Merge pull request #26086 from akien-mga/drop-rtaudio

Drop RtAudio driver on Windows

CODEOWNERS

@@ -18,7 +18,6 @@ doc_classes/*        @godotengine/documentation
 /drivers/coreaudio/  @marcelofg55
 /drivers/coremidi/   @marcelofg55
 /drivers/pulseaudio/ @marcelofg55
-/drivers/rtaudio/    @marcelofg55
 /drivers/wasapi/     @marcelofg55
 /drivers/winmidi/    @marcelofg55
 /drivers/xaudio2/    @marcelofg55

COPYRIGHT.txt

@@ -362,11 +362,6 @@ Comment: Recast
 Copyright: 2009, Mikko Mononen
 License: Zlib
 
-Files: ./thirdparty/rtaudio/
-Comment: RtAudio
-Copyright: 2001-2016, Gary P. Scavone
-License: Expat
-
 Files: ./thirdparty/squish/
 Comment: libSquish
 Copyright: 2006, Simon Brown

drivers/SCsub

@@ -13,7 +13,6 @@ SConscript('alsa/SCsub')
 SConscript('coreaudio/SCsub')
 SConscript('pulseaudio/SCsub')
 if (env["platform"] == "windows"):
-    SConscript("rtaudio/SCsub")
     SConscript("wasapi/SCsub")
 if env['xaudio2']:
     SConscript("xaudio2/SCsub")

drivers/rtaudio/SCsub

@@ -1,23 +0,0 @@
-#!/usr/bin/env python
-
-Import('env')
-
-# Not cloning the env, the includes need to be accessible for platform/
-
-# Thirdparty source files
-thirdparty_dir = "#thirdparty/rtaudio/"
-thirdparty_sources = [
-    "RtAudio.cpp",
-]
-thirdparty_sources = [thirdparty_dir + file for file in thirdparty_sources]
-
-env.Append(CPPPATH=[thirdparty_dir])
-
-env_thirdparty = env.Clone()
-env_thirdparty.disable_warnings()
-env_thirdparty.add_source_files(env.drivers_sources, thirdparty_sources)
-
-# Driver source files
-env.add_source_files(env.drivers_sources, "*.cpp")
-
-Export('env')

drivers/rtaudio/audio_driver_rtaudio.cpp

@@ -1,205 +0,0 @@
-/*************************************************************************/
-/*  audio_driver_rtaudio.cpp                                             */
-/*************************************************************************/
-/*                       This file is part of:                           */
-/*                           GODOT ENGINE                                */
-/*                      https://godotengine.org                          */
-/*************************************************************************/
-/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur.                 */
-/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md)    */
-/*                                                                       */
-/* Permission is hereby granted, free of charge, to any person obtaining */
-/* a copy of this software and associated documentation files (the       */
-/* "Software"), to deal in the Software without restriction, including   */
-/* without limitation the rights to use, copy, modify, merge, publish,   */
-/* distribute, sublicense, and/or sell copies of the Software, and to    */
-/* permit persons to whom the Software is furnished to do so, subject to */
-/* the following conditions:                                             */
-/*                                                                       */
-/* The above copyright notice and this permission notice shall be        */
-/* included in all copies or substantial portions of the Software.       */
-/*                                                                       */
-/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,       */
-/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF    */
-/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
-/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY  */
-/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,  */
-/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE     */
-/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.                */
-/*************************************************************************/
-
-#include "audio_driver_rtaudio.h"
-
-#include "core/os/os.h"
-#include "core/project_settings.h"
-
-#ifdef RTAUDIO_ENABLED
-
-const char *AudioDriverRtAudio::get_name() const {
-
-#ifdef OSX_ENABLED
-	return "RtAudio-OSX";
-#elif defined(UNIX_ENABLED)
-	return "RtAudio-ALSA";
-#elif defined(WINDOWS_ENABLED)
-	return "RtAudio-DirectSound";
-#else
-	return "RtAudio-None";
-#endif
-}
-
-int AudioDriverRtAudio::callback(void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames, double streamTime, RtAudioStreamStatus status, void *userData) {
-
-	if (status) {
-		if (status & RTAUDIO_INPUT_OVERFLOW) {
-			WARN_PRINT("RtAudio input overflow!");
-		}
-		if (status & RTAUDIO_OUTPUT_UNDERFLOW) {
-			WARN_PRINT("RtAudio output underflow!");
-		}
-	}
-	int32_t *buffer = (int32_t *)outputBuffer;
-
-	AudioDriverRtAudio *self = (AudioDriverRtAudio *)userData;
-
-	if (self->mutex->try_lock() != OK) {
-		// what should i do..
-		for (unsigned int i = 0; i < nBufferFrames; i++)
-			buffer[i] = 0;
-
-		return 0;
-	}
-
-	self->audio_server_process(nBufferFrames, buffer);
-
-	self->mutex->unlock();
-
-	return 0;
-}
-
-Error AudioDriverRtAudio::init() {
-
-	active = false;
-	mutex = Mutex::create(true);
-	dac = memnew(RtAudio);
-
-	ERR_EXPLAIN("Cannot initialize RtAudio audio driver: No devices present.")
-	ERR_FAIL_COND_V(dac->getDeviceCount() < 1, ERR_UNAVAILABLE);
-
-	// FIXME: Adapt to the OutputFormat -> SpeakerMode change
-	/*
-	String channels = GLOBAL_DEF_RST("audio/output","stereo");
-
-	if (channels=="5.1")
-		output_format=OUTPUT_5_1;
-	else if (channels=="quad")
-		output_format=OUTPUT_QUAD;
-	else if (channels=="mono")
-		output_format=OUTPUT_MONO;
-	else
-		output_format=OUTPUT_STEREO;
-	*/
-
-	RtAudio::StreamParameters parameters;
-	parameters.deviceId = dac->getDefaultOutputDevice();
-	RtAudio::StreamOptions options;
-
-	// set the desired numberOfBuffers
-	options.numberOfBuffers = 4;
-
-	parameters.firstChannel = 0;
-	mix_rate = GLOBAL_DEF_RST("audio/mix_rate", DEFAULT_MIX_RATE);
-
-	int latency = GLOBAL_DEF("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
-	unsigned int buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
-	print_verbose("Audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");
-
-	short int tries = 4;
-
-	while (tries > 0) {
-		switch (speaker_mode) {
-			case SPEAKER_MODE_STEREO: parameters.nChannels = 2; break;
-			case SPEAKER_SURROUND_31: parameters.nChannels = 4; break;
-			case SPEAKER_SURROUND_51: parameters.nChannels = 6; break;
-			case SPEAKER_SURROUND_71: parameters.nChannels = 8; break;
-		};
-
-		try {
-			dac->openStream(&parameters, NULL, RTAUDIO_SINT32, mix_rate, &buffer_frames, &callback, this, &options);
-			active = true;
-
-			break;
-		} catch (RtAudioError) {
-			// try with less channels
-			ERR_PRINT("Unable to open audio, retrying with fewer channels...");
-
-			switch (speaker_mode) {
-				case SPEAKER_MODE_STEREO: break; // Required to silence unhandled enum value warning.
-				case SPEAKER_SURROUND_31: speaker_mode = SPEAKER_MODE_STEREO; break;
-				case SPEAKER_SURROUND_51: speaker_mode = SPEAKER_SURROUND_31; break;
-				case SPEAKER_SURROUND_71: speaker_mode = SPEAKER_SURROUND_51; break;
-			}
-
-			tries--;
-		}
-	}
-
-	return active ? OK : ERR_UNAVAILABLE;
-}
-
-int AudioDriverRtAudio::get_mix_rate() const {
-
-	return mix_rate;
-}
-
-AudioDriver::SpeakerMode AudioDriverRtAudio::get_speaker_mode() const {
-
-	return speaker_mode;
-}
-
-void AudioDriverRtAudio::start() {
-
-	if (active)
-		dac->startStream();
-}
-
-void AudioDriverRtAudio::lock() {
-
-	if (mutex)
-		mutex->lock();
-}
-
-void AudioDriverRtAudio::unlock() {
-
-	if (mutex)
-		mutex->unlock();
-}
-
-void AudioDriverRtAudio::finish() {
-
-	lock();
-	if (active && dac->isStreamOpen()) {
-		dac->closeStream();
-		active = false;
-	}
-	unlock();
-
-	if (mutex) {
-		memdelete(mutex);
-		mutex = NULL;
-	}
-	if (dac) {
-		memdelete(dac);
-		dac = NULL;
-	}
-}
-
-AudioDriverRtAudio::AudioDriverRtAudio() :
-		speaker_mode(SPEAKER_MODE_STEREO),
-		mutex(NULL),
-		dac(NULL),
-		mix_rate(DEFAULT_MIX_RATE),
-		active(false) {
-}
-
-#endif

drivers/rtaudio/audio_driver_rtaudio.h

@@ -1,65 +0,0 @@
-/*************************************************************************/
-/*  audio_driver_rtaudio.h                                               */
-/*************************************************************************/
-/*                       This file is part of:                           */
-/*                           GODOT ENGINE                                */
-/*                      https://godotengine.org                          */
-/*************************************************************************/
-/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur.                 */
-/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md)    */
-/*                                                                       */
-/* Permission is hereby granted, free of charge, to any person obtaining */
-/* a copy of this software and associated documentation files (the       */
-/* "Software"), to deal in the Software without restriction, including   */
-/* without limitation the rights to use, copy, modify, merge, publish,   */
-/* distribute, sublicense, and/or sell copies of the Software, and to    */
-/* permit persons to whom the Software is furnished to do so, subject to */
-/* the following conditions:                                             */
-/*                                                                       */
-/* The above copyright notice and this permission notice shall be        */
-/* included in all copies or substantial portions of the Software.       */
-/*                                                                       */
-/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,       */
-/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF    */
-/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
-/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY  */
-/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,  */
-/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE     */
-/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.                */
-/*************************************************************************/
-
-#ifndef AUDIO_DRIVER_RTAUDIO_H
-#define AUDIO_DRIVER_RTAUDIO_H
-
-#ifdef RTAUDIO_ENABLED
-
-#include "servers/audio_server.h"
-
-#include <RtAudio.h>
-
-class AudioDriverRtAudio : public AudioDriver {
-
-	static int callback(void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
-			double streamTime, RtAudioStreamStatus status, void *userData);
-	SpeakerMode speaker_mode;
-	Mutex *mutex;
-	RtAudio *dac;
-	int mix_rate;
-	bool active;
-
-public:
-	virtual const char *get_name() const;
-
-	virtual Error init();
-	virtual void start();
-	virtual int get_mix_rate() const;
-	virtual SpeakerMode get_speaker_mode() const;
-	virtual void lock();
-	virtual void unlock();
-	virtual void finish();
-
-	AudioDriverRtAudio();
-};
-
-#endif // AUDIO_DRIVER_RTAUDIO_H
-#endif

platform/server/os_server.h

@@ -32,7 +32,6 @@
 #define OS_SERVER_H
 
 #include "drivers/dummy/texture_loader_dummy.h"
-#include "drivers/rtaudio/audio_driver_rtaudio.h"
 #include "drivers/unix/os_unix.h"
 #include "main/input_default.h"
 #ifdef __APPLE__

platform/windows/crash_handler_windows.cpp

@@ -30,9 +30,9 @@
 
 #include "crash_handler_windows.h"
 
+#include "core/os/os.h"
 #include "core/project_settings.h"
 #include "main/main.h"
-#include "os_windows.h"
 
