int) /* in=dev#, out=memsize */+#define SNDCTL_FM_4OP_ENABLE	_IOW ('Q',15, int) /* in=dev# */+#define SNDCTL_PMGR_ACCESS	_IOWR('Q',16, struct patmgr_info)+#define SNDCTL_SEQ_PANIC	_IO  ('Q',17)+#define SNDCTL_SEQ_OUTOFBAND	_IOW ('Q',18, struct seq_event_rec)+#define SNDCTL_SEQ_GETTIME	_IOR ('Q',19, int)++struct seq_event_rec {+	u_char arr[8];+};++#define SNDCTL_TMR_TIMEBASE	_IOWR('T', 1, int)+#define SNDCTL_TMR_START	_IO  ('T', 2)+#define SNDCTL_TMR_STOP		_IO  ('T', 3)+#define SNDCTL_TMR_CONTINUE	_IO  ('T', 4)+#define SNDCTL_TMR_TEMPO	_IOWR('T', 5, int)+#define SNDCTL_TMR_SOURCE	_IOWR('T', 6, int)+#   define TMR_INTERNAL		0x00000001+#   define TMR_EXTERNAL		0x00000002+#	define TMR_MODE_MIDI	0x00000010+#	define TMR_MODE_FSK	0x00000020+#	define TMR_MODE_CLS	0x00000040+#	define TMR_MODE_SMPTE	0x00000080+#define SNDCTL_TMR_METRONOME	_IOW ('T', 7, int)+#define SNDCTL_TMR_SELECT	_IOW ('T', 8, int)++/*+ *	Endian aware patch key generation algorithm.+ */++#if defined(_AIX) || defined(AIX)+#  define _PATCHKEY(id) (0xfd00|id)+#else+#  define _PATCHKEY(id) ((id<<8)|0xfd)+#endif++/*+ *	Sample loading mechanism for internal synthesizers (/dev/sequencer)+ *	The following patch_info structure has been designed to support+ *	Gravis UltraSound. It tries to be universal format for uploading+ *	sample based patches but is probably too limited.+ */++struct patch_info {+/*		u_short key;		 Use GUS_PATCH here */+	short key;		 /* Use GUS_PATCH here */+#define GUS_PATCH	_PATCHKEY(0x04)+#define OBSOLETE_GUS_PATCH	_PATCHKEY(0x02)++	short device_no;	/* Synthesizer number */+	short instr_no;		/* Midi pgm# */++	u_long mode;+/*+ * The least significant byte has the same format than the GUS .PAT+ * files+ */+#define WAVE_16_BITS	0x01	/* bit 0 = 8 or 16 bit wave data. */+#define WAVE_UNSIGNED	0x02	/* bit 1 = Signed - Unsigned data. */+#define WAVE_LOOPING	0x04	/* bit 2 = looping enabled-1. */+#define WAVE_BIDIR_LOOP	0x08	/* bit 3 = Set is bidirectional looping. */+#define WAVE_LOOP_BACK	0x10	/* bit 4 = Set is looping backward. */+#define WAVE_SUSTAIN_ON	0x20	/* bit 5 = Turn sustaining on. (Env. pts. 3)*/+#define WAVE_ENVELOPES	0x40	/* bit 6 = Enable envelopes - 1 */+				/* 	(use the env_rate/env_offs fields). */+/* Linux specific bits */+#define WAVE_VIBRATO	0x00010000	/* The vibrato info is valid */+#define WAVE_TREMOLO	0x00020000	/* The tremolo info is valid */+#define WAVE_SCALE	0x00040000	/* The scaling info is valid */+/* Other bits must be zeroed */++	long len;	/* Size of the wave data in bytes */+	long loop_start, loop_end; /* Byte offsets from the beginning */++/*+ * The base_freq and base_note fields are used when computing the+ * playback speed for a note. The base_note defines the tone frequency+ * which is heard if the sample is played using the base_freq as the+ * playback speed.+ *+ * The low_note and high_note fields define the minimum and maximum note+ * frequencies for which this sample is valid. It is possible to define+ * more than one samples for an instrument number at the same time. The+ * low_note and high_note fields are used to select the most suitable one.+ *+ * The fields base_note, high_note and low_note should contain+ * the note frequency multiplied by 1000. For example value for the+ * middle A is 440*1000.+ */++	u_int base_freq;+	u_long base_note;+	u_long high_note;+	u_long low_note;+	int panning;	/* -128=left, 127=right */+	int detuning;++/*	New fields introduced in version 1.99.5	*/++       /* Envelope. Enabled by mode bit WAVE_ENVELOPES	*/+	u_char	env_rate[ 6 ];	 /* GUS HW ramping rate */+	u_char	env_offset[ 6 ]; /* 255 == 100% */++	/*+	 * The tremolo, vibrato and scale info are not supported yet.+	 * Enable by setting the mode bits WAVE_TREMOLO, WAVE_VIBRATO or+	 * WAVE_SCALE+	 */++	u_char	tremolo_sweep;+	u_char	tremolo_rate;+	u_char	tremolo_depth;++	u_char	vibrato_sweep;+	u_char	vibrato_rate;+	u_char	vibrato_depth;++	int		scale_frequency;+	u_int	scale_factor;		/* from 0 to 2048 or 0 to 2 */++	int		volume;+	int		spare[4];+	char data[1];	/* The waveform data starts here */+};++struct sysex_info {+	short key;		/* Use GUS_PATCH here */+#define SYSEX_PATCH	_PATCHKEY(0x05)+#define MAUI_PATCH	_PATCHKEY(0x06)+	short device_no;	/* Synthesizer number */+	long len;	/* Size of the sysex data in bytes */+	u_char data[1];	/* Sysex data starts here */+};++/*+ * Patch management interface (/dev/sequencer, /dev/patmgr#)+ * Don't use these calls if you want to maintain compatibility with+ * the future versions of the driver.+ */++#define PS_NO_PATCHES		0	/* No patch support on device */+#define	PS_MGR_NOT_OK		1	/* Plain patch support (no mgr) */+#define	PS_MGR_OK		2	/* Patch manager supported */+#define	PS_MANAGED		3	/* Patch manager running */++#define SNDCTL_PMGR_IFACE		_IOWR('P', 1, struct patmgr_info)++/*+ * The patmgr_info is a fixed size structure which is used for two+ * different purposes. The intended use is for communication between+ * the application using /dev/sequencer and the patch manager daemon+ * associated with a synthesizer device (ioctl(SNDCTL_PMGR_ACCESS)).+ *+ * This structure is also used with ioctl(SNDCTL_PGMR_IFACE) which allows+ * a patch manager daemon to read and write device parameters. This+ * ioctl available through /dev/sequencer also. Avoid using it since it's+ * extremely hardware dependent. In addition access trough /dev/sequencer+ * may confuse the patch manager daemon.+ */++struct patmgr_info {	/* Note! size must be < 4k since kmalloc() is used */+	  u_long key;	/* Don't worry. Reserved for communication+	  			   between the patch manager and the driver. */+#define PM_K_EVENT		1 /* Event from the /dev/sequencer driver */+#define PM_K_COMMAND		2 /* Request from an application */+#define PM_K_RESPONSE		3 /* From patmgr to application */+#define PM_ERROR		4 /* Error returned by the patmgr */+	  int device;+	  int command;++/*+ * Commands 0x000 to 0xfff reserved for patch manager programs+ */+#define PM_GET_DEVTYPE	1	/* Returns type of the patch mgr interface of dev */+#define		PMTYPE_FM2	1	/* 2 OP fm */+#define		PMTYPE_FM4	2	/* Mixed 4 or 2 op FM (OPL-3) */+#define		PMTYPE_WAVE	3	/* Wave table synthesizer (GUS) */+#define PM_GET_NRPGM	2	/* Returns max # of midi programs in parm1 */+#define PM_GET_PGMMAP	3	/* Returns map of loaded midi programs in data8 */+#define PM_GET_PGM_PATCHES 4	/* Return list of patches of a program (parm1) */+#define PM_GET_PATCH	5	/* Return patch header of patch parm1 */+#define PM_SET_PATCH	6	/* Set patch header of patch parm1 */+#define PM_READ_PATCH	7	/* Read patch (wave) data */+#define PM_WRITE_PATCH	8	/* Write patch (wave) data */++/*+ * Commands 0x1000 to 0xffff are for communication between the patch manager+ * and the client+ */+#define _PM_LOAD_PATCH	0x100++/*+ * Commands above 0xffff reserved for device specific use+ */++	long parm1;+	long parm2;+	long parm3;++	union {+		u_char data8[4000];+		u_short data16[2000];+		u_long data32[1000];+		struct patch_info patch;+	} data;+};++/*+ * When a patch manager daemon is present, it will be informed by the+ * driver when something important happens. For example when the+ * /dev/sequencer is opened or closed. A record with key == PM_K_EVENT is+ * returned. The command field contains the event type:+ */+#define PM_E_OPENED		1	/* /dev/sequencer opened */+#define PM_E_CLOSED		2	/* /dev/sequencer closed */+#define PM_E_PATCH_RESET	3	/* SNDCTL_RESETSAMPLES called */+#define PM_E_PATCH_LOADED	4	/* A patch has been loaded by appl */++/*+ * /dev/sequencer input events.+ *+ * The data written to the /dev/sequencer is a stream of events. Events+ * are records of 4 or 8 bytes. The first byte defines the size.+ * Any number of events can be written with a write call. There+ * is a set of macros for sending these events. Use these macros if you+ * want to maximize portability of your program.+ *+ * Events SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO. Are also input events.+ * (All input events are currently 4 bytes long. Be prepared to support+ * 8 byte events also. If you receive any event having first byte >= 128,+ * it's a 8 byte event.+ *+ * The events are documented at the end of this file.+ *+ * Normal events (4 bytes)+ * There is also a 8 byte version of most of the 4 byte events. The+ * 8 byte one is recommended.+ */+#define SEQ_NOTEOFF		0+#define SEQ_FMNOTEOFF		SEQ_NOTEOFF	/* Just old name */+#define SEQ_NOTEON		1+#define	SEQ_FMNOTEON		SEQ_NOTEON+#define SEQ_WAIT		TMR_WAIT_ABS+#define SEQ_PGMCHANGE		3+#define SEQ_FMPGMCHANGE		SEQ_PGMCHANGE+#define SEQ_SYNCTIMER		TMR_START+#define SEQ_MIDIPUTC		5+#define SEQ_DRUMON		6	/*** OBSOLETE ***/+#define SEQ_DRUMOFF		7	/*** OBSOLETE ***/+#define SEQ_ECHO		TMR_ECHO	/* For synching programs with output */+#define SEQ_AFTERTOUCH		9+#define SEQ_CONTROLLER		10++/*+ *	Midi controller numbers+ *+ * Controllers 0 to 31 (0x00 to 0x1f) and 32 to 63 (0x20 to 0x3f)+ * are continuous controllers.+ * In the MIDI 1.0 these controllers are sent using two messages.+ * Controller numbers 0 to 31 are used to send the MSB and the+ * controller numbers 32 to 63 are for the LSB. Note that just 7 bits+ * are used in MIDI bytes.+ */++#define	CTL_BANK_SELECT		0x00+#define	CTL_MODWHEEL		0x01+#define CTL_BREATH		0x02+/*	undefined		0x03 */+#define CTL_FOOT		0x04+#define CTL_PORTAMENTO_TIME	0x05+#define CTL_DATA_ENTRY		0x06+#define CTL_MAIN_VOLUME		0x07+#define CTL_BALANCE		0x08+/*	undefined		0x09 */+#define CTL_PAN			0x0a+#define CTL_EXPRESSION		0x0b+/*	undefined		0x0c - 0x0f */+#define CTL_GENERAL_PURPOSE1	0x10+#define CTL_GENERAL_PURPOSE2	0x11+#define CTL_GENERAL_PURPOSE3	0x12+#define CTL_GENERAL_PURPOSE4	0x13+/*	undefined		0x14 - 0x1f */++/*	undefined		0x20 */++/*+ * The controller numbers 0x21 to 0x3f are reserved for the+ * least significant bytes of the controllers 0x00 to 0x1f.+ * These controllers are not recognised by the driver.+ *+ * Controllers 64 to 69 (0x40 to 0x45) are on/off switches.+ * 0=OFF and 127=ON (intermediate values are possible)+ */+#define CTL_DAMPER_PEDAL	0x40+#define CTL_SUSTAIN		CTL_DAMPER_PEDAL	/* Alias */+#define CTL_HOLD		CTL_DAMPER_PEDAL	/* Alias */+#define CTL_PORTAMENTO		0x41+#define CTL_SOSTENUTO		0x42+#define CTL_SOFT_PEDAL		0x43+/*	undefined		0x44 */+#define CTL_HOLD2		0x45+/*	undefined		0x46 - 0x4f */++#define CTL_GENERAL_PURPOSE5	0x50+#define CTL_GENERAL_PURPOSE6	0x51+#define CTL_GENERAL_PURPOSE7	0x52+#define CTL_GENERAL_PURPOSE8	0x53+/*	undefined		0x54 - 0x5a */+#define CTL_EXT_EFF_DEPTH	0x5b+#define CTL_TREMOLO_DEPTH	0x5c+#define CTL_CHORUS_DEPTH	0x5d+#define CTL_DETUNE_DEPTH	0x5e+#define CTL_CELESTE_DEPTH	CTL_DETUNE_DEPTH /* Alias for the above one */+#define CTL_PHASER_DEPTH	0x5f+#define CTL_DATA_INCREMENT	0x60+#define CTL_DATA_DECREMENT	0x61+#define CTL_NONREG_PARM_NUM_LSB	0x62+#define CTL_NONREG_PARM_NUM_MSB	0x63+#define CTL_REGIST_PARM_NUM_LSB	0x64+#define CTL_REGIST_PARM_NUM_MSB	0x65+/*	undefined		0x66 - 0x78 */+/*	reserved		0x79 - 0x7f */++/* Pseudo controllers (not midi compatible) */+#define CTRL_PITCH_BENDER	255+#define CTRL_PITCH_BENDER_RANGE	254+#define CTRL_EXPRESSION		253	/* Obsolete */+#define CTRL_MAIN_VOLUME	252	/* Obsolete */++#define SEQ_BALANCE		11+#define SEQ_VOLMODE             12++/*+ * Volume mode decides how volumes are used+ */++#define VOL_METHOD_ADAGIO	1+#define VOL_METHOD_LINEAR	2++/*+ * Note! SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO are used also as+ *	 input events.+ */++/*+ * Event codes 0xf0 to 0xfc are reserved for future extensions.+ */++#define SEQ_FULLSIZE		0xfd	/* Long events */+/*+ * SEQ_FULLSIZE events are used for loading patches/samples to the+ * synthesizer devices. These events are passed directly to the driver+ * of the associated synthesizer device. There is no limit to the size+ * of the extended events. These events are not queued but executed+ * immediately when the write() is called (execution can take several+ * seconds of time).+ *+ * When a SEQ_FULLSIZE message is written to the device, it must+ * be written using exactly one write() call. Other events cannot+ * be mixed to the same write.+ *+ * For FM synths (YM3812/OPL3) use struct sbi_instrument and write+ * it to the /dev/sequencer. Don't write other data together with+ * the instrument structure Set the key field of the structure to+ * FM_PATCH. The device field is used to route the patch to the+ * corresponding device.+ *+ * For Gravis UltraSound use struct patch_info. Initialize the key field+ * to GUS_PATCH.+ */+#define SEQ_PRIVATE	0xfe	/* Low level HW dependent events (8 bytes) */+#define SEQ_EXTENDED	0xff	/* Extended events (8 bytes) OBSOLETE */++/*+ * Record for FM patches+ */++typedef u_char sbi_instr_data[32];++struct sbi_instrument {+	u_short	key;	/* FM_PATCH or OPL3_PATCH */+#define FM_PATCH	_PATCHKEY(0x01)+#define OPL3_PATCH	_PATCHKEY(0x03)+	short		device;		/* Synth# (0-4)	*/+	int 		channel;	/* Program# to be initialized  */+	sbi_instr_data	operators;	/* Reg. settings for operator cells+					 * (.SBI format)	*/+};++struct synth_info {	/* Read only */+	char	name[30];+	int	device;		/* 0-N. INITIALIZE BEFORE CALLING */+	int	synth_type;+#define SYNTH_TYPE_FM			0+#define SYNTH_TYPE_SAMPLE		1+#define SYNTH_TYPE_MIDI			2	/* Midi interface */++	int	synth_subtype;+#define FM_TYPE_ADLIB			0x00+#define FM_TYPE_OPL3			0x01+#define MIDI_TYPE_MPU401		0x401++#define SAMPLE_TYPE_BASIC		0x10+#define SAMPLE_TYPE_GUS			SAMPLE_TYPE_BASIC+#define SAMPLE_TYPE_AWE32		0x20++	int	perc_mode;	/* No longer supported */+	int	nr_voices;+	int	nr_drums;	/* Obsolete field */+	int	instr_bank_size;+	u_long	capabilities;+#define SYNTH_CAP_PERCMODE	0x00000001 /* No longer used */+#define SYNTH_CAP_OPL3		0x00000002 /* Set if OPL3 supported */+#define SYNTH_CAP_INPUT		0x00000004 /* Input (MIDI) device */+	int	dummies[19];	/* Reserve space */+};++struct sound_timer_info {+	char name[32];+	int caps;+};++struct midi_info {+	char		name[30];+	int		device;		/* 0-N. INITIALIZE BEFORE CALLING */+	u_long	capabilities;	/* To be defined later */+	int		dev_type;+	int		dummies[18];	/* Reserve space */+};++/*+ * ioctl commands for the /dev/midi##+ */+typedef struct {+	u_char cmd;+	char nr_args, nr_returns;+	u_char data[30];+} mpu_command_rec;++#define SNDCTL_MIDI_PRETIME	_IOWR('m', 0, int)+#define SNDCTL_MIDI_MPUMODE	_IOWR('m', 1, int)+#define SNDCTL_MIDI_MPUCMD	_IOWR('m', 2, mpu_command_rec)+#define MIOSPASSTHRU		_IOWR('m', 3, int)+#define MIOGPASSTHRU		_IOWR('m', 4, int)++/*+ * IOCTL commands for /dev/dsp and /dev/audio+ */++#define SNDCTL_DSP_RESET	_IO  ('P', 0)+#define SNDCTL_DSP_SYNC		_IO  ('P', 1)+#define SNDCTL_DSP_SPEED	_IOWR('P', 2, int)+#define SNDCTL_DSP_STEREO	_IOWR('P', 3, int)+#define SNDCTL_DSP_GETBLKSIZE	_IOR('P', 4, int)+#define SNDCTL_DSP_SETBLKSIZE   _IOW('P', 4, int)+#define SNDCTL_DSP_SETFMT	_IOWR('P',5, int) /* Selects ONE fmt*/++/*+ * SOUND_PCM_WRITE_CHANNELS is not that different+ * from SNDCTL_DSP_STEREO+ */+#define SOUND_PCM_WRITE_CHANNELS	_IOWR('P', 6, int)+#define SNDCTL_DSP_CHANNELS	SOUND_PCM_WRITE_CHANNELS+#define SOUND_PCM_WRITE_FILTER	_IOWR('P', 7, int)+#define SNDCTL_DSP_POST		_IO  ('P', 8)++/*+ * SNDCTL_DSP_SETBLKSIZE and the following two calls mostly do+ * the same thing, i.e. set the block size used in DMA transfers.+ */+#define SNDCTL_DSP_SUBDIVIDE	_IOWR('P', 9, int)+#define SNDCTL_DSP_SETFRAGMENT	_IOWR('P',10, int)+++#define SNDCTL_DSP_GETFMTS	_IOR ('P',11, int) /* Returns a mask */+/*+ * Buffer status queries.+ */+typedef struct audio_buf_info {+    int fragments;	/* # of avail. frags (partly used ones not counted) */+    int fragstotal;	/* Total # of fragments allocated */+    int fragsize;	/* Size of a fragment in bytes */++    int bytes;	/* Avail. space in bytes (includes partly used fragments) */+		/* Note! 'bytes' could be more than fragments*fragsize */+} audio_buf_info;++#define SNDCTL_DSP_GETOSPACE	_IOR ('P',12, audio_buf_info)+#define SNDCTL_DSP_GETISPACE	_IOR ('P',13, audio_buf_info)++/*+ * SNDCTL_DSP_NONBLOCK is the same (but less powerful, since the+ * action cannot be undone) of FIONBIO. The same can be achieved+ * by opening the device with O_NDELAY+ */+#define SNDCTL_DSP_NONBLOCK	_IO  ('P',14)++#define SNDCTL_DSP_GETCAPS	_IOR ('P',15, int)+#define DSP_CAP_REVISION	0x000000ff /* revision level (0 to 255) */+#define DSP_CAP_DUPLEX		0x00000100 /* Full duplex record/playback */+#define DSP_CAP_REALTIME	0x00000200 /* Real time capability */+#define DSP_CAP_BATCH		0x00000400+    /*+     * Device has some kind of internal buffers which may+     * cause some delays and decrease precision of timing+     */+#define DSP_CAP_COPROC		0x00000800+    /* Has a coprocessor, sometimes it's a DSP but usually not */+#define DSP_CAP_TRIGGER		0x00001000 /* Supports SETTRIGGER */+#define DSP_CAP_MMAP 0x00002000 /* Supports mmap() */++/*+ * What do these function do ?+ */+#define SNDCTL_DSP_GETTRIGGER	_IOR ('P',16, int)+#define SNDCTL_DSP_SETTRIGGER	_IOW ('P',16, int)+#define PCM_ENABLE_INPUT	0x00000001+#define PCM_ENABLE_OUTPUT	0x00000002++typedef struct count_info {+	int bytes;	/* Total # of bytes processed */+	int blocks;	/* # of fragment transitions since last time */+	int ptr;	/* Current DMA pointer value */+} count_info;++/*+ * GETIPTR and GETISPACE are not that different... same for out.+ */+#define SNDCTL_DSP_GETIPTR	_IOR ('P',17, count_info)+#define SNDCTL_DSP_GETOPTR	_IOR ('P',18, count_info)++typedef struct buffmem_desc {+	caddr_t buffer;+	int size;+} buffmem_desc;++#define SNDCTL_DSP_MAPINBUF	_IOR ('P', 19, buffmem_desc)+#define SNDCTL_DSP_MAPOUTBUF	_IOR ('P', 20, buffmem_desc)+#define SNDCTL_DSP_SETSYNCRO	_IO  ('P', 21)+#define SNDCTL_DSP_SETDUPLEX	_IO  ('P', 22)+#define SNDCTL_DSP_GETODELAY	_IOR ('P', 23, int)++/*+ * I guess these are the readonly version of the same+ * functions that exist above as SNDCTL_DSP_...+ */+#define SOUND_PCM_READ_RATE	_IOR ('P', 2, int)+#define SOUND_PCM_READ_CHANNELS	_IOR ('P', 6, int)+#define SOUND_PCM_READ_BITS	_IOR ('P', 5, int)+#define SOUND_PCM_READ_FILTER	_IOR ('P', 7, int)++/*+ * ioctl calls to be used in communication with coprocessors and+ * DSP chips.+ */++typedef struct copr_buffer {+	int command;	/* Set to 0 if not used */+	int flags;+#define CPF_NONE		0x0000+#define CPF_FIRST		0x0001	/* First block */+#define CPF_LAST		0x0002	/* Last block */+	int len;+	int offs;	/* If required by the device (0 if not used) */++	u_char data[4000]; /* NOTE! 4000 is not 4k */+} copr_buffer;++typedef struct copr_debug_buf {+	int command;	/* Used internally. Set to 0 */+	int parm1;+	int parm2;+	int flags;+	int len;	/* Length of data in bytes */+} copr_debug_buf;++typedef struct copr_msg {+	int len;+	u_char data[4000];+} copr_msg;++#define SNDCTL_COPR_RESET       _IO  ('C',  0)+#define SNDCTL_COPR_LOAD	_IOWR('C',  1, copr_buffer)+#define SNDCTL_COPR_RDATA	_IOWR('C',  2, copr_debug_buf)+#define SNDCTL_COPR_RCODE	_IOWR('C',  3, copr_debug_buf)+#define SNDCTL_COPR_WDATA	_IOW ('C',  4, copr_debug_buf)+#define SNDCTL_COPR_WCODE	_IOW ('C',  5, copr_debug_buf)+#define SNDCTL_COPR_RUN		_IOWR('C',  6, copr_debug_buf)+#define SNDCTL_COPR_HALT	_IOWR('C',  7, copr_debug_buf)+#define SNDCTL_COPR_SENDMSG	_IOW ('C',  8, copr_msg)+#define SNDCTL_COPR_RCVMSG	_IOR ('C',  9, copr_msg)++/*+ * IOCTL commands for /dev/mixer+ */++/*+ * Mixer devices+ *+ * There can be up to 20 different analog mixer channels. The+ * SOUND_MIXER_NRDEVICES gives the currently supported maximum.+ * The SOUND_MIXER_READ_DEVMASK returns a bitmask which tells+ * the devices supported by the particular mixer.+ */++#define SOUND_MIXER_NRDEVICES	25+#define SOUND_MIXER_VOLUME	0	/* Master output level */+#define SOUND_MIXER_BASS	1	/* Treble level of all output channels */+#define SOUND_MIXER_TREBLE	2	/* Bass level of all output channels */+#define SOUND_MIXER_SYNTH	3	/* Volume of synthesier input */+#define SOUND_MIXER_PCM		4	/* Output level for the audio device */+#define SOUND_MIXER_SPEAKER	5	/* Output level for the PC speaker+					 * signals */+#define SOUND_MIXER_LINE	6	/* Volume level for the line in jack */+#define SOUND_MIXER_MIC		7	/* Volume for the signal coming from+					 * the microphone jack */+#define SOUND_MIXER_CD		8	/* Volume level for the input signal+					 * connected to the CD audio input */+#define SOUND_MIXER_IMIX	9	/* Recording monitor. It controls the+					 * output volume of the selected+					 * recording sources while recording */+#define SOUND_MIXER_ALTPCM	10	/* Volume of the alternative codec+					 * device */+#define SOUND_MIXER_RECLEV	11	/* Global recording level */+#define SOUND_MIXER_IGAIN	12	/* Input gain */+#define SOUND_MIXER_OGAIN	13	/* Output gain */+/*+ * The AD1848 codec and compatibles have three line level inputs+ * (line, aux1 and aux2). Since each card manufacturer have assigned+ * different meanings to these inputs, it's inpractical to assign+ * specific meanings (line, cd, synth etc.) to them.+ */+#define SOUND_MIXER_LINE1	14	/* Input source 1  (aux1) */+#define SOUND_MIXER_LINE2	15	/* Input source 2  (aux2) */+#define SOUND_MIXER_LINE3	16	/* Input source 3  (line) */+#define SOUND_MIXER_DIGITAL1    17      /* Digital (input) 1 */+#define SOUND_MIXER_DIGITAL2    18      /* Digital (input) 2 */+#define SOUND_MIXER_DIGITAL3    19      /* Digital (input) 3 */+#define SOUND_MIXER_PHONEIN     20      /* Phone input */+#define SOUND_MIXER_PHONEOUT    21      /* Phone output */+#define SOUND_MIXER_VIDEO       22      /* Video/TV (audio) in */+#define SOUND_MIXER_RADIO       23      /* Radio in */+#define SOUND_MIXER_MONITOR     24      /* Monitor (usually mic) volume */+++/*+ * Some on/off settings (SOUND_SPECIAL_MIN - SOUND_SPECIAL_MAX)+ * Not counted to SOUND_MIXER_NRDEVICES, but use the same number space+ */+#define SOUND_ONOFF_MIN		28+#define SOUND_ONOFF_MAX		30+#define SOUND_MIXER_MUTE	28	/* 0 or 1 */+#define SOUND_MIXER_ENHANCE	29	/* Enhanced stereo (0, 40, 60 or 80) */+#define SOUND_MIXER_LOUD	30	/* 0 or 1 */++/* Note!	Number 31 cannot be used since the sign bit is reserved */+#define SOUND_MIXER_NONE        31++#define SOUND_DEVICE_LABELS	{ \+	"Vol  ", "Bass ", "Trebl", "Synth", "Pcm  ", "Spkr ", "Line ", \+	"Mic  ", "CD   ", "Mix  ", "Pcm2 ", "Rec  ", "IGain", "OGain", \+	"Line1", "Line2", "Line3", "Digital1", "Digital2", "Digital3", \+	"PhoneIn", "PhoneOut", "Video", "Radio", "Monitor"}++#define SOUND_DEVICE_NAMES	{ \+	"vol", "bass", "treble", "synth", "pcm", "speaker", "line", \+	"mic", "cd", "mix", "pcm2", "rec", "igain", "ogain", \+	"line1", "line2", "line3", "dig1", "dig2", "dig3", \+	"phin", "phout", "video", "radio", "monitor"}++/*	Device bitmask identifiers	*/++#define SOUND_MIXER_RECSRC	0xff	/* 1 bit per recording source */+#define SOUND_MIXER_DEVMASK	0xfe	/* 1 bit per supported device */+#define SOUND_MIXER_RECMASK	0xfd	/* 1 bit per supp. recording source */+#define SOUND_MIXER_CAPS	0xfc+#define SOUND_CAP_EXCL_INPUT	0x00000001	/* Only 1 rec. src at a time */+#define SOUND_MIXER_STEREODEVS	0xfb	/* Mixer channels supporting stereo */++/*	Device mask bits	*/++#define SOUND_MASK_VOLUME	(1 << SOUND_MIXER_VOLUME)+#define SOUND_MASK_BASS		(1 << SOUND_MIXER_BASS)+#define SOUND_MASK_TREBLE	(1 << SOUND_MIXER_TREBLE)+#define SOUND_MASK_SYNTH	(1 << SOUND_MIXER_SYNTH)+#define SOUND_MASK_PCM		(1 << SOUND_MIXER_PCM)+#define SOUND_MASK_SPEAKER	(1 << SOUND_MIXER_SPEAKER)+#define SOUND_MASK_LINE		(1 << SOUND_MIXER_LINE)+#define SOUND_MASK_MIC		(1 << SOUND_MIXER_MIC)+#define SOUND_MASK_CD		(1 << SOUND_MIXER_CD)+#define SOUND_MASK_IMIX		(1 << SOUND_MIXER_IMIX)+#define SOUND_MASK_ALTPCM	(1 << SOUND_MIXER_ALTPCM)+#define SOUND_MASK_RECLEV	(1 << SOUND_MIXER_RECLEV)+#define SOUND_MASK_IGAIN	(1 << SOUND_MIXER_IGAIN)+#define SOUND_MASK_OGAIN	(1 << SOUND_MIXER_OGAIN)+#define SOUND_MASK_LINE1	(1 << SOUND_MIXER_LINE1)+#define SOUND_MASK_LINE2	(1 << SOUND_MIXER_LINE2)+#define SOUND_MASK_LINE3	(1 << SOUND_MIXER_LINE3)+#define SOUND_MASK_DIGITAL1     (1 << SOUND_MIXER_DIGITAL1)+#define SOUND_MASK_DIGITAL2     (1 << SOUND_MIXER_DIGITAL2)+#define SOUND_MASK_DIGITAL3     (1 << SOUND_MIXER_DIGITAL3)+#define SOUND_MASK_PHONEIN      (1 << SOUND_MIXER_PHONEIN)+#define SOUND_MASK_PHONEOUT     (1 << SOUND_MIXER_PHONEOUT)+#define SOUND_MASK_RADIO        (1 << SOUND_MIXER_RADIO)+#define SOUND_MASK_VIDEO        (1 << SOUND_MIXER_VIDEO)+#define SOUND_MASK_MONITOR      (1 << SOUND_MIXER_MONITOR)++/* Obsolete macros */+#define SOUND_MASK_MUTE		(1 << SOUND_MIXER_MUTE)+#define SOUND_MASK_ENHANCE	(1 << SOUND_MIXER_ENHANCE)+#define SOUND_MASK_LOUD		(1 << SOUND_MIXER_LOUD)++#define MIXER_READ(dev)		_IOR('M', dev, int)+#define SOUND_MIXER_READ_VOLUME		MIXER_READ(SOUND_MIXER_VOLUME)+#define SOUND_MIXER_READ_BASS		MIXER_READ(SOUND_MIXER_BASS)+#define SOUND_MIXER_READ_TREBLE		MIXER_READ(SOUND_MIXER_TREBLE)+#define SOUND_MIXER_READ_SYNTH		MIXER_READ(SOUND_MIXER_SYNTH)+#define SOUND_MIXER_READ_PCM		MIXER_READ(SOUND_MIXER_PCM)+#define SOUND_MIXER_READ_SPEAKER	MIXER_READ(SOUND_MIXER_SPEAKER)+#define SOUND_MIXER_READ_LINE		MIXER_READ(SOUND_MIXER_LINE)+#define SOUND_MIXER_READ_MIC		MIXER_READ(SOUND_MIXER_MIC)+#define SOUND_MIXER_READ_CD		MIXER_READ(SOUND_MIXER_CD)+#define SOUND_MIXER_READ_IMIX		MIXER_READ(SOUND_MIXER_IMIX)+#define SOUND_MIXER_READ_ALTPCM		MIXER_READ(SOUND_MIXER_ALTPCM)+#define SOUND_MIXER_READ_RECLEV		MIXER_READ(SOUND_MIXER_RECLEV)+#define SOUND_MIXER_READ_IGAIN		MIXER_READ(SOUND_MIXER_IGAIN)+#define SOUND_MIXER_READ_OGAIN		MIXER_READ(SOUND_MIXER_OGAIN)+#define SOUND_MIXER_READ_LINE1		MIXER_READ(SOUND_MIXER_LINE1)+#define SOUND_MIXER_READ_LINE2		MIXER_READ(SOUND_MIXER_LINE2)+#define SOUND_MIXER_READ_LINE3		MIXER_READ(SOUND_MIXER_LINE3)+#define SOUND_MIXER_READ_DIGITAL1	MIXER_READ(SOUND_MIXER_DIGITAL1)+#define SOUND_MIXER_READ_DIGITAL2	MIXER_READ(SOUND_MIXER_DIGITAL2)+#define SOUND_MIXER_READ_DIGITAL3	MIXER_READ(SOUND_MIXER_DIGITAL3)+#define SOUND_MIXER_READ_PHONEIN      	MIXER_READ(SOUND_MIXER_PHONEIN)+#define SOUND_MIXER_READ_PHONEOUT	MIXER_READ(SOUND_MIXER_PHONEOUT)+#define SOUND_MIXER_READ_RADIO		MIXER_READ(SOUND_MIXER_RADIO)+#define SOUND_MIXER_READ_VIDEO		MIXER_READ(SOUND_MIXER_VIDEO)+#define SOUND_MIXER_READ_MONITOR	MIXER_READ(SOUND_MIXER_MONITOR)++/* Obsolete macros */+#define SOUND_MIXER_READ_MUTE		MIXER_READ(SOUND_MIXER_MUTE)+#define SOUND_MIXER_READ_ENHANCE	MIXER_READ(SOUND_MIXER_ENHANCE)+#define SOUND_MIXER_READ_LOUD		MIXER_READ(SOUND_MIXER_LOUD)++#define SOUND_MIXER_READ_RECSRC		MIXER_READ(SOUND_MIXER_RECSRC)+#define SOUND_MIXER_READ_DEVMASK	MIXER_READ(SOUND_MIXER_DEVMASK)+#define SOUND_MIXER_READ_RECMASK	MIXER_READ(SOUND_MIXER_RECMASK)+#define SOUND_MIXER_READ_STEREODEVS	MIXER_READ(SOUND_MIXER_STEREODEVS)+#define SOUND_MIXER_READ_CAPS		MIXER_READ(SOUND_MIXER_CAPS)++#define MIXER_WRITE(dev)		_IOWR('M', dev, int)+#define SOUND_MIXER_WRITE_VOLUME	MIXER_WRITE(SOUND_MIXER_VOLUME)+#define SOUND_MIXER_WRITE_BASS		MIXER_WRITE(SOUND_MIXER_BASS)+#define SOUND_MIXER_WRITE_TREBLE	MIXER_WRITE(SOUND_MIXER_TREBLE)+#define SOUND_MIXER_WRITE_SYNTH		MIXER_WRITE(SOUND_MIXER_SYNTH)+#define SOUND_MIXER_WRITE_PCM		MIXER_WRITE(SOUND_MIXER_PCM)+#define SOUND_MIXER_WRITE_SPEAKER	MIXER_WRITE(SOUND_MIXER_SPEAKER)+#define SOUND_MIXER_WRITE_LINE		MIXER_WRITE(SOUND_MIXER_LINE)+#define SOUND_MIXER_WRITE_MIC		MIXER_WRITE(SOUND_MIXER_MIC)+#define SOUND_MIXER_WRITE_CD		MIXER_WRITE(SOUND_MIXER_CD)+#define SOUND_MIXER_WRITE_IMIX		MIXER_WRITE(SOUND_MIXER_IMIX)+#define SOUND_MIXER_WRITE_ALTPCM	MIXER_WRITE(SOUND_MIXER_ALTPCM)+#define SOUND_MIXER_WRITE_RECLEV	MIXER_WRITE(SOUND_MIXER_RECLEV)+#define SOUND_MIXER_WRITE_IGAIN		MIXER_WRITE(SOUND_MIXER_IGAIN)+#define SOUND_MIXER_WRITE_OGAIN		MIXER_WRITE(SOUND_MIXER_OGAIN)+#define SOUND_MIXER_WRITE_LINE1		MIXER_WRITE(SOUND_MIXER_LINE1)+#define SOUND_MIXER_WRITE_LINE2		MIXER_WRITE(SOUND_MIXER_LINE2)+#define SOUND_MIXER_WRITE_LINE3		MIXER_WRITE(SOUND_MIXER_LINE3)+#define SOUND_MIXER_WRITE_DIGITAL1	MIXER_WRITE(SOUND_MIXER_DIGITAL1)+#define SOUND_MIXER_WRITE_DIGITAL2	MIXER_WRITE(SOUND_MIXER_DIGITAL2)+#define SOUND_MIXER_WRITE_DIGITAL3	MIXER_WRITE(SOUND_MIXER_DIGITAL3)+#define SOUND_MIXER_WRITE_PHONEIN      	MIXER_WRITE(SOUND_MIXER_PHONEIN)+#define SOUND_MIXER_WRITE_PHONEOUT	MIXER_WRITE(SOUND_MIXER_PHONEOUT)+#define SOUND_MIXER_WRITE_RADIO		MIXER_WRITE(SOUND_MIXER_RADIO)+#define SOUND_MIXER_WRITE_VIDEO		MIXER_WRITE(SOUND_MIXER_VIDEO)+#define SOUND_MIXER_WRITE_MONITOR	MIXER_WRITE(SOUND_MIXER_MONITOR)++#define SOUND_MIXER_WRITE_MUTE		MIXER_WRITE(SOUND_MIXER_MUTE)+#define SOUND_MIXER_WRITE_ENHANCE	MIXER_WRITE(SOUND_MIXER_ENHANCE)+#define SOUND_MIXER_WRITE_LOUD		MIXER_WRITE(SOUND_MIXER_LOUD)++#define SOUND_MIXER_WRITE_RECSRC	MIXER_WRITE(SOUND_MIXER_RECSRC)++typedef struct mixer_info {+  char id[16];+  char name[32];+  int  modify_counter;+  int fillers[10];+} mixer_info;++#define SOUND_MIXER_INFO		_IOR('M', 101, mixer_info)++#define LEFT_CHN	0+#define RIGHT_CHN	1++/*+ * Level 2 event types for /dev/sequencer+ */++/*+ * The 4 most significant bits of byte 0 specify the class of+ * the event:+ *+ *	0x8X = system level events,+ *	0x9X = device/port specific events, event[1] = device/port,+ *		The last 4 bits give the subtype:+ *			0x02	= Channel event (event[3] = chn).+ *			0x01	= note event (event[4] = note).+ *			(0x01 is not used alone but always with bit 0x02).+ *	       event[2] = MIDI message code (0x80=note off etc.)+ *+ */++#define EV_SEQ_LOCAL		0x80+#define EV_TIMING		0x81+#define EV_CHN_COMMON		0x92+#define EV_CHN_VOICE		0x93+#define EV_SYSEX		0x94+/*+ * Event types 200 to 220 are reserved for application use.+ * These numbers will not be used by the driver.+ */++/*+ * Events for event type EV_CHN_VOICE+ */++#define MIDI_NOTEOFF		0x80+#define MIDI_NOTEON		0x90+#define MIDI_KEY_PRESSURE	0xA0++/*+ * Events for event type EV_CHN_COMMON+ */++#define MIDI_CTL_CHANGE		0xB0+#define MIDI_PGM_CHANGE		0xC0+#define MIDI_CHN_PRESSURE	0xD0+#define MIDI_PITCH_BEND		0xE0++#define MIDI_SYSTEM_PREFIX	0xF0++/*+ * Timer event types+ */+#define TMR_WAIT_REL		1	/* Time relative to the prev time */+#define TMR_WAIT_ABS		2	/* Absolute time since TMR_START */+#define TMR_STOP		3+#define TMR_START		4+#define TMR_CONTINUE		5+#define TMR_TEMPO		6+#define TMR_ECHO		8+#define TMR_CLOCK		9	/* MIDI clock */+#define TMR_SPP			10	/* Song position pointer */+#define TMR_TIMESIG		11	/* Time signature */++/*+ *	Local event types+ */+#define LOCL_STARTAUDIO		1++#if (!defined(_KERNEL) && !defined(INKERNEL)) || defined(USE_SEQ_MACROS)+/*+ *	Some convenience macros to simplify programming of the+ *	/dev/sequencer interface+ *+ *	These macros define the API which should be used when possible.+ */++#ifndef USE_SIMPLE_MACROS+void seqbuf_dump(void);	/* This function must be provided by programs */++/* Sample seqbuf_dump() implementation:+ *+ *	SEQ_DEFINEBUF (2048);	-- Defines a buffer for 2048 bytes+ *+ *	int seqfd;		-- The file descriptor for /dev/sequencer.+ *+ *	void+ *	seqbuf_dump ()+ *	{+ *	  if (_seqbufptr)+ *	    if (write (seqfd, _seqbuf, _seqbufptr) == -1)+ *	      {+ *		perror ("write /dev/sequencer");+ *		exit (-1);+ *	      }+ *	  _seqbufptr = 0;+ *	}+ */++#define SEQ_DEFINEBUF(len)		\+	u_char _seqbuf[len]; int _seqbuflen = len;int _seqbufptr = 0+#define SEQ_USE_EXTBUF()		\+	extern u_char _seqbuf[]; \+	extern int _seqbuflen;extern int _seqbufptr+#define SEQ_DECLAREBUF()		SEQ_USE_EXTBUF()+#define SEQ_PM_DEFINES			struct patmgr_info _pm_info+#define _SEQ_NEEDBUF(len)		\+	if ((_seqbufptr+(len)) > _seqbuflen) \+		seqbuf_dump()+#define _SEQ_ADVBUF(len)		_seqbufptr += len+#define SEQ_DUMPBUF			seqbuf_dump+#else+/*+ * This variation of the sequencer macros is used just to format one event+ * using fixed buffer.+ *+ * The program using the macro library must define the following macros before+ * using this library.+ *+ * #define _seqbuf 		 name of the buffer (u_char[])+ * #define _SEQ_ADVBUF(len)	 If the applic needs to know the exact+ *				 size of the event, this macro can be used.+ *				 Otherwise this must be defined as empty.+ * #define _seqbufptr		 Define the name of index variable or 0 if+ *				 not required.+ */+#define _SEQ_NEEDBUF(len)	/* empty */+#endif++#define PM_LOAD_PATCH(dev, bank, pgm)	\+	(SEQ_DUMPBUF(), _pm_info.command = _PM_LOAD_PATCH, \+	_pm_info.device=dev, _pm_info.data.data8[0]=pgm, \+	_pm_info.parm1 = bank, _pm_info.parm2 = 1, \+	ioctl(seqfd, SNDCTL_PMGR_ACCESS, &_pm_info))+#define PM_LOAD_PATCHES(dev, bank, pgm) \+	(SEQ_DUMPBUF(), _pm_info.command = _PM_LOAD_PATCH, \+	_pm_info.device=dev, bcopy( pgm, _pm_info.data.data8,  128), \+	_pm_info.parm1 = bank, _pm_info.parm2 = 128, \+	ioctl(seqfd, SNDCTL_PMGR_ACCESS, &_pm_info))++#define SEQ_VOLUME_MODE(dev, mode)	{ \+	_SEQ_NEEDBUF(8);\+	_seqbuf[_seqbufptr] = SEQ_EXTENDED;\+	_seqbuf[_seqbufptr+1] = SEQ_VOLMODE;\+	_seqbuf[_seqbufptr+2] = (dev);\+	_seqbuf[_seqbufptr+3] = (mode);\+	_seqbuf[_seqbufptr+4] = 0;\+	_seqbuf[_seqbufptr+5] = 0;\+	_seqbuf[_seqbufptr+6] = 0;\+	_seqbuf[_seqbufptr+7] = 0;\+	_SEQ_ADVBUF(8);}++/*+ * Midi voice messages+ */++#define _CHN_VOICE(dev, event, chn, note, parm)  { \+	_SEQ_NEEDBUF(8);\+	_seqbuf[_seqbufptr] = EV_CHN_VOICE;\+	_seqbuf[_seqbufptr+1] = (dev);\+	_seqbuf[_seqbufptr+2] = (event);\+	_seqbuf[_seqbufptr+3] = (chn);\+	_seqbuf[_seqbufptr+4] = (note);\+	_seqbuf[_seqbufptr+5] = (parm);\+	_seqbuf[_seqbufptr+6] = (0);\+	_seqbuf[_seqbufptr+7] = 0;\+	_SEQ_ADVBUF(8);}++#define SEQ_START_NOTE(dev, chn, note, vol) \+		_CHN_VOICE(dev, MIDI_NOTEON, chn, note, vol)++#define SEQ_STOP_NOTE(dev, chn, note, vol) \+		_CHN_VOICE(dev, MIDI_NOTEOFF, chn, note, vol)++#define SEQ_KEY_PRESSURE(dev, chn, note, pressure) \+		_CHN_VOICE(dev, MIDI_KEY_PRESSURE, chn, note, pressure)++/*+ * Midi channel messages+ */++#define _CHN_COMMON(dev, event, chn, p1, p2, w14) { \+	_SEQ_NEEDBUF(8);\+	_seqbuf[_seqbufptr] = EV_CHN_COMMON;\+	_seqbuf[_seqbufptr+1] = (dev);\+	_seqbuf[_seqbufptr+2] = (event);\+	_seqbuf[_seqbufptr+3] = (chn);\+	_seqbuf[_seqbufptr+4] = (p1);\+	_seqbuf[_seqbufptr+5] = (p2);\+	*(short *)&_seqbuf[_seqbufptr+6] = (w14);\+	_SEQ_ADVBUF(8);}+/*+ * SEQ_SYSEX permits sending of sysex messages. (It may look that it permits+ * sending any MIDI bytes but it's absolutely not possible. Trying to do+ * so _will_ cause problems with MPU401 intelligent mode).+ *+ * Sysex messages are sent in blocks of 1 to 6 bytes. Longer messages must be+ * sent by calling SEQ_SYSEX() several times (there must be no other events+ * between them). First sysex fragment must have 0xf0 in the first byte+ * and the last byte (buf[len-1] of the last fragment must be 0xf7. No byte+ * between these sysex start and end markers cannot be larger than 0x7f. Also+ * lengths of each fragments (except the last one) must be 6.+ *+ * Breaking the above rules may work with some MIDI ports but is likely to+ * cause fatal problems with some other devices (such as MPU401).+ */+#define SEQ_SYSEX(dev, buf, len) { \+	int i, l=(len); if (l>6)l=6;\+	_SEQ_NEEDBUF(8);\+	_seqbuf[_seqbufptr] = EV_SYSEX;\+	for(i=0;i<l;i++)_seqbuf[_seqbufptr+i+1] = (buf)[i];\+	for(i=l;i<6;i++)_seqbuf[_seqbufptr+i+1] = 0xff;\+	_SEQ_ADVBUF(8);}++#define SEQ_CHN_PRESSURE(dev, chn, pressure) \+	_CHN_COMMON(dev, MIDI_CHN_PRESSURE, chn, pressure, 0, 0)++#define SEQ_SET_PATCH(dev, chn, patch) \+	_CHN_COMMON(dev, MIDI_PGM_CHANGE, chn, patch, 0, 0)++#define SEQ_CONTROL(dev, chn, controller, value) \+	_CHN_COMMON(dev, MIDI_CTL_CHANGE, chn, controller, 0, value)++#define SEQ_BENDER(dev, chn, value) \+	_CHN_COMMON(dev, MIDI_PITCH_BEND, chn, 0, 0, value)+++#define SEQ_V2_X_CONTROL(dev, voice, controller, value)	{ \+	_SEQ_NEEDBUF(8);\+	_seqbuf[_seqbufptr] = SEQ_EXTENDED;\+	_seqbuf[_seqbufptr+1] = SEQ_CONTROLLER;\+	_seqbuf[_seqbufptr+2] = (dev);\+	_seqbuf[_seqbufptr+3] = (voice);\+	_seqbuf[_seqbufptr+4] = (controller);\+	*(short *)&_seqbuf[_seqbufptr+5] = (value);\+	_seqbuf[_seqbufptr+7] = 0;\+	_SEQ_ADVBUF(8);}++/*+ * The following 5 macros are incorrectly implemented and obsolete.+ * Use SEQ_BENDER and SEQ_CONTROL (with proper controller) instead.+ */++#define SEQ_PITCHBEND(dev, voice, value) \+	SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER, value)+#define SEQ_BENDER_RANGE(dev, voice, value) \+	SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER_RANGE, value)+#define SEQ_EXPRESSION(dev, voice, value) \+	SEQ_CONTROL(dev, voice, CTL_EXPRESSION, value*128)+#define SEQ_MAIN_VOLUME(dev, voice, value) \+	SEQ_CONTROL(dev, voice, CTL_MAIN_VOLUME, (value*16383)/100)+#define SEQ_PANNING(dev, voice, pos) \+	SEQ_CONTROL(dev, voice, CTL_PAN, (pos+128) / 2)++/*+ * Timing and syncronization macros+ */++#define _TIMER_EVENT(ev, parm)		{ \+	_SEQ_NEEDBUF(8);\+	_seqbuf[_seqbufptr+0] = EV_TIMING; \+	_seqbuf[_seqbufptr+1] = (ev); \+	_seqbuf[_seqbufptr+2] = 0;\+	_seqbuf[_seqbufptr+3] = 0;\+	*(u_int *)&_seqbuf[_seqbufptr+4] = (parm); \+	_SEQ_ADVBUF(8); \+	}++#define SEQ_START_TIMER()		_TIMER_EVENT(TMR_START, 0)+#define SEQ_STOP_TIMER()		_TIMER_EVENT(TMR_STOP, 0)+#define SEQ_CONTINUE_TIMER()		_TIMER_EVENT(TMR_CONTINUE, 0)+#define SEQ_WAIT_TIME(ticks)		_TIMER_EVENT(TMR_WAIT_ABS, ticks)+#define SEQ_DELTA_TIME(ticks)		_TIMER_EVENT(TMR_WAIT_REL, ticks)+#define SEQ_ECHO_BACK(key)		_TIMER_EVENT(TMR_ECHO, key)+#define SEQ_SET_TEMPO(value)		_TIMER_EVENT(TMR_TEMPO, value)+#define SEQ_SONGPOS(pos)		_TIMER_EVENT(TMR_SPP, pos)+#define SEQ_TIME_SIGNATURE(sig)		_TIMER_EVENT(TMR_TIMESIG, sig)++/*+ * Local control events+ */++#define _LOCAL_EVENT(ev, parm)		{ \+	_SEQ_NEEDBUF(8);\+	_seqbuf[_seqbufptr+0] = EV_SEQ_LOCAL; \+	_seqbuf[_seqbufptr+1] = (ev); \+	_seqbuf[_seqbufptr+2] = 0;\+	_seqbuf[_seqbufptr+3] = 0;\+	*(u_int *)&_seqbuf[_seqbufptr+4] = (parm); \+	_SEQ_ADVBUF(8); \+	}++#define SEQ_PLAYAUDIO(devmask)		_LOCAL_EVENT(LOCL_STARTAUDIO, devmask)+/*+ * Events for the level 1 interface only+ */++#define SEQ_MIDIOUT(device, byte)	{ \+	_SEQ_NEEDBUF(4);\+	_seqbuf[_seqbufptr] = SEQ_MIDIPUTC;\+	_seqbuf[_seqbufptr+1] = (byte);\+	_seqbuf[_seqbufptr+2] = (device);\+	_seqbuf[_seqbufptr+3] = 0;\+	_SEQ_ADVBUF(4);}++/*+ * Patch loading.+ */+#define SEQ_WRPATCH(patchx, len)	{ \+	if (_seqbufptr) seqbuf_dump(); \+	if (write(seqfd, (char*)(patchx), len)==-1) \+	   perror("Write patch: /dev/sequencer"); \+	}++#define SEQ_WRPATCH2(patchx, len)	\+	( seqbuf_dump(), write(seqfd, (char*)(patchx), len) )++#endif++/*+ * Here I have moved all the aliases for ioctl names.+ */++#define SNDCTL_DSP_SAMPLESIZE	SNDCTL_DSP_SETFMT+#define SOUND_PCM_WRITE_BITS	SNDCTL_DSP_SETFMT+#define SOUND_PCM_SETFMT	SNDCTL_DSP_SETFMT++#define SOUND_PCM_WRITE_RATE	SNDCTL_DSP_SPEED+#define SOUND_PCM_POST		SNDCTL_DSP_POST+#define SOUND_PCM_RESET		SNDCTL_DSP_RESET+#define SOUND_PCM_SYNC		SNDCTL_DSP_SYNC+#define SOUND_PCM_SUBDIVIDE	SNDCTL_DSP_SUBDIVIDE+#define SOUND_PCM_SETFRAGMENT	SNDCTL_DSP_SETFRAGMENT+#define SOUND_PCM_GETFMTS	SNDCTL_DSP_GETFMTS+#define SOUND_PCM_GETOSPACE	SNDCTL_DSP_GETOSPACE+#define SOUND_PCM_GETISPACE	SNDCTL_DSP_GETISPACE+#define SOUND_PCM_NONBLOCK	SNDCTL_DSP_NONBLOCK+#define SOUND_PCM_GETCAPS	SNDCTL_DSP_GETCAPS+#define SOUND_PCM_GETTRIGGER	SNDCTL_DSP_GETTRIGGER+#define SOUND_PCM_SETTRIGGER	SNDCTL_DSP_SETTRIGGER+#define SOUND_PCM_SETSYNCRO	SNDCTL_DSP_SETSYNCRO+#define SOUND_PCM_GETIPTR	SNDCTL_DSP_GETIPTR+#define SOUND_PCM_GETOPTR	SNDCTL_DSP_GETOPTR+#define SOUND_PCM_MAPINBUF	SNDCTL_DSP_MAPINBUF+#define SOUND_PCM_MAPOUTBUF	SNDCTL_DSP_MAPOUTBUF++/***********************************************************************/++/**+ * XXX OSSv4 defines -- some bits taken straight out of the new+ * sys/soundcard.h bundled with recent OSS releases.+ *+ * NB:  These macros and structures will be reorganized and inserted+ * 	in appropriate places throughout this file once the code begins+ * 	to take shape.+ *+ * @todo reorganize layout more like the 4Front version+ * @todo ask about maintaining __SIOWR vs. _IOWR ioctl cmd defines+ */++/**+ * @note The @c OSSV4_EXPERIMENT macro is meant to wrap new development code+ * in the sound system relevant to adopting 4Front's OSSv4 specification.+ * Users should not enable this!  Really!+ */+#if 0+# define OSSV4_EXPERIMENT 1+#else+# undef OSSV4_EXPERIMENT+#endif++#ifdef SOUND_VERSION+# undef SOUND_VERSION+# define SOUND_VERSION	0x040000+#endif	/* !SOUND_VERSION */++#define OSS_LONGNAME_SIZE	64+#define OSS_LABEL_SIZE		16+#define OSS_DEVNODE_SIZE        32+typedef char oss_longname_t[OSS_LONGNAME_SIZE];+typedef char oss_label_t[OSS_LABEL_SIZE];+typedef char oss_devnode_t[OSS_DEVNODE_SIZE];++typedef struct audio_errinfo+{+	int		play_underruns;+	int		rec_overruns;+	unsigned int	play_ptradjust;+	unsigned int	rec_ptradjust;+	int		play_errorcount;+	int		rec_errorcount;+	int		play_lasterror;+	int		rec_lasterror;+	long		play_errorparm;+	long		rec_errorparm;+	int		filler[16];+} audio_errinfo;++#define SNDCTL_DSP_GETPLAYVOL           _IOR ('P', 24, int)+#define SNDCTL_DSP_SETPLAYVOL           _IOWR('P', 24, int)+#define SNDCTL_DSP_GETERROR             _IOR ('P', 25, audio_errinfo)+++/*+ ****************************************************************************+ * Sync groups for audio devices+ */+typedef struct oss_syncgroup+{+  int id;+  int mode;+  int filler[16];+} oss_syncgroup;++#define SNDCTL_DSP_SYNCGROUP            _IOWR('P', 28, oss_syncgroup)+#define SNDCTL_DSP_SYNCSTART            _IOW ('P', 29, int)++/*+ **************************************************************************+ * "cooked" mode enables software based conversions for sample rate, sample+ * format (bits) and number of channels (mono/stereo). These conversions are+ * required with some devices that support only one sample rate or just stereo+ * to let the applications to use other formats. The cooked mode is enabled by+ * default. However it's necessary to disable this mode when mmap() is used or+ * when very deterministic timing is required. SNDCTL_DSP_COOKEDMODE is an+ * optional call introduced in OSS 3.9.6f. It's _error return must be ignored_+ * since normally this call will return erno=EINVAL.+ *+ * SNDCTL_DSP_COOKEDMODE must be called immediately after open before doing+ * anything else. Otherwise the call will not have any effect.+ */+#define SNDCTL_DSP_COOKEDMODE           _IOW ('P', 30, int)++/*+ **************************************************************************+ * SNDCTL_DSP_SILENCE and SNDCTL_DSP_SKIP are new calls in OSS 3.99.0+ * that can be used to implement pause/continue during playback (no effect+ * on recording).+ */+#define SNDCTL_DSP_SILENCE              _IO  ('P', 31)+#define SNDCTL_DSP_SKIP                 _IO  ('P', 32)++/*+ ****************************************************************************+ * Abort transfer (reset) functions for input and output+ */+#define SNDCTL_DSP_HALT_INPUT		_IO  ('P', 33)+#define SNDCTL_DSP_RESET_INPUT	SNDCTL_DSP_HALT_INPUT	/* Old name */+#define SNDCTL_DSP_HALT_OUTPUT		_IO  ('P', 34)+#define SNDCTL_DSP_RESET_OUTPUT	SNDCTL_DSP_HALT_OUTPUT	/* Old name */++/*+ ****************************************************************************+ * Low water level control+ */+#define SNDCTL_DSP_LOW_WATER		_IOW ('P', 34, int)++/** @todo Get rid of OSS_NO_LONG_LONG references? */++/*+ ****************************************************************************+ * 64 bit pointer support. Only available in environments that support+ * the 64 bit (long long) integer type.+ */+#ifndef OSS_NO_LONG_LONG+typedef struct+{+  long long samples;+  int fifo_samples;+  int filler[32];		/* For future use */+} oss_count_t;++#define SNDCTL_DSP_CURRENT_IPTR		_IOR ('P', 35, oss_count_t)+#define SNDCTL_DSP_CURRENT_OPTR		_IOR ('P', 36, oss_count_t)+#endif++/*+ ****************************************************************************+ * Interface for selecting recording sources and playback output routings.+ */+#define SNDCTL_DSP_GET_RECSRC_NAMES     _IOR ('P', 37, oss_mixer_enuminfo)+#define SNDCTL_DSP_GET_RECSRC           _IOR ('P', 38, int)+#define SNDCTL_DSP_SET_RECSRC           _IOWR('P', 38, int)++#define SNDCTL_DSP_GET_PLAYTGT_NAMES    _IOR ('P', 39, oss_mixer_enuminfo)+#define SNDCTL_DSP_GET_PLAYTGT          _IOR ('P', 40, int)+#define SNDCTL_DSP_SET_PLAYTGT          _IOWR('P', 40, int)+#define SNDCTL_DSP_GETRECVOL            _IOR ('P', 41, int)+#define SNDCTL_DSP_SETRECVOL            _IOWR('P', 41, int)++/*+ ***************************************************************************+ * Some calls for setting the channel assignment with multi channel devices+ * (see the manual for details).                                                 */+#define SNDCTL_DSP_GET_CHNORDER         _IOR ('P', 42, unsigned long long)+#define SNDCTL_DSP_SET_CHNORDER         _IOWR('P', 42, unsigned long long)+#       define CHID_UNDEF       0+#       define CHID_L           1                                               #       define CHID_R           2+#       define CHID_C           3+#       define CHID_LFE         4+#       define CHID_LS          5+#       define CHID_RS          6+#       define CHID_LR          7+#       define CHID_RR          8+#define CHNORDER_UNDEF          0x0000000000000000ULL+#define CHNORDER_NORMAL         0x0000000087654321ULL++#define MAX_PEAK_CHANNELS	128+typedef unsigned short oss_peaks_t[MAX_PEAK_CHANNELS];+#define SNDCTL_DSP_GETIPEAKS		_IOR('P', 43, oss_peaks_t)+#define SNDCTL_DSP_GETOPEAKS		_IOR('P', 44, oss_peaks_t)+#define SNDCTL_DSP_POLICY               _IOW('P', 45, int)    /* See the manual */++/*+ * OSS_SYSIFO is obsolete. Use SNDCTL_SYSINFO insteads.+ */+#define OSS_GETVERSION                  _IOR ('M', 118, int)++/**+ * @brief	Argument for SNDCTL_SYSINFO ioctl.+ *+ * For use w/ the SNDCTL_SYSINFO ioctl available on audio (/dev/dsp*),+ * mixer, and MIDI devices.+ */+typedef struct oss_sysinfo+{+	char	product[32];	/* For example OSS/Free, OSS/Linux or+				   OSS/Solaris */+	char	version[32];	/* For example 4.0a */+	int	versionnum;	/* See OSS_GETVERSION */+	char	options[128];	/* Reserved */++	int	numaudios;	/* # of audio/dsp devices */+	int	openedaudio[8];	/* Bit mask telling which audio devices+				   are busy */++	int	numsynths;	/* # of availavle synth devices */+	int	nummidis;	/* # of available MIDI ports */+	int	numtimers;	/* # of available timer devices */+	int	nummixers;	/* # of mixer devices */++	int	openedmidi[8];	/* Bit mask telling which midi devices+				   are busy */+	int	numcards;	/* Number of sound cards in the system */+	int	filler[241];	/* For future expansion (set to -1) */+} oss_sysinfo;++typedef struct oss_mixext+{+  int dev;			/* Mixer device number */+  int ctrl;			/* Controller number */+  int type;			/* Entry type */+#	define MIXT_DEVROOT	 0	/* Device root entry */+#	define MIXT_GROUP	 1	/* Controller group */+#	define MIXT_ONOFF	 2	/* OFF (0) or ON (1) */+#	define MIXT_ENUM	 3	/* Enumerated (0 to maxvalue) */+#	define MIXT_MONOSLIDER	 4	/* Mono slider (0 to 100) */+#	define MIXT_STEREOSLIDER 5	/* Stereo slider (dual 0 to 100) */+#	define MIXT_MESSAGE	 6	/* (Readable) textual message */+#	define MIXT_MONOVU	 7	/* VU meter value (mono) */+#	define MIXT_STEREOVU	 8	/* VU meter value (stereo) */+#	define MIXT_MONOPEAK	 9	/* VU meter peak value (mono) */+#	define MIXT_STEREOPEAK	10	/* VU meter peak value (stereo) */+#	define MIXT_RADIOGROUP	11	/* Radio button group */+#	define MIXT_MARKER	12	/* Separator between normal and extension entries */+#	define MIXT_VALUE	13	/* Decimal value entry */+#	define MIXT_HEXVALUE	14	/* Hexadecimal value entry */+#	define MIXT_MONODB	15	/* Mono atten. slider (0 to -144) */+#	define MIXT_STEREODB	16	/* Stereo atten. slider (dual 0 to -144) */+#	define MIXT_SLIDER	17	/* Slider (mono) with full integer range */+#	define MIXT_3D		18++  /* Possible value range (minvalue to maxvalue) */+  /* Note that maxvalue may also be smaller than minvalue */+  int maxvalue;+  int minvalue;++  int flags;+#	define MIXF_READABLE	0x00000001	/* Has readable value */+#	define MIXF_WRITEABLE	0x00000002	/* Has writeable value */+#	define MIXF_POLL	0x00000004	/* May change itself */+#	define MIXF_HZ		0x00000008	/* Herz scale */+#	define MIXF_STRING	0x00000010	/* Use dynamic extensions for value */+#	define MIXF_DYNAMIC	0x00000010	/* Supports dynamic extensions */+#	define MIXF_OKFAIL	0x00000020	/* Interpret value as 1=OK, 0=FAIL */+#	define MIXF_FLAT	0x00000040	/* Flat vertical space requirements */+#	define MIXF_LEGACY	0x00000080	/* Legacy mixer control group */+  char id[16];			/* Mnemonic ID (mainly for internal use) */+  int parent;			/* Entry# of parent (group) node (-1 if root) */++  int dummy;			/* Internal use */++  int timestamp;++  char data[64];		/* Misc data (entry type dependent) */+  unsigned char enum_present[32];	/* Mask of allowed enum values */+  int control_no;		/* SOUND_MIXER_VOLUME..SOUND_MIXER_MIDI */+  /* (-1 means not indicated) */++/*+ * The desc field is reserved for internal purposes of OSS. It should not be + * used by applications.+ */+  unsigned int desc;+#define MIXEXT_SCOPE_MASK			0x0000003f+#define MIXEXT_SCOPE_OTHER			0x00000000+#define MIXEXT_SCOPE_INPUT			0x00000001+#define MIXEXT_SCOPE_OUTPUT			0x00000002+#define MIXEXT_SCOPE_MONITOR			0x00000003+#define MIXEXT_SCOPE_RECSWITCH			0x00000004++  char extname[32];+  int update_counter;+  int filler[7];+} oss_mixext;++typedef struct oss_mixext_root+{+  char id[16];+  char name[48];+} oss_mixext_root;++typedef struct oss_mixer_value+{+  int dev;+  int ctrl;+  int value;+  int flags;			/* Reserved for future use. Initialize to 0 */+  int timestamp;		/* Must be set to oss_mixext.timestamp */+  int filler[8];		/* Reserved for future use. Initialize to 0 */+} oss_mixer_value;++#define OSS_ENUM_MAXVALUE       255+typedef struct oss_mixer_enuminfo+{+	int	dev;+	int	ctrl;+	int	nvalues;+	int	version;                  /* Read the manual */+	short	strindex[OSS_ENUM_MAXVALUE];+	char	strings[3000];+} oss_mixer_enuminfo;++#define OPEN_READ       PCM_ENABLE_INPUT+#define OPEN_WRITE      PCM_ENABLE_OUTPUT+#define OPEN_READWRITE  (OPEN_READ|OPEN_WRITE)++/**+ * @brief	Argument for SNDCTL_AUDIOINFO ioctl.+ *+ * For use w/ the SNDCTL_AUDIOINFO ioctl available on audio (/dev/dsp*)+ * devices.+ */+typedef struct oss_audioinfo+{+	int	dev;		/* Audio device number */+	char	name[64];+	int	busy;		/* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */+	int	pid;+	int	caps;		/* DSP_CAP_INPUT, DSP_CAP_OUTPUT */+	int	iformats;+	int	oformats;+	int	magic;		/* Reserved for internal use */+	char 	cmd[64];	/* Command using the device (if known) */+	int	card_number;+	int	port_number;+	int	mixer_dev;+	int	real_device;	/* Obsolete field. Replaced by devnode */+	int	enabled;	/* 1=enabled, 0=device not ready at this+				   moment */+	int	flags;		/* For internal use only - no practical+				   meaning */+	int	min_rate;	/* Sample rate limits */+	int	max_rate;+	int	min_channels;	/* Number of channels supported */+	int	max_channels;+	int	binding;	/* DSP_BIND_FRONT, etc. 0 means undefined */+	int	rate_source;+	char	handle[32];+	#define OSS_MAX_SAMPLE_RATES	20	/* Cannot be changed  */+	unsigned int nrates;+	unsigned int rates[OSS_MAX_SAMPLE_RATES]; /* Please read the manual before using these */+	oss_longname_t	song_name;	/* Song name (if given) */+	oss_label_t	label;		/* Device label (if given) */+	int		latency;	/* In usecs, -1=unknown */+	oss_devnode_t	devnode;	/* Device special file name (inside+					   /dev) */+	int filler[186];+} oss_audioinfo;++typedef struct oss_mixerinfo+{+  int dev;+  char id[16];+  char name[32];+  int modify_counter;+  int card_number;+  int port_number;+  char handle[32];+  int magic;			/* Reserved */+  int enabled;			/* Reserved */+  int caps;+#define MIXER_CAP_VIRTUAL				0x00000001+  int flags;			/* Reserved */+  int nrext;+  /*+   * The priority field can be used to select the default (motherboard)+   * mixer device. The mixer with the highest priority is the+   * most preferred one. -2 or less means that this device cannot be used+   * as the default mixer.+   */+  int priority;+  int filler[254];		/* Reserved */+} oss_mixerinfo;++typedef struct oss_midi_info+{+  int dev;			/* Midi device number */+  char name[64];+  int busy;			/* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */+  int pid;+  char cmd[64];			/* Command using the device (if known) */+  int caps;+#define MIDI_CAP_MPU401		0x00000001	/**** OBSOLETE ****/+#define MIDI_CAP_INPUT		0x00000002+#define MIDI_CAP_OUTPUT		0x00000004+#define MIDI_CAP_INOUT		(MIDI_CAP_INPUT|MIDI_CAP_OUTPUT)+#define MIDI_CAP_VIRTUAL	0x00000008	/* Pseudo device */+#define MIDI_CAP_MTCINPUT	0x00000010	/* Supports SNDCTL_MIDI_MTCINPUT */+#define MIDI_CAP_CLIENT		0x00000020	/* Virtual client side device */+#define MIDI_CAP_SERVER		0x00000040	/* Virtual server side device */+#define MIDI_CAP_INTERNAL	0x00000080	/* Internal (synth) device */+#define MIDI_CAP_EXTERNAL	0x00000100	/* external (MIDI port) device */+#define MIDI_CAP_PTOP		0x00000200	/* Point to point link to one device */+#define MIDI_CAP_MTC		0x00000400	/* MTC/SMPTE (control) device */+  int magic;			/* Reserved for internal use */+  int card_number;+  int port_number;+  int enabled;			/* 1=enabled, 0=device not ready at this moment */+  int flags;			/* For internal use only - no practical meaning */+  char handle[32];+  oss_longname_t song_name;	/* Song name (if known) */+  oss_label_t label;		/* Device label (if given) */+  int latency;			/* In usecs, -1=unknown */+  int filler[244];+} oss_midi_info;++typedef struct oss_card_info+{+  int card;+  char shortname[16];+  char longname[128];+  int flags;+  int filler[256];+} oss_card_info;++#define SNDCTL_SYSINFO          _IOR ('X', 1, oss_sysinfo)+#define OSS_SYSINFO             SNDCTL_SYSINFO /* Old name */++#define SNDCTL_MIX_NRMIX	_IOR ('X', 2, int)+#define SNDCTL_MIX_NREXT	_IOWR('X', 3, int)+#define SNDCTL_MIX_EXTINFO	_IOWR('X', 4, oss_mixext)+#define SNDCTL_MIX_READ		_IOWR('X', 5, oss_mixer_value)+#define SNDCTL_MIX_WRITE	_IOWR('X', 6, oss_mixer_value)++#define SNDCTL_AUDIOINFO	_IOWR('X', 7, oss_audioinfo)+#define SNDCTL_MIX_ENUMINFO	_IOWR('X', 8, oss_mixer_enuminfo)+#define SNDCTL_MIDIINFO		_IOWR('X', 9, oss_midi_info)+#define SNDCTL_MIXERINFO	_IOWR('X',10, oss_mixerinfo)+#define SNDCTL_CARDINFO		_IOWR('X',11, oss_card_info)++/*+ * Few more "globally" available ioctl calls.+ */+#define SNDCTL_SETSONG          _IOW ('Y', 2, oss_longname_t)+#define SNDCTL_GETSONG          _IOR ('Y', 2, oss_longname_t)+#define SNDCTL_SETNAME          _IOW ('Y', 3, oss_longname_t)+#define SNDCTL_SETLABEL         _IOW ('Y', 4, oss_label_t)+#define SNDCTL_GETLABEL         _IOR ('Y', 4, oss_label_t)++#endif	/* !_SYS_SOUNDCARD_H_ */
+ cbits/proAudio.cpp view
@@ -0,0 +1,167 @@+#include "proAudio.h"+#include <cstdio>+#include <cstdlib>+#include <cstring>+#include <climits>++using namespace std;++//--- class AudioSample --------------------------------------------+AudioSample::AudioSample(const AudioSample & source) :+	m_size(source.m_size), m_channels(source.m_channels), m_sampleRate(source.m_sampleRate), m_bitsPerSample(source.m_bitsPerSample) { +	m_data = new unsigned char [m_size]; memcpy(m_data,source.m_data, m_size); +}++bool AudioSample::bitsPerSample(unsigned short bits) {+	if(bits==16) {+		if(m_bitsPerSample==8) {+			unsigned char* data = new unsigned char[2*m_size];+			 for(unsigned int i=0; i<m_size; ++i) {+				 signed short *ptr =(signed short*)data+i;+				 *ptr = m_data[i]*255;+			 }+			 delete [] m_data;+			 m_data = data;+			 m_size*=2;+			 return true;+		}+		else if(m_bitsPerSample==16) {+			return true; // nothing to do+		}+		else if(m_bitsPerSample==32) { // float, normalized from -1.0f to 1.0f+			unsigned char* data = new unsigned char[m_size/2];+			 for(unsigned int i=0; i<m_size/4; ++i) {+				 signed short *ptr =(signed short*)data+i;+				 float* src=(float*)m_data+i;+				 *ptr =(signed short)(*src*SHRT_MAX);+			 }+			 delete [] m_data;+			 m_data = data;+			 m_size/=2;+			 return true;+		}+	}+	fprintf(stderr,"AudioSample::bitsPerSample ERROR: conversion from %i to %i bits not supported.\n", m_bitsPerSample, bits);+	return false;+}++void AudioSample::volume(float f) {+	if(m_bitsPerSample==8) for(signed char *ptr =(signed char *)m_data; ptr<(signed char *)m_data+m_size; ++ptr) {+		float value=*ptr * f;+		if(value>CHAR_MAX) *ptr =CHAR_MAX;+		else if(value<CHAR_MIN) *ptr =CHAR_MIN;+		else *ptr =(signed char)value;+	}+	else if(m_bitsPerSample==16) for(signed short *ptr =(signed short*)m_data; ptr<(signed short*)m_data+m_size/2; ++ptr) {+		float value=*ptr * f;+		if(value>SHRT_MAX) *ptr =SHRT_MAX;+		else if(value<SHRT_MIN) *ptr =SHRT_MIN;+		else *ptr =(signed short)value;+	}+	else if(m_bitsPerSample==32) for(float *ptr =(float*)m_data; ptr<(float*)m_data+m_size/4; ++ptr) {+		*ptr *= f;+		if(*ptr>1.0f) *ptr=1.0f;			+		else if(*ptr<-1.0f) *ptr=-1.0f;+	}+	else fprintf(stderr,"AudioSample::changeVolume ERROR: %i bits per sample not supported.\n",m_bitsPerSample);+}++AudioSample* AudioSample::readWav(FILE* stream, size_t (*readFunc)( void *, size_t, size_t, FILE *)) {+	char id[4]; //four unsigned chars to hold chunk IDs+	readFunc(id,sizeof(unsigned char),4,stream);+	if (strncmp(id,"RIFF",4)!=0) return 0;++	unsigned int size;+	readFunc(&size,sizeof(unsigned int),1,stream);++	readFunc(id,sizeof(unsigned char),4,stream);+	if (strncmp(id,"WAVE",4)!=0) return 0;++	unsigned short encoding, block_align, channels, bitsPerSample;+	unsigned int chunk_length, byte_rate, sampleRate;++	readFunc(id, sizeof(unsigned char), 4, stream); //read ID 'fmt ';+	readFunc(&chunk_length, sizeof(unsigned int),1,stream); // header length, 16 expected			+	readFunc(&encoding, sizeof(short), 1, stream); // should be "1" for simple PCM data+	if(encoding!=1) return 0;++	readFunc(&channels, sizeof(short),1,stream);+	readFunc(&sampleRate, sizeof(unsigned int), 1, stream);+	readFunc(&byte_rate, sizeof(unsigned int), 1, stream);+	readFunc(&block_align, sizeof(short), 1, stream);+	readFunc(&bitsPerSample, sizeof(short), 1, stream);+	+	readFunc(id, sizeof(unsigned char), 4, stream); // read ID 'data'+	readFunc(&size, sizeof(unsigned int), 1, stream);+	unsigned char *data = new unsigned char[size];+	readFunc(data, sizeof(unsigned char), size, stream);+	+	return new AudioSample(data,size, channels, sampleRate, bitsPerSample);+}++AudioSample* AudioSample::loadWav(const std::string & fname) {+#ifdef _MSC_VER+	FILE *fp = 0;+	fopen_s(&fp, fname.c_str(), "rb");+#else+	FILE *fp = fopen(fname.c_str(), "rb");+#endif+	if (!fp) return 0;+	AudioSample * pSample = readWav(fp, fread);+	fclose(fp);+	return pSample;+}++//--- class DeviceAudio --------------------------------------------++DeviceAudio* DeviceAudio::s_instance=0;++extern "C" {+extern int stb_vorbis_decode_filename(char *filename, int *channels, int* sample_rate, short **output);+};++static AudioSample* loadOgg(const std::string & fname) {+	int channels, sampleRate;+	short *decoded;+	int len = stb_vorbis_decode_filename(const_cast<char*>(fname.c_str()), &channels, &sampleRate, &decoded);+	if(len<0) return 0;+	// convert to AudioSample:+	unsigned int size = len*channels*sizeof(short);+	unsigned char * data = new unsigned char[size];+	if(!data) return 0;+	memcpy(data,decoded, size);+	free(decoded);+	return new AudioSample(data, size, channels, sampleRate, 16);+}++static string toLower(const string & s) {+    string retStr(s);+    for(size_t i=0; i<s.size(); ++i)+        retStr[i]=static_cast<char>(tolower(retStr[i]));+    return retStr;+}++DeviceAudio::DeviceAudio() : m_freqOut(0), m_volL(1.0f), m_volR(1.0f) {+	loaderRegister(AudioSample::loadWav,"wav");+	loaderRegister(loadOgg,"ogg");+}++unsigned int DeviceAudio::sampleFromFile(const std::string & filename, float volume) { +	if(filename.rfind('.')>filename.size()) return 0;+	string suffix=toLower(filename.substr(filename.rfind('.')+1));+	map<string, AudioSample * (*)(const string &)>::iterator it = mm_loader.find(suffix);+	if(it==mm_loader.end()) return 0;+	AudioSample* pSample = (*(it->second))(filename);+	if(!pSample) return 0;+	unsigned int ret = sampleFromMemory(*pSample, volume);+	delete pSample;+	return ret; +}++bool DeviceAudio::loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string & suffix) {+	return mm_loader.insert(std::make_pair(toLower(suffix),loadFunc)).second;+}++bool DeviceAudio::loaderAvailable(const std::string & suffix) const {+	return mm_loader.find(toLower(suffix))!=mm_loader.end();+}