 #ifdef CRASH_HANDLER_EXCEPTION
 
@@ -41,6 +41,7 @@
 #include <psapi.h>
 #include <algorithm>
 #include <iterator>
+#include <vector>
 
 #pragma comment(lib, "psapi.lib")
 #pragma comment(lib, "dbghelp.lib")

platform/windows/detect.py

@@ -205,8 +205,8 @@ def configure_msvc(env, manual_msvc_config):
             print("Missing environment variable: WindowsSdkDir")
 
     env.AppendUnique(CPPDEFINES = ['WINDOWS_ENABLED', 'OPENGL_ENABLED',
-                                   'RTAUDIO_ENABLED', 'WASAPI_ENABLED',
-                                   'WINMIDI_ENABLED', 'TYPED_METHOD_BIND',
+                                   'WASAPI_ENABLED', 'WINMIDI_ENABLED',
+                                   'TYPED_METHOD_BIND',
                                    'WIN32', 'MSVC',
                                    'WINVER=%s' % env["target_win_version"],
                                    '_WIN32_WINNT=%s' % env["target_win_version"]])
@@ -326,8 +326,8 @@ def configure_mingw(env):
 
     env.Append(CCFLAGS=['-DWINDOWS_ENABLED', '-mwindows'])
     env.Append(CCFLAGS=['-DOPENGL_ENABLED'])
-    env.Append(CCFLAGS=['-DRTAUDIO_ENABLED'])
     env.Append(CCFLAGS=['-DWASAPI_ENABLED'])
+    env.Append(CCFLAGS=['-DWINMIDI_ENABLED'])
     env.Append(CCFLAGS=['-DWINVER=%s' % env['target_win_version'], '-D_WIN32_WINNT=%s' % env['target_win_version']])
     env.Append(LIBS=['mingw32', 'opengl32', 'dsound', 'ole32', 'd3d9', 'winmm', 'gdi32', 'iphlpapi', 'shlwapi', 'wsock32', 'ws2_32', 'kernel32', 'oleaut32', 'dinput8', 'dxguid', 'ksuser', 'imm32', 'bcrypt','avrt'])
 

platform/windows/os_windows.cpp

@@ -52,6 +52,7 @@
 #include "windows_terminal_logger.h"
 
 #include <avrt.h>
+#include <direct.h>
 #include <process.h>
 #include <regstr.h>
 #include <shlobj.h>
@@ -3021,9 +3022,6 @@ OS_Windows::OS_Windows(HINSTANCE _hInstance) {
 #ifdef WASAPI_ENABLED
 	AudioDriverManager::add_driver(&driver_wasapi);
 #endif
-#ifdef RTAUDIO_ENABLED
-	AudioDriverManager::add_driver(&driver_rtaudio);
-#endif
 #ifdef XAUDIO2_ENABLED
 	AudioDriverManager::add_driver(&driver_xaudio2);
 #endif

platform/windows/os_windows.h

@@ -36,7 +36,6 @@
 #include "core/os/os.h"
 #include "core/project_settings.h"
 #include "crash_handler_windows.h"
-#include "drivers/rtaudio/audio_driver_rtaudio.h"
 #include "drivers/unix/ip_unix.h"
 #include "drivers/wasapi/audio_driver_wasapi.h"
 #include "drivers/winmidi/midi_driver_winmidi.h"
@@ -142,9 +141,6 @@ class OS_Windows : public OS {
 #ifdef WASAPI_ENABLED
 	AudioDriverWASAPI driver_wasapi;
 #endif
-#ifdef RTAUDIO_ENABLED
-	AudioDriverRtAudio driver_rtaudio;
-#endif
 #ifdef XAUDIO2_ENABLED
 	AudioDriverXAudio2 driver_xaudio2;
 #endif

thirdparty/README.md

@@ -470,17 +470,6 @@ Files extracted from upstream source:
 - License.txt
 
 
-## rtaudio
-
-- Upstream: http://www.music.mcgill.ca/~gary/rtaudio/
-- Version: 4.1.2
-- License: MIT-like
-
-Files extracted from upstream source:
-
-- `RtAudio.{cpp,h}`
-
-
 ## squish
 