+ cbits/proAudio.h view
@@ -0,0 +1,169 @@+#ifndef _PRO_AUDIO+#define _PRO_AUDIO++#include <string>+#include <map>++/** @file proAudio.h+ \brief Public interface of proteaAudio+ + Contains the declaration of the audio sample class and the abstract base class for audio mixer/playback devices+ + \author  Gerald Franz, www.viremo.de+ \version 0.6+ + License notice (zlib license):++ (c) 2009 by Gerald Franz, www.viremo.de++  This software is provided 'as-is', without any express or implied+  warranty.  In no event will the author be held liable for any damages+  arising from the use of this software.++  Permission is granted to anyone to use this software for any purpose,+  including commercial applications, and to alter it and redistribute it+  freely, subject to the following restrictions:++  1. The origin of this software must not be misrepresented; you must not+     claim that you wrote the original software. If you use this software+     in a product, an acknowledgment in the product documentation would be+     appreciated but is not required.+  2. Altered source versions must be plainly marked as such, and must not be+     misrepresented as being the original software.+  3. This notice may not be removed or altered from any source distribution.+*/++//--- class AudioSample --------------------------------------------+/// class representing an audio sample+class AudioSample {+public:+	/// constructor from memory data+	AudioSample(unsigned char * data, unsigned int size, unsigned short channels, unsigned int sampleRate, unsigned short bitsPerSample) :+		m_data(data), m_size(size), m_channels(channels), m_sampleRate(sampleRate), m_bitsPerSample(bitsPerSample) { }+	/// copy constructor+	AudioSample(const AudioSample & source);+	/// destructor+	~AudioSample() { delete[] m_data; }+	+	/// allows accessing sample data+	unsigned char * data() { return m_data; };+	/// allows reading sample data+	const unsigned char * data() const { return m_data; };+	/// returns sample size in bytes+	unsigned int size() const { return m_size; }+	/// returns sample size in number of frames+	unsigned int frames() const { return m_size/m_channels/(m_bitsPerSample>>3); }+	/// returns size of a single frame in bytes+	unsigned int sizeFrame() const { return m_channels*(m_bitsPerSample>>3); }+	/// returns number of parallel channels, 1 mono, 2 stereo+	unsigned short channels() const { return m_channels; }+	/// returns number of frames per second, e.g., 44100, 22050+	unsigned int sampleRate() const { return m_sampleRate; }+	/// returns number of bits per mono sample, e.g., 8, 16+	unsigned short bitsPerSample() const { return m_bitsPerSample; }+	/// converts to a different bit rate, e.g., 8, 16+	bool bitsPerSample(unsigned short bits);+	/// returns number of bytes per sample, e.g., 1, 2+	unsigned short bytesPerSample() const { return m_bitsPerSample>>3; }++	/// changes volume by given factor+	void volume(float f);+	+	/// loads a WAV file+	static AudioSample* loadWav(const std::string & fname);+	/// reads WAV data from a stream via a function compatible to std::fread+	static AudioSample* readWav(FILE* stream, size_t (*readFunc)( void *, size_t, size_t, FILE *));+protected:+	/// stores sample data+	unsigned char * m_data;+	/// sample size in bytes+	unsigned int m_size;+	/// number of parallel channels, 1 mono, 2 stereo+	unsigned short m_channels;+	/// number of samples per second, e.g., 44100, 22050+	unsigned int m_sampleRate;+	/// number of bits per sample, e.g., 8, 16+	unsigned short m_bitsPerSample;+};++//--- class DeviceAudio --------------------------------------------++/// abstract base class for stereo audio mixer/playback devices+class DeviceAudio {+public:+    /// returns singleton object+	/** This call is only allowed after a successful precedent creation of an audio device */+    static DeviceAudio& singleton() { return *s_instance; }+    /// calls the destructor of the singleton object+    static void destroy() { if(s_instance) delete s_instance; s_instance=0; };++	/// sets master volume+    void volume(float left, float right) { m_volL=left; m_volR=right; }+    /// sets master volume+    void volume(float leftAndRight) { m_volL=m_volR=leftAndRight; }+	/// registers an audio sample loader function handling a file type identified by suffix+	/** The function has to be of type AudioSample * loadXYZ(const std::string & filename).*/+	bool loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string & suffix);+	/// returns true in case a loader for this file type is available+	bool loaderAvailable(const std::string & suffix) const;++    /// loads a sound sample from file, optionally adjusts volume, returns handle+    virtual unsigned int sampleFromFile(const std::string & filename, float volume=1.0f);+    /// converts a sound sample to internal audio format, returns handle+    virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f)=0;+	/// deletes a previously created sound sample resource identified by its handle+	virtual bool sampleDestroy(unsigned int sample)=0;+	/// allows read access to a sample identified by its handle+	virtual const AudioSample* sample(unsigned int handle) const { return 0; }+	+    /// plays a specified sound sample once and sets its parameters+    /** \param sample  handle of a previously loaded sample+     \param volumeL (optional) left volume+     \param volumeR (optional) right volume+     \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+     \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+     \return a handle to the currently played sound or -1 in case of error */+    virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0;+    /// plays a specified sound sample continuously and sets its parameters+     /** \param sample  handle of a previously loaded sample+     \param volumeL (optional) left volume+     \param volumeR (optional) right volume+     \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+     \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+     \return a handle to the currently played sound or -1 in case of error */+    virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0;+    /// updates parameters of a specified sound+     /** \param sound  handle of a currently active sound+     \param volumeL left volume+     \param volumeR right volume+     \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+     \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+     \return true in case the parameters have been updated successfully */+    virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f )=0;+    /// stops a specified sound immediately+    virtual bool soundStop(unsigned int sound)=0;+    /// stops all sounds immediately+    virtual void soundStop()=0;+	/// returns number of currently active sounds+	virtual unsigned int soundActive() const=0;++protected:+	/// constructor+	DeviceAudio();+    /// destructor+    virtual ~DeviceAudio() { s_instance = 0; }+	+	/// stores output stream frequency+	unsigned int m_freqOut;+    /// stores left master volume+    float m_volL;+    /// stores right master volume+    float m_volR;+	/// map associating suffixes to loader functions+	std::map<std::string, AudioSample * (*)(const std::string &)> mm_loader;+	+    /// pointer to singleton+    static DeviceAudio * s_instance;+};++#endif // _PRO_AUDIO
+ cbits/proAudioRt.cpp view
@@ -0,0 +1,242 @@+#include "proAudioRt.h"+#include <cmath>+#include <cstdio>+#include <climits>+#include <cstring>+#include <cstdlib>++using namespace std;++struct _AudioTrack {+	/// sample+	AudioSample * sample;+	+	/// position in sample in frames+	unsigned int dpos;+    /// length of sample in frames+    unsigned int dlen;+    /// disparity in seconds between left and right, normally 0.0f+    float disparity;+    /// left volume+    float volL;+    /// right volume+    float volR;+    /// pitch factor, normally 1.0f+    float pitch;+    /// stores whether sample has to be looped+    bool isLoop;+    /// stores whether sample is currently playing+    bool isPlaying;+};++DeviceAudio* DeviceAudioRt::create(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize) {+    if(!s_instance) {+		DeviceAudioRt* pAudio = new DeviceAudioRt(nTracks,frequency,chunkSize);+		if(!pAudio->m_freqOut) delete pAudio;+		else s_instance = pAudio;+	}+    return s_instance;+}++DeviceAudioRt::DeviceAudioRt(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize) : DeviceAudio() {+	if ( m_dac.getDeviceCount() < 1 ) {+		fprintf(stderr,"DeviceAudioRt ERROR: No audio devices found!\n");+		return;+	}+	// Set our stream parameters for output only.+	RtAudio::StreamParameters oParams;+	oParams.deviceId = m_dac.getDefaultOutputDevice(); // default device+	oParams.nChannels = 2; // stereo+	oParams.firstChannel = 0;++	try {+		m_dac.openStream( &oParams, NULL, RTAUDIO_SINT16, frequency, &chunkSize, &cbMix, (void *)this );+		m_dac.startStream();+	}+	catch ( RtError& e ) {+		fprintf(stderr,"%s\n", e.getMessage().c_str());+		if(m_dac.isStreamOpen()) m_dac.closeStream();+		return;+	}++    // initialize tracks:+    m_nSound=nTracks;+    ma_sound=new _AudioTrack[m_nSound];+	memset(ma_sound,0,m_nSound*sizeof(_AudioTrack));+	m_freqOut = frequency;+}++DeviceAudioRt::~DeviceAudioRt() {+    if(m_dac.isStreamOpen()) m_dac.closeStream();+    delete [] ma_sound;+	for( map<unsigned int,AudioSample*>::iterator it=mm_sample.begin(); it!=mm_sample.end(); ++it)+		delete it->second;+	mm_sample.clear();+}++unsigned int DeviceAudioRt::sampleFromMemory(const AudioSample & sample, float volume) {+	AudioSample * pSample = new AudioSample(sample);+	if(volume!=1.0f) pSample->volume(volume);+	pSample->bitsPerSample(16);+    mm_sample.insert(make_pair(++m_sampleCounter,pSample));+    return m_sampleCounter;+}++bool DeviceAudioRt::sampleDestroy(unsigned int sample) {+    // look for sample:+    map<unsigned int,AudioSample*>::iterator iter=mm_sample.find(sample);+    if( iter == mm_sample.end() ) return false;+	// stop currently playing sounds referring to this sample:+	for (unsigned int i=0; i<m_nSound; ++i ) if(ma_sound[i].sample == iter->second)+		ma_sound[i].isPlaying=false;+	// cleanup:+	delete iter->second;+	if(iter->first==m_sampleCounter) --m_sampleCounter;+	mm_sample.erase(iter);+	return true;+}++const AudioSample* DeviceAudioRt::sample(unsigned int handle) const { +    map<unsigned int,AudioSample*>::const_iterator it=mm_sample.find(handle);+    if( it == mm_sample.end() ) return 0;+	return it->second;+}+++unsigned int DeviceAudioRt::soundPlay(unsigned int sample, float volumeL, float volumeR, float disparity, float pitch ) {+    // look for sample:+    map<unsigned int,AudioSample*>::iterator iter=mm_sample.find(sample);+    if( iter == mm_sample.end() ) return 0; // no sample found+    // look for an empty (or finished) sound track+    unsigned int i;+    for ( i=0; i<m_nSound; ++i )+        if (!ma_sound[i].isPlaying) break;+    if ( i == m_nSound ) return 0; // no empty slot found++	unsigned int sampleRate = iter->second->sampleRate();+	if(sampleRate!=m_freqOut) pitch*=(float)sampleRate/(float)m_freqOut;+	+    // put the sample data in the slot and play it+    ma_sound[i].sample = iter->second;+    ma_sound[i].dlen = iter->second->frames();+    ma_sound[i].dpos = 0;+    ma_sound[i].volL=volumeL;+    ma_sound[i].volR=volumeR;+    ma_sound[i].disparity=disparity;+    ma_sound[i].pitch=fabs(pitch);+    ma_sound[i].isLoop=false;+    ma_sound[i].isPlaying=true;+    return i+1;+}++unsigned int DeviceAudioRt::soundLoop(unsigned int sample, float volumeL, float volumeR, float disparity, float pitch ) {+    unsigned int ret=soundPlay(sample,volumeL,volumeR,disparity, pitch);+    if(ret) ma_sound[ret-1].isLoop=true;+    return ret;+}++bool DeviceAudioRt::soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity, float pitch ) {+    if(!sound || (sound>m_nSound) || !ma_sound[sound-1].isPlaying) return false;+    ma_sound[--sound].volL=volumeL;+    ma_sound[sound].volR=volumeR;+    ma_sound[sound].disparity=disparity;+	unsigned int sampleRate = ma_sound[sound].sample->sampleRate();+	if(sampleRate!=m_freqOut) pitch*=(float)sampleRate/(float)m_freqOut;+    ma_sound[sound].pitch=fabs(pitch);+    return true;+}++bool DeviceAudioRt::soundStop(unsigned int sound) {+    if(!sound||(sound>m_nSound)||!ma_sound[sound-1].isPlaying) return false;+    ma_sound[sound-1].isPlaying=false;+    return true;+}++void DeviceAudioRt::soundStop() {+	for (unsigned int i=0; i<m_nSound; ++i )+		ma_sound[i].isPlaying=false;+}++unsigned int DeviceAudioRt::soundActive() const {+	if(!const_cast<RtAudio*>(&m_dac)->isStreamRunning()	) return 0;+    unsigned int ret = 0, i;+    for ( i=0; i<m_nSound; ++i )+        if (ma_sound[i].isPlaying) ++ret;+    return ret;+}++int DeviceAudioRt::mixOutputFloat(signed short *outputBuffer, unsigned int nFrames) {+    for(unsigned int j=0; j<nFrames; ++j) {+        float left=0.0f;+        float right=0.0f;+        for (unsigned int i=0; i<m_nSound; ++i ) if(ma_sound[i].isPlaying) {+			unsigned int nChannels = ma_sound[i].sample->channels();+            if((ma_sound[i].pitch==1.0f)&&!ma_sound[i].disparity) { // use optimized default mixing:+                unsigned int currPos=ma_sound[i].dpos+j;+                if(ma_sound[i].isLoop) currPos%=ma_sound[i].dlen;+                else if(currPos >= ma_sound[i].dlen) continue;+				currPos*=ma_sound[i].sample->sizeFrame();+				float dataL = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPos])));+                left += dataL * m_volL*ma_sound[i].volL;+                float dataR = (nChannels>1) ? (float)(*((signed short *)(&ma_sound[i].sample->data()[currPos+2]))) : dataL;+				right+= dataR * m_volR*ma_sound[i].volR;				+            }+            else { // use nearest sample and disparity:+                double fract=ma_sound[i].dpos+j*ma_sound[i].pitch;+                unsigned int currPos=(unsigned int)fract;+				fract = fmod(fract,1.0);+                int currPosL= (ma_sound[i].disparity<0.0f) ? currPos+int(m_freqOut*ma_sound[i].disparity) : currPos;+                int currPosR= (ma_sound[i].disparity>0.0f) ? currPos-int(m_freqOut*ma_sound[i].disparity) : currPos;+				if(nChannels>1) currPosR+=sizeof(signed short); // use second channel+                if(ma_sound[i].isLoop) {+					currPosL+=ma_sound[i].dlen;+					currPosL%=ma_sound[i].dlen;+					currPosR+=ma_sound[i].dlen;+					currPosR%=ma_sound[i].dlen;+				}+				if(currPosL<0) {+					// do nothing+				}+				else if((unsigned int)currPosL+1 < ma_sound[i].dlen) {+					currPosL*=ma_sound[i].sample->sizeFrame();+					float dataL = (1.0f-(float)fract)*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL])))+						+ (float)fract*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL+ma_sound[i].sample->sizeFrame()])));+					left += dataL * m_volL*ma_sound[i].volL;+				}+				else if((unsigned int)currPosL+1 == ma_sound[i].dlen) {+					currPosL*=ma_sound[i].sample->sizeFrame();+					float dataL = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL])));+					left += dataL * m_volL*ma_sound[i].volL;+				}+				+				if(currPosR<0) {+					// do nothing+				}+				else if((unsigned int)currPosR+1 < ma_sound[i].dlen) {+					currPosR*=ma_sound[i].sample->sizeFrame();+					float dataR = (1.0f-(float)fract)*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR])))+						+ (float)fract*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR+ma_sound[i].sample->sizeFrame()])));+					right += dataR * m_volR*ma_sound[i].volR;+				}+				else if((unsigned int)currPosR+1 == ma_sound[i].dlen) {+					currPosR*=ma_sound[i].sample->sizeFrame();+					float dataR = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR])));+					right += dataR * m_volR*ma_sound[i].volR;+				}+            }+        }+        // clamp and set output:+        outputBuffer[2*j] = left>SHRT_MAX ? SHRT_MAX : left<SHRT_MIN ? SHRT_MIN : (signed short)left;+        outputBuffer[2*j+1] = right>SHRT_MAX ? SHRT_MAX : right<SHRT_MIN ? SHRT_MIN : (signed short)right;+    }+	// calculate new pos:+    for (unsigned int i=0; i<m_nSound; ++i ) {+        if(ma_sound[i].pitch==1.0f) ma_sound[i].dpos += nFrames;+        else ma_sound[i].dpos += (unsigned int)(nFrames*ma_sound[i].pitch);++        if(ma_sound[i].isLoop) ma_sound[i].dpos%=ma_sound[i].dlen;+        else if(ma_sound[i].dpos>ma_sound[i].dlen+2*abs(int(m_freqOut*-ma_sound[i].disparity)))+            ma_sound[i].isPlaying=false;+    }+	return 0;+}
+ cbits/proAudioRt.h view
@@ -0,0 +1,88 @@+#include "proAudio.h"+#include <RtAudio.h>+#include <map>++/** @file proAudioRt.h+ \brief RtAudio backend of proteaAudio+ \author  Gerald Franz, www.viremo.de+ \version 0.6+*/ ++struct _AudioTrack;++/// an rtAudio based stereo audio mixer/playback device+/** DeviceAudioRt offers some advanced features such as dynamic pitch,+ independent volume control for both channels, and user-defined time shifts between the channels. */+class DeviceAudioRt : public DeviceAudio {+public:+	///creates audio device+    /** Use this method instead of a constructor.+     \param nTracks (optional) the maximum number of sounds that are played parallely. Computation time is linearly correlated to this factor.+     \param frequency (optional) sample frequency of the playback in Hz. 22050 corresponds to FM radio 44100 is CD quality. Computation time is linearly correlated to this factor.+     \param chunkSize (optional) the number of bytes that are sent to the sound card at once. Low numbers lead to smaller latencies but need more computation time (thread switches). If a too small number is chosen, the sounds might not be played continuously. The default value 512 guarantees a good latency below 40 ms at 22050 Hz sample frequency.+     \return a pointer to an audio device object in case of success+	Note that the parameters are only handled when calling for the first time. Afterwards always the same object is returned until an explicit destroy() is called.+     */+    static DeviceAudio* create(unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024);++    /// converts a sound sample to internal audio format, returns handle+    virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f);+	/// deletes a previously created sound sample resource identified by its handle+	virtual bool sampleDestroy(unsigned int sample);+	/// allows read access to a sample identified by its handle+	virtual const AudioSample* sample(unsigned int handle) const;++	/// plays a specified sample once and sets its parameters+    /** \param sample a sample handle returned by a previous load() call+     \param volumeL (optional) left volume+     \param volumeR (optional) right volume+     \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+     \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+     \return a handle to the currently played sound or 0 in case of error */+    virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );+    /** plays a specified sample continuously and sets its parameters+     \param sample a sample handle returned by a previous load() call+     \param volumeL (optional) left volume+     \param volumeR (optional) right volume+     \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+     \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+     \return a handle to the currently played sound or 0 in case of error */+    virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );+    /// updates parameters of a specified sound+     /** \param sound  handle of a currently active sound+     \param volumeL left volume+     \param volumeR right volume+     \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+     \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+     \return true in case the parameters have been updated successfully */+    virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f );+    /// stops a specified sound immediately+    virtual bool soundStop(unsigned int sound);+    /// stops all sounds immediately+    virtual void soundStop();+	/// returns number of currently active sounds+	virtual unsigned soundActive() const;+protected:+    /// constructor. Use the create() method instead+    DeviceAudioRt(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize);+    /// destructor. Use the destroy() method instead+    virtual ~DeviceAudioRt();+	/// mixes tracks to a single output stream+	int mixOutputFloat(signed short *outputBuffer, unsigned int nFrames);++	/// stores loaded sound samples+    std::map<unsigned int, AudioSample*> mm_sample;+    /// stores maximum sample id+    unsigned int m_sampleCounter;++    /// stores sounds to be mixed+    _AudioTrack * ma_sound;+    /// stores number of parallel sounds+    unsigned int m_nSound;+	/// audio manager+	RtAudio m_dac;++    /// mixer callback+    static int cbMix(void *outputBuffer, void *inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void *data) {+		return static_cast<DeviceAudioRt*>(data)->mixOutputFloat((signed short*)outputBuffer, nFrames); }+};
+ cbits/proteaaudio_binding.cpp view
@@ -0,0 +1,67 @@+#include "proteaaudio_binding.h"+#include "proAudioRt.h"++// generic+int initAudio(int nTracks, int frequency, int chunkSize) {+    DeviceAudio* pAudio = DeviceAudioRt::create(nTracks, frequency, chunkSize);+    return pAudio != 0;+}++void finishAudio() {+    DeviceAudio::destroy();+}++int loaderAvailable(char* suffix) {+    DeviceAudio & audio = DeviceAudio::singleton();+    if(!&audio) return 0;+    return audio.loaderAvailable(suffix);+}++void volume(float left, float right) {+    DeviceAudio & audio = DeviceAudio::singleton();+    if(!&audio) return;+    audio.volume(left,right);+}++sample_t sampleFromFile(char* filename, float volume) {+    DeviceAudio & audio = DeviceAudio::singleton();+    if(!&audio) return 0;+    return (int)audio.sampleFromFile(filename, volume);+}++int soundActive() {+    DeviceAudio & audio = DeviceAudio::singleton();+    if(!&audio) return 0;+    return (int)audio.soundActive();+}++void soundStopAll() {+    DeviceAudio & audio = DeviceAudio::singleton();+    if(!&audio) return;+    audio.soundStop();+}++// sound+void soundLoop(sample_t sample, float volumeL, float volumeR, float disparity, float pitch) {+    DeviceAudio & audio = DeviceAudio::singleton();+    if(!&audio) return;+    audio.soundLoop(sample, volumeL,volumeR,disparity,pitch);+}++void soundPlay(sample_t sample, float volumeL, float volumeR, float disparity, float pitch) {+    DeviceAudio & audio = DeviceAudio::singleton();+    if(!&audio) return;+    audio.soundPlay(sample, volumeL,volumeR,disparity,pitch);+}++int soundUpdate(sample_t sample, float volumeL, float volumeR, float disparity, float pitch) {+    DeviceAudio & audio = DeviceAudio::singleton();+    if(!&audio) return 0;+    return audio.soundUpdate(sample, volumeL,volumeR,disparity,pitch);+}++int soundStop(sample_t sample) {+    DeviceAudio & audio = DeviceAudio::singleton();+    if(!&audio) return 0;+    return audio.soundStop(sample);+}
+ cbits/proteaaudio_binding.h view
@@ -0,0 +1,20 @@+#ifdef __cplusplus+extern "C" { +#endif++typedef int sample_t;++int initAudio(int nTracks, int frequency, int chunkSize);+void finishAudio();+int loaderAvailable(char* suffix);+void volume(float left, float right);+sample_t sampleFromFile(char* filename, float volume);+int soundActive();+void soundStopAll();+void soundLoop(sample_t sample, float volumeL, float volumeR, float disparity, float pitch);+void soundPlay(sample_t sample, float volumeL, float volumeR, float disparity, float pitch);+int soundUpdate(sample_t sample, float volumeL, float volumeR, float disparity, float pitch);+int soundStop(sample_t sample);+#ifdef __cplusplus+}+#endif
+ cbits/stb_vorbis.c view
@@ -0,0 +1,5349 @@+// Ogg Vorbis I audio decoder  -- version 0.99996
+//
+// Written in April 2007 by Sean Barrett, sponsored by RAD Game Tools.
+//
+// Placed in the public domain April 2007 by the author: no copyright is
+// claimed, and you may use it for any purpose you like.
+//
+// No warranty for any purpose is expressed or implied by the author (nor
+// by RAD Game Tools). Report bugs and send enhancements to the author.
+//
+// Get the latest version and other information at:
+//     http://nothings.org/stb_vorbis/
+
+
+// Todo:
+//
+//   - seeking (note you can seek yourself using the pushdata API)
+//
+// Limitations:
+//
+//   - floor 0 not supported (used in old ogg vorbis files)
+//   - lossless sample-truncation at beginning ignored
+//   - cannot concatenate multiple vorbis streams
+//   - sample positions are 32-bit, limiting seekable 192Khz
+//       files to around 6 hours (Ogg supports 64-bit)
+// 
+// All of these limitations may be removed in future versions.
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+//  HEADER BEGINS HERE
+//
+
+#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H
+#define STB_VORBIS_INCLUDE_STB_VORBIS_H
+
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+#define STB_VORBIS_NO_STDIO 1
+#endif
+
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+///////////   THREAD SAFETY
+
+// Individual stb_vorbis* handles are not thread-safe; you cannot decode from
+// them from multiple threads at the same time. However, you can have multiple
+// stb_vorbis* handles and decode from them independently in multiple thrads.
+
+
+///////////   MEMORY ALLOCATION
+
+// normally stb_vorbis uses malloc() to allocate memory at startup,
+// and alloca() to allocate temporary memory during a frame on the
+// stack. (Memory consumption will depend on the amount of setup
+// data in the file and how you set the compile flags for speed
+// vs. size. In my test files the maximal-size usage is ~150KB.)
+//
+// You can modify the wrapper functions in the source (setup_malloc,
+// setup_temp_malloc, temp_malloc) to change this behavior, or you
+// can use a simpler allocation model: you pass in a buffer from
+// which stb_vorbis will allocate _all_ its memory (including the
+// temp memory). "open" may fail with a VORBIS_outofmem if you
+// do not pass in enough data; there is no way to determine how
+// much you do need except to succeed (at which point you can
+// query get_info to find the exact amount required. yes I know
+// this is lame).