 - Upstream: https://sourceforge.net/projects/libsquish

thirdparty/rtaudio/RtAudio.cpp

@@ -1,10232 +0,0 @@
-#ifdef RTAUDIO_ENABLED // -GODOT-
-
-/************************************************************************/
-/*! \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, ASIO and WASAPI) operating systems.
-
-    RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
-
-    RtAudio: realtime audio i/o C++ classes
-    Copyright (c) 2001-2016 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 4.1.2
-
-#include "RtAudio.h"
-#include <iostream>
-#include <cstdlib>
-#include <cstring>
-#include <climits>
-#include <algorithm>
-
-// 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__) || defined(__WINDOWS_WASAPI__)
-  #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
-  #define MUTEX_DESTROY(A)    DeleteCriticalSection(A)
-  #define MUTEX_LOCK(A)       EnterCriticalSection(A)
-  #define MUTEX_UNLOCK(A)     LeaveCriticalSection(A)
-
-  #include "tchar.h"
-
-  static std::string convertCharPointerToStdString(const char *text)
-  {
-    return std::string(text);
-  }
-
-  static std::string convertCharPointerToStdString(const wchar_t *text)
-  {
-    int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
-    std::string s( length-1, '\0' );
-    WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
-    return s;
-  }
-
-#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.
-//
-// *************************************************** //
-
-std::string RtAudio :: getVersion( void ) throw()
-{
-  return RTAUDIO_VERSION;
-}
-
-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_WASAPI__)
-  apis.push_back( WINDOWS_WASAPI );
-#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_WASAPI__)
-  if ( api == WINDOWS_WASAPI )
-    rtapi_ = new RtApiWasapi();
-#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 )
-{
-  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_ && 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 thow an error.
-  std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
-  throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
-}
-
-RtAudio :: ~RtAudio() throw()
-{
-  if ( rtapi_ )
-    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,
-                            RtAudioErrorCallback errorCallback )
-{
-  return rtapi_->openStream( outputParameters, inputParameters, format,
-                             sampleRate, bufferFrames, callback,
-                             userData, options, errorCallback );
-}
-
-// *************************************************** //
-//
-// 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;
-  firstErrorOccurred_ = false;
-}
-
-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,
-                          RtAudioErrorCallback errorCallback )
-{
-  if ( stream_.state != STREAM_CLOSED ) {
-    errorText_ = "RtApi::openStream: a stream is already open!";
-    error( RtAudioError::INVALID_USE );
-    return;
-  }
-
-  // Clear stream information potentially left from a previously open stream.
-  clearStreamInfo();
-
-  if ( oParams && oParams->nChannels < 1 ) {
-    errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
-    error( RtAudioError::INVALID_USE );
-    return;
-  }
-
-  if ( iParams && iParams->nChannels < 1 ) {
-    errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
-    error( RtAudioError::INVALID_USE );
-    return;
-  }
-
-  if ( oParams == NULL && iParams == NULL ) {
-    errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
-    error( RtAudioError::INVALID_USE );
-    return;
-  }
-
-  if ( formatBytes(format) == 0 ) {
-    errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
-    error( RtAudioError::INVALID_USE );
-    return;
-  }
-
-  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( RtAudioError::INVALID_USE );
-      return;
-    }
-  }
-
-  unsigned int iChannels = 0;
-  if ( iParams ) {
-    iChannels = iParams->nChannels;
-    if ( iParams->deviceId >= nDevices ) {
-      errorText_ = "RtApi::openStream: input device parameter value is invalid.";
-      error( RtAudioError::INVALID_USE );
-      return;
-    }
-  }
-
-  bool result;
-
-  if ( oChannels > 0 ) {
-
-    result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
-                              sampleRate, format, bufferFrames, options );
-    if ( result == false ) {
-      error( RtAudioError::SYSTEM_ERROR );
-      return;
-    }
-  }
-
-  if ( iChannels > 0 ) {
-
-    result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
-                              sampleRate, format, bufferFrames, options );
-    if ( result == false ) {
-      if ( oChannels > 0 ) closeStream();
-      error( RtAudioError::SYSTEM_ERROR );
-      return;
-    }
-  }
-
-  stream_.callbackInfo.callback = (void *) callback;
-  stream_.callbackInfo.userData = userData;
-  stream_.callbackInfo.errorCallback = (void *) errorCallback;
-
-  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
-}
-
-void RtApi :: setStreamTime( double time )
-{
-  verifyStream();
-
-  if ( time >= 0.0 )
-    stream_.streamTime = time;
-}
-
-unsigned int RtApi :: getStreamSampleRate( void )
-{
- verifyStream();
-
- return stream_.sampleRate;
-}
-
-
-// *************************************************** //
-//
-// OS/API-specific methods.
-//
-// *************************************************** //
-
-#if defined(__MACOSX_CORE__)
-
-// The OS X CoreAudio API is designed to use a separate callback
-// procedure for each of its audio devices.  A single RtAudio duplex
-// stream using two different devices is supported here, though it
-// cannot be guaranteed to always behave correctly because we cannot
-// synchronize these two callbacks.
-//
-// A property listener is installed for over/underrun information.
-// However, no functionality is currently provided to allow property
-// listeners to trigger user handlers because it is unclear what could
-// be done if a critical stream parameter (buffer size, sample rate,
-// device disconnect) notification arrived.  The listeners entail
-// quite a bit of extra code and most likely, a user program wouldn't
-// be prepared for the result anyway.  However, we do provide a flag
-// to the client callback function to inform of an over/underrun.
-
-// A structure to hold various information related to the CoreAudio API
-// implementation.
-struct CoreHandle {
-  AudioDeviceID id[2];    // device ids
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
-  AudioDeviceIOProcID procId[2];
-#endif
-  UInt32 iStream[2];      // device stream index (or first if using multiple)
-  UInt32 nStreams[2];     // number of streams to use
-  bool xrun[2];
-  char *deviceBuffer;
-  pthread_cond_t condition;
-  int drainCounter;       // Tracks callback counts when draining
-  bool internalDrain;     // Indicates if stop is initiated from callback or not.
-
-  CoreHandle()
-    :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
-};
-
-RtApiCore:: RtApiCore()
-{
-#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
-  // This is a largely undocumented but absolutely necessary
-  // requirement starting with OS-X 10.6.  If not called, queries and
-  // updates to various audio device properties are not handled
-  // correctly.
-  CFRunLoopRef theRunLoop = NULL;
-  AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
-                                          kAudioObjectPropertyScopeGlobal,
-                                          kAudioObjectPropertyElementMaster };
-  OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
-  if ( result != noErr ) {
-    errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
-    error( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::INVALID_USE );
-    return info;
-  }
-
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
-    error( RtAudioError::INVALID_USE );
-    return info;
-  }
-
-  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( RtAudioError::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( RtAudioError::WARNING );
-    return info;
-  }
-
-  //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
-  int length = CFStringGetLength(cfname);
-  char *mname = (char *)malloc(length * 3 + 1);
-#if defined( UNICODE ) || defined( _UNICODE )
-  CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
-#else
-  CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
-#endif
-  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( RtAudioError::WARNING );
-    return info;
-  }
-
-  //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
-  length = CFStringGetLength(cfname);
-  char *name = (char *)malloc(length * 3 + 1);
-#if defined( UNICODE ) || defined( _UNICODE )
-  CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
-#else
-  CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
-#endif
-  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( RtAudioError::WARNING );
-    return info;
-  }
-
-  // Allocate the AudioBufferList.
-  bufferList = (AudioBufferList *) malloc( dataSize );
-  if ( bufferList == NULL ) {
-    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
-    error( RtAudioError::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( RtAudioError::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( RtAudioError::WARNING );
-    return info;
-  }
-
-  // Allocate the AudioBufferList.
-  bufferList = (AudioBufferList *) malloc( dataSize );
-  if ( bufferList == NULL ) {
-    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
-    error( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::WARNING );
-    return info;
-  }
-
-  // The sample rate reporting mechanism is a bit of a mystery.  It
-  // seems that it can either return individual rates or a range of
-  // rates.  I assume that if the min / max range values are the same,
-  // then that represents a single supported rate and if the min / max
-  // range values are different, the device supports an arbitrary
-  // range of values (though there might be multiple ranges, so we'll
-  // use the most conservative range).
-  Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
-  bool haveValueRange = false;
-  info.sampleRates.clear();
-  for ( UInt32 i=0; i<nRanges; i++ ) {
-    if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
-      unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
-      info.sampleRates.push_back( tmpSr );
-
-      if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
-        info.preferredSampleRate = tmpSr;
-
-    } else {
-      haveValueRange = true;
-      if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
-      if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
-    }
-  }
-
-  if ( haveValueRange ) {
-    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.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
-          info.preferredSampleRate = SAMPLE_RATES[k];
-      }
-    }
-  }
-
-  // Sort and remove any redundant values
-  std::sort( info.sampleRates.begin(), info.sampleRates.end() );
-  info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
-
-  if ( info.sampleRates.size() == 0 ) {
-    errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
-    errorText_ = errorStream_.str();
-    error( RtAudioError::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;
-}
-
-static 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;
-}
-
-static 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;
-}
-
-static 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) {
-    free( bufferList );
-    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 ) {
-      AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
-      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( RtAudioError::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;
-  property.mScope = kAudioObjectPropertyScopeGlobal;
-  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;
-  }
-
-  stream_.state = STREAM_CLOSED;
-  return FAILURE;
-}
-
-void RtApiCore :: closeStream( void )
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiCore::closeStream(): no open stream to close!";
-    error( RtAudioError::WARNING );
-    return;
-  }
-
-  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    if (handle) {
-      AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
-        kAudioObjectPropertyScopeGlobal,
-        kAudioObjectPropertyElementMaster };
-
-      property.mSelector = kAudioDeviceProcessorOverload;
-      property.mScope = kAudioObjectPropertyScopeGlobal;
-      if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
-        errorText_ = "RtApiCore::closeStream(): error removing property listener!";
-        error( RtAudioError::WARNING );
-      }
-    }
-    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 (handle) {
-      AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
-        kAudioObjectPropertyScopeGlobal,
-        kAudioObjectPropertyElementMaster };
-
-      property.mSelector = kAudioDeviceProcessorOverload;
-      property.mScope = kAudioObjectPropertyScopeGlobal;
-      if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
-        errorText_ = "RtApiCore::closeStream(): error removing property listener!";
-        error( RtAudioError::WARNING );
-      }
-    }
-    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( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiCore :: stopStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
-    error( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiCore :: abortStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
-    error( RtAudioError::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.
-static 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( RtAudioError::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 ) {
-    ThreadHandle threadId;
-
-    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;
-        }
-      }
-    }
-  }
-
-  // Don't bother draining input
-  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; }
-};
-
-static 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( RtAudioError::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( RtAudioError::INVALID_USE );
-    return info;
-  }
-
-  // Get the current jack server sample rate.
-  info.sampleRates.clear();
-
-  info.preferredSampleRate = jack_get_sample_rate( client );
-  info.sampleRates.push_back( info.preferredSampleRate );
-
-  // 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( RtAudioError::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;
-}
-
-static 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.
-static void *jackCloseStream( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiJack *object = (RtApiJack *) info->object;
-
-  object->closeStream();
-
-  pthread_exit( NULL );
-}
-static 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;
-
-  ThreadHandle threadId;
-  pthread_create( &threadId, NULL, jackCloseStream, info );
-  std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
-}
-
-static 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( RtAudioError::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 ] ) {
-    // Added by Ge Wang
-    jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
-    // the range (usually the min and max are equal)
-    jack_latency_range_t latrange; latrange.min = latrange.max = 0;
-    // get the latency range
-    jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
-    // be optimistic, use the min!
-    stream_.latency[mode] = latrange.min;
-    //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( RtAudioError::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( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiJack :: stopStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
-    error( RtAudioError::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( RtAudioError::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.
-static 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( RtAudioError::WARNING );
-    return FAILURE;
-  }
-  if ( stream_.bufferSize != nframes ) {
-    errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
-    error( RtAudioError::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 ) {
-    ThreadHandle threadId;
-
-    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 );
-      }
-    }
-  }
-
-  // Don't bother draining input
-  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>
-
-static AsioDrivers drivers;
-static ASIOCallbacks asioCallbacks;
-static ASIODriverInfo driverInfo;
-static CallbackInfo *asioCallbackInfo;
-static 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 );
-static void sampleRateChanged( ASIOSampleRate sRate );
-static 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( RtAudioError::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( RtAudioError::INVALID_USE );
-    return info;
-  }
-
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
-    error( RtAudioError::INVALID_USE );
-    return info;
-  }
-
-  // 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( RtAudioError::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( RtAudioError::WARNING );
-    return info;
-  }
-
-  info.name = driverName;
-
-  if ( !drivers.loadDriver( driverName ) ) {
-    errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtAudioError::WARNING );
-    return info;
-  }
-
-  result = ASIOInit( &driverInfo );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtAudioError::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( RtAudioError::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] );
-
-      if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
-        info.preferredSampleRate = 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( RtAudioError::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;
-  else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
-    info.nativeFormats |= RTAUDIO_SINT24;
-
-  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;
-}
-
-static 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 )
-{////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
-
-  bool isDuplexInput =  mode == INPUT && stream_.mode == OUTPUT;
-
-  // For ASIO, a duplex stream MUST use the same driver.
-  if ( isDuplexInput && 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 ( !isDuplexInput ) {
-    // 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;
-    }
-  }
-
-  // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
-  bool buffersAllocated = false;
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  unsigned int nChannels;
-
-
-  // Check the device channel count.
-  long inputChannels, outputChannels;
-  result = ASIOGetChannels( &inputChannels, &outputChannels );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-