+//
+// If you pass in a non-NULL buffer of the type below, allocation
+// will occur from it as described above. Otherwise just pass NULL
+// to use malloc()/alloca()
+
+typedef struct
+{
+   char *alloc_buffer;
+   int   alloc_buffer_length_in_bytes;
+} stb_vorbis_alloc;
+
+
+///////////   FUNCTIONS USEABLE WITH ALL INPUT MODES
+
+typedef struct stb_vorbis stb_vorbis;
+
+typedef struct
+{
+   unsigned int sample_rate;
+   int channels;
+
+   unsigned int setup_memory_required;
+   unsigned int setup_temp_memory_required;
+   unsigned int temp_memory_required;
+
+   int max_frame_size;
+} stb_vorbis_info;
+
+// get general information about the file
+extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f);
+
+// get the last error detected (clears it, too)
+extern int stb_vorbis_get_error(stb_vorbis *f);
+
+// close an ogg vorbis file and free all memory in use
+extern void stb_vorbis_close(stb_vorbis *f);
+
+// this function returns the offset (in samples) from the beginning of the
+// file that will be returned by the next decode, if it is known, or -1
+// otherwise. after a flush_pushdata() call, this may take a while before
+// it becomes valid again.
+// NOT WORKING YET after a seek with PULLDATA API
+extern int stb_vorbis_get_sample_offset(stb_vorbis *f);
+
+// returns the current seek point within the file, or offset from the beginning
+// of the memory buffer. In pushdata mode it returns 0.
+extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f);
+
+///////////   PUSHDATA API
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+
+// this API allows you to get blocks of data from any source and hand
+// them to stb_vorbis. you have to buffer them; stb_vorbis will tell
+// you how much it used, and you have to give it the rest next time;
+// and stb_vorbis may not have enough data to work with and you will
+// need to give it the same data again PLUS more. Note that the Vorbis
+// specification does not bound the size of an individual frame.
+
+extern stb_vorbis *stb_vorbis_open_pushdata(
+         unsigned char *datablock, int datablock_length_in_bytes,
+         int *datablock_memory_consumed_in_bytes,
+         int *error,
+         stb_vorbis_alloc *alloc_buffer);
+// create a vorbis decoder by passing in the initial data block containing
+//    the ogg&vorbis headers (you don't need to do parse them, just provide
+//    the first N bytes of the file--you're told if it's not enough, see below)
+// on success, returns an stb_vorbis *, does not set error, returns the amount of
+//    data parsed/consumed on this call in *datablock_memory_consumed_in_bytes;
+// on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed
+// if returns NULL and *error is VORBIS_need_more_data, then the input block was
+//       incomplete and you need to pass in a larger block from the start of the file
+
+extern int stb_vorbis_decode_frame_pushdata(
+         stb_vorbis *f, unsigned char *datablock, int datablock_length_in_bytes,
+         int *channels,             // place to write number of float * buffers
+         float ***output,           // place to write float ** array of float * buffers
+         int *samples               // place to write number of output samples
+     );
+// decode a frame of audio sample data if possible from the passed-in data block
+//
+// return value: number of bytes we used from datablock
+// possible cases:
+//     0 bytes used, 0 samples output (need more data)
+//     N bytes used, 0 samples output (resynching the stream, keep going)
+//     N bytes used, M samples output (one frame of data)
+// note that after opening a file, you will ALWAYS get one N-bytes,0-sample
+// frame, because Vorbis always "discards" the first frame.
+//
+// Note that on resynch, stb_vorbis will rarely consume all of the buffer,
+// instead only datablock_length_in_bytes-3 or less. This is because it wants
+// to avoid missing parts of a page header if they cross a datablock boundary,
+// without writing state-machiney code to record a partial detection.
+//
+// The number of channels returned are stored in *channels (which can be
+// NULL--it is always the same as the number of channels reported by
+// get_info). *output will contain an array of float* buffers, one per
+// channel. In other words, (*output)[0][0] contains the first sample from
+// the first channel, and (*output)[1][0] contains the first sample from
+// the second channel.
+
+extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
+// inform stb_vorbis that your next datablock will not be contiguous with
+// previous ones (e.g. you've seeked in the data); future attempts to decode
+// frames will cause stb_vorbis to resynchronize (as noted above), and
+// once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it
+// will begin decoding the _next_ frame.
+//
+// if you want to seek using pushdata, you need to seek in your file, then
+// call stb_vorbis_flush_pushdata(), then start calling decoding, then once
+// decoding is returning you data, call stb_vorbis_get_sample_offset, and
+// if you don't like the result, seek your file again and repeat.
+#endif
+
+
+//////////   PULLING INPUT API
+
+#ifndef STB_VORBIS_NO_PULLDATA_API
+// This API assumes stb_vorbis is allowed to pull data from a source--
+// either a block of memory containing the _entire_ vorbis stream, or a
+// FILE * that you or it create, or possibly some other reading mechanism
+// if you go modify the source to replace the FILE * case with some kind
+// of callback to your code. (But if you don't support seeking, you may
+// just want to go ahead and use pushdata.)
+
+#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
+extern int stb_vorbis_decode_filename(char *filename, int *channels, int* sample_rate, short **output);
+#endif
+extern int stb_vorbis_decode_memory(unsigned char *mem, int len, int *channels, int* sample_rate, short **output);
+// decode an entire file and output the data interleaved into a malloc()ed
+// buffer stored in *output. The return value is the number of samples
+// decoded, or -1 if the file could not be opened or was not an ogg vorbis file.
+// When you're done with it, just free() the pointer returned in *output.
+
+extern stb_vorbis * stb_vorbis_open_memory(unsigned char *data, int len,
+                                  int *error, stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from an ogg vorbis stream in memory (note
+// this must be the entire stream!). on failure, returns NULL and sets *error
+
+#ifndef STB_VORBIS_NO_STDIO
+extern stb_vorbis * stb_vorbis_open_filename(char *filename,
+                                  int *error, stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from a filename via fopen(). on failure,
+// returns NULL and sets *error (possibly to VORBIS_file_open_failure).
+
+extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close,
+                                  int *error, stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+// the _current_ seek point (ftell). on failure, returns NULL and sets *error.
+// note that stb_vorbis must "own" this stream; if you seek it in between
+// calls to stb_vorbis, it will become confused. Morever, if you attempt to
+// perform stb_vorbis_seek_*() operations on this file, it will assume it
+// owns the _entire_ rest of the file after the start point. Use the next
+// function, stb_vorbis_open_file_section(), to limit it.
+
+extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close,
+                int *error, stb_vorbis_alloc *alloc_buffer, unsigned int len);
+// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+// the _current_ seek point (ftell); the stream will be of length 'len' bytes.
+// on failure, returns NULL and sets *error. note that stb_vorbis must "own"
+// this stream; if you seek it in between calls to stb_vorbis, it will become
+// confused.
+#endif
+
+extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number);
+extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number);
+// NOT WORKING YET
+// these functions seek in the Vorbis file to (approximately) 'sample_number'.
+// after calling seek_frame(), the next call to get_frame_*() will include
+// the specified sample. after calling stb_vorbis_seek(), the next call to
+// stb_vorbis_get_samples_* will start with the specified sample. If you
+// do not need to seek to EXACTLY the target sample when using get_samples_*,
+// you can also use seek_frame().
+
+extern void stb_vorbis_seek_start(stb_vorbis *f);
+// this function is equivalent to stb_vorbis_seek(f,0), but it
+// actually works
+
+extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f);
+extern float        stb_vorbis_stream_length_in_seconds(stb_vorbis *f);
+// these functions return the total length of the vorbis stream
+
+extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output);
+// decode the next frame and return the number of samples. the number of
+// channels returned are stored in *channels (which can be NULL--it is always
+// the same as the number of channels reported by get_info). *output will
+// contain an array of float* buffers, one per channel. These outputs will
+// be overwritten on the next call to stb_vorbis_get_frame_*.
+//
+// You generally should not intermix calls to stb_vorbis_get_frame_*()
+// and stb_vorbis_get_samples_*(), since the latter calls the former.
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts);
+extern int stb_vorbis_get_frame_short            (stb_vorbis *f, int num_c, short **buffer, int num_samples);
+#endif
+// decode the next frame and return the number of samples per channel. the
+// data is coerced to the number of channels you request according to the
+// channel coercion rules (see below). You must pass in the size of your
+// buffer(s) so that stb_vorbis will not overwrite the end of the buffer.
+// The maximum buffer size needed can be gotten from get_info(); however,
+// the Vorbis I specification implies an absolute maximum of 4096 samples
+// per channel. Note that for interleaved data, you pass in the number of
+// shorts (the size of your array), but the return value is the number of
+// samples per channel, not the total number of samples.
+
+// Channel coercion rules:
+//    Let M be the number of channels requested, and N the number of channels present,
+//    and Cn be the nth channel; let stereo L be the sum of all L and center channels,
+//    and stereo R be the sum of all R and center channels (channel assignment from the
+//    vorbis spec).
+//        M    N       output
+//        1    k      sum(Ck) for all k
+//        2    *      stereo L, stereo R
+//        k    l      k > l, the first l channels, then 0s
+//        k    l      k <= l, the first k channels
+//    Note that this is not _good_ surround etc. mixing at all! It's just so
+//    you get something useful.
+
+extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats);
+extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples);
+// gets num_samples samples, not necessarily on a frame boundary--this requires
+// buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES.
+// Returns the number of samples stored per channel; it may be less than requested
+// at the end of the file. If there are no more samples in the file, returns 0.
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts);
+extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples);
+#endif
+// gets num_samples samples, not necessarily on a frame boundary--this requires
+// buffering so you have to supply the buffers. Applies the coercion rules above
+// to produce 'channels' channels. Returns the number of samples stored per channel;
+// it may be less than requested at the end of the file. If there are no more
+// samples in the file, returns 0.
+
+#endif
+
+////////   ERROR CODES
+
+enum STBVorbisError
+{
+   VORBIS__no_error,
+
+   VORBIS_need_more_data=1,             // not a real error
+
+   VORBIS_invalid_api_mixing,           // can't mix API modes
+   VORBIS_outofmem,                     // not enough memory
+   VORBIS_feature_not_supported,        // uses floor 0
+   VORBIS_too_many_channels,            // STB_VORBIS_MAX_CHANNELS is too small
+   VORBIS_file_open_failure,            // fopen() failed
+   VORBIS_seek_without_length,          // can't seek in unknown-length file
+
+   VORBIS_unexpected_eof=10,            // file is truncated?
+   VORBIS_seek_invalid,                 // seek past EOF
+
+   // decoding errors (corrupt/invalid stream) -- you probably
+   // don't care about the exact details of these
+
+   // vorbis errors:
+   VORBIS_invalid_setup=20,
+   VORBIS_invalid_stream,
+
+   // ogg errors:
+   VORBIS_missing_capture_pattern=30,
+   VORBIS_invalid_stream_structure_version,
+   VORBIS_continued_packet_flag_invalid,
+   VORBIS_incorrect_stream_serial_number,
+   VORBIS_invalid_first_page,
+   VORBIS_bad_packet_type,
+   VORBIS_cant_find_last_page,
+   VORBIS_seek_failed,
+};
+
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H
+//
+//  HEADER ENDS HERE
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#ifndef STB_VORBIS_HEADER_ONLY
+
+// global configuration settings (e.g. set these in the project/makefile),
+// or just set them in this file at the top (although ideally the first few
+// should be visible when the header file is compiled too, although it's not
+// crucial)
+
+// STB_VORBIS_NO_PUSHDATA_API
+//     does not compile the code for the various stb_vorbis_*_pushdata()
+//     functions
+// #define STB_VORBIS_NO_PUSHDATA_API
+
+// STB_VORBIS_NO_PULLDATA_API
+//     does not compile the code for the non-pushdata APIs
+// #define STB_VORBIS_NO_PULLDATA_API
+
+// STB_VORBIS_NO_STDIO
+//     does not compile the code for the APIs that use FILE *s internally
+//     or externally (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_STDIO
+
+// STB_VORBIS_NO_INTEGER_CONVERSION
+//     does not compile the code for converting audio sample data from
+//     float to integer (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_INTEGER_CONVERSION
+
+// STB_VORBIS_NO_FAST_SCALED_FLOAT
+//      does not use a fast float-to-int trick to accelerate float-to-int on
+//      most platforms which requires endianness be defined correctly.
+//#define STB_VORBIS_NO_FAST_SCALED_FLOAT
+
+
+// STB_VORBIS_MAX_CHANNELS [number]
+//     globally define this to the maximum number of channels you need.
+//     The spec does not put a restriction on channels except that
+//     the count is stored in a byte, so 255 is the hard limit.
+//     Reducing this saves about 16 bytes per value, so using 16 saves
+//     (255-16)*16 or around 4KB. Plus anything other memory usage
+//     I forgot to account for. Can probably go as low as 8 (7.1 audio),
+//     6 (5.1 audio), or 2 (stereo only).
+#ifndef STB_VORBIS_MAX_CHANNELS
+#define STB_VORBIS_MAX_CHANNELS    16  // enough for anyone?
+#endif
+
+// STB_VORBIS_PUSHDATA_CRC_COUNT [number]
+//     after a flush_pushdata(), stb_vorbis begins scanning for the
+//     next valid page, without backtracking. when it finds something
+//     that looks like a page, it streams through it and verifies its
+//     CRC32. Should that validation fail, it keeps scanning. But it's
+//     possible that _while_ streaming through to check the CRC32 of
+//     one candidate page, it sees another candidate page. This #define
+//     determines how many "overlapping" candidate pages it can search
+//     at once. Note that "real" pages are typically ~4KB to ~8KB, whereas
+//     garbage pages could be as big as 64KB, but probably average ~16KB.
+//     So don't hose ourselves by scanning an apparent 64KB page and
+//     missing a ton of real ones in the interim; so minimum of 2
+#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT
+#define STB_VORBIS_PUSHDATA_CRC_COUNT  4
+#endif
+
+// STB_VORBIS_FAST_HUFFMAN_LENGTH [number]
+//     sets the log size of the huffman-acceleration table.  Maximum
+//     supported value is 24. with larger numbers, more decodings are O(1),
+//     but the table size is larger so worse cache missing, so you'll have
+//     to probe (and try multiple ogg vorbis files) to find the sweet spot.
+#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH
+#define STB_VORBIS_FAST_HUFFMAN_LENGTH   10
+#endif
+
+// STB_VORBIS_FAST_BINARY_LENGTH [number]
+//     sets the log size of the binary-search acceleration table. this
+//     is used in similar fashion to the fast-huffman size to set initial
+//     parameters for the binary search
+
+// STB_VORBIS_FAST_HUFFMAN_INT
+//     The fast huffman tables are much more efficient if they can be
+//     stored as 16-bit results instead of 32-bit results. This restricts
+//     the codebooks to having only 65535 possible outcomes, though.
+//     (At least, accelerated by the huffman table.)
+#ifndef STB_VORBIS_FAST_HUFFMAN_INT
+#define STB_VORBIS_FAST_HUFFMAN_SHORT
+#endif
+
+// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+//     If the 'fast huffman' search doesn't succeed, then stb_vorbis falls
+//     back on binary searching for the correct one. This requires storing
+//     extra tables with the huffman codes in sorted order. Defining this
+//     symbol trades off space for speed by forcing a linear search in the
+//     non-fast case, except for "sparse" codebooks.
+// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+
+// STB_VORBIS_DIVIDES_IN_RESIDUE
+//     stb_vorbis precomputes the result of the scalar residue decoding
+//     that would otherwise require a divide per chunk. you can trade off
+//     space for time by defining this symbol.
+// #define STB_VORBIS_DIVIDES_IN_RESIDUE
+
+// STB_VORBIS_DIVIDES_IN_CODEBOOK
+//     vorbis VQ codebooks can be encoded two ways: with every case explicitly
+//     stored, or with all elements being chosen from a small range of values,
+//     and all values possible in all elements. By default, stb_vorbis expands
+//     this latter kind out to look like the former kind for ease of decoding,
+//     because otherwise an integer divide-per-vector-element is required to
+//     unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can
+//     trade off storage for speed.
+//#define STB_VORBIS_DIVIDES_IN_CODEBOOK
+
+// STB_VORBIS_CODEBOOK_SHORTS
+//     The vorbis file format encodes VQ codebook floats as ax+b where a and
+//     b are floating point per-codebook constants, and x is a 16-bit int.
+//     Normally, stb_vorbis decodes them to floats rather than leaving them
+//     as 16-bit ints and computing ax+b while decoding. This is a speed/space
+//     tradeoff; you can save space by defining this flag.
+#ifndef STB_VORBIS_CODEBOOK_SHORTS
+#define STB_VORBIS_CODEBOOK_FLOATS
+#endif
+
+// STB_VORBIS_DIVIDE_TABLE
+//     this replaces small integer divides in the floor decode loop with
+//     table lookups. made less than 1% difference, so disabled by default.
+
+// STB_VORBIS_NO_INLINE_DECODE
+//     disables the inlining of the scalar codebook fast-huffman decode.
+//     might save a little codespace; useful for debugging
+// #define STB_VORBIS_NO_INLINE_DECODE
+
+// STB_VORBIS_NO_DEFER_FLOOR
+//     Normally we only decode the floor without synthesizing the actual
+//     full curve. We can instead synthesize the curve immediately. This
+//     requires more memory and is very likely slower, so I don't think
+//     you'd ever want to do it except for debugging.
+// #define STB_VORBIS_NO_DEFER_FLOOR
+
+
+
+
+//////////////////////////////////////////////////////////////////////////////
+
+#ifdef STB_VORBIS_NO_PULLDATA_API
+   #define STB_VORBIS_NO_INTEGER_CONVERSION
+   #define STB_VORBIS_NO_STDIO
+#endif
+
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+   #define STB_VORBIS_NO_STDIO 1
+#endif
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
+
+   // only need endianness for fast-float-to-int, which we don't
+   // use for pushdata
+
+   #ifndef STB_VORBIS_BIG_ENDIAN
+     #define STB_VORBIS_ENDIAN  0
+   #else
+     #define STB_VORBIS_ENDIAN  1
+   #endif
+
+#endif
+#endif
+
+
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+
+#ifndef STB_VORBIS_NO_CRT
+#include <stdlib.h>
+#include <string.h>
+#include <assert.h>
+#include <math.h>
+#if !(defined(__APPLE__) || defined(MACOSX) || defined(macintosh) || defined(Macintosh))
+#include <malloc.h>
+#endif
+#else
+#define NULL 0
+#endif
+
+#ifndef _MSC_VER
+   #if __GNUC__
+      #define __forceinline inline
+   #else
+      #define __forceinline
+   #endif
+#endif
+
+#if STB_VORBIS_MAX_CHANNELS > 256
+#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range"
+#endif
+
+#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24
+#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range"
+#endif
+
+
+#define MAX_BLOCKSIZE_LOG  13   // from specification
+#define MAX_BLOCKSIZE      (1 << MAX_BLOCKSIZE_LOG)
+
+
+typedef unsigned char  uint8;
+typedef   signed char   int8;
+typedef unsigned short uint16;
+typedef   signed short  int16;
+typedef unsigned int   uint32;
+typedef   signed int    int32;
+
+#ifndef TRUE
+#define TRUE 1
+#define FALSE 0
+#endif
+
+#ifdef STB_VORBIS_CODEBOOK_FLOATS
+typedef float codetype;
+#else
+typedef uint16 codetype;
+#endif
+
+// @NOTE
+//
+// Some arrays below are tagged "//varies", which means it's actually
+// a variable-sized piece of data, but rather than malloc I assume it's
+// small enough it's better to just allocate it all together with the
+// main thing
+//
+// Most of the variables are specified with the smallest size I could pack
+// them into. It might give better performance to make them all full-sized
+// integers. It should be safe to freely rearrange the structures or change
+// the sizes larger--nothing relies on silently truncating etc., nor the
+// order of variables.
+
+#define FAST_HUFFMAN_TABLE_SIZE   (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH)
+#define FAST_HUFFMAN_TABLE_MASK   (FAST_HUFFMAN_TABLE_SIZE - 1)
+
+typedef struct
+{
+   int dimensions, entries;
+   uint8 *codeword_lengths;
+   float  minimum_value;
+   float  delta_value;
+   uint8  value_bits;
+   uint8  lookup_type;
+   uint8  sequence_p;
+   uint8  sparse;
+   uint32 lookup_values;
+   codetype *multiplicands;
+   uint32 *codewords;
+   #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+    int16  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+   #else
+    int32  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+   #endif
+   uint32 *sorted_codewords;
+   int    *sorted_values;
+   int     sorted_entries;
+} Codebook;
+
+typedef struct
+{
+   uint8 order;
+   uint16 rate;
+   uint16 bark_map_size;
+   uint8 amplitude_bits;
+   uint8 amplitude_offset;
+   uint8 number_of_books;
+   uint8 book_list[16]; // varies
+} Floor0;
+
+typedef struct
+{
+   uint8 partitions;
+   uint8 partition_class_list[32]; // varies
+   uint8 class_dimensions[16]; // varies
+   uint8 class_subclasses[16]; // varies
+   uint8 class_masterbooks[16]; // varies
+   int16 subclass_books[16][8]; // varies
+   uint16 Xlist[31*8+2]; // varies
+   uint8 sorted_order[31*8+2];
+   uint8 neighbors[31*8+2][2];
+   uint8 floor1_multiplier;
+   uint8 rangebits;
+   int values;
+} Floor1;
+
+typedef union
+{
+   Floor0 floor0;
+   Floor1 floor1;
+} Floor;
+
+typedef struct
+{
+   uint32 begin, end;
+   uint32 part_size;
+   uint8 classifications;
+   uint8 classbook;
+   uint8 **classdata;
+   int16 (*residue_books)[8];
+} Residue;
+
+typedef struct
+{
+   uint8 magnitude;
+   uint8 angle;
+   uint8 mux;
+} MappingChannel;
+
+typedef struct
+{
+   uint16 coupling_steps;
+   MappingChannel *chan;
+   uint8  submaps;
+   uint8  submap_floor[15]; // varies
+   uint8  submap_residue[15]; // varies
+} Mapping;
+
+typedef struct
+{
+   uint8 blockflag;
+   uint8 mapping;
+   uint16 windowtype;
+   uint16 transformtype;
+} Mode;
+
+typedef struct
+{
+   uint32  goal_crc;    // expected crc if match
+   int     bytes_left;  // bytes left in packet
+   uint32  crc_so_far;  // running crc
+   int     bytes_done;  // bytes processed in _current_ chunk
+   uint32  sample_loc;  // granule pos encoded in page
+} CRCscan;
+
+typedef struct
+{
+   uint32 page_start, page_end;
+   uint32 after_previous_page_start;
+   uint32 first_decoded_sample;
+   uint32 last_decoded_sample;
+} ProbedPage;
+
+struct stb_vorbis
+{
+  // user-accessible info
+   unsigned int sample_rate;
+   int channels;
+
+   unsigned int setup_memory_required;
+   unsigned int temp_memory_required;
+   unsigned int setup_temp_memory_required;
+
+  // input config
+#ifndef STB_VORBIS_NO_STDIO
+   FILE *f;
+   uint32 f_start;
+   int close_on_free;
+#endif
+
+   uint8 *stream;
+   uint8 *stream_start;
+   uint8 *stream_end;
+
+   uint32 stream_len;
+
+   uint8  push_mode;
+
+   uint32 first_audio_page_offset;
+
+   ProbedPage p_first, p_last;
+
+  // memory management
+   stb_vorbis_alloc alloc;
+   int setup_offset;
+   int temp_offset;
+
+  // run-time results
+   int eof;
+   enum STBVorbisError error;
+
+  // user-useful data
+
+  // header info
+   int blocksize[2];
+   int blocksize_0, blocksize_1;
+   int codebook_count;
+   Codebook *codebooks;
+   int floor_count;
+   uint16 floor_types[64]; // varies
+   Floor *floor_config;
+   int residue_count;
+   uint16 residue_types[64]; // varies
+   Residue *residue_config;
+   int mapping_count;
+   Mapping *mapping;
+   int mode_count;
+   Mode mode_config[64];  // varies
+
+   uint32 total_samples;
+
+  // decode buffer
+   float *channel_buffers[STB_VORBIS_MAX_CHANNELS];
+   float *outputs        [STB_VORBIS_MAX_CHANNELS];
+
+   float *previous_window[STB_VORBIS_MAX_CHANNELS];
+   int previous_length;
+
+   #ifndef STB_VORBIS_NO_DEFER_FLOOR
+   int16 *finalY[STB_VORBIS_MAX_CHANNELS];
+   #else
+   float *floor_buffers[STB_VORBIS_MAX_CHANNELS];
+   #endif
+
+   uint32 current_loc; // sample location of next frame to decode
+   int    current_loc_valid;
+
+  // per-blocksize precomputed data
+   
+   // twiddle factors
+   float *A[2],*B[2],*C[2];
+   float *window[2];
+   uint16 *bit_reverse[2];
+
+  // current page/packet/segment streaming info
+   uint32 serial; // stream serial number for verification
+   int last_page;
+   int segment_count;
+   uint8 segments[255];
+   uint8 page_flag;
+   uint8 bytes_in_seg;
+   uint8 first_decode;
+   int next_seg;
+   int last_seg;  // flag that we're on the last segment
+   int last_seg_which; // what was the segment number of the last seg?
+   uint32 acc;
+   int valid_bits;
+   int packet_bytes;
+   int end_seg_with_known_loc;
+   uint32 known_loc_for_packet;
+   int discard_samples_deferred;
+   uint32 samples_output;
+
+  // push mode scanning
+   int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+   CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT];
+#endif
+
+  // sample-access
+   int channel_buffer_start;
+   int channel_buffer_end;
+};
+
+#if defined(STB_VORBIS_NO_PUSHDATA_API)
+   #define IS_PUSH_MODE(f)   FALSE
+#elif defined(STB_VORBIS_NO_PULLDATA_API)
+   #define IS_PUSH_MODE(f)   TRUE
+#else
+   #define IS_PUSH_MODE(f)   ((f)->push_mode)
+#endif
+
+typedef struct stb_vorbis vorb;
+
+static int error(vorb *f, enum STBVorbisError e)
+{
+   f->error = e;
+   if (!f->eof && e != VORBIS_need_more_data) {
+      f->error=e; // breakpoint for debugging
+   }
+   return 0;
+}
+
+
+// these functions are used for allocating temporary memory
+// while decoding. if you can afford the stack space, use
+// alloca(); otherwise, provide a temp buffer and it will
+// allocate out of those.
+
+#define array_size_required(count,size)  (count*(sizeof(void *)+(size)))
+
+#define temp_alloc(f,size)              (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size))
+#ifdef dealloca
+#define temp_free(f,p)                  (f->alloc.alloc_buffer ? 0 : dealloca(size))
+#else
+#define temp_free(f,p)                  0
+#endif
+#define temp_alloc_save(f)              ((f)->temp_offset)
+#define temp_alloc_restore(f,p)         ((f)->temp_offset = (p))
+
+#define temp_block_array(f,count,size)  make_block_array(temp_alloc(f,array_size_required(count,size)), count, size)
+
+// given a sufficiently large block of memory, make an array of pointers to subblocks of it
+static void *make_block_array(void *mem, int count, int size)
+{
+   int i;
+   void ** p = (void **) mem;
+   char *q = (char *) (p + count);
+   for (i=0; i < count; ++i) {
+      p[i] = q;
+      q += size;
+   }
+   return p;
+}
+
+static void *setup_malloc(vorb *f, int sz)
+{
+   sz = (sz+3) & ~3;
+   f->setup_memory_required += sz;
+   if (f->alloc.alloc_buffer) {
+      void *p = (char *) f->alloc.alloc_buffer + f->setup_offset;
+      if (f->setup_offset + sz > f->temp_offset) return NULL;
+      f->setup_offset += sz;
+      return p;
+   }
+   return sz ? malloc(sz) : NULL;
+}
+
+static void setup_free(vorb *f, void *p)
+{
+   if (f->alloc.alloc_buffer) return; // do nothing; setup mem is not a stack
+   free(p);
+}
+
+static void *setup_temp_malloc(vorb *f, int sz)
+{
+   sz = (sz+3) & ~3;
+   if (f->alloc.alloc_buffer) {
+      if (f->temp_offset - sz < f->setup_offset) return NULL;
+      f->temp_offset -= sz;
+      return (char *) f->alloc.alloc_buffer + f->temp_offset;
+   }
+   return malloc(sz);
+}
+
+static void setup_temp_free(vorb *f, void *p, size_t sz)
+{
+   if (f->alloc.alloc_buffer) {
+      f->temp_offset += (sz+3)&~3;
+      return;
+   }
+   free(p);
+}
+
+#define CRC32_POLY    0x04c11db7   // from spec
+
+static uint32 crc_table[256];
+static void crc32_init(void)
+{
+   int i,j;
+   uint32 s;
+   for(i=0; i < 256; i++) {
+      for (s=i<<24, j=0; j < 8; ++j)
+         s = (s << 1) ^ (s >= (1<<31) ? CRC32_POLY : 0);
+      crc_table[i] = s;
+   }
+}
+
+static __forceinline uint32 crc32_update(uint32 crc, uint8 byte)
+{
+   return (crc << 8) ^ crc_table[byte ^ (crc >> 24)];
+}
+
+
+// used in setup, and for huffman that doesn't go fast path
+static unsigned int bit_reverse(unsigned int n)
+{
+  n = ((n & 0xAAAAAAAA) >>  1) | ((n & 0x55555555) << 1);
+  n = ((n & 0xCCCCCCCC) >>  2) | ((n & 0x33333333) << 2);
+  n = ((n & 0xF0F0F0F0) >>  4) | ((n & 0x0F0F0F0F) << 4);
+  n = ((n & 0xFF00FF00) >>  8) | ((n & 0x00FF00FF) << 8);
+  return (n >> 16) | (n << 16);
+}
+
+static float square(float x)
+{
+   return x*x;
+}
+
+// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3
+// as required by the specification. fast(?) implementation from stb.h
+// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup
+static int ilog(int32 n)
+{
+   static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 };
+
+   // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29)
+   if (n < (1U << 14))
+        if (n < (1U <<  4))        return     0 + log2_4[n      ];
+        else if (n < (1U <<  9))      return  5 + log2_4[n >>  5];
+             else                     return 10 + log2_4[n >> 10];
+   else if (n < (1U << 24))
+             if (n < (1U << 19))      return 15 + log2_4[n >> 15];
+             else                     return 20 + log2_4[n >> 20];
+        else if (n < (1U << 29))      return 25 + log2_4[n >> 25];
+             else if (n < (1U << 31)) return 30 + log2_4[n >> 30];
+                  else                return 0; // signed n returns 0
+}
+
+#ifndef M_PI
+  #define M_PI  3.14159265358979323846264f  // from CRC
+#endif
+
+// code length assigned to a value with no huffman encoding
+#define NO_CODE   255
+
+/////////////////////// LEAF SETUP FUNCTIONS //////////////////////////
+//
+// these functions are only called at setup, and only a few times
+// per file
+
+static float float32_unpack(uint32 x)
+{
+   // from the specification
+   uint32 mantissa = x & 0x1fffff;
+   uint32 sign = x & 0x80000000;
+   uint32 exp = (x & 0x7fe00000) >> 21;
+   double res = sign ? -(double)mantissa : (double)mantissa;
+   return (float) ldexp((float)res, exp-788);
+}
+
+
+// zlib & jpeg huffman tables assume that the output symbols
+// can either be arbitrarily arranged, or have monotonically
+// increasing frequencies--they rely on the lengths being sorted;
+// this makes for a very simple generation algorithm.
+// vorbis allows a huffman table with non-sorted lengths. This
+// requires a more sophisticated construction, since symbols in
+// order do not map to huffman codes "in order".
+static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values)
+{
+   if (!c->sparse) {
+      c->codewords      [symbol] = huff_code;
+   } else {
+      c->codewords       [count] = huff_code;
+      c->codeword_lengths[count] = len;
+      values             [count] = symbol;
+   }
+}
+
+static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
+{
+   int i,k,m=0;
+   uint32 available[32];
+
+   memset(available, 0, sizeof(available));
+   // find the first entry
+   for (k=0; k < n; ++k) if (len[k] < NO_CODE) break;
+   if (k == n) { assert(c->sorted_entries == 0); return TRUE; }
+   // add to the list
+   add_entry(c, 0, k, m++, len[k], values);
+   // add all available leaves
+   for (i=1; i <= len[k]; ++i)
+      available[i] = 1 << (32-i);
+   // note that the above code treats the first case specially,
+   // but it's really the same as the following code, so they
+   // could probably be combined (except the initial code is 0,
+   // and I use 0 in available[] to mean 'empty')
+   for (i=k+1; i < n; ++i) {
+      uint32 res;
+      int z = len[i], y;
+      if (z == NO_CODE) continue;
+      // find lowest available leaf (should always be earliest,
+      // which is what the specification calls for)
+      // note that this property, and the fact we can never have
+      // more than one free leaf at a given level, isn't totally
+      // trivial to prove, but it seems true and the assert never
+      // fires, so!
+      while (z > 0 && !available[z]) --z;
+      if (z == 0) { assert(0); return FALSE; }
+      res = available[z];
+      available[z] = 0;
+      add_entry(c, bit_reverse(res), i, m++, len[i], values);
+      // propogate availability up the tree
+      if (z != len[i]) {
+         for (y=len[i]; y > z; --y) {
+            assert(available[y] == 0);
+            available[y] = res + (1 << (32-y));
+         }
+      }
+   }
+   return TRUE;
+}
+
+// accelerated huffman table allows fast O(1) match of all symbols
+// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH
+static void compute_accelerated_huffman(Codebook *c)
+{
+   int i, len;
+   for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i)
+      c->fast_huffman[i] = -1;
+
+   len = c->sparse ? c->sorted_entries : c->entries;
+   #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+   if (len > 32767) len = 32767; // largest possible value we can encode!
+   #endif
+   for (i=0; i < len; ++i) {
+      if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) {
+         uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i];
+         // set table entries for all bit combinations in the higher bits
+         while (z < FAST_HUFFMAN_TABLE_SIZE) {
+             c->fast_huffman[z] = i;
+             z += 1 << c->codeword_lengths[i];
+         }
+      }
+   }
+}
+
+static int uint32_compare(const void *p, const void *q)
+{
+   uint32 x = * (uint32 *) p;
+   uint32 y = * (uint32 *) q;
+   return x < y ? -1 : x > y;
+}
+
+static int include_in_sort(Codebook *c, uint8 len)
+{
+   if (c->sparse) { assert(len != NO_CODE); return TRUE; }
+   if (len == NO_CODE) return FALSE;
+   if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE;
+   return FALSE;
+}
+
+// if the fast table above doesn't work, we want to binary
+// search them... need to reverse the bits
+static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values)
+{
+   int i, len;
+   // build a list of all the entries
+   // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN.
+   // this is kind of a frivolous optimization--I don't see any performance improvement,
+   // but it's like 4 extra lines of code, so.