-  if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
-       ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-  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 ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-
-  // Get the current sample rate
-  ASIOSampleRate currentRate;
-  result = ASIOGetSampleRate( &currentRate );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-
-  // Set the sample rate only if necessary
-  if ( currentRate != sampleRate ) {
-    result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
-    if ( result != ASE_OK ) {
-      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
-      errorText_ = errorStream_.str();
-      goto error;
-    }
-  }
-
-  // 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 ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-
-  // 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;
-  }
-  else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-    if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
-  }
-
-  if ( stream_.deviceFormat[mode] == 0 ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-
-  // 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 ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-
-  if ( isDuplexInput ) {
-    // When this is the duplex input (output was opened before), then we have to use the same
-    // buffersize as the output, because it might use the preferred buffer size, which most
-    // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
-    // So instead of throwing an error, make them equal. The caller uses the reference
-    // to the "bufferSize" param as usual to set up processing buffers.
-
-    *bufferSize = stream_.bufferSize;
-
-  } else {
-    if ( *bufferSize == 0 ) *bufferSize = preferSize;
-    else 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;
-    }
-  }
-
-  /*
-  // we don't use it anymore, see above!
-  // Just left it here for the case...
-  if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
-    errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
-    goto error;
-  }
-  */
-
-  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.
-  if ( handle == 0 ) {
-    try {
-      handle = new AsioHandle;
-    }
-    catch ( std::bad_alloc& ) {
-      errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
-      goto error;
-    }
-    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.
-  if ( mode == INPUT && stream_.mode == OUTPUT ) {
-    ASIODisposeBuffers();
-    if ( handle->bufferInfos ) free( handle->bufferInfos );
-  }
-
-  // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
-  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;
-  }
-
-  // prepare for callbacks
-  stream_.sampleRate = sampleRate;
-  stream_.device[mode] = device;
-  stream_.mode = isDuplexInput ? DUPLEX : mode;
-
-  // store this class instance before registering callbacks, that are going to use it
-  asioCallbackInfo = &stream_.callbackInfo;
-  stream_.callbackInfo.object = (void *) this;
-
-  // 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 ) {
-    // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
-    // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver
-    // in that case, let's be naïve and try that instead
-    *bufferSize = preferSize;
-    stream_.bufferSize = *bufferSize;
-    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;
-  stream_.state = STREAM_STOPPED;
-
-  // 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 ( isDuplexInput && 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;
-      }
-    }
-  }
-
-  // Determine device latencies
-  long inputLatency, outputLatency;
-  result = ASIOGetLatencies( &inputLatency, &outputLatency );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
-    errorText_ = errorStream_.str();
-    error( RtAudioError::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 ( !isDuplexInput ) {
-    // the cleanup for error in the duplex input, is done by RtApi::openStream
-    // So we clean up for single channel only
-
-    if ( buffersAllocated )
-      ASIODisposeBuffers();
-
-    drivers.removeCurrentDriver();
-
-    if ( handle ) {
-      CloseHandle( handle->condition );
-      if ( handle->bufferInfos )
-        free( handle->bufferInfos );
-
-      delete handle;
-      stream_.apiHandle = 0;
-    }
-
-
-    if ( stream_.userBuffer[mode] ) {
-      free( stream_.userBuffer[mode] );
-      stream_.userBuffer[mode] = 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( RtAudioError::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( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiAsio :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
-    error( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiAsio :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
-    error( RtAudioError::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.
-static 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( RtAudioError::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 );
-      }
-
-    }
-  }
-
-  // Don't bother draining input
-  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;
-}
-
-static 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 ( RtAudioError &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;
-}
-
-static 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 const 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_WASAPI__) // Windows WASAPI API
-
-// Authored by Marcus Tomlinson <[email protected]>, April 2014
-// - Introduces support for the Windows WASAPI API
-// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
-// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
-// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
-
-#ifndef INITGUID
-  #define INITGUID
-#endif
-#include <audioclient.h>
-#include <avrt.h>
-#include <mmdeviceapi.h>
-#include <functiondiscoverykeys_devpkey.h>
-
-//=============================================================================
-
-#define SAFE_RELEASE( objectPtr )\
-if ( objectPtr )\
-{\
-  objectPtr->Release();\
-  objectPtr = NULL;\
-}
-
-typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
-
-//-----------------------------------------------------------------------------
-
-// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
-// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
-// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
-// provide intermediate storage for read / write synchronization.
-class WasapiBuffer
-{
-public:
-  WasapiBuffer()
-    : buffer_( NULL ),
-      bufferSize_( 0 ),
-      inIndex_( 0 ),
-      outIndex_( 0 ) {}
-
-  ~WasapiBuffer() {
-    free( buffer_ );
-  }
-
-  // sets the length of the internal ring buffer
-  void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
-    free( buffer_ );
-
-    buffer_ = ( char* ) calloc( bufferSize, formatBytes );
-
-    bufferSize_ = bufferSize;
-    inIndex_ = 0;
-    outIndex_ = 0;
-  }
-
-  // attempt to push a buffer into the ring buffer at the current "in" index
-  bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
-  {
-    if ( !buffer ||                 // incoming buffer is NULL
-         bufferSize == 0 ||         // incoming buffer has no data
-         bufferSize > bufferSize_ ) // incoming buffer too large
-    {
-      return false;
-    }
-
-    unsigned int relOutIndex = outIndex_;
-    unsigned int inIndexEnd = inIndex_ + bufferSize;
-    if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
-      relOutIndex += bufferSize_;
-    }
-
-    // "in" index can end on the "out" index but cannot begin at it
-    if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
-      return false; // not enough space between "in" index and "out" index
-    }
-
-    // copy buffer from external to internal
-    int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
-    fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
-    int fromInSize = bufferSize - fromZeroSize;
-
-    switch( format )
-      {
-      case RTAUDIO_SINT8:
-        memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
-        memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
-        break;
-      case RTAUDIO_SINT16:
-        memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
-        memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
-        break;
-      case RTAUDIO_SINT24:
-        memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
-        memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
-        break;
-      case RTAUDIO_SINT32:
-        memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
-        memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
-        break;
-      case RTAUDIO_FLOAT32:
-        memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
-        memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
-        break;
-      case RTAUDIO_FLOAT64:
-        memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
-        memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
-        break;
-    }
-
-    // update "in" index
-    inIndex_ += bufferSize;
-    inIndex_ %= bufferSize_;
-
-    return true;
-  }
-
-  // attempt to pull a buffer from the ring buffer from the current "out" index
-  bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
-  {
-    if ( !buffer ||                 // incoming buffer is NULL
-         bufferSize == 0 ||         // incoming buffer has no data
-         bufferSize > bufferSize_ ) // incoming buffer too large
-    {
-      return false;
-    }
-
-    unsigned int relInIndex = inIndex_;
-    unsigned int outIndexEnd = outIndex_ + bufferSize;
-    if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
-      relInIndex += bufferSize_;
-    }
-
-    // "out" index can begin at and end on the "in" index
-    if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
-      return false; // not enough space between "out" index and "in" index
-    }
-
-    // copy buffer from internal to external
-    int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
-    fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
-    int fromOutSize = bufferSize - fromZeroSize;
-
-    switch( format )
-    {
-      case RTAUDIO_SINT8:
-        memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
-        memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
-        break;
-      case RTAUDIO_SINT16:
-        memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
-        memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
-        break;
-      case RTAUDIO_SINT24:
-        memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
-        memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
-        break;
-      case RTAUDIO_SINT32:
-        memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
-        memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
-        break;
-      case RTAUDIO_FLOAT32:
-        memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
-        memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
-        break;
-      case RTAUDIO_FLOAT64:
-        memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
-        memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
-        break;
-    }
-
-    // update "out" index
-    outIndex_ += bufferSize;
-    outIndex_ %= bufferSize_;
-
-    return true;
-  }
-
-private:
-  char* buffer_;
-  unsigned int bufferSize_;
-  unsigned int inIndex_;
-  unsigned int outIndex_;
-};
-
-//-----------------------------------------------------------------------------
-
-// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
-// between HW and the user. The convertBufferWasapi function is used to perform this conversion
-// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
-// This sample rate converter favors speed over quality, and works best with conversions between
-// one rate and its multiple.
-void convertBufferWasapi( char* outBuffer,
-                          const char* inBuffer,
-                          const unsigned int& channelCount,
-                          const unsigned int& inSampleRate,
-                          const unsigned int& outSampleRate,
-                          const unsigned int& inSampleCount,
-                          unsigned int& outSampleCount,
-                          const RtAudioFormat& format )
-{
-  // calculate the new outSampleCount and relative sampleStep
-  float sampleRatio = ( float ) outSampleRate / inSampleRate;
-  float sampleStep = 1.0f / sampleRatio;
-  float inSampleFraction = 0.0f;
-
-  outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );
-
-  // frame-by-frame, copy each relative input sample into it's corresponding output sample
-  for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
-  {
-    unsigned int inSample = ( unsigned int ) inSampleFraction;
-
-    switch ( format )
-    {
-      case RTAUDIO_SINT8:
-        memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
-        break;
-      case RTAUDIO_SINT16:
-        memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
-        break;
-      case RTAUDIO_SINT24:
-        memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
-        break;
-      case RTAUDIO_SINT32:
-        memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
-        break;
-      case RTAUDIO_FLOAT32:
-        memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
-        break;
-      case RTAUDIO_FLOAT64:
-        memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
-        break;
-    }
-
-    // jump to next in sample
-    inSampleFraction += sampleStep;
-  }
-}
-
-//-----------------------------------------------------------------------------
-
-// A structure to hold various information related to the WASAPI implementation.
-struct WasapiHandle
-{
-  IAudioClient* captureAudioClient;
-  IAudioClient* renderAudioClient;
-  IAudioCaptureClient* captureClient;
-  IAudioRenderClient* renderClient;
-  HANDLE captureEvent;
-  HANDLE renderEvent;
-
-  WasapiHandle()
-  : captureAudioClient( NULL ),
-    renderAudioClient( NULL ),
-    captureClient( NULL ),
-    renderClient( NULL ),
-    captureEvent( NULL ),
-    renderEvent( NULL ) {}
-};
-
-//=============================================================================
-
-RtApiWasapi::RtApiWasapi()
-  : coInitialized_( false ), deviceEnumerator_( NULL )
-{
-  // WASAPI can run either apartment or multi-threaded
-  HRESULT hr = CoInitialize( NULL );
-  if ( !FAILED( hr ) )
-    coInitialized_ = true;
-
-  // Instantiate device enumerator
-  hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
-                         CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
-                         ( void** ) &deviceEnumerator_ );
-
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
-    error( RtAudioError::DRIVER_ERROR );
-  }
-}
-
-//-----------------------------------------------------------------------------
-
-RtApiWasapi::~RtApiWasapi()
-{
-  if ( stream_.state != STREAM_CLOSED )
-    closeStream();
-
-  SAFE_RELEASE( deviceEnumerator_ );
-
-  // If this object previously called CoInitialize()
-  if ( coInitialized_ )
-    CoUninitialize();
-}
-
-//=============================================================================
-
-unsigned int RtApiWasapi::getDeviceCount( void )
-{
-  unsigned int captureDeviceCount = 0;
-  unsigned int renderDeviceCount = 0;
-
-  IMMDeviceCollection* captureDevices = NULL;
-  IMMDeviceCollection* renderDevices = NULL;
-
-  // Count capture devices
-  errorText_.clear();
-  HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
-    goto Exit;
-  }
-
-  hr = captureDevices->GetCount( &captureDeviceCount );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
-    goto Exit;
-  }
-
-  // Count render devices
-  hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
-    goto Exit;
-  }
-
-  hr = renderDevices->GetCount( &renderDeviceCount );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
-    goto Exit;
-  }
-
-Exit:
-  // release all references
-  SAFE_RELEASE( captureDevices );
-  SAFE_RELEASE( renderDevices );
-
-  if ( errorText_.empty() )
-    return captureDeviceCount + renderDeviceCount;
-
-  error( RtAudioError::DRIVER_ERROR );
-  return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  unsigned int captureDeviceCount = 0;
-  unsigned int renderDeviceCount = 0;
-  std::string defaultDeviceName;
-  bool isCaptureDevice = false;
-
-  PROPVARIANT deviceNameProp;
-  PROPVARIANT defaultDeviceNameProp;
-
-  IMMDeviceCollection* captureDevices = NULL;
-  IMMDeviceCollection* renderDevices = NULL;
-  IMMDevice* devicePtr = NULL;
-  IMMDevice* defaultDevicePtr = NULL;
-  IAudioClient* audioClient = NULL;
-  IPropertyStore* devicePropStore = NULL;
-  IPropertyStore* defaultDevicePropStore = NULL;
-
-  WAVEFORMATEX* deviceFormat = NULL;
-  WAVEFORMATEX* closestMatchFormat = NULL;
-
-  // probed
-  info.probed = false;
-
-  // Count capture devices
-  errorText_.clear();
-  RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
-  HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
-    goto Exit;
-  }
-
-  hr = captureDevices->GetCount( &captureDeviceCount );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
-    goto Exit;
-  }
-
-  // Count render devices
-  hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
-    goto Exit;
-  }
-
-  hr = renderDevices->GetCount( &renderDeviceCount );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
-    goto Exit;
-  }
-
-  // validate device index
-  if ( device >= captureDeviceCount + renderDeviceCount ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
-    errorType = RtAudioError::INVALID_USE;
-    goto Exit;
-  }
-
-  // determine whether index falls within capture or render devices
-  if ( device >= renderDeviceCount ) {
-    hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
-      goto Exit;
-    }
-    isCaptureDevice = true;
-  }
-  else {
-    hr = renderDevices->Item( device, &devicePtr );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
-      goto Exit;
-    }
-    isCaptureDevice = false;
-  }
-
-  // get default device name
-  if ( isCaptureDevice ) {
-    hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
-      goto Exit;
-    }
-  }
-  else {
-    hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
-      goto Exit;
-    }
-  }
-
-  hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
-    goto Exit;
-  }
-  PropVariantInit( &defaultDeviceNameProp );
-
-  hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
-    goto Exit;
-  }
-
-  defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
-
-  // name
-  hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
-    goto Exit;
-  }
-
-  PropVariantInit( &deviceNameProp );
-