+   if (!c->sparse) {
+      int k = 0;
+      for (i=0; i < c->entries; ++i)
+         if (include_in_sort(c, lengths[i])) 
+            c->sorted_codewords[k++] = bit_reverse(c->codewords[i]);
+      assert(k == c->sorted_entries);
+   } else {
+      for (i=0; i < c->sorted_entries; ++i)
+         c->sorted_codewords[i] = bit_reverse(c->codewords[i]);
+   }
+
+   qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare);
+   c->sorted_codewords[c->sorted_entries] = 0xffffffff;
+
+   len = c->sparse ? c->sorted_entries : c->entries;
+   // now we need to indicate how they correspond; we could either
+   //   #1: sort a different data structure that says who they correspond to
+   //   #2: for each sorted entry, search the original list to find who corresponds
+   //   #3: for each original entry, find the sorted entry
+   // #1 requires extra storage, #2 is slow, #3 can use binary search!
+   for (i=0; i < len; ++i) {
+      int huff_len = c->sparse ? lengths[values[i]] : lengths[i];
+      if (include_in_sort(c,huff_len)) {
+         uint32 code = bit_reverse(c->codewords[i]);
+         int x=0, n=c->sorted_entries;
+         while (n > 1) {
+            // invariant: sc[x] <= code < sc[x+n]
+            int m = x + (n >> 1);
+            if (c->sorted_codewords[m] <= code) {
+               x = m;
+               n -= (n>>1);
+            } else {
+               n >>= 1;
+            }
+         }
+         assert(c->sorted_codewords[x] == code);
+         if (c->sparse) {
+            c->sorted_values[x] = values[i];
+            c->codeword_lengths[x] = huff_len;
+         } else {
+            c->sorted_values[x] = i;
+         }
+      }
+   }
+}
+
+// only run while parsing the header (3 times)
+static int vorbis_validate(uint8 *data)
+{
+   static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' };
+   return memcmp(data, vorbis, 6) == 0;
+}
+
+// called from setup only, once per code book
+// (formula implied by specification)
+static int lookup1_values(int entries, int dim)
+{
+   int r = (int) floor(exp((float) log((float) entries) / dim));
+   if ((int) floor(pow((float) r+1, dim)) <= entries)   // (int) cast for MinGW warning;
+      ++r;                                              // floor() to avoid _ftol() when non-CRT
+   assert(pow((float) r+1, dim) > entries);
+   assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above
+   return r;
+}
+
+// called twice per file
+static void compute_twiddle_factors(int n, float *A, float *B, float *C)
+{
+   int n4 = n >> 2, n8 = n >> 3;
+   int k,k2;
+
+   for (k=k2=0; k < n4; ++k,k2+=2) {
+      A[k2  ] = (float)  cos(4*k*M_PI/n);
+      A[k2+1] = (float) -sin(4*k*M_PI/n);
+      B[k2  ] = (float)  cos((k2+1)*M_PI/n/2) * 0.5f;
+      B[k2+1] = (float)  sin((k2+1)*M_PI/n/2) * 0.5f;
+   }
+   for (k=k2=0; k < n8; ++k,k2+=2) {
+      C[k2  ] = (float)  cos(2*(k2+1)*M_PI/n);
+      C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
+   }
+}
+
+static void compute_window(int n, float *window)
+{
+   int n2 = n >> 1, i;
+   for (i=0; i < n2; ++i)
+      window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI)));
+}
+
+static void compute_bitreverse(int n, uint16 *rev)
+{
+   int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+   int i, n8 = n >> 3;
+   for (i=0; i < n8; ++i)
+      rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2;
+}
+
+static int init_blocksize(vorb *f, int b, int n)
+{
+   int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3;
+   f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+   f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+   f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4);
+   if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem);
+   compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]);
+   f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+   if (!f->window[b]) return error(f, VORBIS_outofmem);
+   compute_window(n, f->window[b]);
+   f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8);
+   if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem);
+   compute_bitreverse(n, f->bit_reverse[b]);
+   return TRUE;
+}
+
+static void neighbors(uint16 *x, int n, int *plow, int *phigh)
+{
+   int low = -1;
+   int high = 65536;
+   int i;
+   for (i=0; i < n; ++i) {
+      if (x[i] > low  && x[i] < x[n]) { *plow  = i; low = x[i]; }
+      if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; }
+   }
+}
+
+// this has been repurposed so y is now the original index instead of y
+typedef struct
+{
+   uint16 x,y;
+} Point;
+
+int point_compare(const void *p, const void *q)
+{
+   Point *a = (Point *) p;
+   Point *b = (Point *) q;
+   return a->x < b->x ? -1 : a->x > b->x;
+}
+
+//
+/////////////////////// END LEAF SETUP FUNCTIONS //////////////////////////
+
+
+#if defined(STB_VORBIS_NO_STDIO)
+   #define USE_MEMORY(z)    TRUE
+#else
+   #define USE_MEMORY(z)    ((z)->stream)
+#endif
+
+static uint8 get8(vorb *z)
+{
+   if (USE_MEMORY(z)) {
+      if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; }
+      return *z->stream++;
+   }
+
+   #ifndef STB_VORBIS_NO_STDIO
+   {
+   int c = fgetc(z->f);
+   if (c == EOF) { z->eof = TRUE; return 0; }
+   return c;
+   }
+   #endif
+}
+
+static uint32 get32(vorb *f)
+{
+   uint32 x;
+   x = get8(f);
+   x += get8(f) << 8;
+   x += get8(f) << 16;
+   x += get8(f) << 24;
+   return x;
+}
+
+static int getn(vorb *z, uint8 *data, int n)
+{
+   if (USE_MEMORY(z)) {
+      if (z->stream+n > z->stream_end) { z->eof = 1; return 0; }
+      memcpy(data, z->stream, n);
+      z->stream += n;
+      return 1;
+   }
+
+   #ifndef STB_VORBIS_NO_STDIO   
+   if (fread(data, n, 1, z->f) == 1)
+      return 1;
+   else {
+      z->eof = 1;
+      return 0;
+   }
+   #endif
+}
+
+static void skip(vorb *z, int n)
+{
+   if (USE_MEMORY(z)) {
+      z->stream += n;
+      if (z->stream >= z->stream_end) z->eof = 1;
+      return;
+   }
+   #ifndef STB_VORBIS_NO_STDIO
+   {
+      long x = ftell(z->f);
+      fseek(z->f, x+n, SEEK_SET);
+   }
+   #endif
+}
+
+static int set_file_offset(stb_vorbis *f, unsigned int loc)
+{
+   #ifndef STB_VORBIS_NO_PUSHDATA_API
+   if (f->push_mode) return 0;
+   #endif
+   f->eof = 0;
+   if (USE_MEMORY(f)) {
+      if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) {
+         f->stream = f->stream_end;
+         f->eof = 1;
+         return 0;
+      } else {
+         f->stream = f->stream_start + loc;
+         return 1;
+      }
+   }
+   #ifndef STB_VORBIS_NO_STDIO
+   if (loc + f->f_start < loc || loc >= 0x80000000) {
+      loc = 0x7fffffff;
+      f->eof = 1;
+   } else {
+      loc += f->f_start;
+   }
+   if (!fseek(f->f, loc, SEEK_SET))
+      return 1;
+   f->eof = 1;
+   fseek(f->f, f->f_start, SEEK_END);
+   return 0;
+   #endif
+}
+
+
+static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 };
+
+static int capture_pattern(vorb *f)
+{
+   if (0x4f != get8(f)) return FALSE;
+   if (0x67 != get8(f)) return FALSE;
+   if (0x67 != get8(f)) return FALSE;
+   if (0x53 != get8(f)) return FALSE;
+   return TRUE;
+}
+
+#define PAGEFLAG_continued_packet   1
+#define PAGEFLAG_first_page         2
+#define PAGEFLAG_last_page          4
+
+static int start_page_no_capturepattern(vorb *f)
+{
+   uint32 loc0,loc1,n,i;
+   // stream structure version
+   if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version);
+   // header flag
+   f->page_flag = get8(f);
+   // absolute granule position
+   loc0 = get32(f); 
+   loc1 = get32(f);
+   // @TODO: validate loc0,loc1 as valid positions?
+   // stream serial number -- vorbis doesn't interleave, so discard
+   get32(f);
+   //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number);
+   // page sequence number
+   n = get32(f);
+   f->last_page = n;
+   // CRC32
+   get32(f);
+   // page_segments
+   f->segment_count = get8(f);
+   if (!getn(f, f->segments, f->segment_count))
+      return error(f, VORBIS_unexpected_eof);
+   // assume we _don't_ know any the sample position of any segments
+   f->end_seg_with_known_loc = -2;
+   if (loc0 != ~0 || loc1 != ~0) {
+      // determine which packet is the last one that will complete
+      for (i=f->segment_count-1; i >= 0; --i)
+         if (f->segments[i] < 255)
+            break;
+      // 'i' is now the index of the _last_ segment of a packet that ends
+      if (i >= 0) {
+         f->end_seg_with_known_loc = i;
+         f->known_loc_for_packet   = loc0;
+      }
+   }
+   if (f->first_decode) {
+      int i,len;
+      ProbedPage p;
+      len = 0;
+      for (i=0; i < f->segment_count; ++i)
+         len += f->segments[i];
+      len += 27 + f->segment_count;
+      p.page_start = f->first_audio_page_offset;
+      p.page_end = p.page_start + len;
+      p.after_previous_page_start = p.page_start;
+      p.first_decoded_sample = 0;
+      p.last_decoded_sample = loc0;
+      f->p_first = p;
+   }
+   f->next_seg = 0;
+   return TRUE;
+}
+
+static int start_page(vorb *f)
+{
+   if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern);
+   return start_page_no_capturepattern(f);
+}
+
+static int start_packet(vorb *f)
+{
+   while (f->next_seg == -1) {
+      if (!start_page(f)) return FALSE;
+      if (f->page_flag & PAGEFLAG_continued_packet)
+         return error(f, VORBIS_continued_packet_flag_invalid);
+   }
+   f->last_seg = FALSE;
+   f->valid_bits = 0;
+   f->packet_bytes = 0;
+   f->bytes_in_seg = 0;
+   // f->next_seg is now valid
+   return TRUE;
+}
+
+static int maybe_start_packet(vorb *f)
+{
+   if (f->next_seg == -1) {
+      int x = get8(f);
+      if (f->eof) return FALSE; // EOF at page boundary is not an error!
+      if (0x4f != x      ) return error(f, VORBIS_missing_capture_pattern);
+      if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+      if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+      if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+      if (!start_page_no_capturepattern(f)) return FALSE;
+      if (f->page_flag & PAGEFLAG_continued_packet) {
+         // set up enough state that we can read this packet if we want,
+         // e.g. during recovery
+         f->last_seg = FALSE;
+         f->bytes_in_seg = 0;
+         return error(f, VORBIS_continued_packet_flag_invalid);
+      }
+   }
+   return start_packet(f);
+}
+
+static int next_segment(vorb *f)
+{
+   int len;
+   if (f->last_seg) return 0;
+   if (f->next_seg == -1) {
+      f->last_seg_which = f->segment_count-1; // in case start_page fails
+      if (!start_page(f)) { f->last_seg = 1; return 0; }
+      if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid);
+   }
+   len = f->segments[f->next_seg++];
+   if (len < 255) {
+      f->last_seg = TRUE;
+      f->last_seg_which = f->next_seg-1;
+   }
+   if (f->next_seg >= f->segment_count)
+      f->next_seg = -1;
+   assert(f->bytes_in_seg == 0);
+   f->bytes_in_seg = len;
+   return len;
+}
+
+#define EOP    (-1)
+#define INVALID_BITS  (-1)
+
+static int get8_packet_raw(vorb *f)
+{
+   if (!f->bytes_in_seg) {
+      if (f->last_seg) return EOP;
+      else if (!next_segment(f)) return EOP;
+	}
+   assert(f->bytes_in_seg > 0);
+   --f->bytes_in_seg;
+   ++f->packet_bytes;
+   return get8(f);
+}
+
+static int get8_packet(vorb *f)
+{
+   int x = get8_packet_raw(f);
+   f->valid_bits = 0;
+   return x;
+}
+
+static void flush_packet(vorb *f)
+{
+   while (get8_packet_raw(f) != EOP);
+}
+
+// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important
+// as the huffman decoder?
+static uint32 get_bits(vorb *f, int n)
+{
+   uint32 z;
+
+   if (f->valid_bits < 0) return 0;
+   if (f->valid_bits < n) {
+      if (n > 24) {
+         // the accumulator technique below would not work correctly in this case
+         z = get_bits(f, 24);
+         z += get_bits(f, n-24) << 24;
+         return z;
+      }
+      if (f->valid_bits == 0) f->acc = 0;
+      while (f->valid_bits < n) {
+         int z = get8_packet_raw(f);
+         if (z == EOP) {
+            f->valid_bits = INVALID_BITS;
+            return 0;
+         }
+         f->acc += z << f->valid_bits;
+         f->valid_bits += 8;
+      }
+   }
+   if (f->valid_bits < 0) return 0;
+   z = f->acc & ((1 << n)-1);
+   f->acc >>= n;
+   f->valid_bits -= n;
+   return z;
+}
++/*
+static int32 get_bits_signed(vorb *f, int n) {
+   uint32 z = get_bits(f, n);
+   if (z & (1 << (n-1)))
+      z += ~((1 << n) - 1);
+   return (int32) z;
+}+*/
+
+// @OPTIMIZE: primary accumulator for huffman
+// expand the buffer to as many bits as possible without reading off end of packet
+// it might be nice to allow f->valid_bits and f->acc to be stored in registers,
+// e.g. cache them locally and decode locally
+static __forceinline void prep_huffman(vorb *f)
+{
+   if (f->valid_bits <= 24) {
+      if (f->valid_bits == 0) f->acc = 0;
+      do {
+         int z;
+         if (f->last_seg && !f->bytes_in_seg) return;
+         z = get8_packet_raw(f);
+         if (z == EOP) return;
+         f->acc += z << f->valid_bits;
+         f->valid_bits += 8;
+      } while (f->valid_bits <= 24);
+   }
+}
+
+enum
+{
+   VORBIS_packet_id = 1,
+   VORBIS_packet_comment = 3,
+   VORBIS_packet_setup = 5,
+};
+
+static int codebook_decode_scalar_raw(vorb *f, Codebook *c)
+{
+   int i;
+   prep_huffman(f);
+
+   assert(c->sorted_codewords || c->codewords);
+   // cases to use binary search: sorted_codewords && !c->codewords
+   //                             sorted_codewords && c->entries > 8
+   if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) {
+      // binary search
+      uint32 code = bit_reverse(f->acc);
+      int x=0, n=c->sorted_entries, len;
+
+      while (n > 1) {
+         // invariant: sc[x] <= code < sc[x+n]
+         int m = x + (n >> 1);
+         if (c->sorted_codewords[m] <= code) {
+            x = m;
+            n -= (n>>1);
+         } else {
+            n >>= 1;
+         }
+      }
+      // x is now the sorted index
+      if (!c->sparse) x = c->sorted_values[x];
+      // x is now sorted index if sparse, or symbol otherwise
+      len = c->codeword_lengths[x];
+      if (f->valid_bits >= len) {
+         f->acc >>= len;
+         f->valid_bits -= len;
+         return x;
+      }
+
+      f->valid_bits = 0;
+      return -1;
+   }
+
+   // if small, linear search
+   assert(!c->sparse);
+   for (i=0; i < c->entries; ++i) {
+      if (c->codeword_lengths[i] == NO_CODE) continue;
+      if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) {
+         if (f->valid_bits >= c->codeword_lengths[i]) {
+            f->acc >>= c->codeword_lengths[i];
+            f->valid_bits -= c->codeword_lengths[i];
+            return i;
+         }
+         f->valid_bits = 0;
+         return -1;
+      }
+   }
+
+   error(f, VORBIS_invalid_stream);
+   f->valid_bits = 0;
+   return -1;
+}
++/*
+static int codebook_decode_scalar(vorb *f, Codebook *c) {
+   int i;
+   if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)
+      prep_huffman(f);
+   // fast huffman table lookup
+   i = f->acc & FAST_HUFFMAN_TABLE_MASK;
+   i = c->fast_huffman[i];
+   if (i >= 0) {
+      f->acc >>= c->codeword_lengths[i];
+      f->valid_bits -= c->codeword_lengths[i];
+      if (f->valid_bits < 0) { f->valid_bits = 0; return -1; }
+      return i;
+   }
+   return codebook_decode_scalar_raw(f,c);
+}
+*/+
+#ifndef STB_VORBIS_NO_INLINE_DECODE
+
+#define DECODE_RAW(var, f,c)                                  \
+   if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)        \
+      prep_huffman(f);                                        \
+   var = f->acc & FAST_HUFFMAN_TABLE_MASK;                    \
+   var = c->fast_huffman[var];                                \
+   if (var >= 0) {                                            \
+      int n = c->codeword_lengths[var];                       \
+      f->acc >>= n;                                           \
+      f->valid_bits -= n;                                     \
+      if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \
+   } else {                                                   \
+      var = codebook_decode_scalar_raw(f,c);                  \
+   }
+
+#else
+
+#define DECODE_RAW(var,f,c)    var = codebook_decode_scalar(f,c);
+
+#endif
+
+#define DECODE(var,f,c)                                       \
+   DECODE_RAW(var,f,c)                                        \
+   if (c->sparse) var = c->sorted_values[var];
+
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+  #define DECODE_VQ(var,f,c)   DECODE_RAW(var,f,c)
+#else
+  #define DECODE_VQ(var,f,c)   DECODE(var,f,c)
+#endif
+
+
+
+
+
+
+// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case
+// where we avoid one addition
+#ifndef STB_VORBIS_CODEBOOK_FLOATS
+   #define CODEBOOK_ELEMENT(c,off)          (c->multiplicands[off] * c->delta_value + c->minimum_value)
+   #define CODEBOOK_ELEMENT_FAST(c,off)     (c->multiplicands[off] * c->delta_value)
+   #define CODEBOOK_ELEMENT_BASE(c)         (c->minimum_value)
+#else
+   #define CODEBOOK_ELEMENT(c,off)          (c->multiplicands[off])
+   #define CODEBOOK_ELEMENT_FAST(c,off)     (c->multiplicands[off])
+   #define CODEBOOK_ELEMENT_BASE(c)         (0)
+#endif
+
+static int codebook_decode_start(vorb *f, Codebook *c, int len)
+{
+   int z = -1;
+
+   // type 0 is only legal in a scalar context
+   if (c->lookup_type == 0)
+      error(f, VORBIS_invalid_stream);
+   else {
+      DECODE_VQ(z,f,c);
+      if (c->sparse) assert(z < c->sorted_entries);
+      if (z < 0) {  // check for EOP
+         if (!f->bytes_in_seg)
+            if (f->last_seg)
+               return z;
+         error(f, VORBIS_invalid_stream);
+      }
+   }
+   return z;
+}
+
+static int codebook_decode(vorb *f, Codebook *c, float *output, int len)
+{
+   int i,z = codebook_decode_start(f,c,len);
+   if (z < 0) return FALSE;
+   if (len > c->dimensions) len = c->dimensions;
+
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+   if (c->lookup_type == 1) {
+      float last = CODEBOOK_ELEMENT_BASE(c);
+      int div = 1;
+      for (i=0; i < len; ++i) {
+         int off = (z / div) % c->lookup_values;
+         float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
+         output[i] += val;
+         if (c->sequence_p) last = val + c->minimum_value;
+         div *= c->lookup_values;
+      }
+      return TRUE;
+   }
+#endif
+
+   z *= c->dimensions;
+   if (c->sequence_p) {
+      float last = CODEBOOK_ELEMENT_BASE(c);
+      for (i=0; i < len; ++i) {
+         float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+         output[i] += val;
+         last = val + c->minimum_value;
+      }
+   } else {
+      float last = CODEBOOK_ELEMENT_BASE(c);
+      for (i=0; i < len; ++i) {
+         output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+      }
+   }
+
+   return TRUE;
+}
+
+static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step)
+{
+   int i,z = codebook_decode_start(f,c,len);
+   float last = CODEBOOK_ELEMENT_BASE(c);
+   if (z < 0) return FALSE;
+   if (len > c->dimensions) len = c->dimensions;
+
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+   if (c->lookup_type == 1) {
+      int div = 1;
+      for (i=0; i < len; ++i) {
+         int off = (z / div) % c->lookup_values;
+         float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
+         output[i*step] += val;
+         if (c->sequence_p) last = val;
+         div *= c->lookup_values;
+      }
+      return TRUE;
+   }
+#endif
+
+   z *= c->dimensions;
+   for (i=0; i < len; ++i) {
+      float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+      output[i*step] += val;
+      if (c->sequence_p) last = val;
+   }
+
+   return TRUE;
+}
+
+static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode)
+{
+   int c_inter = *c_inter_p;
+   int p_inter = *p_inter_p;
+   int i,z, effective = c->dimensions;
+
+   // type 0 is only legal in a scalar context
+   if (c->lookup_type == 0)   return error(f, VORBIS_invalid_stream);
+
+   while (total_decode > 0) {
+      float last = CODEBOOK_ELEMENT_BASE(c);
+      DECODE_VQ(z,f,c);
+      #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+      assert(!c->sparse || z < c->sorted_entries);
+      #endif
+      if (z < 0) {
+         if (!f->bytes_in_seg)
+            if (f->last_seg) return FALSE;
+         return error(f, VORBIS_invalid_stream);
+      }
+
+      // if this will take us off the end of the buffers, stop short!
+      // we check by computing the length of the virtual interleaved
+      // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter),
+      // and the length we'll be using (effective)
+      if (c_inter + p_inter*ch + effective > len * ch) {
+         effective = len*ch - (p_inter*ch - c_inter);
+      }
+
+   #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+      if (c->lookup_type == 1) {
+         int div = 1;
+         for (i=0; i < effective; ++i) {
+            int off = (z / div) % c->lookup_values;
+            float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
+            outputs[c_inter][p_inter] += val;
+            if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+            if (c->sequence_p) last = val;
+            div *= c->lookup_values;
+         }
+      } else
+   #endif
+      {
+         z *= c->dimensions;
+         if (c->sequence_p) {
+            for (i=0; i < effective; ++i) {
+               float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+               outputs[c_inter][p_inter] += val;
+               if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+               last = val;
+            }
+         } else {
+            for (i=0; i < effective; ++i) {
+               float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+               outputs[c_inter][p_inter] += val;
+               if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+            }
+         }
+      }
+
+      total_decode -= effective;
+   }
+   *c_inter_p = c_inter;
+   *p_inter_p = p_inter;
+   return TRUE;
+}
+
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+static int codebook_decode_deinterleave_repeat_2(vorb *f, Codebook *c, float **outputs, int *c_inter_p, int *p_inter_p, int len, int total_decode)
+{
+   int c_inter = *c_inter_p;
+   int p_inter = *p_inter_p;
+   int i,z, effective = c->dimensions;
+
+   // type 0 is only legal in a scalar context
+   if (c->lookup_type == 0)   return error(f, VORBIS_invalid_stream);
+
+   while (total_decode > 0) {
+      float last = CODEBOOK_ELEMENT_BASE(c);
+      DECODE_VQ(z,f,c);
+
+      if (z < 0) {
+         if (!f->bytes_in_seg)
+            if (f->last_seg) return FALSE;
+         return error(f, VORBIS_invalid_stream);
+      }
+
+      // if this will take us off the end of the buffers, stop short!
+      // we check by computing the length of the virtual interleaved
+      // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter),
+      // and the length we'll be using (effective)
+      if (c_inter + p_inter*2 + effective > len * 2) {
+         effective = len*2 - (p_inter*2 - c_inter);
+      }
+
+      {
+         z *= c->dimensions;
+         if (c->sequence_p) {
+            // haven't optimized this case because I don't have any examples
+            for (i=0; i < effective; ++i) {
+               float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+               outputs[c_inter][p_inter] += val;
+               if (++c_inter == 2) { c_inter = 0; ++p_inter; }
+               last = val;
+            }
+         } else {
+            i=0;
+            if (c_inter == 1) {
+               float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+               outputs[c_inter][p_inter] += val;
+               c_inter = 0; ++p_inter;
+               ++i;
+            }
+            {
+               float *z0 = outputs[0];
+               float *z1 = outputs[1];
+               for (; i+1 < effective;) {
+                  z0[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+                  z1[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i+1) + last;
+                  ++p_inter;
+                  i += 2;
+               }
+            }
+            if (i < effective) {
+               float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+               outputs[c_inter][p_inter] += val;
+               if (++c_inter == 2) { c_inter = 0; ++p_inter; }
+            }
+         }
+      }
+
+      total_decode -= effective;
+   }
+   *c_inter_p = c_inter;
+   *p_inter_p = p_inter;
+   return TRUE;
+}
+#endif
+
+static int predict_point(int x, int x0, int x1, int y0, int y1)
+{
+   int dy = y1 - y0;
+   int adx = x1 - x0;
+   // @OPTIMIZE: force int division to round in the right direction... is this necessary on x86?
+   int err = abs(dy) * (x - x0);
+   int off = err / adx;
+   return dy < 0 ? y0 - off : y0 + off;
+}
+
+// the following table is block-copied from the specification
+static float inverse_db_table[256] =
+{
+  1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f, 
+  1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f, 
+  1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f, 
+  2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f, 
+  2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f, 
+  3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f, 
+  4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f, 
+  6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f, 
+  7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f, 
+  1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f, 
+  1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f, 
+  1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f, 
+  2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f, 
+  2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f, 
+  3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f, 
+  4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f, 
+  5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f, 
+  7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f, 
+  9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f, 
+  1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f, 
+  1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f, 
+  2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f, 
+  2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f, 
+  3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f, 
+  4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f, 
+  5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f, 
+  7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f, 
+  9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f, 
+  0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f, 
+  0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f, 
+  0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f, 
+  0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f, 
+  0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f, 
+  0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f, 
+  0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f, 
+  0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f, 
+  0.00092223983f, 0.00098217216f, 0.0010459992f,  0.0011139742f, 
+  0.0011863665f,  0.0012634633f,  0.0013455702f,  0.0014330129f, 
+  0.0015261382f,  0.0016253153f,  0.0017309374f,  0.0018434235f, 
+  0.0019632195f,  0.0020908006f,  0.0022266726f,  0.0023713743f, 
+  0.0025254795f,  0.0026895994f,  0.0028643847f,  0.0030505286f, 
+  0.0032487691f,  0.0034598925f,  0.0036847358f,  0.0039241906f, 
+  0.0041792066f,  0.0044507950f,  0.0047400328f,  0.0050480668f, 
+  0.0053761186f,  0.0057254891f,  0.0060975636f,  0.0064938176f, 
+  0.0069158225f,  0.0073652516f,  0.0078438871f,  0.0083536271f, 
+  0.0088964928f,  0.009474637f,   0.010090352f,   0.010746080f, 
+  0.011444421f,   0.012188144f,   0.012980198f,   0.013823725f, 
+  0.014722068f,   0.015678791f,   0.016697687f,   0.017782797f, 
+  0.018938423f,   0.020169149f,   0.021479854f,   0.022875735f, 
+  0.024362330f,   0.025945531f,   0.027631618f,   0.029427276f, 
+  0.031339626f,   0.033376252f,   0.035545228f,   0.037855157f, 
+  0.040315199f,   0.042935108f,   0.045725273f,   0.048696758f, 
+  0.051861348f,   0.055231591f,   0.058820850f,   0.062643361f, 
+  0.066714279f,   0.071049749f,   0.075666962f,   0.080584227f, 
+  0.085821044f,   0.091398179f,   0.097337747f,   0.10366330f, 
+  0.11039993f,    0.11757434f,    0.12521498f,    0.13335215f, 
+  0.14201813f,    0.15124727f,    0.16107617f,    0.17154380f, 
+  0.18269168f,    0.19456402f,    0.20720788f,    0.22067342f, 
+  0.23501402f,    0.25028656f,    0.26655159f,    0.28387361f, 
+  0.30232132f,    0.32196786f,    0.34289114f,    0.36517414f, 
+  0.38890521f,    0.41417847f,    0.44109412f,    0.46975890f, 
+  0.50028648f,    0.53279791f,    0.56742212f,    0.60429640f, 
+  0.64356699f,    0.68538959f,    0.72993007f,    0.77736504f, 
+  0.82788260f,    0.88168307f,    0.9389798f,     1.0f
+};
+
+
+// @OPTIMIZE: if you want to replace this bresenham line-drawing routine,
+// note that you must produce bit-identical output to decode correctly;
+// this specific sequence of operations is specified in the spec (it's
+// drawing integer-quantized frequency-space lines that the encoder
+// expects to be exactly the same)
+//     ... also, isn't the whole point of Bresenham's algorithm to NOT
+// have to divide in the setup? sigh.
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+#define LINE_OP(a,b)   a *= b
+#else
+#define LINE_OP(a,b)   a = b
+#endif
+
+#ifdef STB_VORBIS_DIVIDE_TABLE
+#define DIVTAB_NUMER   32
+#define DIVTAB_DENOM   64
+int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB
+#endif
+
+static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n)
+{
+   int dy = y1 - y0;
+   int adx = x1 - x0;
+   int ady = abs(dy);
+   int base;
+   int x=x0,y=y0;
+   int err = 0;
+   int sy;
+
+#ifdef STB_VORBIS_DIVIDE_TABLE
+   if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) {
+      if (dy < 0) {
+         base = -integer_divide_table[ady][adx];
+         sy = base-1;
+      } else {
+         base =  integer_divide_table[ady][adx];
+         sy = base+1;
+      }
+   } else {
+      base = dy / adx;
+      if (dy < 0)
+         sy = base - 1;
+      else
+         sy = base+1;
+   }
+#else
+   base = dy / adx;
+   if (dy < 0)
+      sy = base - 1;
+   else
+      sy = base+1;
+#endif
+   ady -= abs(base) * adx;
+   if (x1 > n) x1 = n;
+   LINE_OP(output[x], inverse_db_table[y]);
+   for (++x; x < x1; ++x) {
+      err += ady;
+      if (err >= adx) {
+         err -= adx;
+         y += sy;
+      } else
+         y += base;
+      LINE_OP(output[x], inverse_db_table[y]);
+   }
+}
+
+static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype)
+{
+   int k;
+   if (rtype == 0) {
+      int step = n / book->dimensions;
+      for (k=0; k < step; ++k)
+         if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step))
+            return FALSE;
+   } else {
+      for (k=0; k < n; ) {
+         if (!codebook_decode(f, book, target+offset, n-k))
+            return FALSE;
+         k += book->dimensions;
+         offset += book->dimensions;
+      }
+   }
+   return TRUE;
+}
+
+static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode)
+{
+   int i,j,pass;
+   Residue *r = f->residue_config + rn;
+   int rtype = f->residue_types[rn];
+   int c = r->classbook;
+   int classwords = f->codebooks[c].dimensions;
+   int n_read = r->end - r->begin;
+   int part_read = n_read / r->part_size;
+   int temp_alloc_point = temp_alloc_save(f);
+   #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+   uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata));
+   #else
+   int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications));
+   #endif
+
+   for (i=0; i < ch; ++i)
+      if (!do_not_decode[i])
+         memset(residue_buffers[i], 0, sizeof(float) * n);
+
+   if (rtype == 2 && ch != 1) {
+      //int len = ch * n;
+      for (j=0; j < ch; ++j)
+         if (!do_not_decode[j])
+            break;
+      if (j == ch)
+         goto done;
+
+      for (pass=0; pass < 8; ++pass) {
+         int pcount = 0, class_set = 0;
+         if (ch == 2) {
+            while (pcount < part_read) {
+               int z = r->begin + pcount*r->part_size;
+               int c_inter = (z & 1), p_inter = z>>1;
+               if (pass == 0) {
+                  Codebook *c = f->codebooks+r->classbook;
+                  int q;
+                  DECODE(q,f,c);
+                  if (q == EOP) goto done;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  part_classdata[0][class_set] = r->classdata[q];
+                  #else
+                  for (i=classwords-1; i >= 0; --i) {
+                     classifications[0][i+pcount] = q % r->classifications;
+                     q /= r->classifications;
+                  }
+                  #endif
+               }
+               for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                  int z = r->begin + pcount*r->part_size;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  int c = part_classdata[0][class_set][i];
+                  #else
+                  int c = classifications[0][pcount];
+                  #endif
+                  int b = r->residue_books[c][pass];
+                  if (b >= 0) {
+                     Codebook *book = f->codebooks + b;
+                     #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+                     if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                        goto done;
+                     #else
+                     // saves 1%
+                     if (!codebook_decode_deinterleave_repeat_2(f, book, residue_buffers, &c_inter, &p_inter, n, r->part_size))
+                        goto done;
+                     #endif
+                  } else {
+                     z += r->part_size;
+                     c_inter = z & 1;
+                     p_inter = z >> 1;
+                  }
+               }
+               #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+               ++class_set;
+               #endif
+            }
+         } else if (ch == 1) {
+            while (pcount < part_read) {
+               int z = r->begin + pcount*r->part_size;
+               int c_inter = 0, p_inter = z;
+               if (pass == 0) {
+                  Codebook *c = f->codebooks+r->classbook;
+                  int q;
+                  DECODE(q,f,c);
+                  if (q == EOP) goto done;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  part_classdata[0][class_set] = r->classdata[q];
+                  #else
+                  for (i=classwords-1; i >= 0; --i) {
+                     classifications[0][i+pcount] = q % r->classifications;
+                     q /= r->classifications;
+                  }
+                  #endif
+               }
+               for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                  int z = r->begin + pcount*r->part_size;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  int c = part_classdata[0][class_set][i];
+                  #else
+                  int c = classifications[0][pcount];
+                  #endif
+                  int b = r->residue_books[c][pass];
+                  if (b >= 0) {
+                     Codebook *book = f->codebooks + b;
+                     if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                        goto done;
+                  } else {
+                     z += r->part_size;
+                     c_inter = 0;
+                     p_inter = z;
+                  }
+               }
+               #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+               ++class_set;
+               #endif
+            }
+         } else {
+            while (pcount < part_read) {
+               int z = r->begin + pcount*r->part_size;
+               int c_inter = z % ch, p_inter = z/ch;
+               if (pass == 0) {
+                  Codebook *c = f->codebooks+r->classbook;
+                  int q;
+                  DECODE(q,f,c);
+                  if (q == EOP) goto done;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  part_classdata[0][class_set] = r->classdata[q];
+                  #else
+                  for (i=classwords-1; i >= 0; --i) {
+                     classifications[0][i+pcount] = q % r->classifications;
+                     q /= r->classifications;
+                  }
+                  #endif
+               }
+               for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                  int z = r->begin + pcount*r->part_size;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  int c = part_classdata[0][class_set][i];
+                  #else
+                  int c = classifications[0][pcount];
+                  #endif
+                  int b = r->residue_books[c][pass];
+                  if (b >= 0) {
+                     Codebook *book = f->codebooks + b;
+                     if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                        goto done;
+                  } else {
+                     z += r->part_size;
+                     c_inter = z % ch;
+                     p_inter = z / ch;
+                  }
+               }
+               #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+               ++class_set;
+               #endif
+            }
+         }
+      }
+      goto done;
+   }
+
+   for (pass=0; pass < 8; ++pass) {
+      int pcount = 0, class_set=0;
+      while (pcount < part_read) {
+         if (pass == 0) {
+            for (j=0; j < ch; ++j) {
+               if (!do_not_decode[j]) {
+                  Codebook *c = f->codebooks+r->classbook;
+                  int temp;
+                  DECODE(temp,f,c);
+                  if (temp == EOP) goto done;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  part_classdata[j][class_set] = r->classdata[temp];
+                  #else
+                  for (i=classwords-1; i >= 0; --i) {
+                     classifications[j][i+pcount] = temp % r->classifications;
+                     temp /= r->classifications;
+                  }
+                  #endif
+               }
+            }
+         }
+         for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+            for (j=0; j < ch; ++j) {
+               if (!do_not_decode[j]) {
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  int c = part_classdata[j][class_set][i];
+                  #else
+                  int c = classifications[j][pcount];
+                  #endif
+                  int b = r->residue_books[c][pass];
+                  if (b >= 0) {
+                     float *target = residue_buffers[j];
+                     int offset = r->begin + pcount * r->part_size;
+                     int n = r->part_size;
+                     Codebook *book = f->codebooks + b;
+                     if (!residue_decode(f, book, target, offset, n, rtype))
+                        goto done;
+                  }
+               }
+            }
+         }
+         #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+         ++class_set;
+         #endif
+      }
+   }
+  done:
+   temp_alloc_restore(f,temp_alloc_point);
+}
+
+
+#if 0
+// slow way for debugging
+void inverse_mdct_slow(float *buffer, int n)
+{
+   int i,j;
+   int n2 = n >> 1;
+   float *x = (float *) malloc(sizeof(*x) * n2);
+   memcpy(x, buffer, sizeof(*x) * n2);
+   for (i=0; i < n; ++i) {
+      float acc = 0;
+      for (j=0; j < n2; ++j)
+         // formula from paper:
+         //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
+         // formula from wikipedia
+         //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
+         // these are equivalent, except the formula from the paper inverts the multiplier!
+         // however, what actually works is NO MULTIPLIER!?!
+         //acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
+         acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
+      buffer[i] = acc;
+   }
+   free(x);
+}
+#elif 0
+// same as above, but just barely able to run in real time on modern machines
+void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
+{
+   float mcos[16384];
+   int i,j;
+   int n2 = n >> 1, nmask = (n << 2) -1;
+   float *x = (float *) malloc(sizeof(*x) * n2);
+   memcpy(x, buffer, sizeof(*x) * n2);
+   for (i=0; i < 4*n; ++i)
+      mcos[i] = (float) cos(M_PI / 2 * i / n);
+
+   for (i=0; i < n; ++i) {
+      float acc = 0;
+      for (j=0; j < n2; ++j)
+         acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask];
+      buffer[i] = acc;
+   }
+   free(x);
+}
+#else
+// transform to use a slow dct-iv; this is STILL basically trivial,
+// but only requires half as many ops
+void dct_iv_slow(float *buffer, int n)
+{
+   float mcos[16384];
+   float x[2048];
+   int i,j;
+   //int n2 = n >> 1;+   int nmask = (n << 3) - 1;
+   memcpy(x, buffer, sizeof(*x) * n);
+   for (i=0; i < 8*n; ++i)
+      mcos[i] = (float) cos(M_PI / 4 * i / n);
+   for (i=0; i < n; ++i) {
+      float acc = 0;
+      for (j=0; j < n; ++j)
+         acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask];
+         //acc += x[j] * cos(M_PI / n * (i + 0.5) * (j + 0.5));
+      buffer[i] = acc;
+   }
+   free(x);
+}
+
+void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
+{
+   int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4;
+   float temp[4096];
+
+   memcpy(temp, buffer, n2 * sizeof(float));
+   dct_iv_slow(temp, n2);  // returns -c'-d, a-b'
+
+   for (i=0; i < n4  ; ++i) buffer[i] = temp[i+n4];            // a-b'
+   for (   ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1];   // b-a', c+d'
+   for (   ; i < n   ; ++i) buffer[i] = -temp[i - n3_4];       // c'+d
+}
+#endif
+
+#ifndef LIBVORBIS_MDCT
+#define LIBVORBIS_MDCT 0
+#endif
+
+#if LIBVORBIS_MDCT
+// directly call the vorbis MDCT using an interface documented
+// by Jeff Roberts... useful for performance comparison
+typedef struct 
+{
+  int n;
+  int log2n;
+  
+  float *trig;
+  int   *bitrev;
+
+  float scale;
+} mdct_lookup;
+
+extern void mdct_init(mdct_lookup *lookup, int n);
+extern void mdct_clear(mdct_lookup *l);
+extern void mdct_backward(mdct_lookup *init, float *in, float *out);
+
+mdct_lookup M1,M2;
+
+void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
+{
+   mdct_lookup *M;
+   if (M1.n == n) M = &M1;
+   else if (M2.n == n) M = &M2;
+   else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; }
+   else { 
+      if (M2.n) __asm int 3;
+      mdct_init(&M2, n);
+      M = &M2;
+   }
+
+   mdct_backward(M, buffer, buffer);
+}
+#endif
+
+
+// the following were split out into separate functions while optimizing;
+// they could be pushed back up but eh. __forceinline showed no change;
+// they're probably already being inlined.
+static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A)
+{
+   float *ee0 = e + i_off;
+   float *ee2 = ee0 + k_off;
+   int i;
+
+   assert((n & 3) == 0);
+   for (i=(n>>2); i > 0; --i) {
+      float k00_20, k01_21;
+      k00_20  = ee0[ 0] - ee2[ 0];
+      k01_21  = ee0[-1] - ee2[-1];
+      ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0];
+      ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1];
+      ee2[ 0] = k00_20 * A[0] - k01_21 * A[1];
+      ee2[-1] = k01_21 * A[0] + k00_20 * A[1];
+      A += 8;
+
+      k00_20  = ee0[-2] - ee2[-2];
+      k01_21  = ee0[-3] - ee2[-3];
+      ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2];
+      ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3];
+      ee2[-2] = k00_20 * A[0] - k01_21 * A[1];
+      ee2[-3] = k01_21 * A[0] + k00_20 * A[1];
+      A += 8;
+
+      k00_20  = ee0[-4] - ee2[-4];
+      k01_21  = ee0[-5] - ee2[-5];
+      ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4];
+      ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5];
+      ee2[-4] = k00_20 * A[0] - k01_21 * A[1];
+      ee2[-5] = k01_21 * A[0] + k00_20 * A[1];
+      A += 8;
+
+      k00_20  = ee0[-6] - ee2[-6];
+      k01_21  = ee0[-7] - ee2[-7];
+      ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6];
+      ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7];
+      ee2[-6] = k00_20 * A[0] - k01_21 * A[1];
+      ee2[-7] = k01_21 * A[0] + k00_20 * A[1];
+      A += 8;
+      ee0 -= 8;
+      ee2 -= 8;
+   }
+}
+
+static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1)
+{
+   int i;
+   float k00_20, k01_21;
+
+   float *e0 = e + d0;
+   float *e2 = e0 + k_off;
+
+   for (i=lim >> 2; i > 0; --i) {
+      k00_20 = e0[-0] - e2[-0];
+      k01_21 = e0[-1] - e2[-1];
+      e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0];
+      e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1];
+      e2[-0] = (k00_20)*A[0] - (k01_21) * A[1];
+      e2[-1] = (k01_21)*A[0] + (k00_20) * A[1];
+
+      A += k1;
+
+      k00_20 = e0[-2] - e2[-2];
+      k01_21 = e0[-3] - e2[-3];
+      e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2];
+      e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3];
+      e2[-2] = (k00_20)*A[0] - (k01_21) * A[1];
+      e2[-3] = (k01_21)*A[0] + (k00_20) * A[1];
+
+      A += k1;
+
+      k00_20 = e0[-4] - e2[-4];
+      k01_21 = e0[-5] - e2[-5];
+      e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4];
+      e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5];
+      e2[-4] = (k00_20)*A[0] - (k01_21) * A[1];
+      e2[-5] = (k01_21)*A[0] + (k00_20) * A[1];
+
+      A += k1;
+
+      k00_20 = e0[-6] - e2[-6];
+      k01_21 = e0[-7] - e2[-7];
+      e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6];
+      e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7];
+      e2[-6] = (k00_20)*A[0] - (k01_21) * A[1];
+      e2[-7] = (k01_21)*A[0] + (k00_20) * A[1];
+
+      e0 -= 8;
+      e2 -= 8;
+
+      A += k1;
+   }
+}
+
+static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0)
+{
+   int i;
+   float A0 = A[0];
+   float A1 = A[0+1];
+   float A2 = A[0+a_off];
+   float A3 = A[0+a_off+1];
+   float A4 = A[0+a_off*2+0];
+   float A5 = A[0+a_off*2+1];
+   float A6 = A[0+a_off*3+0];
+   float A7 = A[0+a_off*3+1];
+
+   float k00,k11;
+
+   float *ee0 = e  +i_off;
+   float *ee2 = ee0+k_off;
+
+   for (i=n; i > 0; --i) {
+      k00     = ee0[ 0] - ee2[ 0];
+      k11     = ee0[-1] - ee2[-1];
+      ee0[ 0] =  ee0[ 0] + ee2[ 0];
+      ee0[-1] =  ee0[-1] + ee2[-1];
+      ee2[ 0] = (k00) * A0 - (k11) * A1;
+      ee2[-1] = (k11) * A0 + (k00) * A1;
+
+      k00     = ee0[-2] - ee2[-2];
+      k11     = ee0[-3] - ee2[-3];
+      ee0[-2] =  ee0[-2] + ee2[-2];
+      ee0[-3] =  ee0[-3] + ee2[-3];
+      ee2[-2] = (k00) * A2 - (k11) * A3;
+      ee2[-3] = (k11) * A2 + (k00) * A3;
+
+      k00     = ee0[-4] - ee2[-4];
+      k11     = ee0[-5] - ee2[-5];
+      ee0[-4] =  ee0[-4] + ee2[-4];
+      ee0[-5] =  ee0[-5] + ee2[-5];
+      ee2[-4] = (k00) * A4 - (k11) * A5;
+      ee2[-5] = (k11) * A4 + (k00) * A5;
+
+      k00     = ee0[-6] - ee2[-6];
+      k11     = ee0[-7] - ee2[-7];
+      ee0[-6] =  ee0[-6] + ee2[-6];
+      ee0[-7] =  ee0[-7] + ee2[-7];
+      ee2[-6] = (k00) * A6 - (k11) * A7;
+      ee2[-7] = (k11) * A6 + (k00) * A7;
+
+      ee0 -= k0;
+      ee2 -= k0;
+   }
+}
+
+static __forceinline void iter_54(float *z)
+{
+   float k00,k11,k22,k33;
+   float y0,y1,y2,y3;
+
+   k00  = z[ 0] - z[-4];
+   y0   = z[ 0] + z[-4];
+   y2   = z[-2] + z[-6];
+   k22  = z[-2] - z[-6];
+
+   z[-0] = y0 + y2;      // z0 + z4 + z2 + z6
+   z[-2] = y0 - y2;      // z0 + z4 - z2 - z6
+
+   // done with y0,y2
+
+   k33  = z[-3] - z[-7];
+
+   z[-4] = k00 + k33;    // z0 - z4 + z3 - z7
+   z[-6] = k00 - k33;    // z0 - z4 - z3 + z7
+
+   // done with k33
+
+   k11  = z[-1] - z[-5];
+   y1   = z[-1] + z[-5];
+   y3   = z[-3] + z[-7];
+
+   z[-1] = y1 + y3;      // z1 + z5 + z3 + z7
+   z[-3] = y1 - y3;      // z1 + z5 - z3 - z7
+   z[-5] = k11 - k22;    // z1 - z5 + z2 - z6
+   z[-7] = k11 + k22;    // z1 - z5 - z2 + z6
+}
+
+static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n) {
+   //int k_off = -8;
+   int a_off = base_n >> 3;
+   float A2 = A[0+a_off];
+   float *z = e + i_off;
+   float *base = z - 16 * n;
+
+   while (z > base) {
+      float k00,k11;
+
+      k00   = z[-0] - z[-8];
+      k11   = z[-1] - z[-9];
+      z[-0] = z[-0] + z[-8];
+      z[-1] = z[-1] + z[-9];
+      z[-8] =  k00;
+      z[-9] =  k11 ;
+
+      k00    = z[ -2] - z[-10];
+      k11    = z[ -3] - z[-11];
+      z[ -2] = z[ -2] + z[-10];
+      z[ -3] = z[ -3] + z[-11];
+      z[-10] = (k00+k11) * A2;
+      z[-11] = (k11-k00) * A2;
+
+      k00    = z[-12] - z[ -4];  // reverse to avoid a unary negation
+      k11    = z[ -5] - z[-13];
+      z[ -4] = z[ -4] + z[-12];
+      z[ -5] = z[ -5] + z[-13];
+      z[-12] = k11;
+      z[-13] = k00;
+
+      k00    = z[-14] - z[ -6];  // reverse to avoid a unary negation
+      k11    = z[ -7] - z[-15];
+      z[ -6] = z[ -6] + z[-14];
+      z[ -7] = z[ -7] + z[-15];
+      z[-14] = (k00+k11) * A2;
+      z[-15] = (k00-k11) * A2;
+
+      iter_54(z);
+      iter_54(z-8);
+      z -= 16;
+   }
+}
+
+static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) {
+   int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
+   //int n3_4 = n - n4;+   int ld;
+   // @OPTIMIZE: reduce register pressure by using fewer variables?
+   int save_point = temp_alloc_save(f);
+   float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2));
+   float *u=NULL,*v=NULL;
+   // twiddle factors
+   float *A = f->A[blocktype];
+
+   // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
+   // See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function.
+
+   // kernel from paper
+
+
+   // merged:
+   //   copy and reflect spectral data
+   //   step 0
+
+   // note that it turns out that the items added together during
+   // this step are, in fact, being added to themselves (as reflected
+   // by step 0). inexplicable inefficiency! this became obvious
+   // once I combined the passes.
+
+   // so there's a missing 'times 2' here (for adding X to itself).
+   // this propogates through linearly to the end, where the numbers
+   // are 1/2 too small, and need to be compensated for.
+
+   {
+      float *d,*e, *AA, *e_stop;
+      d = &buf2[n2-2];
+      AA = A;
+      e = &buffer[0];
+      e_stop = &buffer[n2];
+      while (e != e_stop) {
+         d[1] = (e[0] * AA[0] - e[2]*AA[1]);
+         d[0] = (e[0] * AA[1] + e[2]*AA[0]);
+         d -= 2;
+         AA += 2;
+         e += 4;
+      }
+
+      e = &buffer[n2-3];
+      while (d >= buf2) {
+         d[1] = (-e[2] * AA[0] - -e[0]*AA[1]);
+         d[0] = (-e[2] * AA[1] + -e[0]*AA[0]);
+         d -= 2;
+         AA += 2;
+         e -= 4;
+      }
+   }
+
+   // now we use symbolic names for these, so that we can
+   // possibly swap their meaning as we change which operations
+   // are in place
+
+   u = buffer;
+   v = buf2;
+
+   // step 2    (paper output is w, now u)
+   // this could be in place, but the data ends up in the wrong
+   // place... _somebody_'s got to swap it, so this is nominated
+   {
+      float *AA = &A[n2-8];
+      float *d0,*d1, *e0, *e1;
+
+      e0 = &v[n4];
+      e1 = &v[0];
+
+      d0 = &u[n4];
+      d1 = &u[0];
+
+      while (AA >= A) {
+         float v40_20, v41_21;
+
+         v41_21 = e0[1] - e1[1];
+         v40_20 = e0[0] - e1[0];
+         d0[1]  = e0[1] + e1[1];
+         d0[0]  = e0[0] + e1[0];
+         d1[1]  = v41_21*AA[4] - v40_20*AA[5];
+         d1[0]  = v40_20*AA[4] + v41_21*AA[5];
+
+         v41_21 = e0[3] - e1[3];
+         v40_20 = e0[2] - e1[2];
+         d0[3]  = e0[3] + e1[3];
+         d0[2]  = e0[2] + e1[2];
+         d1[3]  = v41_21*AA[0] - v40_20*AA[1];
+         d1[2]  = v40_20*AA[0] + v41_21*AA[1];
+
+         AA -= 8;
+
+         d0 += 4;
+         d1 += 4;
+         e0 += 4;
+         e1 += 4;
+      }
+   }
+
+   // step 3
+   ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+
+   // optimized step 3:
+
+   // the original step3 loop can be nested r inside s or s inside r;
+   // it's written originally as s inside r, but this is dumb when r
+   // iterates many times, and s few. So I have two copies of it and
+   // switch between them halfway.
+
+   // this is iteration 0 of step 3
+   imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A);
+   imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A);
+
+   // this is iteration 1 of step 3
+   imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16);
+   imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16);
+   imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16);
+   imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16);
+
+   l=2;
+   for (; l < (ld-3)>>1; ++l) {
+      int k0 = n >> (l+2), k0_2 = k0>>1;
+      int lim = 1 << (l+1);
+      int i;
+      for (i=0; i < lim; ++i)
+         imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3));
+   }
+
+   for (; l < ld-6; ++l) {
+      int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1;
+      int rlim = n >> (l+6), r;
+      int lim = 1 << (l+1);
+      int i_off;
+      float *A0 = A;
+      i_off = n2-1;
+      for (r=rlim; r > 0; --r) {
+         imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0);
+         A0 += k1*4;
+         i_off -= 8;
+      }
+   }
+
+   // iterations with count:
+   //   ld-6,-5,-4 all interleaved together
+   //       the big win comes from getting rid of needless flops
+   //         due to the constants on pass 5 & 4 being all 1 and 0;
+   //       combining them to be simultaneous to improve cache made little difference
+   imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n);
+
+   // output is u
+
+   // step 4, 5, and 6
+   // cannot be in-place because of step 5
+   {
+      uint16 *bitrev = f->bit_reverse[blocktype];
+      // weirdly, I'd have thought reading sequentially and writing
+      // erratically would have been better than vice-versa, but in
+      // fact that's not what my testing showed. (That is, with
+      // j = bitreverse(i), do you read i and write j, or read j and write i.)
+
+      float *d0 = &v[n4-4];
+      float *d1 = &v[n2-4];
+      while (d0 >= v) {
+         int k4;
+
+         k4 = bitrev[0];
+         d1[3] = u[k4+0];
+         d1[2] = u[k4+1];
+         d0[3] = u[k4+2];
+         d0[2] = u[k4+3];
+
+         k4 = bitrev[1];
+         d1[1] = u[k4+0];
+         d1[0] = u[k4+1];
+         d0[1] = u[k4+2];
+         d0[0] = u[k4+3];
+         
+         d0 -= 4;
+         d1 -= 4;
+         bitrev += 2;
+      }
+   }
+   // (paper output is u, now v)
+
+
+   // data must be in buf2
+   assert(v == buf2);
+
+   // step 7   (paper output is v, now v)
+   // this is now in place
+   {
+      float *C = f->C[blocktype];
+      float *d, *e;
+
+      d = v;
+      e = v + n2 - 4;
+
+      while (d < e) {
+         float a02,a11,b0,b1,b2,b3;
+
+         a02 = d[0] - e[2];
+         a11 = d[1] + e[3];
+
+         b0 = C[1]*a02 + C[0]*a11;
+         b1 = C[1]*a11 - C[0]*a02;
+
+         b2 = d[0] + e[ 2];
+         b3 = d[1] - e[ 3];
+
+         d[0] = b2 + b0;
+         d[1] = b3 + b1;
+         e[2] = b2 - b0;
+         e[3] = b1 - b3;
+
+         a02 = d[2] - e[0];
+         a11 = d[3] + e[1];
+
+         b0 = C[3]*a02 + C[2]*a11;
+         b1 = C[3]*a11 - C[2]*a02;
+
+         b2 = d[2] + e[ 0];
+         b3 = d[3] - e[ 1];
+
+         d[2] = b2 + b0;
+         d[3] = b3 + b1;
+         e[0] = b2 - b0;
+         e[1] = b1 - b3;
+
+         C += 4;
+         d += 4;
+         e -= 4;
+      }
+   }
+
+   // data must be in buf2
+
+
+   // step 8+decode   (paper output is X, now buffer)
+   // this generates pairs of data a la 8 and pushes them directly through
+   // the decode kernel (pushing rather than pulling) to avoid having
+   // to make another pass later
+
+   // this cannot POSSIBLY be in place, so we refer to the buffers directly
+
+   {
+      float *d0,*d1,*d2,*d3;
+
+      float *B = f->B[blocktype] + n2 - 8;
+      float *e = buf2 + n2 - 8;
+      d0 = &buffer[0];
+      d1 = &buffer[n2-4];
+      d2 = &buffer[n2];
+      d3 = &buffer[n-4];
+      while (e >= v) {
+         float p0,p1,p2,p3;
+
+         p3 =  e[6]*B[7] - e[7]*B[6];
+         p2 = -e[6]*B[6] - e[7]*B[7]; 
+
+         d0[0] =   p3;
+         d1[3] = - p3;
+         d2[0] =   p2;
+         d3[3] =   p2;
+
+         p1 =  e[4]*B[5] - e[5]*B[4];
+         p0 = -e[4]*B[4] - e[5]*B[5]; 
+
+         d0[1] =   p1;
+         d1[2] = - p1;
+         d2[1] =   p0;
+         d3[2] =   p0;
+
+         p3 =  e[2]*B[3] - e[3]*B[2];
+         p2 = -e[2]*B[2] - e[3]*B[3]; 
+
+         d0[2] =   p3;
+         d1[1] = - p3;
+         d2[2] =   p2;
+         d3[1] =   p2;
+
+         p1 =  e[0]*B[1] - e[1]*B[0];
+         p0 = -e[0]*B[0] - e[1]*B[1]; 
+
+         d0[3] =   p1;
+         d1[0] = - p1;
+         d2[3] =   p0;
+         d3[0] =   p0;
+
+         B -= 8;
+         e -= 8;
+         d0 += 4;
+         d2 += 4;
+         d1 -= 4;
+         d3 -= 4;
+      }
+   }
+
+   temp_alloc_restore(f,save_point);
+}
+
+#if 0
+// this is the original version of the above code, if you want to optimize it from scratch
+void inverse_mdct_naive(float *buffer, int n)
+{
+   float s;
+   float A[1 << 12], B[1 << 12], C[1 << 11];
+   int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
+   int n3_4 = n - n4, ld;
+   // how can they claim this only uses N words?!
+   // oh, because they're only used sparsely, whoops
+   float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13];
+   // set up twiddle factors
+
+   for (k=k2=0; k < n4; ++k,k2+=2) {
+      A[k2  ] = (float)  cos(4*k*M_PI/n);
+      A[k2+1] = (float) -sin(4*k*M_PI/n);
+      B[k2  ] = (float)  cos((k2+1)*M_PI/n/2);
+      B[k2+1] = (float)  sin((k2+1)*M_PI/n/2);
+   }
+   for (k=k2=0; k < n8; ++k,k2+=2) {
+      C[k2  ] = (float)  cos(2*(k2+1)*M_PI/n);
+      C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
+   }
+
+   // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
+   // Note there are bugs in that pseudocode, presumably due to them attempting
+   // to rename the arrays nicely rather than representing the way their actual
+   // implementation bounces buffers back and forth. As a result, even in the
+   // "some formulars corrected" version, a direct implementation fails. These
+   // are noted below as "paper bug".
+
+   // copy and reflect spectral data
+   for (k=0; k < n2; ++k) u[k] = buffer[k];
+   for (   ; k < n ; ++k) u[k] = -buffer[n - k - 1];
+   // kernel from paper
+   // step 1
+   for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) {
+      v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2]   - (u[k4+2] - u[n-k4-3])*A[k2+1];
+      v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2];
+   }
+   // step 2
+   for (k=k4=0; k < n8; k+=1, k4+=4) {
+      w[n2+3+k4] = v[n2+3+k4] + v[k4+3];
+      w[n2+1+k4] = v[n2+1+k4] + v[k4+1];
+      w[k4+3]    = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4];
+      w[k4+1]    = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4];
+   }
+   // step 3
+   ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+   for (l=0; l < ld-3; ++l) {
+      int k0 = n >> (l+2), k1 = 1 << (l+3);
+      int rlim = n >> (l+4), r4, r;
+      int s2lim = 1 << (l+2), s2;
+      for (r=r4=0; r < rlim; r4+=4,++r) {
+         for (s2=0; s2 < s2lim; s2+=2) {
+            u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4];
+            u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4];
+            u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1]
+                                - (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1];
+            u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1]
+                                + (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1];
+         }
+      }
+      if (l+1 < ld-3) {
+         // paper bug: ping-ponging of u&w here is omitted
+         memcpy(w, u, sizeof(u));
+      }
+   }
+
+   // step 4
+   for (i=0; i < n8; ++i) {
+      int j = bit_reverse(i) >> (32-ld+3);
+      assert(j < n8);
+      if (i == j) {
+         // paper bug: original code probably swapped in place; if copying,
+         //            need to directly copy in this case
+         int i8 = i << 3;
+         v[i8+1] = u[i8+1];
+         v[i8+3] = u[i8+3];
+         v[i8+5] = u[i8+5];
+         v[i8+7] = u[i8+7];
+      } else if (i < j) {
+         int i8 = i << 3, j8 = j << 3;
+         v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1];
+         v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3];
+         v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5];
+         v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7];
+      }
+   }
+   // step 5
+   for (k=0; k < n2; ++k) {
+      w[k] = v[k*2+1];
+   }
+   // step 6
+   for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) {
+      u[n-1-k2] = w[k4];
+      u[n-2-k2] = w[k4+1];
+      u[n3_4 - 1 - k2] = w[k4+2];
+      u[n3_4 - 2 - k2] = w[k4+3];
+   }
+   // step 7
+   for (k=k2=0; k < n8; ++k, k2 += 2) {
+      v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
+      v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
+      v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
+      v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
+   }
+   // step 8
+   for (k=k2=0; k < n4; ++k,k2 += 2) {
+      X[k]      = v[k2+n2]*B[k2  ] + v[k2+1+n2]*B[k2+1];
+      X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2  ];
+   }
+
+   // decode kernel to output
+   // determined the following value experimentally
+   // (by first figuring out what made inverse_mdct_slow work); then matching that here
+   // (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?)
+   s = 0.5; // theoretically would be n4
+
+   // [[[ note! the s value of 0.5 is compensated for by the B[] in the current code,
+   //     so it needs to use the "old" B values to behave correctly, or else
+   //     set s to 1.0 ]]]
+   for (i=0; i < n4  ; ++i) buffer[i] = s * X[i+n4];
+   for (   ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1];
+   for (   ; i < n   ; ++i) buffer[i] = -s * X[i - n3_4];
+}
+#endif
+
+static float *get_window(vorb *f, int len)
+{
+   len <<= 1;
+   if (len == f->blocksize_0) return f->window[0];
+   if (len == f->blocksize_1) return f->window[1];
+   assert(0);
+   return NULL;
+}
+
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+typedef int16 YTYPE;
+#else
+typedef int YTYPE;
+#endif
+static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag)
+{
+   int n2 = n >> 1;
+   int s = map->chan[i].mux, floor;
+   floor = map->submap_floor[s];
+   if (f->floor_types[floor] == 0) {
+      return error(f, VORBIS_invalid_stream);
+   } else {
+      Floor1 *g = &f->floor_config[floor].floor1;
+      int j,q;
+      int lx = 0, ly = finalY[0] * g->floor1_multiplier;
+      for (q=1; q < g->values; ++q) {
+         j = g->sorted_order[q];
+         #ifndef STB_VORBIS_NO_DEFER_FLOOR
+         if (finalY[j] >= 0)
+         #else
+         if (step2_flag[j])
+         #endif
+         {
+            int hy = finalY[j] * g->floor1_multiplier;
+            int hx = g->Xlist[j];
+            draw_line(target, lx,ly, hx,hy, n2);
+            lx = hx, ly = hy;
+         }
+      }
+      if (lx < n2)
+         // optimization of: draw_line(target, lx,ly, n,ly, n2);
+         for (j=lx; j < n2; ++j)
+            LINE_OP(target[j], inverse_db_table[ly]);
+   }
+   return TRUE;
+}
+
+static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
+{
+   Mode *m;
+   int i, n, prev, next, window_center;
+   f->channel_buffer_start = f->channel_buffer_end = 0;
+
+  retry:
+   if (f->eof) return FALSE;
+   if (!maybe_start_packet(f))
+      return FALSE;
+   // check packet type
+   if (get_bits(f,1) != 0) {
+      if (IS_PUSH_MODE(f))
+         return error(f,VORBIS_bad_packet_type);
+      while (EOP != get8_packet(f));
+      goto retry;
+   }
+
+   if (f->alloc.alloc_buffer)
+      assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+
+   i = get_bits(f, ilog(f->mode_count-1));
+   if (i == EOP) return FALSE;
+   if (i >= f->mode_count) return FALSE;
+   *mode = i;
+   m = f->mode_config + i;
+   if (m->blockflag) {
+      n = f->blocksize_1;
+      prev = get_bits(f,1);
+      next = get_bits(f,1);
+   } else {
+      prev = next = 0;
+      n = f->blocksize_0;
+   }
+
+// WINDOWING
+
+   window_center = n >> 1;
+   if (m->blockflag && !prev) {
+      *p_left_start = (n - f->blocksize_0) >> 2;
+      *p_left_end   = (n + f->blocksize_0) >> 2;
+   } else {
+      *p_left_start = 0;
+      *p_left_end   = window_center;
+   }
+   if (m->blockflag && !next) {
+      *p_right_start = (n*3 - f->blocksize_0) >> 2;
+      *p_right_end   = (n*3 + f->blocksize_0) >> 2;
+   } else {
+      *p_right_start = window_center;
+      *p_right_end   = n;
+   }
+   return TRUE;
+}
+
+static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left)
+{
+   Mapping *map;
+   int i,j,k,n,n2;
+   int zero_channel[256];
+   int really_zero_channel[256];
+   int window_center;
+
+// WINDOWING
+
+   n = f->blocksize[m->blockflag];
+   window_center = n >> 1;
+
+   map = &f->mapping[m->mapping];
+
+// FLOORS
+   n2 = n >> 1;
+
+   for (i=0; i < f->channels; ++i) {
+      int s = map->chan[i].mux, floor;
+      zero_channel[i] = FALSE;
+      floor = map->submap_floor[s];
+      if (f->floor_types[floor] == 0) {
+         return error(f, VORBIS_invalid_stream);
+      } else {
+         Floor1 *g = &f->floor_config[floor].floor1;
+         if (get_bits(f, 1)) {
+            short *finalY;
+            uint8 step2_flag[256];
+            static int range_list[4] = { 256, 128, 86, 64 };
+            int range = range_list[g->floor1_multiplier-1];
+            int offset = 2;
+            finalY = f->finalY[i];
+            finalY[0] = get_bits(f, ilog(range)-1);
+            finalY[1] = get_bits(f, ilog(range)-1);
+            for (j=0; j < g->partitions; ++j) {
+               int pclass = g->partition_class_list[j];
+               int cdim = g->class_dimensions[pclass];
+               int cbits = g->class_subclasses[pclass];
+               int csub = (1 << cbits)-1;
+               int cval = 0;
+               if (cbits) {
+                  Codebook *c = f->codebooks + g->class_masterbooks[pclass];
+                  DECODE(cval,f,c);
+               }
+               for (k=0; k < cdim; ++k) {
+                  int book = g->subclass_books[pclass][cval & csub];
+                  cval = cval >> cbits;
+                  if (book >= 0) {
+                     int temp;
+                     Codebook *c = f->codebooks + book;
+                     DECODE(temp,f,c);
+                     finalY[offset++] = temp;
+                  } else
+                     finalY[offset++] = 0;
+               }
+            }
+            if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec
+            step2_flag[0] = step2_flag[1] = 1;
+            for (j=2; j < g->values; ++j) {
+               int low, high, pred, highroom, lowroom, room, val;
+               low = g->neighbors[j][0];
+               high = g->neighbors[j][1];
+               //neighbors(g->Xlist, j, &low, &high);
+               pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]);
+               val = finalY[j];
+               highroom = range - pred;
+               lowroom = pred;
+               if (highroom < lowroom)
+                  room = highroom * 2;
+               else
+                  room = lowroom * 2;
+               if (val) {
+                  step2_flag[low] = step2_flag[high] = 1;
+                  step2_flag[j] = 1;
+                  if (val >= room)
+                     if (highroom > lowroom)
+                        finalY[j] = val - lowroom + pred;
+                     else
+                        finalY[j] = pred - val + highroom - 1;
+                  else
+                     if (val & 1)
+                        finalY[j] = pred - ((val+1)>>1);
+                     else
+                        finalY[j] = pred + (val>>1);
+               } else {
+                  step2_flag[j] = 0;
+                  finalY[j] = pred;
+               }
+            }
+
+#ifdef STB_VORBIS_NO_DEFER_FLOOR
+            do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag);
+#else
+            // defer final floor computation until _after_ residue
+            for (j=0; j < g->values; ++j) {
+               if (!step2_flag[j])
+                  finalY[j] = -1;
+            }
+#endif
+         } else {
+           error:
+            zero_channel[i] = TRUE;
+         }
+         // So we just defer everything else to later
+
+         // at this point we've decoded the floor into buffer
+      }
+   }
+   // at this point we've decoded all floors
+
+   if (f->alloc.alloc_buffer)
+      assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+
+   // re-enable coupled channels if necessary
+   memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels);
+   for (i=0; i < map->coupling_steps; ++i)
+      if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) {
+         zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE;
+      }
+
+// RESIDUE DECODE
+   for (i=0; i < map->submaps; ++i) {
+      float *residue_buffers[STB_VORBIS_MAX_CHANNELS];
+      int r,t;
+      uint8 do_not_decode[256];
+      int ch = 0;
+      for (j=0; j < f->channels; ++j) {
+         if (map->chan[j].mux == i) {
+            if (zero_channel[j]) {
+               do_not_decode[ch] = TRUE;
+               residue_buffers[ch] = NULL;
+            } else {
+               do_not_decode[ch] = FALSE;
+               residue_buffers[ch] = f->channel_buffers[j];
+            }
+            ++ch;
+         }
+      }
+      r = map->submap_residue[i];
+      t = f->residue_types[r];
+      decode_residue(f, residue_buffers, ch, n2, r, do_not_decode);
+   }
+
+   if (f->alloc.alloc_buffer)
+      assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+
+// INVERSE COUPLING
+   for (i = map->coupling_steps-1; i >= 0; --i) {
+      int n2 = n >> 1;
+      float *m = f->channel_buffers[map->chan[i].magnitude];
+      float *a = f->channel_buffers[map->chan[i].angle    ];
+      for (j=0; j < n2; ++j) {
+         float a2,m2;
+         if (m[j] > 0)
+            if (a[j] > 0)
+               m2 = m[j], a2 = m[j] - a[j];
+            else
+               a2 = m[j], m2 = m[j] + a[j];
+         else
+            if (a[j] > 0)
+               m2 = m[j], a2 = m[j] + a[j];
+            else
+               a2 = m[j], m2 = m[j] - a[j];
+         m[j] = m2;
+         a[j] = a2;
+      }
+   }
+
+   // finish decoding the floors
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+   for (i=0; i < f->channels; ++i) {
+      if (really_zero_channel[i]) {
+         memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
+      } else {
+         do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL);
+      }
+   }
+#else
+   for (i=0; i < f->channels; ++i) {
+      if (really_zero_channel[i]) {
+         memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
+      } else {
+         for (j=0; j < n2; ++j)
+            f->channel_buffers[i][j] *= f->floor_buffers[i][j];
+      }
+   }
+#endif
+
+// INVERSE MDCT
+   for (i=0; i < f->channels; ++i)
+      inverse_mdct(f->channel_buffers[i], n, f, m->blockflag);
+
+   // this shouldn't be necessary, unless we exited on an error
+   // and want to flush to get to the next packet
+   flush_packet(f);
+
+   if (f->first_decode) {
+      // assume we start so first non-discarded sample is sample 0
+      // this isn't to spec, but spec would require us to read ahead
+      // and decode the size of all current frames--could be done,
+      // but presumably it's not a commonly used feature
+      f->current_loc = -n2; // start of first frame is positioned for discard
+      // we might have to discard samples "from" the next frame too,
+      // if we're lapping a large block then a small at the start?