-  hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
-    goto Exit;
-  }
-
-  info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
-
-  // is default
-  if ( isCaptureDevice ) {
-    info.isDefaultInput = info.name == defaultDeviceName;
-    info.isDefaultOutput = false;
-  }
-  else {
-    info.isDefaultInput = false;
-    info.isDefaultOutput = info.name == defaultDeviceName;
-  }
-
-  // channel count
-  hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
-    goto Exit;
-  }
-
-  hr = audioClient->GetMixFormat( &deviceFormat );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
-    goto Exit;
-  }
-
-  if ( isCaptureDevice ) {
-    info.inputChannels = deviceFormat->nChannels;
-    info.outputChannels = 0;
-    info.duplexChannels = 0;
-  }
-  else {
-    info.inputChannels = 0;
-    info.outputChannels = deviceFormat->nChannels;
-    info.duplexChannels = 0;
-  }
-
-  // sample rates
-  info.sampleRates.clear();
-
-  // allow support for all sample rates as we have a built-in sample rate converter
-  for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
-    info.sampleRates.push_back( SAMPLE_RATES[i] );
-  }
-  info.preferredSampleRate = deviceFormat->nSamplesPerSec;
-
-  // native format
-  info.nativeFormats = 0;
-
-  if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
-       ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
-         ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
-  {
-    if ( deviceFormat->wBitsPerSample == 32 ) {
-      info.nativeFormats |= RTAUDIO_FLOAT32;
-    }
-    else if ( deviceFormat->wBitsPerSample == 64 ) {
-      info.nativeFormats |= RTAUDIO_FLOAT64;
-    }
-  }
-  else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
-           ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
-             ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
-  {
-    if ( deviceFormat->wBitsPerSample == 8 ) {
-      info.nativeFormats |= RTAUDIO_SINT8;
-    }
-    else if ( deviceFormat->wBitsPerSample == 16 ) {
-      info.nativeFormats |= RTAUDIO_SINT16;
-    }
-    else if ( deviceFormat->wBitsPerSample == 24 ) {
-      info.nativeFormats |= RTAUDIO_SINT24;
-    }
-    else if ( deviceFormat->wBitsPerSample == 32 ) {
-      info.nativeFormats |= RTAUDIO_SINT32;
-    }
-  }
-
-  // probed
-  info.probed = true;
-
-Exit:
-  // release all references
-  PropVariantClear( &deviceNameProp );
-  PropVariantClear( &defaultDeviceNameProp );
-
-  SAFE_RELEASE( captureDevices );
-  SAFE_RELEASE( renderDevices );
-  SAFE_RELEASE( devicePtr );
-  SAFE_RELEASE( defaultDevicePtr );
-  SAFE_RELEASE( audioClient );
-  SAFE_RELEASE( devicePropStore );
-  SAFE_RELEASE( defaultDevicePropStore );
-
-  CoTaskMemFree( deviceFormat );
-  CoTaskMemFree( closestMatchFormat );
-
-  if ( !errorText_.empty() )
-    error( errorType );
-  return info;
-}
-
-//-----------------------------------------------------------------------------
-
-unsigned int RtApiWasapi::getDefaultOutputDevice( void )
-{
-  for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
-    if ( getDeviceInfo( i ).isDefaultOutput ) {
-      return i;
-    }
-  }
-
-  return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-unsigned int RtApiWasapi::getDefaultInputDevice( void )
-{
-  for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
-    if ( getDeviceInfo( i ).isDefaultInput ) {
-      return i;
-    }
-  }
-
-  return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::closeStream( void )
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
-    error( RtAudioError::WARNING );
-    return;
-  }
-
-  if ( stream_.state != STREAM_STOPPED )
-    stopStream();
-
-  // clean up stream memory
-  SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
-  SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
-
-  SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
-  SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
-
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
-    CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
-
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
-    CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
-
-  delete ( WasapiHandle* ) stream_.apiHandle;
-  stream_.apiHandle = NULL;
-
-  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;
-  }
-
-  // update stream state
-  stream_.state = STREAM_CLOSED;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::startStream( void )
-{
-  verifyStream();
-
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiWasapi::startStream: The stream is already running.";
-    error( RtAudioError::WARNING );
-    return;
-  }
-
-  // update stream state
-  stream_.state = STREAM_RUNNING;
-
-  // create WASAPI stream thread
-  stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
-
-  if ( !stream_.callbackInfo.thread ) {
-    errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
-    error( RtAudioError::THREAD_ERROR );
-  }
-  else {
-    SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
-    ResumeThread( ( void* ) stream_.callbackInfo.thread );
-  }
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::stopStream( void )
-{
-  verifyStream();
-
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
-    error( RtAudioError::WARNING );
-    return;
-  }
-
-  // inform stream thread by setting stream state to STREAM_STOPPING
-  stream_.state = STREAM_STOPPING;
-
-  // wait until stream thread is stopped
-  while( stream_.state != STREAM_STOPPED ) {
-    Sleep( 1 );
-  }
-
-  // Wait for the last buffer to play before stopping.
-  Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
-
-  // stop capture client if applicable
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
-    HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
-      error( RtAudioError::DRIVER_ERROR );
-      return;
-    }
-  }
-
-  // stop render client if applicable
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
-    HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
-      error( RtAudioError::DRIVER_ERROR );
-      return;
-    }
-  }
-
-  // close thread handle
-  if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
-    errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
-    error( RtAudioError::THREAD_ERROR );
-    return;
-  }
-
-  stream_.callbackInfo.thread = (ThreadHandle) NULL;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::abortStream( void )
-{
-  verifyStream();
-
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
-    error( RtAudioError::WARNING );
-    return;
-  }
-
-  // inform stream thread by setting stream state to STREAM_STOPPING
-  stream_.state = STREAM_STOPPING;
-
-  // wait until stream thread is stopped
-  while ( stream_.state != STREAM_STOPPED ) {
-    Sleep( 1 );
-  }
-
-  // stop capture client if applicable
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
-    HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
-      error( RtAudioError::DRIVER_ERROR );
-      return;
-    }
-  }
-
-  // stop render client if applicable
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
-    HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
-      error( RtAudioError::DRIVER_ERROR );
-      return;
-    }
-  }
-
-  // close thread handle
-  if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
-    errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
-    error( RtAudioError::THREAD_ERROR );
-    return;
-  }
-
-  stream_.callbackInfo.thread = (ThreadHandle) NULL;
-}
-
-//-----------------------------------------------------------------------------
-
-bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                   unsigned int firstChannel, unsigned int sampleRate,
-                                   RtAudioFormat format, unsigned int* bufferSize,
-                                   RtAudio::StreamOptions* options )
-{
-  bool methodResult = FAILURE;
-  unsigned int captureDeviceCount = 0;
-  unsigned int renderDeviceCount = 0;
-
-  IMMDeviceCollection* captureDevices = NULL;
-  IMMDeviceCollection* renderDevices = NULL;
-  IMMDevice* devicePtr = NULL;
-  WAVEFORMATEX* deviceFormat = NULL;
-  unsigned int bufferBytes;
-  stream_.state = STREAM_STOPPED;
-
-  // create API Handle if not already created
-  if ( !stream_.apiHandle )
-    stream_.apiHandle = ( void* ) new WasapiHandle();
-
-  // Count capture devices
-  errorText_.clear();
-  RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
-  HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
-    goto Exit;
-  }
-
-  hr = captureDevices->GetCount( &captureDeviceCount );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
-    goto Exit;
-  }
-
-  // Count render devices
-  hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
-    goto Exit;
-  }
-
-  hr = renderDevices->GetCount( &renderDeviceCount );
-  if ( FAILED( hr ) ) {
-    errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
-    goto Exit;
-  }
-
-  // validate device index
-  if ( device >= captureDeviceCount + renderDeviceCount ) {
-    errorType = RtAudioError::INVALID_USE;
-    errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
-    goto Exit;
-  }
-
-  // determine whether index falls within capture or render devices
-  if ( device >= renderDeviceCount ) {
-    if ( mode != INPUT ) {
-      errorType = RtAudioError::INVALID_USE;
-      errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
-      goto Exit;
-    }
-
-    // retrieve captureAudioClient from devicePtr
-    IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
-
-    hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
-      goto Exit;
-    }
-
-    hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
-                              NULL, ( void** ) &captureAudioClient );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
-      goto Exit;
-    }
-
-    hr = captureAudioClient->GetMixFormat( &deviceFormat );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
-      goto Exit;
-    }
-
-    stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
-    captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
-  }
-  else {
-    if ( mode != OUTPUT ) {
-      errorType = RtAudioError::INVALID_USE;
-      errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
-      goto Exit;
-    }
-
-    // retrieve renderAudioClient from devicePtr
-    IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
-
-    hr = renderDevices->Item( device, &devicePtr );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
-      goto Exit;
-    }
-
-    hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
-                              NULL, ( void** ) &renderAudioClient );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
-      goto Exit;
-    }
-
-    hr = renderAudioClient->GetMixFormat( &deviceFormat );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
-      goto Exit;
-    }
-
-    stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
-    renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
-  }
-
-  // fill stream data
-  if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
-       ( stream_.mode == INPUT && mode == OUTPUT ) ) {
-    stream_.mode = DUPLEX;
-  }
-  else {
-    stream_.mode = mode;
-  }
-
-  stream_.device[mode] = device;
-  stream_.doByteSwap[mode] = false;
-  stream_.sampleRate = sampleRate;
-  stream_.bufferSize = *bufferSize;
-  stream_.nBuffers = 1;
-  stream_.nUserChannels[mode] = channels;
-  stream_.channelOffset[mode] = firstChannel;
-  stream_.userFormat = format;
-  stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
-
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
-    stream_.userInterleaved = false;
-  else
-    stream_.userInterleaved = true;
-  stream_.deviceInterleaved[mode] = true;
-
-  // Set flags for buffer conversion.
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] ||
-       stream_.nUserChannels != stream_.nDeviceChannels )
-    stream_.doConvertBuffer[mode] = true;
-  else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-            stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
-
-  if ( stream_.doConvertBuffer[mode] )
-    setConvertInfo( mode, 0 );
-
-  // Allocate necessary internal buffers
-  bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
-
-  stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
-  if ( !stream_.userBuffer[mode] ) {
-    errorType = RtAudioError::MEMORY_ERROR;
-    errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
-    goto Exit;
-  }
-
-  if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
-    stream_.callbackInfo.priority = 15;
-  else
-    stream_.callbackInfo.priority = 0;
-
-  ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
-  ///! TODO: RTAUDIO_HOG_DEVICE       // Exclusive mode
-
-  methodResult = SUCCESS;
-
-Exit:
-  //clean up
-  SAFE_RELEASE( captureDevices );
-  SAFE_RELEASE( renderDevices );
-  SAFE_RELEASE( devicePtr );
-  CoTaskMemFree( deviceFormat );
-
-  // if method failed, close the stream
-  if ( methodResult == FAILURE )
-    closeStream();
-
-  if ( !errorText_.empty() )
-    error( errorType );
-  return methodResult;
-}
-
-//=============================================================================
-
-DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
-{
-  if ( wasapiPtr )
-    ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
-
-  return 0;
-}
-
-DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
-{
-  if ( wasapiPtr )
-    ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
-
-  return 0;
-}
-
-DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
-{
-  if ( wasapiPtr )
-    ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
-
-  return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::wasapiThread()
-{
-  // as this is a new thread, we must CoInitialize it
-  CoInitialize( NULL );
-
-  HRESULT hr;
-
-  IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
-  IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
-  IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
-  IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
-  HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
-  HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
-
-  WAVEFORMATEX* captureFormat = NULL;
-  WAVEFORMATEX* renderFormat = NULL;
-  float captureSrRatio = 0.0f;
-  float renderSrRatio = 0.0f;
-  WasapiBuffer captureBuffer;
-  WasapiBuffer renderBuffer;
-
-  // declare local stream variables
-  RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
-  BYTE* streamBuffer = NULL;
-  unsigned long captureFlags = 0;
-  unsigned int bufferFrameCount = 0;
-  unsigned int numFramesPadding = 0;
-  unsigned int convBufferSize = 0;
-  bool callbackPushed = false;
-  bool callbackPulled = false;
-  bool callbackStopped = false;
-  int callbackResult = 0;
-
-  // convBuffer is used to store converted buffers between WASAPI and the user
-  char* convBuffer = NULL;
-  unsigned int convBuffSize = 0;
-  unsigned int deviceBuffSize = 0;
-
-  errorText_.clear();
-  RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
-
-  // Attempt to assign "Pro Audio" characteristic to thread
-  HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
-  if ( AvrtDll ) {
-    DWORD taskIndex = 0;
-    TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
-    AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
-    FreeLibrary( AvrtDll );
-  }
-
-  // start capture stream if applicable
-  if ( captureAudioClient ) {
-    hr = captureAudioClient->GetMixFormat( &captureFormat );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
-      goto Exit;
-    }
-
-    captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
-
-    // initialize capture stream according to desire buffer size
-    float desiredBufferSize = stream_.bufferSize * captureSrRatio;
-    REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
-
-    if ( !captureClient ) {
-      hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
-                                           AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
-                                           desiredBufferPeriod,
-                                           desiredBufferPeriod,
-                                           captureFormat,
-                                           NULL );
-      if ( FAILED( hr ) ) {
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
-        goto Exit;
-      }
-
-      hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
-                                           ( void** ) &captureClient );
-      if ( FAILED( hr ) ) {
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
-        goto Exit;
-      }
-
-      // configure captureEvent to trigger on every available capture buffer
-      captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
-      if ( !captureEvent ) {
-        errorType = RtAudioError::SYSTEM_ERROR;
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
-        goto Exit;
-      }
-
-      hr = captureAudioClient->SetEventHandle( captureEvent );
-      if ( FAILED( hr ) ) {
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
-        goto Exit;
-      }
-
-      ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
-      ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
-    }
-
-    unsigned int inBufferSize = 0;
-    hr = captureAudioClient->GetBufferSize( &inBufferSize );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
-      goto Exit;
-    }
-
-    // scale outBufferSize according to stream->user sample rate ratio
-    unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
-    inBufferSize *= stream_.nDeviceChannels[INPUT];
-
-    // set captureBuffer size
-    captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
-
-    // reset the capture stream
-    hr = captureAudioClient->Reset();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
-      goto Exit;
-    }
-
-    // start the capture stream
-    hr = captureAudioClient->Start();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
-      goto Exit;
-    }
-  }
-
-  // start render stream if applicable
-  if ( renderAudioClient ) {
-    hr = renderAudioClient->GetMixFormat( &renderFormat );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
-      goto Exit;
-    }
-
-    renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
-
-    // initialize render stream according to desire buffer size
-    float desiredBufferSize = stream_.bufferSize * renderSrRatio;
-    REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
-
-    if ( !renderClient ) {
-      hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
-                                          AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
-                                          desiredBufferPeriod,
-                                          desiredBufferPeriod,