+      f->discard_samples_deferred = n - right_end;
+      f->current_loc_valid = TRUE;
+      f->first_decode = FALSE;
+   } else if (f->discard_samples_deferred) {
+      left_start += f->discard_samples_deferred;
+      *p_left = left_start;
+      f->discard_samples_deferred = 0;
+   } else if (f->previous_length == 0 && f->current_loc_valid) {
+      // we're recovering from a seek... that means we're going to discard
+      // the samples from this packet even though we know our position from
+      // the last page header, so we need to update the position based on
+      // the discarded samples here
+      // but wait, the code below is going to add this in itself even
+      // on a discard, so we don't need to do it here...
+   }
+
+   // check if we have ogg information about the sample # for this packet
+   if (f->last_seg_which == f->end_seg_with_known_loc) {
+      // if we have a valid current loc, and this is final:
+      if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) {
+         uint32 current_end = f->known_loc_for_packet - (n-right_end);
+         // then let's infer the size of the (probably) short final frame
+         if (current_end < f->current_loc + right_end) {
+            if (current_end < f->current_loc) {
+               // negative truncation, that's impossible!
+               *len = 0;
+            } else {
+               *len = current_end - f->current_loc;
+            }
+            *len += left_start;
+            f->current_loc += *len;
+            return TRUE;
+         }
+      }
+      // otherwise, just set our sample loc
+      // guess that the ogg granule pos refers to the _middle_ of the
+      // last frame?
+      // set f->current_loc to the position of left_start
+      f->current_loc = f->known_loc_for_packet - (n2-left_start);
+      f->current_loc_valid = TRUE;
+   }
+   if (f->current_loc_valid)
+      f->current_loc += (right_start - left_start);
+
+   if (f->alloc.alloc_buffer)
+      assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+   *len = right_end;  // ignore samples after the window goes to 0
+   return TRUE;
+}
+
+static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right)
+{
+   int mode, left_end, right_end;
+   if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0;
+   return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left);
+}
+
+static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right)
+{
+   int prev,i,j;
+   // we use right&left (the start of the right- and left-window sin()-regions)
+   // to determine how much to return, rather than inferring from the rules
+   // (same result, clearer code); 'left' indicates where our sin() window
+   // starts, therefore where the previous window's right edge starts, and
+   // therefore where to start mixing from the previous buffer. 'right'
+   // indicates where our sin() ending-window starts, therefore that's where
+   // we start saving, and where our returned-data ends.
+
+   // mixin from previous window
+   if (f->previous_length) {
+      int i,j, n = f->previous_length;
+      float *w = get_window(f, n);
+      for (i=0; i < f->channels; ++i) {
+         for (j=0; j < n; ++j)
+            f->channel_buffers[i][left+j] =
+               f->channel_buffers[i][left+j]*w[    j] +
+               f->previous_window[i][     j]*w[n-1-j];
+      }
+   }
+
+   prev = f->previous_length;
+
+   // last half of this data becomes previous window
+   f->previous_length = len - right;
+
+   // @OPTIMIZE: could avoid this copy by double-buffering the
+   // output (flipping previous_window with channel_buffers), but
+   // then previous_window would have to be 2x as large, and
+   // channel_buffers couldn't be temp mem (although they're NOT
+   // currently temp mem, they could be (unless we want to level
+   // performance by spreading out the computation))
+   for (i=0; i < f->channels; ++i)
+      for (j=0; right+j < len; ++j)
+         f->previous_window[i][j] = f->channel_buffers[i][right+j];
+
+   if (!prev)
+      // there was no previous packet, so this data isn't valid...
+      // this isn't entirely true, only the would-have-overlapped data
+      // isn't valid, but this seems to be what the spec requires
+      return 0;
+
+   // truncate a short frame
+   if (len < right) right = len;
+
+   f->samples_output += right-left;
+
+   return right - left;
+}
+
+static void vorbis_pump_first_frame(stb_vorbis *f)
+{
+   int len, right, left;
+   if (vorbis_decode_packet(f, &len, &left, &right))
+      vorbis_finish_frame(f, len, left, right);
+}
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+static int is_whole_packet_present(stb_vorbis *f, int end_page)
+{
+   // make sure that we have the packet available before continuing...
+   // this requires a full ogg parse, but we know we can fetch from f->stream
+
+   // instead of coding this out explicitly, we could save the current read state,
+   // read the next packet with get8() until end-of-packet, check f->eof, then
+   // reset the state? but that would be slower, esp. since we'd have over 256 bytes
+   // of state to restore (primarily the page segment table)
+
+   int s = f->next_seg, first = TRUE;
+   uint8 *p = f->stream;
+
+   if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag
+      for (; s < f->segment_count; ++s) {
+         p += f->segments[s];
+         if (f->segments[s] < 255)               // stop at first short segment
+            break;
+      }
+      // either this continues, or it ends it...
+      if (end_page)
+         if (s < f->segment_count-1)             return error(f, VORBIS_invalid_stream);
+      if (s == f->segment_count)
+         s = -1; // set 'crosses page' flag
+      if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+      first = FALSE;
+   }
+   for (; s == -1;) {
+      uint8 *q; 
+      int n;
+
+      // check that we have the page header ready
+      if (p + 26 >= f->stream_end)               return error(f, VORBIS_need_more_data);
+      // validate the page
+      if (memcmp(p, ogg_page_header, 4))         return error(f, VORBIS_invalid_stream);
+      if (p[4] != 0)                             return error(f, VORBIS_invalid_stream);
+      if (first) { // the first segment must NOT have 'continued_packet', later ones MUST
+         if (f->previous_length)
+            if ((p[5] & PAGEFLAG_continued_packet))  return error(f, VORBIS_invalid_stream);
+         // if no previous length, we're resynching, so we can come in on a continued-packet,
+         // which we'll just drop
+      } else {
+         if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream);
+      }
+      n = p[26]; // segment counts
+      q = p+27;  // q points to segment table
+      p = q + n; // advance past header
+      // make sure we've read the segment table
+      if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+      for (s=0; s < n; ++s) {
+         p += q[s];
+         if (q[s] < 255)
+            break;
+      }
+      if (end_page)
+         if (s < n-1)                            return error(f, VORBIS_invalid_stream);
+      if (s == f->segment_count)
+         s = -1; // set 'crosses page' flag
+      if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+      first = FALSE;
+   }
+   return TRUE;
+}
+#endif // !STB_VORBIS_NO_PUSHDATA_API
+
+static int start_decoder(vorb *f)
+{
+   uint8 header[6], x,y;
+   int len,i,j,k, max_submaps = 0;
+   int longest_floorlist=0;
+
+   // first page, first packet
+
+   if (!start_page(f))                              return FALSE;
+   // validate page flag
+   if (!(f->page_flag & PAGEFLAG_first_page))       return error(f, VORBIS_invalid_first_page);
+   if (f->page_flag & PAGEFLAG_last_page)           return error(f, VORBIS_invalid_first_page);
+   if (f->page_flag & PAGEFLAG_continued_packet)    return error(f, VORBIS_invalid_first_page);
+   // check for expected packet length
+   if (f->segment_count != 1)                       return error(f, VORBIS_invalid_first_page);
+   if (f->segments[0] != 30)                        return error(f, VORBIS_invalid_first_page);
+   // read packet
+   // check packet header
+   if (get8(f) != VORBIS_packet_id)                 return error(f, VORBIS_invalid_first_page);
+   if (!getn(f, header, 6))                         return error(f, VORBIS_unexpected_eof);
+   if (!vorbis_validate(header))                    return error(f, VORBIS_invalid_first_page);
+   // vorbis_version
+   if (get32(f) != 0)                               return error(f, VORBIS_invalid_first_page);
+   f->channels = get8(f); if (!f->channels)         return error(f, VORBIS_invalid_first_page);
+   if (f->channels > STB_VORBIS_MAX_CHANNELS)       return error(f, VORBIS_too_many_channels);
+   f->sample_rate = get32(f); if (!f->sample_rate)  return error(f, VORBIS_invalid_first_page);
+   get32(f); // bitrate_maximum
+   get32(f); // bitrate_nominal
+   get32(f); // bitrate_minimum
+   x = get8(f);
+   { int log0,log1;
+   log0 = x & 15;
+   log1 = x >> 4;
+   f->blocksize_0 = 1 << log0;
+   f->blocksize_1 = 1 << log1;
+   if (log0 < 6 || log0 > 13)                       return error(f, VORBIS_invalid_setup);
+   if (log1 < 6 || log1 > 13)                       return error(f, VORBIS_invalid_setup);
+   if (log0 > log1)                                 return error(f, VORBIS_invalid_setup);
+   }
+
+   // framing_flag
+   x = get8(f);
+   if (!(x & 1))                                    return error(f, VORBIS_invalid_first_page);
+
+   // second packet!
+   if (!start_page(f))                              return FALSE;
+
+   if (!start_packet(f))                            return FALSE;
+   do {
+      len = next_segment(f);
+      skip(f, len);
+      f->bytes_in_seg = 0;
+   } while (len);
+
+   // third packet!
+   if (!start_packet(f))                            return FALSE;
+
+   #ifndef STB_VORBIS_NO_PUSHDATA_API
+   if (IS_PUSH_MODE(f)) {
+      if (!is_whole_packet_present(f, TRUE)) {
+         // convert error in ogg header to write type
+         if (f->error == VORBIS_invalid_stream)
+            f->error = VORBIS_invalid_setup;
+         return FALSE;
+      }
+   }
+   #endif
+
+   crc32_init(); // always init it, to avoid multithread race conditions
+
+   if (get8_packet(f) != VORBIS_packet_setup)       return error(f, VORBIS_invalid_setup);
+   for (i=0; i < 6; ++i) header[i] = get8_packet(f);
+   if (!vorbis_validate(header))                    return error(f, VORBIS_invalid_setup);
+
+   // codebooks
+
+   f->codebook_count = get_bits(f,8) + 1;
+   f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count);
+   if (f->codebooks == NULL)                        return error(f, VORBIS_outofmem);
+   memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count);
+   for (i=0; i < f->codebook_count; ++i) {
+      uint32 *values;
+      int ordered, sorted_count;
+      int total=0;
+      uint8 *lengths;
+      Codebook *c = f->codebooks+i;
+      x = get_bits(f, 8); if (x != 0x42)            return error(f, VORBIS_invalid_setup);
+      x = get_bits(f, 8); if (x != 0x43)            return error(f, VORBIS_invalid_setup);
+      x = get_bits(f, 8); if (x != 0x56)            return error(f, VORBIS_invalid_setup);
+      x = get_bits(f, 8);
+      c->dimensions = (get_bits(f, 8)<<8) + x;
+      x = get_bits(f, 8);
+      y = get_bits(f, 8);
+      c->entries = (get_bits(f, 8)<<16) + (y<<8) + x;
+      ordered = get_bits(f,1);
+      c->sparse = ordered ? 0 : get_bits(f,1);
+
+      if (c->sparse)
+         lengths = (uint8 *) setup_temp_malloc(f, c->entries);
+      else
+         lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
+
+      if (!lengths) return error(f, VORBIS_outofmem);
+
+      if (ordered) {
+         int current_entry = 0;
+         int current_length = get_bits(f,5) + 1;
+         while (current_entry < c->entries) {
+            int limit = c->entries - current_entry;
+            int n = get_bits(f, ilog(limit));
+            if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); }
+            memset(lengths + current_entry, current_length, n);
+            current_entry += n;
+            ++current_length;
+         }
+      } else {
+         for (j=0; j < c->entries; ++j) {
+            int present = c->sparse ? get_bits(f,1) : 1;
+            if (present) {
+               lengths[j] = get_bits(f, 5) + 1;
+               ++total;
+            } else {
+               lengths[j] = NO_CODE;
+            }
+         }
+      }
+
+      if (c->sparse && total >= c->entries >> 2) {
+         // convert sparse items to non-sparse!
+         if (c->entries > (int) f->setup_temp_memory_required)
+            f->setup_temp_memory_required = c->entries;
+
+         c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
+         memcpy(c->codeword_lengths, lengths, c->entries);
+         setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs!
+         lengths = c->codeword_lengths;
+         c->sparse = 0;
+      }
+
+      // compute the size of the sorted tables
+      if (c->sparse) {
+         sorted_count = total;
+         //assert(total != 0);
+      } else {
+         sorted_count = 0;
+         #ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+         for (j=0; j < c->entries; ++j)
+            if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE)
+               ++sorted_count;
+         #endif
+      }
+
+      c->sorted_entries = sorted_count;
+      values = NULL;
+
+      if (!c->sparse) {
+         c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries);
+         if (!c->codewords)                  return error(f, VORBIS_outofmem);
+      } else {
+         unsigned int size;
+         if (c->sorted_entries) {
+            c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries);
+            if (!c->codeword_lengths)           return error(f, VORBIS_outofmem);
+            c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries);
+            if (!c->codewords)                  return error(f, VORBIS_outofmem);
+            values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries);
+            if (!values)                        return error(f, VORBIS_outofmem);
+         }
+         size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries;
+         if (size > f->setup_temp_memory_required)
+            f->setup_temp_memory_required = size;
+      }
+
+      if (!compute_codewords(c, lengths, c->entries, values)) {
+         if (c->sparse) setup_temp_free(f, values, 0);
+         return error(f, VORBIS_invalid_setup);
+      }
+
+      if (c->sorted_entries) {
+         // allocate an extra slot for sentinels
+         c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1));
+         // allocate an extra slot at the front so that c->sorted_values[-1] is defined
+         // so that we can catch that case without an extra if
+         c->sorted_values    = ( int   *) setup_malloc(f, sizeof(*c->sorted_values   ) * (c->sorted_entries+1));
+         if (c->sorted_values) { ++c->sorted_values; c->sorted_values[-1] = -1; }
+         compute_sorted_huffman(c, lengths, values);
+      }
+
+      if (c->sparse) {
+         setup_temp_free(f, values, sizeof(*values)*c->sorted_entries);
+         setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries);
+         setup_temp_free(f, lengths, c->entries);
+         c->codewords = NULL;
+      }
+
+      compute_accelerated_huffman(c);
+
+      c->lookup_type = get_bits(f, 4);
+      if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup);
+      if (c->lookup_type > 0) {
+         uint16 *mults;
+         c->minimum_value = float32_unpack(get_bits(f, 32));
+         c->delta_value = float32_unpack(get_bits(f, 32));
+         c->value_bits = get_bits(f, 4)+1;
+         c->sequence_p = get_bits(f,1);
+         if (c->lookup_type == 1) {
+            c->lookup_values = lookup1_values(c->entries, c->dimensions);
+         } else {
+            c->lookup_values = c->entries * c->dimensions;
+         }
+         mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values);
+         if (mults == NULL) return error(f, VORBIS_outofmem);
+         for (j=0; j < (int) c->lookup_values; ++j) {
+            int q = get_bits(f, c->value_bits);
+            if (q == EOP) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); }
+            mults[j] = q;
+         }
+
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+         if (c->lookup_type == 1) {
+            int len, sparse = c->sparse;
+            // pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop
+            if (sparse) {
+               if (c->sorted_entries == 0) goto skip;
+               c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions);
+            } else
+               c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries        * c->dimensions);
+            if (c->multiplicands == NULL) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
+            len = sparse ? c->sorted_entries : c->entries;
+            for (j=0; j < len; ++j) {
+               int z = sparse ? c->sorted_values[j] : j, div=1;
+               for (k=0; k < c->dimensions; ++k) {
+                  int off = (z / div) % c->lookup_values;
+                  c->multiplicands[j*c->dimensions + k] =
+                         #ifndef STB_VORBIS_CODEBOOK_FLOATS
+                            mults[off];
+                         #else
+                            mults[off]*c->delta_value + c->minimum_value;
+                            // in this case (and this case only) we could pre-expand c->sequence_p,
+                            // and throw away the decode logic for it; have to ALSO do
+                            // it in the case below, but it can only be done if
+                            //    STB_VORBIS_CODEBOOK_FLOATS
+                            //   !STB_VORBIS_DIVIDES_IN_CODEBOOK
+                         #endif
+                  div *= c->lookup_values;
+               }
+            }
+            setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values);
+            c->lookup_type = 2;
+         }
+         else
+#endif
+         {
+            c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values);
+            #ifndef STB_VORBIS_CODEBOOK_FLOATS
+            memcpy(c->multiplicands, mults, sizeof(c->multiplicands[0]) * c->lookup_values);
+            #else
+            for (j=0; j < (int) c->lookup_values; ++j)
+               c->multiplicands[j] = mults[j] * c->delta_value + c->minimum_value;
+            setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values);
+            #endif
+         }
+        skip:;
+
+         #ifdef STB_VORBIS_CODEBOOK_FLOATS
+         if (c->lookup_type == 2 && c->sequence_p) {
+            for (j=1; j < (int) c->lookup_values; ++j)
+               c->multiplicands[j] = c->multiplicands[j-1];
+            c->sequence_p = 0;
+         }
+         #endif
+      }
+   }
+
+   // time domain transfers (notused)
+
+   x = get_bits(f, 6) + 1;
+   for (i=0; i < x; ++i) {
+      uint32 z = get_bits(f, 16);
+      if (z != 0) return error(f, VORBIS_invalid_setup);
+   }
+
+   // Floors
+   f->floor_count = get_bits(f, 6)+1;
+   f->floor_config = (Floor *)  setup_malloc(f, f->floor_count * sizeof(*f->floor_config));
+   for (i=0; i < f->floor_count; ++i) {
+      f->floor_types[i] = get_bits(f, 16);
+      if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup);
+      if (f->floor_types[i] == 0) {
+         Floor0 *g = &f->floor_config[i].floor0;
+         g->order = get_bits(f,8);
+         g->rate = get_bits(f,16);
+         g->bark_map_size = get_bits(f,16);
+         g->amplitude_bits = get_bits(f,6);
+         g->amplitude_offset = get_bits(f,8);
+         g->number_of_books = get_bits(f,4) + 1;
+         for (j=0; j < g->number_of_books; ++j)
+            g->book_list[j] = get_bits(f,8);
+         return error(f, VORBIS_feature_not_supported);
+      } else {
+         Point p[31*8+2];
+         Floor1 *g = &f->floor_config[i].floor1;
+         int max_class = -1; 
+         g->partitions = get_bits(f, 5);
+         for (j=0; j < g->partitions; ++j) {
+            g->partition_class_list[j] = get_bits(f, 4);
+            if (g->partition_class_list[j] > max_class)
+               max_class = g->partition_class_list[j];
+         }
+         for (j=0; j <= max_class; ++j) {
+            g->class_dimensions[j] = get_bits(f, 3)+1;
+            g->class_subclasses[j] = get_bits(f, 2);
+            if (g->class_subclasses[j]) {
+               g->class_masterbooks[j] = get_bits(f, 8);
+               if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+            }
+            for (k=0; k < 1 << g->class_subclasses[j]; ++k) {
+               g->subclass_books[j][k] = get_bits(f,8)-1;
+               if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+            }
+         }
+         g->floor1_multiplier = get_bits(f,2)+1;
+         g->rangebits = get_bits(f,4);
+         g->Xlist[0] = 0;
+         g->Xlist[1] = 1 << g->rangebits;
+         g->values = 2;
+         for (j=0; j < g->partitions; ++j) {
+            int c = g->partition_class_list[j];
+            for (k=0; k < g->class_dimensions[c]; ++k) {
+               g->Xlist[g->values] = get_bits(f, g->rangebits);
+               ++g->values;
+            }
+         }
+         // precompute the sorting
+         for (j=0; j < g->values; ++j) {
+            p[j].x = g->Xlist[j];
+            p[j].y = j;
+         }
+         qsort(p, g->values, sizeof(p[0]), point_compare);
+         for (j=0; j < g->values; ++j)
+            g->sorted_order[j] = (uint8) p[j].y;
+         // precompute the neighbors
+         for (j=2; j < g->values; ++j) {
+            int low,hi;
+            neighbors(g->Xlist, j, &low,&hi);
+            g->neighbors[j][0] = low;
+            g->neighbors[j][1] = hi;
+         }
+
+         if (g->values > longest_floorlist)
+            longest_floorlist = g->values;
+      }
+   }
+
+   // Residue
+   f->residue_count = get_bits(f, 6)+1;
+   f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(*f->residue_config));
+   for (i=0; i < f->residue_count; ++i) {
+      uint8 residue_cascade[64];
+      Residue *r = f->residue_config+i;
+      f->residue_types[i] = get_bits(f, 16);
+      if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup);
+      r->begin = get_bits(f, 24);
+      r->end = get_bits(f, 24);
+      r->part_size = get_bits(f,24)+1;
+      r->classifications = get_bits(f,6)+1;
+      r->classbook = get_bits(f,8);
+      for (j=0; j < r->classifications; ++j) {
+         uint8 high_bits=0;
+         uint8 low_bits=get_bits(f,3);
+         if (get_bits(f,1))
+            high_bits = get_bits(f,5);
+         residue_cascade[j] = high_bits*8 + low_bits;
+      }
+      r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications);
+      for (j=0; j < r->classifications; ++j) {
+         for (k=0; k < 8; ++k) {
+            if (residue_cascade[j] & (1 << k)) {
+               r->residue_books[j][k] = get_bits(f, 8);
+               if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+            } else {
+               r->residue_books[j][k] = -1;
+            }
+         }
+      }
+      // precompute the classifications[] array to avoid inner-loop mod/divide
+      // call it 'classdata' since we already have r->classifications
+      r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
+      if (!r->classdata) return error(f, VORBIS_outofmem);
+      memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
+      for (j=0; j < f->codebooks[r->classbook].entries; ++j) {
+         int classwords = f->codebooks[r->classbook].dimensions;
+         int temp = j;
+         r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords);
+         for (k=classwords-1; k >= 0; --k) {
+            r->classdata[j][k] = temp % r->classifications;
+            temp /= r->classifications;
+         }
+      }
+   }
+
+   f->mapping_count = get_bits(f,6)+1;
+   f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping));
+   for (i=0; i < f->mapping_count; ++i) {
+      Mapping *m = f->mapping + i;      
+      int mapping_type = get_bits(f,16);
+      if (mapping_type != 0) return error(f, VORBIS_invalid_setup);
+      m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan));
+      if (get_bits(f,1))
+         m->submaps = get_bits(f,4);
+      else
+         m->submaps = 1;
+      if (m->submaps > max_submaps)
+         max_submaps = m->submaps;
+      if (get_bits(f,1)) {
+         m->coupling_steps = get_bits(f,8)+1;
+         for (k=0; k < m->coupling_steps; ++k) {
+            m->chan[k].magnitude = get_bits(f, ilog(f->channels)-1);
+            m->chan[k].angle = get_bits(f, ilog(f->channels)-1);
+            if (m->chan[k].magnitude >= f->channels)        return error(f, VORBIS_invalid_setup);
+            if (m->chan[k].angle     >= f->channels)        return error(f, VORBIS_invalid_setup);
+            if (m->chan[k].magnitude == m->chan[k].angle)   return error(f, VORBIS_invalid_setup);
+         }
+      } else
+         m->coupling_steps = 0;
+
+      // reserved field
+      if (get_bits(f,2)) return error(f, VORBIS_invalid_setup);
+      if (m->submaps > 1) {
+         for (j=0; j < f->channels; ++j) {
+            m->chan[j].mux = get_bits(f, 4);
+            if (m->chan[j].mux >= m->submaps)                return error(f, VORBIS_invalid_setup);
+         }
+      } else
+         // @SPECIFICATION: this case is missing from the spec
+         for (j=0; j < f->channels; ++j)
+            m->chan[j].mux = 0;
+
+      for (j=0; j < m->submaps; ++j) {
+         get_bits(f,8); // discard
+         m->submap_floor[j] = get_bits(f,8);
+         m->submap_residue[j] = get_bits(f,8);
+         if (m->submap_floor[j] >= f->floor_count)      return error(f, VORBIS_invalid_setup);
+         if (m->submap_residue[j] >= f->residue_count)  return error(f, VORBIS_invalid_setup);
+      }
+   }
+
+   // Modes
+   f->mode_count = get_bits(f, 6)+1;
+   for (i=0; i < f->mode_count; ++i) {
+      Mode *m = f->mode_config+i;
+      m->blockflag = get_bits(f,1);
+      m->windowtype = get_bits(f,16);
+      m->transformtype = get_bits(f,16);
+      m->mapping = get_bits(f,8);
+      if (m->windowtype != 0)                 return error(f, VORBIS_invalid_setup);
+      if (m->transformtype != 0)              return error(f, VORBIS_invalid_setup);
+      if (m->mapping >= f->mapping_count)     return error(f, VORBIS_invalid_setup);
+   }
+
+   flush_packet(f);
+
+   f->previous_length = 0;
+
+   for (i=0; i < f->channels; ++i) {
+      f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1);
+      f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
+      f->finalY[i]          = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist);
+      #ifdef STB_VORBIS_NO_DEFER_FLOOR
+      f->floor_buffers[i]   = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
+      #endif
+   }
+
+   if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE;
+   if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE;
+   f->blocksize[0] = f->blocksize_0;
+   f->blocksize[1] = f->blocksize_1;
+
+#ifdef STB_VORBIS_DIVIDE_TABLE
+   if (integer_divide_table[1][1]==0)
+      for (i=0; i < DIVTAB_NUMER; ++i)
+         for (j=1; j < DIVTAB_DENOM; ++j)
+            integer_divide_table[i][j] = i / j;
+#endif
+
+   // compute how much temporary memory is needed
+
+   // 1.
+   {
+      uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1);
+      uint32 classify_mem;
+      int i,max_part_read=0;
+      for (i=0; i < f->residue_count; ++i) {
+         Residue *r = f->residue_config + i;
+         int n_read = r->end - r->begin;
+         int part_read = n_read / r->part_size;
+         if (part_read > max_part_read)
+            max_part_read = part_read;
+      }
+      #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+      classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *));
+      #else
+      classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *));
+      #endif
+
+      f->temp_memory_required = classify_mem;
+      if (imdct_mem > f->temp_memory_required)
+         f->temp_memory_required = imdct_mem;
+   }
+
+   f->first_decode = TRUE;
+
+   if (f->alloc.alloc_buffer) {
+      assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes);
+      // check if there's enough temp memory so we don't error later
+      if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset)
+         return error(f, VORBIS_outofmem);
+   }
+
+   f->first_audio_page_offset = stb_vorbis_get_file_offset(f);
+
+   return TRUE;
+}
+
+static void vorbis_deinit(stb_vorbis *p)
+{
+   int i,j;
+   for (i=0; i < p->residue_count; ++i) {
+      Residue *r = p->residue_config+i;
+      if (r->classdata) {
+         for (j=0; j < p->codebooks[r->classbook].entries; ++j)
+            setup_free(p, r->classdata[j]);
+         setup_free(p, r->classdata);
+      }
+      setup_free(p, r->residue_books);
+   }
+
+   if (p->codebooks) {
+      for (i=0; i < p->codebook_count; ++i) {
+         Codebook *c = p->codebooks + i;
+         setup_free(p, c->codeword_lengths);
+         setup_free(p, c->multiplicands);
+         setup_free(p, c->codewords);
+         setup_free(p, c->sorted_codewords);
+         // c->sorted_values[-1] is the first entry in the array
+         setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL);
+      }
+      setup_free(p, p->codebooks);
+   }
+   setup_free(p, p->floor_config);
+   setup_free(p, p->residue_config);
+   for (i=0; i < p->mapping_count; ++i)
+      setup_free(p, p->mapping[i].chan);
+   setup_free(p, p->mapping);
+   for (i=0; i < p->channels; ++i) {
+      setup_free(p, p->channel_buffers[i]);
+      setup_free(p, p->previous_window[i]);
+      #ifdef STB_VORBIS_NO_DEFER_FLOOR
+      setup_free(p, p->floor_buffers[i]);
+      #endif
+      setup_free(p, p->finalY[i]);
+   }
+   for (i=0; i < 2; ++i) {
+      setup_free(p, p->A[i]);
+      setup_free(p, p->B[i]);
+      setup_free(p, p->C[i]);
+      setup_free(p, p->window[i]);
+   }
+   #ifndef STB_VORBIS_NO_STDIO
+   if (p->close_on_free) fclose(p->f);
+   #endif
+}
+
+void stb_vorbis_close(stb_vorbis *p)
+{
+   if (p == NULL) return;
+   vorbis_deinit(p);
+   setup_free(p,p);
+}
+
+static void vorbis_init(stb_vorbis *p, stb_vorbis_alloc *z)
+{
+   memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start
+   if (z) {
+      p->alloc = *z;
+      p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes+3) & ~3;
+      p->temp_offset = p->alloc.alloc_buffer_length_in_bytes;
+   }
+   p->eof = 0;
+   p->error = VORBIS__no_error;
+   p->stream = NULL;
+   p->codebooks = NULL;
+   p->page_crc_tests = -1;
+   #ifndef STB_VORBIS_NO_STDIO
+   p->close_on_free = FALSE;
+   p->f = NULL;
+   #endif
+}
+
+int stb_vorbis_get_sample_offset(stb_vorbis *f)
+{
+   if (f->current_loc_valid)
+      return f->current_loc;
+   else
+      return -1;
+}
+
+stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f)
+{
+   stb_vorbis_info d;
+   d.channels = f->channels;
+   d.sample_rate = f->sample_rate;
+   d.setup_memory_required = f->setup_memory_required;
+   d.setup_temp_memory_required = f->setup_temp_memory_required;
+   d.temp_memory_required = f->temp_memory_required;
+   d.max_frame_size = f->blocksize_1 >> 1;
+   return d;
+}
+
+int stb_vorbis_get_error(stb_vorbis *f)
+{
+   int e = f->error;
+   f->error = VORBIS__no_error;
+   return e;
+}
+
+static stb_vorbis * vorbis_alloc(stb_vorbis *f)
+{
+   stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p));
+   return p;
+}
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+
+void stb_vorbis_flush_pushdata(stb_vorbis *f)
+{
+   f->previous_length = 0;
+   f->page_crc_tests  = 0;
+   f->discard_samples_deferred = 0;
+   f->current_loc_valid = FALSE;
+   f->first_decode = FALSE;
+   f->samples_output = 0;
+   f->channel_buffer_start = 0;
+   f->channel_buffer_end = 0;
+}
+
+static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len)
+{
+   int i,n;
+   for (i=0; i < f->page_crc_tests; ++i)
+      f->scan[i].bytes_done = 0;
+
+   // if we have room for more scans, search for them first, because
+   // they may cause us to stop early if their header is incomplete
+   if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) {
+      if (data_len < 4) return 0;
+      data_len -= 3; // need to look for 4-byte sequence, so don't miss
+                     // one that straddles a boundary
+      for (i=0; i < data_len; ++i) {
+         if (data[i] == 0x4f) {
+            if (0==memcmp(data+i, ogg_page_header, 4)) {
+               int j,len;
+               uint32 crc;
+               // make sure we have the whole page header
+               if (i+26 >= data_len || i+27+data[i+26] >= data_len) {
+                  // only read up to this page start, so hopefully we'll
+                  // have the whole page header start next time
+                  data_len = i;
+                  break;
+               }
+               // ok, we have it all; compute the length of the page
+               len = 27 + data[i+26];
+               for (j=0; j < data[i+26]; ++j)
+                  len += data[i+27+j];
+               // scan everything up to the embedded crc (which we must 0)
+               crc = 0;
+               for (j=0; j < 22; ++j)
+                  crc = crc32_update(crc, data[i+j]);
+               // now process 4 0-bytes
+               for (   ; j < 26; ++j)
+                  crc = crc32_update(crc, 0);
+               // len is the total number of bytes we need to scan
+               n = f->page_crc_tests++;
+               f->scan[n].bytes_left = len-j;
+               f->scan[n].crc_so_far = crc;
+               f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24);
+               // if the last frame on a page is continued to the next, then
+               // we can't recover the sample_loc immediately
+               if (data[i+27+data[i+26]-1] == 255)
+                  f->scan[n].sample_loc = ~0;
+               else
+                  f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24);
+               f->scan[n].bytes_done = i+j;
+               if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT)
+                  break;
+               // keep going if we still have room for more
+            }
+         }
+      }
+   }
+
+   for (i=0; i < f->page_crc_tests;) {
+      uint32 crc;
+      int j;
+      int n = f->scan[i].bytes_done;
+      int m = f->scan[i].bytes_left;
+      if (m > data_len - n) m = data_len - n;
+      // m is the bytes to scan in the current chunk
+      crc = f->scan[i].crc_so_far;
+      for (j=0; j < m; ++j)
+         crc = crc32_update(crc, data[n+j]);
+      f->scan[i].bytes_left -= m;
+      f->scan[i].crc_so_far = crc;
+      if (f->scan[i].bytes_left == 0) {
+         // does it match?
+         if (f->scan[i].crc_so_far == f->scan[i].goal_crc) {
+            // Houston, we have page
+            data_len = n+m; // consumption amount is wherever that scan ended
+            f->page_crc_tests = -1; // drop out of page scan mode
+            f->previous_length = 0; // decode-but-don't-output one frame
+            f->next_seg = -1;       // start a new page
+            f->current_loc = f->scan[i].sample_loc; // set the current sample location
+                                    // to the amount we'd have decoded had we decoded this page
+            f->current_loc_valid = f->current_loc != ~0;
+            return data_len;
+         }
+         // delete entry
+         f->scan[i] = f->scan[--f->page_crc_tests];
+      } else {
+         ++i;
+      }
+   }
+
+   return data_len;
+}
+
+// return value: number of bytes we used
+int stb_vorbis_decode_frame_pushdata(
+         stb_vorbis *f,                 // the file we're decoding
+         uint8 *data, int data_len,     // the memory available for decoding
+         int *channels,                 // place to write number of float * buffers
+         float ***output,               // place to write float ** array of float * buffers
+         int *samples                   // place to write number of output samples
+     )
+{
+   int i;
+   int len,right,left;
+
+   if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+   if (f->page_crc_tests >= 0) {
+      *samples = 0;
+      return vorbis_search_for_page_pushdata(f, data, data_len);
+   }
+
+   f->stream     = data;
+   f->stream_end = data + data_len;
+   f->error      = VORBIS__no_error;
+
+   // check that we have the entire packet in memory
+   if (!is_whole_packet_present(f, FALSE)) {
+      *samples = 0;
+      return 0;
+   }
+
+   if (!vorbis_decode_packet(f, &len, &left, &right)) {
+      // save the actual error we encountered
+      enum STBVorbisError error = f->error;
+      if (error == VORBIS_bad_packet_type) {
+         // flush and resynch
+         f->error = VORBIS__no_error;
+         while (get8_packet(f) != EOP)
+            if (f->eof) break;
+         *samples = 0;
+         return f->stream - data;
+      }
+      if (error == VORBIS_continued_packet_flag_invalid) {
+         if (f->previous_length == 0) {
+            // we may be resynching, in which case it's ok to hit one
+            // of these; just discard the packet
+            f->error = VORBIS__no_error;
+            while (get8_packet(f) != EOP)
+               if (f->eof) break;
+            *samples = 0;
+            return f->stream - data;
+         }
+      }
+      // if we get an error while parsing, what to do?