-                                          renderFormat,
-                                          NULL );
-      if ( FAILED( hr ) ) {
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
-        goto Exit;
-      }
-
-      hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
-                                          ( void** ) &renderClient );
-      if ( FAILED( hr ) ) {
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
-        goto Exit;
-      }
-
-      // configure renderEvent to trigger on every available render buffer
-      renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
-      if ( !renderEvent ) {
-        errorType = RtAudioError::SYSTEM_ERROR;
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
-        goto Exit;
-      }
-
-      hr = renderAudioClient->SetEventHandle( renderEvent );
-      if ( FAILED( hr ) ) {
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
-        goto Exit;
-      }
-
-      ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
-      ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
-    }
-
-    unsigned int outBufferSize = 0;
-    hr = renderAudioClient->GetBufferSize( &outBufferSize );
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
-      goto Exit;
-    }
-
-    // scale inBufferSize according to user->stream sample rate ratio
-    unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
-    outBufferSize *= stream_.nDeviceChannels[OUTPUT];
-
-    // set renderBuffer size
-    renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
-
-    // reset the render stream
-    hr = renderAudioClient->Reset();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
-      goto Exit;
-    }
-
-    // start the render stream
-    hr = renderAudioClient->Start();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
-      goto Exit;
-    }
-  }
-
-  if ( stream_.mode == INPUT ) {
-    convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
-    deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
-  }
-  else if ( stream_.mode == OUTPUT ) {
-    convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
-    deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
-  }
-  else if ( stream_.mode == DUPLEX ) {
-    convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
-                             ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
-    deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
-                               stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
-  }
-
-  convBuffer = ( char* ) malloc( convBuffSize );
-  stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
-  if ( !convBuffer || !stream_.deviceBuffer ) {
-    errorType = RtAudioError::MEMORY_ERROR;
-    errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
-    goto Exit;
-  }
-
-  // stream process loop
-  while ( stream_.state != STREAM_STOPPING ) {
-    if ( !callbackPulled ) {
-      // Callback Input
-      // ==============
-      // 1. Pull callback buffer from inputBuffer
-      // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
-      //                          Convert callback buffer to user format
-
-      if ( captureAudioClient ) {
-        // Pull callback buffer from inputBuffer
-        callbackPulled = captureBuffer.pullBuffer( convBuffer,
-                                                   ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
-                                                   stream_.deviceFormat[INPUT] );
-
-        if ( callbackPulled ) {
-          // Convert callback buffer to user sample rate
-          convertBufferWasapi( stream_.deviceBuffer,
-                               convBuffer,
-                               stream_.nDeviceChannels[INPUT],
-                               captureFormat->nSamplesPerSec,
-                               stream_.sampleRate,
-                               ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
-                               convBufferSize,
-                               stream_.deviceFormat[INPUT] );
-
-          if ( stream_.doConvertBuffer[INPUT] ) {
-            // Convert callback buffer to user format
-            convertBuffer( stream_.userBuffer[INPUT],
-                           stream_.deviceBuffer,
-                           stream_.convertInfo[INPUT] );
-          }
-          else {
-            // no further conversion, simple copy deviceBuffer to userBuffer
-            memcpy( stream_.userBuffer[INPUT],
-                    stream_.deviceBuffer,
-                    stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
-          }
-        }
-      }
-      else {
-        // if there is no capture stream, set callbackPulled flag
-        callbackPulled = true;
-      }
-
-      // Execute Callback
-      // ================
-      // 1. Execute user callback method
-      // 2. Handle return value from callback
-
-      // if callback has not requested the stream to stop
-      if ( callbackPulled && !callbackStopped ) {
-        // Execute user callback method
-        callbackResult = callback( stream_.userBuffer[OUTPUT],
-                                   stream_.userBuffer[INPUT],
-                                   stream_.bufferSize,
-                                   getStreamTime(),
-                                   captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
-                                   stream_.callbackInfo.userData );
-
-        // Handle return value from callback
-        if ( callbackResult == 1 ) {
-          // instantiate a thread to stop this thread
-          HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
-          if ( !threadHandle ) {
-            errorType = RtAudioError::THREAD_ERROR;
-            errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
-            goto Exit;
-          }
-          else if ( !CloseHandle( threadHandle ) ) {
-            errorType = RtAudioError::THREAD_ERROR;
-            errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
-            goto Exit;
-          }
-
-          callbackStopped = true;
-        }
-        else if ( callbackResult == 2 ) {
-          // instantiate a thread to stop this thread
-          HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
-          if ( !threadHandle ) {
-            errorType = RtAudioError::THREAD_ERROR;
-            errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
-            goto Exit;
-          }
-          else if ( !CloseHandle( threadHandle ) ) {
-            errorType = RtAudioError::THREAD_ERROR;
-            errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
-            goto Exit;
-          }
-
-          callbackStopped = true;
-        }
-      }
-    }
-
-    // Callback Output
-    // ===============
-    // 1. Convert callback buffer to stream format
-    // 2. Convert callback buffer to stream sample rate and channel count
-    // 3. Push callback buffer into outputBuffer
-
-    if ( renderAudioClient && callbackPulled ) {
-      if ( stream_.doConvertBuffer[OUTPUT] ) {
-        // Convert callback buffer to stream format
-        convertBuffer( stream_.deviceBuffer,
-                       stream_.userBuffer[OUTPUT],
-                       stream_.convertInfo[OUTPUT] );
-
-      }
-
-      // Convert callback buffer to stream sample rate
-      convertBufferWasapi( convBuffer,
-                           stream_.deviceBuffer,
-                           stream_.nDeviceChannels[OUTPUT],
-                           stream_.sampleRate,
-                           renderFormat->nSamplesPerSec,
-                           stream_.bufferSize,
-                           convBufferSize,
-                           stream_.deviceFormat[OUTPUT] );
-
-      // Push callback buffer into outputBuffer
-      callbackPushed = renderBuffer.pushBuffer( convBuffer,
-                                                convBufferSize * stream_.nDeviceChannels[OUTPUT],
-                                                stream_.deviceFormat[OUTPUT] );
-    }
-    else {
-      // if there is no render stream, set callbackPushed flag
-      callbackPushed = true;
-    }
-
-    // Stream Capture
-    // ==============
-    // 1. Get capture buffer from stream
-    // 2. Push capture buffer into inputBuffer
-    // 3. If 2. was successful: Release capture buffer
-
-    if ( captureAudioClient ) {
-      // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
-      if ( !callbackPulled ) {
-        WaitForSingleObject( captureEvent, INFINITE );
-      }
-
-      // Get capture buffer from stream
-      hr = captureClient->GetBuffer( &streamBuffer,
-                                     &bufferFrameCount,
-                                     &captureFlags, NULL, NULL );
-      if ( FAILED( hr ) ) {
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
-        goto Exit;
-      }
-
-      if ( bufferFrameCount != 0 ) {
-        // Push capture buffer into inputBuffer
-        if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
-                                       bufferFrameCount * stream_.nDeviceChannels[INPUT],
-                                       stream_.deviceFormat[INPUT] ) )
-        {
-          // Release capture buffer
-          hr = captureClient->ReleaseBuffer( bufferFrameCount );
-          if ( FAILED( hr ) ) {
-            errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
-            goto Exit;
-          }
-        }
-        else
-        {
-          // Inform WASAPI that capture was unsuccessful
-          hr = captureClient->ReleaseBuffer( 0 );
-          if ( FAILED( hr ) ) {
-            errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
-            goto Exit;
-          }
-        }
-      }
-      else
-      {
-        // Inform WASAPI that capture was unsuccessful
-        hr = captureClient->ReleaseBuffer( 0 );
-        if ( FAILED( hr ) ) {
-          errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
-          goto Exit;
-        }
-      }
-    }
-
-    // Stream Render
-    // =============
-    // 1. Get render buffer from stream
-    // 2. Pull next buffer from outputBuffer
-    // 3. If 2. was successful: Fill render buffer with next buffer
-    //                          Release render buffer
-
-    if ( renderAudioClient ) {
-      // if the callback output buffer was not pushed to renderBuffer, wait for next render event
-      if ( callbackPulled && !callbackPushed ) {
-        WaitForSingleObject( renderEvent, INFINITE );
-      }
-
-      // Get render buffer from stream
-      hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
-      if ( FAILED( hr ) ) {
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
-        goto Exit;
-      }
-
-      hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
-      if ( FAILED( hr ) ) {
-        errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
-        goto Exit;
-      }
-
-      bufferFrameCount -= numFramesPadding;
-
-      if ( bufferFrameCount != 0 ) {
-        hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
-        if ( FAILED( hr ) ) {
-          errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
-          goto Exit;
-        }
-
-        // Pull next buffer from outputBuffer
-        // Fill render buffer with next buffer
-        if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
-                                      bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
-                                      stream_.deviceFormat[OUTPUT] ) )
-        {
-          // Release render buffer
-          hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
-          if ( FAILED( hr ) ) {
-            errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
-            goto Exit;
-          }
-        }
-        else
-        {
-          // Inform WASAPI that render was unsuccessful
-          hr = renderClient->ReleaseBuffer( 0, 0 );
-          if ( FAILED( hr ) ) {
-            errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
-            goto Exit;
-          }
-        }
-      }
-      else
-      {
-        // Inform WASAPI that render was unsuccessful
-        hr = renderClient->ReleaseBuffer( 0, 0 );
-        if ( FAILED( hr ) ) {
-          errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
-          goto Exit;
-        }
-      }
-    }
-
-    // if the callback buffer was pushed renderBuffer reset callbackPulled flag
-    if ( callbackPushed ) {
-      callbackPulled = false;
-      // tick stream time
-      RtApi::tickStreamTime();
-    }
-
-  }
-
-Exit:
-  // clean up
-  CoTaskMemFree( captureFormat );
-  CoTaskMemFree( renderFormat );
-
-  free ( convBuffer );
-
-  CoUninitialize();
-
-  // update stream state
-  stream_.state = STREAM_STOPPED;
-
-  if ( errorText_.empty() )
-    return;
-  else
-    error( errorType );
-}
-
-//******************** End of __WINDOWS_WASAPI__ *********************//
-#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 );
-
-static 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; }
-};
-
-struct DsProbeData {
-  bool isInput;
-  std::vector<struct 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.
-  struct DsProbeData probeInfo;
-  probeInfo.isInput = false;
-  probeInfo.dsDevices = &dsDevices;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
-    errorText_ = errorStream_.str();
-    error( RtAudioError::WARNING );
-  }
-
-  // Query DirectSoundCapture devices.
-  probeInfo.isInput = true;
-  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
-    errorText_ = errorStream_.str();
-    error( RtAudioError::WARNING );
-  }
-
-  // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
-  for ( unsigned int i=0; i<dsDevices.size(); ) {
-    if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
-    else i++;
-  }
-
-  return static_cast<unsigned int>(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( RtAudioError::INVALID_USE );
-      return info;
-    }
-  }
-
-  if ( device >= dsDevices.size() ) {
-    errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
-    error( RtAudioError::INVALID_USE );
-    return info;
-  }
-
-  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( RtAudioError::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( RtAudioError::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] );
-
-      if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
-        info.preferredSampleRate = 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( RtAudioError::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( RtAudioError::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;
-  }
-
-  size_t 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;
-  }
-
-  stream_.state = STREAM_CLOSED;
-  return FAILURE;
-}
-
-void RtApiDs :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiDs::closeStream(): no open stream to close!";
-    error( RtAudioError::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( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiDs :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
-    error( RtAudioError::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;
-
-    MUTEX_LOCK( &stream_.mutex );
-
-    // Stop the buffer and clear memory
-    LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-    result = buffer->Stop();
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // Lock the buffer and clear it so that if we start to play again,
-    // we won't have old data playing.
-    result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // Zero the DS buffer
-    ZeroMemory( audioPtr, dataLen );
-
-    // Unlock the DS buffer
-    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // If we start playing again, we must begin at beginning of buffer.
-    handle->bufferPointer[0] = 0;
-  }
-
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-    LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-    audioPtr = NULL;
-    dataLen = 0;
-
-    stream_.state = STREAM_STOPPED;
-
-    if ( stream_.mode != DUPLEX )
-      MUTEX_LOCK( &stream_.mutex );
-
-    result = buffer->Stop();
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // Lock the buffer and clear it so that if we start to play again,
-    // we won't have old data playing.
-    result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // Zero the DS buffer
-    ZeroMemory( audioPtr, dataLen );
-
-    // Unlock the DS buffer
-    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // If we start recording again, we must begin at beginning of buffer.
-    handle->bufferPointer[1] = 0;
-  }
-
- unlock:
-  timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiDs :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
-    error( RtAudioError::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( RtAudioError::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;
-
-  MUTEX_LOCK( &stream_.mutex );
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  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();
-        MUTEX_UNLOCK( &stream_.mutex );
-        error( RtAudioError::SYSTEM_ERROR );
-        return;
-      }
-      result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
-      if ( FAILED( result ) ) {
-        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
-        errorText_ = errorStream_.str();
-        MUTEX_UNLOCK( &stream_.mutex );
-        error( RtAudioError::SYSTEM_ERROR );
-        return;
-      }
-      while ( true ) {
-        result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
-        if ( FAILED( result ) ) {
-          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
-          errorText_ = errorStream_.str();
-          MUTEX_UNLOCK( &stream_.mutex );
-          error( RtAudioError::SYSTEM_ERROR );
-          return;
-        }
-        result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
-        if ( FAILED( result ) ) {
-          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
-          errorText_ = errorStream_.str();
-          MUTEX_UNLOCK( &stream_.mutex );
-          error( RtAudioError::SYSTEM_ERROR );
-          return;
-        }
-        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();
-        MUTEX_UNLOCK( &stream_.mutex );
-        error( RtAudioError::SYSTEM_ERROR );
-        return;
-      }
-      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();
-        MUTEX_UNLOCK( &stream_.mutex );
-        error( RtAudioError::SYSTEM_ERROR );
-        return;
-      }
-
-      // 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();
-      MUTEX_UNLOCK( &stream_.mutex );
-      error( RtAudioError::SYSTEM_ERROR );
-      return;
-    }
-
-    // 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();
-      MUTEX_UNLOCK( &stream_.mutex );
-      error( RtAudioError::SYSTEM_ERROR );
-      return;
-    }
-    nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
-    handle->bufferPointer[0] = nextWritePointer;
-  }
-
-  // Don't bother draining input
-  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();
-      MUTEX_UNLOCK( &stream_.mutex );
-      error( RtAudioError::SYSTEM_ERROR );
-      return;
-    }
-
-    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();
-          MUTEX_UNLOCK( &stream_.mutex );
-          error( RtAudioError::SYSTEM_ERROR );
-          return;
-        }
-
-        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();
-      MUTEX_UNLOCK( &stream_.mutex );
-      error( RtAudioError::SYSTEM_ERROR );
-      return;
-    }
-
-    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();
-      MUTEX_UNLOCK( &stream_.mutex );
-      error( RtAudioError::SYSTEM_ERROR );
-      return;
-    }
-    handle->bufferPointer[1] = nextReadPointer;
-
-    // No byte swapping necessary in DirectSound implementation.
-
-    // If necessary, convert 8-bit data from unsigned to signed.
-    if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
-      for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
-
-    // Do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[1] )
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-  }
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-  RtApi::tickStreamTime();
-}
-
-// Definitions for utility functions and callbacks
-// specific to the DirectSound implementation.
-
-static 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;
-}
-
-static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
-                                          LPCTSTR description,
-                                          LPCTSTR /*module*/,
-                                          LPVOID lpContext )
-{
-  struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
-  std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
-
-  HRESULT hr;
-  bool validDevice = false;
-  if ( probeInfo.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 = convertCharPointerToStdString( description );
-  //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
-  if ( lpguid == NULL )
-    name = "Default Device";
-  if ( validDevice ) {
-    for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
-      if ( dsDevices[i].name == name ) {
-        dsDevices[i].found = true;
-        if ( probeInfo.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 ( probeInfo.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; }
-};
-
-static 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( RtAudioError::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( RtAudioError::WARNING );
-        break;
-      }
-      if ( subdevice < 0 )
-        break;
-      nDevices++;
-    }
-  nextcard:
-    snd_ctl_close( handle );
-    snd_card_next( &card );
-  }
-
-  result = snd_ctl_open( &handle, "default", 0 );
-  if (result == 0) {
-    nDevices++;
-    snd_ctl_close( handle );
-  }
-
-  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;
-  subdevice = -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( RtAudioError::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( RtAudioError::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 );
-  }
-
-  result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
-  if ( result == 0 ) {
-    if ( nDevices == device ) {
-      strcpy( name, "default" );
-      goto foundDevice;
-    }
-    nDevices++;
-  }
-
-  if ( nDevices == 0 ) {
-    errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
-    error( RtAudioError::INVALID_USE );
-    return info;
-  }
-
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
-    error( RtAudioError::INVALID_USE );
-    return info;
-  }
-
- 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( RtAudioError::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 unless default device (which has subdev -1)
-  stream = SND_PCM_STREAM_PLAYBACK;
-  snd_pcm_info_set_stream( pcminfo, stream );
-  if ( subdevice != -1 ) {
-    snd_pcm_info_set_device( pcminfo, subdevice );
-    snd_pcm_info_set_subdevice( pcminfo, 0 );
-
-    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( RtAudioError::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( RtAudioError::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( RtAudioError::WARNING );
-    goto captureProbe;
-  }
-  info.outputChannels = value;
-  snd_pcm_close( phandle );
-
- captureProbe:
-  stream = SND_PCM_STREAM_CAPTURE;
-  snd_pcm_info_set_stream( pcminfo, stream );
-
-  // Now try for capture unless default device (with subdev = -1)
-  if ( subdevice != -1 ) {
-    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;
-    }
-  }
-  else
-    snd_ctl_close( chandle );
-
-  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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
-        info.preferredSampleRate = 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( RtAudioError::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 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    error( RtAudioError::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 );
-    free( cardname );
-  }
-  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 );
-    }
-
-    result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
-    if ( result == 0 ) {
-      if ( nDevices == device ) {
-        strcpy( name, "default" );
-        goto foundDevice;
-      }
-      nDevices++;
-    }
-
-    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( RtAudioError::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 ) {
-      // We previously attempted to increase the audio callback priority
-      // to SCHED_RR here via the attributes.  However, while no errors
-      // were reported in doing so, it did not work.  So, now this is
-      // done in the alsaCallbackHandler function.
-      stream_.callbackInfo.doRealtime = true;
-      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;
-      stream_.callbackInfo.priority = priority;
-    }
-#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;
-  }
-
-  stream_.state = STREAM_CLOSED;
-  return FAILURE;
-}
-
-void RtApiAlsa :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
-    error( RtAudioError::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( RtAudioError::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 ) {
-    result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
-    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( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiAlsa :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
-    error( RtAudioError::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:
-  apiInfo->runnable = false; // fixes high CPU usage when stopped
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result >= 0 ) return;
-  error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiAlsa :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
-    error( RtAudioError::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:
-  apiInfo->runnable = false; // fixes high CPU usage when stopped
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result >= 0 ) return;
-  error( RtAudioError::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( RtAudioError::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( RtAudioError::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
-            errorText_ =  "RtApiAlsa::callbackEvent: audio write error, underrun.";
-        }
-        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( RtAudioError::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();
-}
-
-static void *alsaCallbackHandler( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiAlsa *object = (RtApiAlsa *) info->object;
-  bool *isRunning = &info->isRunning;
-
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
-  if ( info->doRealtime ) {
-    pthread_t tID = pthread_self();	 // ID of this thread
-    sched_param prio = { info->priority }; // scheduling priority of thread
-    pthread_setschedparam( tID, SCHED_RR, &prio );
-  }
-#endif
-
-  while ( *isRunning == true ) {
-    pthread_testcancel();
-    object->callbackEvent();
-  }
-
-  pthread_exit( NULL );
-}
-
-//******************** End of __LINUX_ALSA__ *********************//
-#endif
-
-#if defined(__LINUX_PULSE__)
-
-// Code written by Peter Meerwald, [email protected]
-// and Tristan Matthews.
-
-#include <pulse/error.h>
-#include <pulse/simple.h>
-#include <cstdio>
-
-static 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.preferredSampleRate = 48000;
-  info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
-
-  return info;
-}
-
-static 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( RtAudioError::WARNING );
-    return;
-  }
-
-  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
-  double streamTime = getStreamTime();
-  RtAudioStreamStatus status = 0;
-  int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
-                               stream_.bufferSize, streamTime, status,
-                               stream_.callbackInfo.userData );
-
-  if ( doStopStream == 2 ) {
-    abortStream();
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-  void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
-  void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
-
-  if ( stream_.state != STREAM_RUNNING )
-    goto unlock;
-
-  int pa_error;
-  size_t bytes;
-  if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    if ( stream_.doConvertBuffer[OUTPUT] ) {
-        convertBuffer( stream_.deviceBuffer,
-                       stream_.userBuffer[OUTPUT],
-                       stream_.convertInfo[OUTPUT] );
-        bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
-                formatBytes( stream_.deviceFormat[OUTPUT] );
-    } else
-        bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
-                formatBytes( stream_.userFormat );
-
-    if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
-      errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
-        pa_strerror( pa_error ) << ".";
-      errorText_ = errorStream_.str();
-      error( RtAudioError::WARNING );
-    }
-  }
-
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
-    if ( stream_.doConvertBuffer[INPUT] )
-      bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
-        formatBytes( stream_.deviceFormat[INPUT] );
-    else
-      bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
-        formatBytes( stream_.userFormat );
-
-    if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
-      errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
-        pa_strerror( pa_error ) << ".";
-      errorText_ = errorStream_.str();
-      error( RtAudioError::WARNING );
-    }
-    if ( stream_.doConvertBuffer[INPUT] ) {
-      convertBuffer( stream_.userBuffer[INPUT],
-                     stream_.deviceBuffer,
-                     stream_.convertInfo[INPUT] );
-    }
-  }
-
- 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( RtAudioError::INVALID_USE );
-    return;
-  }
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiPulse::startStream(): the stream is already running!";
-    error( RtAudioError::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( RtAudioError::INVALID_USE );
-    return;
-  }
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
-    error( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
-      return;
-    }
-  }
-
-  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( RtAudioError::INVALID_USE );
-    return;
-  }
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
-    error( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
-      return;
-    }
-  }
-
-  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;
-      stream_.deviceFormat[mode] = stream_.userFormat;
-      ss.format = sf->pa_format;
-      break;
-    }
-  }
-  if ( !sf_found ) { // Use internal data format conversion.
-    stream_.userFormat = format;
-    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-    ss.format = PA_SAMPLE_FLOAT32LE;
-  }
-
-  // Set other stream parameters.
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-  else stream_.userInterleaved = true;
-  stream_.deviceInterleaved[mode] = true;
-  stream_.nBuffers = 1;
-  stream_.doByteSwap[mode] = false;
-  stream_.nUserChannels[mode] = channels;
-  stream_.nDeviceChannels[mode] = channels + firstChannel;
-  stream_.channelOffset[mode] = 0;
-  std::string streamName = "RtAudio";
-
-  // 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;
-
-  // 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_.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_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
-      }
-    }
-  }
-
-  stream_.device[mode] = device;
-
-  // Setup the buffer conversion information structure.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
-
-  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;
-  if ( options && !options->streamName.empty() ) streamName = options->streamName;
-  switch ( mode ) {
-  case INPUT:
-    pa_buffer_attr buffer_attr;
-    buffer_attr.fragsize = bufferBytes;
-    buffer_attr.maxlength = -1;
-
-    pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &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, streamName.c_str(), 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;
-
-  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;
-    }
-  }
-
-  stream_.state = STREAM_STOPPED;
-  return true;
-
- error:
-  if ( pah && stream_.callbackInfo.isRunning ) {
-    pthread_cond_destroy( &pah->runnable_cv );
-    delete pah;
-    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;
-}
-
-//******************** End of __LINUX_PULSE__ *********************//
-#endif
-
-#if defined(__LINUX_OSS__)
-
-#include <unistd.h>
-#include <sys/ioctl.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/soundcard.h>
-#include <errno.h>
-#include <math.h>
-
-static 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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::WARNING );
-    return info;
-  }
-
-  unsigned nDevices = sysinfo.numaudios;
-  if ( nDevices == 0 ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
-    error( RtAudioError::INVALID_USE );
-    return info;
-  }
-
-  if ( device >= nDevices ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
-    error( RtAudioError::INVALID_USE );
-    return info;
-  }
-
-  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( RtAudioError::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( RtAudioError::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] );
-
-          if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
-            info.preferredSampleRate = 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.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
-          info.preferredSampleRate = SAMPLE_RATES[k];
-      }
-    }
-  }
-
-  if ( info.sampleRates.size() == 0 ) {
-    errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    error( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiOss :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
-    error( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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();
-}
-
-static 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( RtAudioError::Type type )
-{
-  errorStream_.str(""); // clear the ostringstream
-
-  RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
-  if ( errorCallback ) {
-    // abortStream() can generate new error messages. Ignore them. Just keep original one.
-
-    if ( firstErrorOccurred_ )
-      return;
-
-    firstErrorOccurred_ = true;
-    const std::string errorMessage = errorText_;
-
-    if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
-      stream_.callbackInfo.isRunning = false; // exit from the thread
-      abortStream();
-    }
-
-    errorCallback( type, errorMessage );
-    firstErrorOccurred_ = false;
-    return;
-  }
-
-  if ( type == RtAudioError::WARNING && showWarnings_ == true )
-    std::cerr << '\n' << errorText_ << "\n\n";
-  else if ( type != RtAudioError::WARNING )
-    throw( RtAudioError( errorText_, type ) );
-}
-
-void RtApi :: verifyStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApi:: a stream is not open!";
-    error( RtAudioError::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;
-  stream_.callbackInfo.errorCallback = 0;
-  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_SINT32 || format == RTAUDIO_FLOAT32 )
-    return 4;
-  else if ( format == RTAUDIO_FLOAT64 )
-    return 8;
-  else if ( format == RTAUDIO_SINT24 )
-    return 3;
-  else if ( format == RTAUDIO_SINT8 )
-    return 1;
-
-  errorText_ = "RtApi::formatBytes: undefined format.";
-  error( RtAudioError::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) {
-      Int24 *in = (Int24 *)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]].asInt());
-          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) {
-      Int24 *in = (Int24 *)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]].asInt());
-          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) {
-      Int24 *in = (Int24 *)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]].asInt();
-          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) {
-    Int24 *out = (Int24 *)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]] << 16);
-          //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]] << 8);
-          //out[info.outOffset[j]] <<= 8;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      // Channel compensation and/or (de)interleaving only.
-      Int24 *in = (Int24 *)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]] >> 8);
-          //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) {
-      Int24 *in = (Int24 *)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]].asInt() >> 8);
-        }
-        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) {
-      Int24 *in = (Int24 *)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]].asInt() >> 16);
-        }
-        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 )
-{
-  char val;
-  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_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_SINT24 ) {
-    for ( unsigned int i=0; i<samples; i++ ) {
-      // Swap 1st and 3rd bytes.
-      val = *(ptr);
-      *(ptr) = *(ptr+2);
-      *(ptr+2) = val;
-
-      // Increment 2 more bytes.
-      ptr += 2;
-    }
-  }
-  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
-
-#endif // RTAUDIO_ENABLED -GODOT-