+      // well, it DEFINITELY won't work to continue from where we are!
+      stb_vorbis_flush_pushdata(f);
+      // restore the error that actually made us bail
+      f->error = error;
+      *samples = 0;
+      return 1;
+   }
+
+   // success!
+   len = vorbis_finish_frame(f, len, left, right);
+   for (i=0; i < f->channels; ++i)
+      f->outputs[i] = f->channel_buffers[i] + left;
+
+   if (channels) *channels = f->channels;
+   *samples = len;
+   *output = f->outputs;
+   return f->stream - data;
+}
+
+stb_vorbis *stb_vorbis_open_pushdata(
+         unsigned char *data, int data_len, // the memory available for decoding
+         int *data_used,              // only defined if result is not NULL
+         int *error, stb_vorbis_alloc *alloc)
+{
+   stb_vorbis *f, p;
+   vorbis_init(&p, alloc);
+   p.stream     = data;
+   p.stream_end = data + data_len;
+   p.push_mode  = TRUE;
+   if (!start_decoder(&p)) {
+      if (p.eof)
+         *error = VORBIS_need_more_data;
+      else
+         *error = p.error;
+      return NULL;
+   }
+   f = vorbis_alloc(&p);
+   if (f) {
+      *f = p;
+      *data_used = f->stream - data;
+      *error = 0;
+      return f;
+   } else {
+      vorbis_deinit(&p);
+      return NULL;
+   }
+}
+#endif // STB_VORBIS_NO_PUSHDATA_API
+
+unsigned int stb_vorbis_get_file_offset(stb_vorbis *f)
+{
+   #ifndef STB_VORBIS_NO_PUSHDATA_API
+   if (f->push_mode) return 0;
+   #endif
+   if (USE_MEMORY(f)) return f->stream - f->stream_start;
+   #ifndef STB_VORBIS_NO_STDIO
+   return ftell(f->f) - f->f_start;
+   #endif
+}
+
+#ifndef STB_VORBIS_NO_PULLDATA_API
+//
+// DATA-PULLING API
+//
+
+static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
+{
+   for(;;) {
+      int n;
+      if (f->eof) return 0;
+      n = get8(f);
+      if (n == 0x4f) { // page header
+         unsigned int retry_loc = stb_vorbis_get_file_offset(f);
+         int i;
+         // check if we're off the end of a file_section stream
+         if (retry_loc - 25 > f->stream_len)
+            return 0;
+         // check the rest of the header
+         for (i=1; i < 4; ++i)
+            if (get8(f) != ogg_page_header[i])
+               break;
+         if (f->eof) return 0;
+         if (i == 4) {
+            uint8 header[27];
+            uint32 i, crc, goal, len;
+            for (i=0; i < 4; ++i)
+               header[i] = ogg_page_header[i];
+            for (; i < 27; ++i)
+               header[i] = get8(f);
+            if (f->eof) return 0;
+            if (header[4] != 0) goto invalid;
+            goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24);
+            for (i=22; i < 26; ++i)
+               header[i] = 0;
+            crc = 0;
+            for (i=0; i < 27; ++i)
+               crc = crc32_update(crc, header[i]);
+            len = 0;
+            for (i=0; i < header[26]; ++i) {
+               int s = get8(f);
+               crc = crc32_update(crc, s);
+               len += s;
+            }
+            if (len && f->eof) return 0;
+            for (i=0; i < len; ++i)
+               crc = crc32_update(crc, get8(f));
+            // finished parsing probable page
+            if (crc == goal) {
+               // we could now check that it's either got the last
+               // page flag set, OR it's followed by the capture
+               // pattern, but I guess TECHNICALLY you could have
+               // a file with garbage between each ogg page and recover
+               // from it automatically? So even though that paranoia
+               // might decrease the chance of an invalid decode by
+               // another 2^32, not worth it since it would hose those
+               // invalid-but-useful files?
+               if (end)
+                  *end = stb_vorbis_get_file_offset(f);
+               if (last) {
+                  if (header[5] & 0x04)
+                     *last = 1;
+                  else
+                     *last = 0;
+			  }
+               set_file_offset(f, retry_loc-1);
+               return 1;
+            }
+         }
+        invalid:
+         // not a valid page, so rewind and look for next one
+         set_file_offset(f, retry_loc);
+      }
+   }
+}
+
+// seek is implemented with 'interpolation search'--this is like
+// binary search, but we use the data values to estimate the likely
+// location of the data item (plus a bit of a bias so when the
+// estimation is wrong we don't waste overly much time)
+
+#define SAMPLE_unknown  0xffffffff
+
+
+// ogg vorbis, in its insane infinite wisdom, only provides
+// information about the sample at the END of the page.
+// therefore we COULD have the data we need in the current
+// page, and not know it. we could just use the end location
+// as our only knowledge for bounds, seek back, and eventually
+// the binary search finds it. or we can try to be smart and
+// not waste time trying to locate more pages. we try to be
+// smart, since this data is already in memory anyway, so
+// doing needless I/O would be crazy!
+static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z)
+{
+   uint8 header[27], lacing[255];
+   uint8 packet_type[255];
+   int num_packet, packet_start, previous =0;
+   int i,len;
+   uint32 samples;
+
+   // record where the page starts
+   z->page_start = stb_vorbis_get_file_offset(f);
+
+   // parse the header
+   getn(f, header, 27);
+   assert(header[0] == 'O' && header[1] == 'g' && header[2] == 'g' && header[3] == 'S');
+   getn(f, lacing, header[26]);
+
+   // determine the length of the payload
+   len = 0;
+   for (i=0; i < header[26]; ++i)
+      len += lacing[i];
+
+   // this implies where the page ends
+   z->page_end = z->page_start + 27 + header[26] + len;
+
+   // read the last-decoded sample out of the data
+   z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 16);
+
+   if (header[5] & 4) {
+      // if this is the last page, it's not possible to work
+      // backwards to figure out the first sample! whoops! fuck.
+      z->first_decoded_sample = SAMPLE_unknown;
+      set_file_offset(f, z->page_start);
+      return 1;
+   }
+
+   // scan through the frames to determine the sample-count of each one...
+   // our goal is the sample # of the first fully-decoded sample on the
+   // page, which is the first decoded sample of the 2nd page
+
+   num_packet=0;
+
+   packet_start = ((header[5] & 1) == 0);
+
+   for (i=0; i < header[26]; ++i) {
+      if (packet_start) {
+         uint8 n,b,m;
+         if (lacing[i] == 0) goto bail; // trying to read from zero-length packet
+         n = get8(f);
+         // if bottom bit is non-zero, we've got corruption
+         if (n & 1) goto bail;
+         n >>= 1;
+         b = ilog(f->mode_count-1);
+         m = n >> b;
+         n &= (1 << b)-1;
+         if (n >= f->mode_count) goto bail;
+         if (num_packet == 0 && f->mode_config[n].blockflag)
+            previous = (m & 1);
+         packet_type[num_packet++] = f->mode_config[n].blockflag;
+         skip(f, lacing[i]-1);
+      } else
+         skip(f, lacing[i]);
+      packet_start = (lacing[i] < 255);
+   }
+
+   // now that we know the sizes of all the pages, we can start determining
+   // how much sample data there is.
+
+   samples = 0;
+
+   // for the last packet, we step by its whole length, because the definition
+   // is that we encoded the end sample loc of the 'last packet completed',
+   // where 'completed' refers to packets being split, and we are left to guess
+   // what 'end sample loc' means. we assume it means ignoring the fact that
+   // the last half of the data is useless without windowing against the next
+   // packet... (so it's not REALLY complete in that sense)
+   if (num_packet > 1)
+      samples += f->blocksize[packet_type[num_packet-1]];
+
+   for (i=num_packet-2; i >= 1; --i) {
+      // now, for this packet, how many samples do we have that
+      // do not overlap the following packet?
+      if (packet_type[i] == 1)
+         if (packet_type[i+1] == 1)
+            samples += f->blocksize_1 >> 1;
+         else
+            samples += ((f->blocksize_1 - f->blocksize_0) >> 2) + (f->blocksize_0 >> 1);
+      else
+         samples += f->blocksize_0 >> 1;
+   }
+   // now, at this point, we've rewound to the very beginning of the
+   // _second_ packet. if we entirely discard the first packet after
+   // a seek, this will be exactly the right sample number. HOWEVER!
+   // we can't as easily compute this number for the LAST page. The
+   // only way to get the sample offset of the LAST page is to use
+   // the end loc from the previous page. But what that returns us
+   // is _exactly_ the place where we get our first non-overlapped
+   // sample. (I think. Stupid spec for being ambiguous.) So for
+   // consistency it's better to do that here, too. However, that
+   // will then require us to NOT discard all of the first frame we
+   // decode, in some cases, which means an even weirder frame size
+   // and extra code. what a fucking pain.
+   
+   // we're going to discard the first packet if we
+   // start the seek here, so we don't care about it. (we could actually
+   // do better; if the first packet is long, and the previous packet
+   // is short, there's actually data in the first half of the first
+   // packet that doesn't need discarding... but not worth paying the
+   // effort of tracking that of that here and in the seeking logic)
+   // except crap, if we infer it from the _previous_ packet's end
+   // location, we DO need to use that definition... and we HAVE to
+   // infer the start loc of the LAST packet from the previous packet's
+   // end location. fuck you, ogg vorbis.
+
+   z->first_decoded_sample = z->last_decoded_sample - samples;
+
+   // restore file state to where we were
+   set_file_offset(f, z->page_start);
+   return 1;
+
+   // restore file state to where we were
+  bail:
+   set_file_offset(f, z->page_start);
+   return 0;
+}
+
+static int vorbis_seek_frame_from_page(stb_vorbis *f, uint32 page_start, uint32 first_sample, uint32 target_sample, int fine)
+{
+   int left_start, left_end, right_start, right_end, mode,i;
+   int frame=0;
+   uint32 frame_start;
+   int frames_to_skip, data_to_skip;
+
+   // first_sample is the sample # of the first sample that doesn't
+   // overlap the previous page... note that this requires us to
+   // _partially_ discard the first packet! bleh.
+   set_file_offset(f, page_start);
+
+   f->next_seg = -1;  // force page resync
+
+   frame_start = first_sample;
+   // frame start is where the previous packet's last decoded sample
+   // was, which corresponds to left_end... EXCEPT if the previous
+   // packet was long and this packet is short? Probably a bug here.
+
+
+   // now, we can start decoding frames... we'll only FAKE decode them,
+   // until we find the frame that contains our sample; then we'll rewind,
+   // and try again
+   for (;;) {
+      int start;
+
+      if (!vorbis_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode))
+         return error(f, VORBIS_seek_failed);
+
+      if (frame == 0)
+         start = left_end;
+      else
+         start = left_start;
+
+      // the window starts at left_start; the last valid sample we generate
+      // before the next frame's window start is right_start-1
+      if (target_sample < frame_start + right_start-start)
+         break;
+
+      flush_packet(f);
+      if (f->eof)
+         return error(f, VORBIS_seek_failed);
+
+      frame_start += right_start - start;
+
+      ++frame;
+   }
+
+   // ok, at this point, the sample we want is contained in frame #'frame'
+
+   // to decode frame #'frame' normally, we have to decode the
+   // previous frame first... but if it's the FIRST frame of the page
+   // we can't. if it's the first frame, it means it falls in the part
+   // of the first frame that doesn't overlap either of the other frames.
+   // so, if we have to handle that case for the first frame, we might
+   // as well handle it for all of them, so:
+   if (target_sample > frame_start + (left_end - left_start)) {
+      // so what we want to do is go ahead and just immediately decode
+      // this frame, but then make it so the next get_frame_float() uses
+      // this already-decoded data? or do we want to go ahead and rewind,
+      // and leave a flag saying to skip the first N data? let's do that
+      frames_to_skip = frame;  // if this is frame #1, skip 1 frame (#0)
+      data_to_skip = left_end - left_start;
+   } else {
+      // otherwise, we want to skip frames 0, 1, 2, ... frame-2
+      // (which means frame-2+1 total frames) then decode frame-1,
+      // then leave frame pending
+      frames_to_skip = frame - 1;
+      assert(frames_to_skip >= 0);
+      data_to_skip = -1;      
+   }
+
+   set_file_offset(f, page_start);
+   f->next_seg = - 1; // force page resync
+
+   for (i=0; i < frames_to_skip; ++i) {
+      maybe_start_packet(f);
+      flush_packet(f);
+   }
+
+   if (data_to_skip >= 0) {
+      int i,j,n = f->blocksize_0 >> 1;
+      f->discard_samples_deferred = data_to_skip;
+      for (i=0; i < f->channels; ++i)
+         for (j=0; j < n; ++j)
+            f->previous_window[i][j] = 0;
+      f->previous_length = n;
+      frame_start += data_to_skip;
+   } else {
+      f->previous_length = 0;
+      vorbis_pump_first_frame(f);
+   }
+
+   // at this point, the NEXT decoded frame will generate the desired sample
+   if (fine) {
+      // so if we're doing sample accurate streaming, we want to go ahead and decode it!
+      if (target_sample != frame_start) {
+         int n;
+         stb_vorbis_get_frame_float(f, &n, NULL);
+         assert(target_sample > frame_start);
+         assert(f->channel_buffer_start + (int) (target_sample-frame_start) < f->channel_buffer_end);
+         f->channel_buffer_start += (target_sample - frame_start);
+      }
+   }
+
+   return 0;
+}
+
+static int vorbis_seek_base(stb_vorbis *f, unsigned int sample_number, int fine)
+{
+   ProbedPage p[2],q;
+   if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+   // do we know the location of the last page?
+   if (f->p_last.page_start == 0) {
+      uint32 z = stb_vorbis_stream_length_in_samples(f);
+      if (z == 0) return error(f, VORBIS_cant_find_last_page);
+   }
+
+   p[0] = f->p_first;
+   p[1] = f->p_last;
+
+   if (sample_number >= f->p_last.last_decoded_sample)
+      sample_number = f->p_last.last_decoded_sample-1;
+
+   if (sample_number < f->p_first.last_decoded_sample) {
+      vorbis_seek_frame_from_page(f, p[0].page_start, 0, sample_number, fine);
+      return 0;
+   } else {
+      int attempts=0;
+      while (p[0].page_end < p[1].page_start) {
+         uint32 probe;
+         uint32 start_offset, end_offset;
+         uint32 start_sample, end_sample;
+
+         // copy these into local variables so we can tweak them
+         // if any are unknown
+         start_offset = p[0].page_end;
+         end_offset   = p[1].after_previous_page_start; // an address known to seek to page p[1]
+         start_sample = p[0].last_decoded_sample;
+         end_sample   = p[1].last_decoded_sample;
+
+         // currently there is no such tweaking logic needed/possible?
+         if (start_sample == SAMPLE_unknown || end_sample == SAMPLE_unknown)
+            return error(f, VORBIS_seek_failed);
+
+         // now we want to lerp between these for the target samples...
+      
+         // step 1: we need to bias towards the page start...
+         if (start_offset + 4000 < end_offset)
+            end_offset -= 4000;
+
+         // now compute an interpolated search loc
+         probe = start_offset + (int) floor((float) (end_offset - start_offset) / (end_sample - start_sample) * (sample_number - start_sample));
+
+         // next we need to bias towards binary search...
+         // code is a little wonky to allow for full 32-bit unsigned values
+         if (attempts >= 4) {
+            uint32 probe2 = start_offset + ((end_offset - start_offset) >> 1);
+            if (attempts >= 8)
+               probe = probe2;
+            else if (probe < probe2)
+               probe = probe + ((probe2 - probe) >> 1);
+            else
+               probe = probe2 + ((probe - probe2) >> 1);
+         }
+         ++attempts;
+
+         set_file_offset(f, probe);
+         if (!vorbis_find_page(f, NULL, NULL))   return error(f, VORBIS_seek_failed);
+         if (!vorbis_analyze_page(f, &q))        return error(f, VORBIS_seek_failed);
+         q.after_previous_page_start = probe;
+
+         // it's possible we've just found the last page again
+         if (q.page_start == p[1].page_start) {
+            p[1] = q;
+            continue;
+         }
+
+         if (sample_number < q.last_decoded_sample)
+            p[1] = q;
+         else
+            p[0] = q;
+      }
+
+      if (p[0].last_decoded_sample <= sample_number && sample_number < p[1].last_decoded_sample) {
+         vorbis_seek_frame_from_page(f, p[1].page_start, p[0].last_decoded_sample, sample_number, fine);
+         return 0;
+      }
+      return error(f, VORBIS_seek_failed);
+   }
+}
+
+int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number)
+{
+   return vorbis_seek_base(f, sample_number, FALSE);
+}
+
+int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number)
+{
+   return vorbis_seek_base(f, sample_number, TRUE);
+}
+
+void stb_vorbis_seek_start(stb_vorbis *f)
+{
+   if (IS_PUSH_MODE(f)) { error(f, VORBIS_invalid_api_mixing); return; }
+   set_file_offset(f, f->first_audio_page_offset);
+   f->previous_length = 0;
+   f->first_decode = TRUE;
+   f->next_seg = -1;
+   vorbis_pump_first_frame(f);
+}
+
+unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f)
+{
+   unsigned int restore_offset, previous_safe;
+   unsigned int end, last_page_loc;
+
+   if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+   if (!f->total_samples) {
+      int last;
+      uint32 lo,hi;
+      char header[6];
+
+      // first, store the current decode position so we can restore it
+      restore_offset = stb_vorbis_get_file_offset(f);
+
+      // now we want to seek back 64K from the end (the last page must
+      // be at most a little less than 64K, but let's allow a little slop)
+      if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset)
+         previous_safe = f->stream_len - 65536;
+      else
+         previous_safe = f->first_audio_page_offset;
+
+      set_file_offset(f, previous_safe);
+      // previous_safe is now our candidate 'earliest known place that seeking
+      // to will lead to the final page'
+
+      if (!vorbis_find_page(f, &end, (int unsigned *)&last)) {
+         // if we can't find a page, we're hosed!
+         f->error = VORBIS_cant_find_last_page;
+         f->total_samples = 0xffffffff;
+         goto done;
+      }
+
+      // check if there are more pages
+      last_page_loc = stb_vorbis_get_file_offset(f);
+
+      // stop when the last_page flag is set, not when we reach eof;
+      // this allows us to stop short of a 'file_section' end without
+      // explicitly checking the length of the section
+      while (!last) {
+         set_file_offset(f, end);
+         if (!vorbis_find_page(f, &end, (int unsigned *)&last)) {
+            // the last page we found didn't have the 'last page' flag
+            // set. whoops!
+            break;
+         }
+         previous_safe = last_page_loc+1;
+         last_page_loc = stb_vorbis_get_file_offset(f);
+      }
+
+      set_file_offset(f, last_page_loc);
+
+      // parse the header
+      getn(f, (unsigned char *)header, 6);
+      // extract the absolute granule position
+      lo = get32(f);
+      hi = get32(f);
+      if (lo == 0xffffffff && hi == 0xffffffff) {
+         f->error = VORBIS_cant_find_last_page;
+         f->total_samples = SAMPLE_unknown;
+         goto done;
+      }
+      if (hi)
+         lo = 0xfffffffe; // saturate
+      f->total_samples = lo;
+
+      f->p_last.page_start = last_page_loc;
+      f->p_last.page_end   = end;
+      f->p_last.last_decoded_sample = lo;
+      f->p_last.first_decoded_sample = SAMPLE_unknown;
+      f->p_last.after_previous_page_start = previous_safe;
+
+     done:
+      set_file_offset(f, restore_offset);
+   }
+   return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples;
+}
+
+float stb_vorbis_stream_length_in_seconds(stb_vorbis *f)
+{
+   return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate;
+}
+
+
+
+int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output)
+{
+   int len, right,left,i;
+   if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+   if (!vorbis_decode_packet(f, &len, &left, &right)) {
+      f->channel_buffer_start = f->channel_buffer_end = 0;
+      return 0;
+   }
+
+   len = vorbis_finish_frame(f, len, left, right);
+   for (i=0; i < f->channels; ++i)
+      f->outputs[i] = f->channel_buffers[i] + left;
+
+   f->channel_buffer_start = left;
+   f->channel_buffer_end   = left+len;
+
+   if (channels) *channels = f->channels;
+   if (output)   *output = f->outputs;
+   return len;
+}
+
+#ifndef STB_VORBIS_NO_STDIO
+
+stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc, unsigned int length)
+{
+   stb_vorbis *f, p;
+   vorbis_init(&p, alloc);
+   p.f = file;
+   p.f_start = ftell(file);
+   p.stream_len   = length;
+   p.close_on_free = close_on_free;
+   if (start_decoder(&p)) {
+      f = vorbis_alloc(&p);
+      if (f) {
+         *f = p;
+         vorbis_pump_first_frame(f);
+         return f;
+      }
+   }
+   if (error) *error = p.error;
+   vorbis_deinit(&p);
+   return NULL;
+}
+
+stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc)
+{
+   unsigned int len, start;
+   start = ftell(file);
+   fseek(file, 0, SEEK_END);
+   len = ftell(file) - start;
+   fseek(file, start, SEEK_SET);
+   return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len);
+}
+
+stb_vorbis * stb_vorbis_open_filename(char *filename, int *error, stb_vorbis_alloc *alloc)
+{
+   FILE *f = fopen(filename, "rb");
+   if (f) 
+      return stb_vorbis_open_file(f, TRUE, error, alloc);
+   if (error) *error = VORBIS_file_open_failure;
+   return NULL;
+}
+#endif // STB_VORBIS_NO_STDIO
+
+stb_vorbis * stb_vorbis_open_memory(unsigned char *data, int len, int *error, stb_vorbis_alloc *alloc)
+{
+   stb_vorbis *f, p;
+   if (data == NULL) return NULL;
+   vorbis_init(&p, alloc);
+   p.stream = data;
+   p.stream_end = data + len;
+   p.stream_start = p.stream;
+   p.stream_len = len;
+   p.push_mode = FALSE;
+   if (start_decoder(&p)) {
+      f = vorbis_alloc(&p);
+      if (f) {
+         *f = p;
+         vorbis_pump_first_frame(f);
+         return f;
+      }
+   }
+   if (error) *error = p.error;
+   vorbis_deinit(&p);
+   return NULL;
+}
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+#define PLAYBACK_MONO     1
+#define PLAYBACK_LEFT     2
+#define PLAYBACK_RIGHT    4
+
+#define L  (PLAYBACK_LEFT  | PLAYBACK_MONO)
+#define C  (PLAYBACK_LEFT  | PLAYBACK_RIGHT | PLAYBACK_MONO)
+#define R  (PLAYBACK_RIGHT | PLAYBACK_MONO)
+
+static int8 channel_position[7][6] =
+{
+   { 0 },
+   { C },
+   { L, R },
+   { L, C, R },
+   { L, R, L, R },
+   { L, C, R, L, R },
+   { L, C, R, L, R, C },
+};
+
+
+#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
+   typedef union {
+      float f;
+      int i;
+   } float_conv;
+   typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4];
+   #define FASTDEF(x) float_conv x
+   // add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round
+   #define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT))
+   #define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22))
+   #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (temp.f = (x) + MAGIC(s), temp.i - ADDEND(s))
+   #define check_endianness()  
+#else
+   #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s))))
+   #define check_endianness()
+   #define FASTDEF(x)
+#endif
+
+static void copy_samples(short *dest, float *src, int len)
+{
+   int i;
+   check_endianness();
+   for (i=0; i < len; ++i) {
+      FASTDEF(temp);
+      int v = FAST_SCALED_FLOAT_TO_INT(temp, src[i],15);
+      if ((unsigned int) (v + 32768) > 65535)
+         v = v < 0 ? -32768 : 32767;
+      dest[i] = v;
+   }
+}
+
+static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len)
+{
+   #define BUFFER_SIZE  32
+   float buffer[BUFFER_SIZE];
+   int i,j,o,n = BUFFER_SIZE;
+   check_endianness();
+   for (o = 0; o < len; o += BUFFER_SIZE) {
+      memset(buffer, 0, sizeof(buffer));
+      if (o + n > len) n = len - o;
+      for (j=0; j < num_c; ++j) {
+         if (channel_position[num_c][j] & mask) {
+            for (i=0; i < n; ++i)
+               buffer[i] += data[j][d_offset+o+i];
+         }
+      }
+      for (i=0; i < n; ++i) {
+         FASTDEF(temp);
+         int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15);
+         if ((unsigned int) (v + 32768) > 65535)
+            v = v < 0 ? -32768 : 32767;
+         output[o+i] = v;
+      }
+   }
+}
+
+//static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} };
+static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len)
+{
+   #define BUFFER_SIZE  32
+   float buffer[BUFFER_SIZE];
+   int i,j,o,n = BUFFER_SIZE >> 1;
+   // o is the offset in the source data
+   check_endianness();
+   for (o = 0; o < len; o += BUFFER_SIZE >> 1) {
+      // o2 is the offset in the output data
+      int o2 = o << 1;
+      memset(buffer, 0, sizeof(buffer));
+      if (o + n > len) n = len - o;
+      for (j=0; j < num_c; ++j) {
+         int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT);
+         if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) {
+            for (i=0; i < n; ++i) {
+               buffer[i*2+0] += data[j][d_offset+o+i];
+               buffer[i*2+1] += data[j][d_offset+o+i];
+            }
+         } else if (m == PLAYBACK_LEFT) {
+            for (i=0; i < n; ++i) {
+               buffer[i*2+0] += data[j][d_offset+o+i];
+            }
+         } else if (m == PLAYBACK_RIGHT) {
+            for (i=0; i < n; ++i) {
+               buffer[i*2+1] += data[j][d_offset+o+i];
+            }
+         }
+      }
+      for (i=0; i < (n<<1); ++i) {
+         FASTDEF(temp);
+         int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15);
+         if ((unsigned int) (v + 32768) > 65535)
+            v = v < 0 ? -32768 : 32767;
+         output[o2+i] = v;
+      }
+   }
+}
+
+static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples)
+{
+   int i;
+   if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
+      static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} };
+      for (i=0; i < buf_c; ++i)
+         compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples);
+   } else {
+      int limit = buf_c < data_c ? buf_c : data_c;
+      for (i=0; i < limit; ++i)
+         copy_samples(buffer[i]+b_offset, data[i], samples);
+      for (   ; i < buf_c; ++i)
+         memset(buffer[i]+b_offset, 0, sizeof(short) * samples);
+   }
+}
+
+int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples)
+{
+   float **output;
+   int len = stb_vorbis_get_frame_float(f, NULL, &output);
+   if (len > num_samples) len = num_samples;
+   if (len)
+      convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len);
+   return len;
+}
+
+static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len)
+{
+   int i;
+   check_endianness();
+   if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
+      assert(buf_c == 2);
+      for (i=0; i < buf_c; ++i)
+         compute_stereo_samples(buffer, data_c, data, d_offset, len);
+   } else {
+      int limit = buf_c < data_c ? buf_c : data_c;
+      int j;
+      for (j=0; j < len; ++j) {
+         for (i=0; i < limit; ++i) {
+            FASTDEF(temp);
+            float f = data[i][d_offset+j];
+            int v = FAST_SCALED_FLOAT_TO_INT(temp, f,15);//data[i][d_offset+j],15);
+            if ((unsigned int) (v + 32768) > 65535)
+               v = v < 0 ? -32768 : 32767;
+            *buffer++ = v;
+         }
+         for (   ; i < buf_c; ++i)
+            *buffer++ = 0;
+      }
+   }
+}
+
+int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts)
+{
+   float **output;
+   int len;
+   if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts);
+   len = stb_vorbis_get_frame_float(f, NULL, &output);
+   if (len) {
+      if (len*num_c > num_shorts) len = num_shorts / num_c;
+      convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len);
+   }
+   return len;
+}
+
+int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts)
+{
+   float **outputs;
+   int len = num_shorts / channels;
+   int n=0;
+   int z = f->channels;
+   if (z > channels) z = channels;
+   while (n < len) {
+      int k = f->channel_buffer_end - f->channel_buffer_start;
+      if (n+k >= len) k = len - n;
+      if (k)
+         convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k);
+      buffer += k*channels;
+      n += k;
+      f->channel_buffer_start += k;
+      if (n == len) break;
+      if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+   }
+   return n;
+}
+
+int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len)
+{
+   float **outputs;
+   int n=0;
+   int z = f->channels;
+   if (z > channels) z = channels;
+   while (n < len) {
+      int k = f->channel_buffer_end - f->channel_buffer_start;
+      if (n+k >= len) k = len - n;
+      if (k)
+         convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k);
+      n += k;
+      f->channel_buffer_start += k;
+      if (n == len) break;
+      if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+   }
+   return n;
+}
+
+#ifndef STB_VORBIS_NO_STDIO
+int stb_vorbis_decode_filename(char *filename, int *channels, int* sample_rate, short **output)
+{
+   int data_len, offset, total, limit, error;
+   short *data;
+   stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL);
+   if (v == NULL) return -1;
+   limit = v->channels * 4096;
+   *channels = v->channels;
+   *sample_rate = v->sample_rate;
+   offset = data_len = 0;
+   total = limit;
+   data = (short *) malloc(total * sizeof(*data));
+   if (data == NULL) {
+      stb_vorbis_close(v);
+      return -2;
+   }
+   for (;;) {
+      int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
+      if (n == 0) break;
+      data_len += n;
+      offset += n * v->channels;
+      if (offset + limit > total) {
+         short *data2;
+         total *= 2;
+         data2 = (short *) realloc(data, total * sizeof(*data));
+         if (data2 == NULL) {
+            free(data);
+            stb_vorbis_close(v);
+            return -2;
+         }
+         data = data2;
+      }
+   }
+   *output = data;
+   return data_len;
+}
+#endif // NO_STDIO
+
+int stb_vorbis_decode_memory(uint8 *mem, int len, int *channels, int* sample_rate, short **output)
+{
+   int data_len, offset, total, limit, error;
+   short *data;
+   stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL);
+   if (v == NULL) return -1;
+   limit = v->channels * 4096;
+   *channels = v->channels;
+   *sample_rate = v->sample_rate;
+   offset = data_len = 0;
+   total = limit;
+   data = (short *) malloc(total * sizeof(*data));
+   if (data == NULL) {
+      stb_vorbis_close(v);
+      return -2;
+   }
+   for (;;) {
+      int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
+      if (n == 0) break;
+      data_len += n;
+      offset += n * v->channels;
+      if (offset + limit > total) {
+         short *data2;
+         total *= 2;
+         data2 = (short *) realloc(data, total * sizeof(*data));
+         if (data2 == NULL) {
+            free(data);
+            stb_vorbis_close(v);
+            return -2;
+         }
+         data = data2;
+      }
+   }
+   *output = data;
+   return data_len;
+}
+#endif
+
+int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats)
+{
+   float **outputs;
+   int len = num_floats / channels;
+   int n=0;
+   int z = f->channels;
+   if (z > channels) z = channels;
+   while (n < len) {
+      int i,j;
+      int k = f->channel_buffer_end - f->channel_buffer_start;
+      if (n+k >= len) k = len - n;
+      for (j=0; j < k; ++j) {
+         for (i=0; i < z; ++i)
+            *buffer++ = f->channel_buffers[i][f->channel_buffer_start+j];
+         for (   ; i < channels; ++i)
+            *buffer++ = 0;
+      }
+      n += k;
+      f->channel_buffer_start += k;
+      if (n == len) break;
+      if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+   }
+   return n;
+}
+
+int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples)
+{
+   float **outputs;
+   int n=0;
+   int z = f->channels;
+   if (z > channels) z = channels;
+   while (n < num_samples) {
+      int i;
+      int k = f->channel_buffer_end - f->channel_buffer_start;
+      if (n+k >= num_samples) k = num_samples - n;
+      if (k) {
+         for (i=0; i < z; ++i)
+            memcpy(buffer[i]+n, f->channel_buffers+f->channel_buffer_start, sizeof(float)*k);
+         for (   ; i < channels; ++i)
+            memset(buffer[i]+n, 0, sizeof(float) * k);
+      }
+      n += k;
+      f->channel_buffer_start += k;
+      if (n == num_samples) break;
+      if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+   }
+   return n;
+}
+#endif // STB_VORBIS_NO_PULLDATA_API
+
+#endif // STB_VORBIS_HEADER_ONLY
+ proteaaudio.cabal view
@@ -0,0 +1,70 @@+Name:                proteaaudio+Version:             0.6.2+Synopsis:            A wrapper for the proteaaudio library.+Description:         A wrapper for the proteaaudio library. http://viremo.eludi.net/proteaAudio/+License:             BSD3+License-file:        LICENSE+Author:              Csaba Hruska+Maintainer:          csaba (dot) hruska (at) gmail (dot) com+Stability:           Experimental+Category:            Sound+Tested-With:         GHC == 7.8.3+Cabal-Version:       >= 1.2+Build-Type:          Simple++Extra-Source-Files:+  cbits/include/asio.cpp+  cbits/include/asio.h+  cbits/include/asiodrivers.cpp+  cbits/include/asiodrivers.h+  cbits/include/asiodrvr.h+  cbits/include/asiolist.cpp+  cbits/include/asiolist.h+  cbits/include/asiosys.h+  cbits/include/dsound.h+  cbits/include/ginclude.h+  cbits/include/iasiodrv.h+  cbits/include/iasiothiscallresolver.cpp+  cbits/include/iasiothiscallresolver.h+  cbits/include/soundcard.h+  cbits/RtAudio.cpp+  cbits/RtAudio.h+  cbits/RtError.h+  cbits/proAudio.cpp+  cbits/proAudio.h+  cbits/proAudioRt.cpp+  cbits/proAudioRt.h+  cbits/proteaaudio_binding.cpp+  cbits/proteaaudio_binding.h+  cbits/stb_vorbis.c++  Sound/ProteaAudio.chs++Library+  Build-Depends:        base >= 4 && < 5++  Build-tools:          c2hs+  Exposed-Modules:      Sound.ProteaAudio+  Hs-Source-Dirs:       .+  Extensions:           ForeignFunctionInterface++  C-Sources:            cbits/RtAudio.cpp+                        cbits/proAudio.cpp+                        cbits/proAudioRt.cpp+                        cbits/proteaaudio_binding.cpp+                        cbits/stb_vorbis.c++  Include-Dirs:         cbits cbits/include++  if os(windows)+    CC-Options:         "-D__WINDOWS_DS__"+    Extra-Libraries:    stdc++ ole32 dsound winmm+  if os(linux)+    CC-Options:         "-D__LINUX_ALSA__ -D__LINUX_PULSE__ -D__LINUX_OSS__"+    Extra-Libraries:    stdc++ pthread asound+  if os(darwin)+    CC-Options:         "-D__MACOSX_CORE__"+    Extra-Libraries:    stdc++ pthread+    Frameworks:         CoreFoundation CoreAudio++  ghc-options: -O2