thirdparty/rtaudio/RtAudio.h

@@ -1,1183 +0,0 @@
-// -GODOT- Start
-
-#ifdef RTAUDIO_ENABLED
-
-#if defined(OSX_ENABLED)
-    #define __MACOSX_CORE__
-#elif defined(UNIX_ENABLED)
-    #define __LINUX_ALSA__
-#elif defined(WINDOWS_ENABLED)
-    #if defined(UWP_ENABLED)
-        #define __RTAUDIO_DUMMY__
-    #else
-        #define __WINDOWS_DS__
-    #endif
-#endif
-
-// -GODOT- End
-
-/************************************************************************/
-/*! \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, ASIO and WASAPI) operating systems.
-
-    RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
-
-    RtAudio: realtime audio i/o C++ classes
-    Copyright (c) 2001-2016 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
- */
-
-#ifndef __RTAUDIO_H
-#define __RTAUDIO_H
-
-#define RTAUDIO_VERSION "4.1.2"
-
-#include <string>
-#include <vector>
-#include <exception>
-#include <iostream>
-
-/*! \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.
-
-    - \e RTAUDIO_SINT8:   8-bit signed integer.
-    - \e RTAUDIO_SINT16:  16-bit signed integer.
-    - \e RTAUDIO_SINT24:  24-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;   // 24-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 );
-
-/************************************************************************/
-/*! \class RtAudioError
-    \brief Exception handling class for RtAudio.
-
-    The RtAudioError class is quite simple but it does allow errors to be
-    "caught" by RtAudioError::Type. See the RtAudio documentation to know
-    which methods can throw an RtAudioError.
-*/
-/************************************************************************/
-
-class RtAudioError : public std::exception
-{
- public:
-  //! Defined RtAudioError 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.
-  RtAudioError( const std::string& message, Type type = RtAudioError::UNSPECIFIED ) throw() : message_(message), type_(type) {}
-
-  //! The destructor.
-  virtual ~RtAudioError( 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_;
-};
-
-//! RtAudio error callback function prototype.
-/*!
-    \param type Type of error.
-    \param errorText Error description.
- */
-typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText );
-
-// **************************************************************** //
-//
-// 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_WASAPI, /*!< The Microsoft WASAPI 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). */
-    unsigned int preferredSampleRate; /*!< Preferred sample rate, eg. for WASAPI the system sample rate. */
-    RtAudioFormat nativeFormats;  /*!< Bit mask of supported data formats. */
-
-    // Default constructor.
-    DeviceInfo()
-      :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
-       isDefaultOutput(false), isDefaultInput(false), preferredSampleRate(0), 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 current RtAudio version.
-  static std::string getVersion( void ) throw();
-
-  //! 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.  An exception
-    can be thrown if no API support is compiled.
-
-    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 );
-
-  //! 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 RtAudioError (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 RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be
-    opened with the specified parameters or an error occurs during
-    processing.  An RtAudioError (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.
-    \param errorCallback A client-defined function that will be invoked
-           when an error has occured.
-  */
-  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, RtAudioErrorCallback errorCallback = 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 RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
-    during processing.  An RtAudioError (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 RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
-    during processing.  An RtAudioError (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 RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
-    during processing.  An RtAudioError (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 RtAudioError (type = INVALID_USE) will be thrown.
-  */
-  double getStreamTime( void );
-
-  //! Set the stream time to a time in seconds greater than or equal to 0.0.
-  /*!
-    If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown.
-  */
-  void setStreamTime( double time );
-
-  //! 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
-    RtAudioError (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 RtAudioError (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__) || defined(__WINDOWS_WASAPI__)
-
-  #ifndef NOMINMAX
-    #define NOMINMAX
-  #endif
-  #include <windows.h>
-  #include <process.h>
-
-  typedef uintptr_t 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 *errorCallback;
-  void *apiInfo;   // void pointer for API specific callback information
-  bool isRunning;
-  bool doRealtime;
-  int priority;
-
-  // Default constructor.
-  CallbackInfo()
-  :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(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).
-//
-// **************************************************************** //
-
-#pragma pack(push, 1)
-class S24 {
-
- protected:
-  unsigned char c3[3];
-
- public:
-  S24() {}
-
-  S24& operator = ( const int& i ) {
-    c3[0] = (i & 0x000000ff);
-    c3[1] = (i & 0x0000ff00) >> 8;
-    c3[2] = (i & 0x00ff0000) >> 16;
-    return *this;
-  }
-
-  S24( const S24& v ) { *this = v; }
-  S24( const double& d ) { *this = (int) d; }
-  S24( const float& f ) { *this = (int) f; }
-  S24( const signed short& s ) { *this = (int) s; }
-  S24( const char& c ) { *this = (int) c; }
-
-  int asInt() {
-    int i = c3[0] | (c3[1] << 8) | (c3[2] << 16);
-    if (i & 0x800000) i |= ~0xffffff;
-    return i;
-  }
-};
-#pragma pack(pop)
-
-#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,
-                   RtAudioErrorCallback errorCallback );
-  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 );
-  virtual void setStreamTime( double time );
-  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 S24 Int24;
-  typedef signed short Int16;
-  typedef signed int Int32;
-  typedef float Float32;
-  typedef double Float64;
-
-  std::ostringstream errorStream_;
-  std::string errorText_;
-  bool showWarnings_;
-  RtApiStream stream_;
-  bool firstErrorOccurred_;
-
-  /*!
-    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 RtAudioError (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( RtAudioError::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 :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); }
-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;
-  std::vector<struct DsDevice> dsDevices;
-  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_WASAPI__)
-
-struct IMMDeviceEnumerator;
-
-class RtApiWasapi : public RtApi
-{
-public:
-  RtApiWasapi();
-  ~RtApiWasapi();
-
-  RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; }
-  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 );
-
-private:
-  bool coInitialized_;
-  IMMDeviceEnumerator* deviceEnumerator_;
-
-  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 DWORD WINAPI runWasapiThread( void* wasapiPtr );
-  static DWORD WINAPI stopWasapiThread( void* wasapiPtr );
-  static DWORD WINAPI abortWasapiThread( void* wasapiPtr );
-  void wasapiThread();
-};
-
-#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( RtAudioError::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
-
-#endif // RTAUDIO_ENABLED -GODOT-