Merge lp:~marcustomlinson/dspatchables/add_basic_components into lp:dspatchables

[Marcus Tomlinson]

=== modified file 'CMakeLists.txt'
--- CMakeLists.txt	2014-12-23 19:18:01 +0000
+++ CMakeLists.txt	2015-04-05 22:23:36 +0000
@@ -4,4 +4,11 @@
 
 project(DSPatchables)
 
+include_directories(${CMAKE_SOURCE_DIR}/include)
+link_directories(${CMAKE_SOURCE_DIR}/link)
+
+add_subdirectory(DspAdder)
+add_subdirectory(DspAudioDevice)
+add_subdirectory(DspGain)
 add_subdirectory(DspOscillator)
+add_subdirectory(DspWaveStreamer)
 
=== added directory 'DspAdder'
=== added file 'DspAdder/CMakeLists.txt'
--- DspAdder/CMakeLists.txt	1970-01-01 00:00:00 +0000
+++ DspAdder/CMakeLists.txt	2015-04-05 22:23:36 +0000
@@ -0,0 +1,17 @@
+project(DspAdder)
+
+file(GLOB srcs *.cpp)
+file(GLOB hdrs *.h)
+
+include_directories(${CMAKE_CURRENT_SOURCE_DIR})
+
+add_library(
+    ${PROJECT_NAME} SHARED
+    ${srcs}
+    ${hdrs}
+)
+
+target_link_libraries(
+    ${PROJECT_NAME}
+    DSPatch
+)
 
=== added file 'DspAdder/DspAdder.cpp'
--- DspAdder/DspAdder.cpp	1970-01-01 00:00:00 +0000
+++ DspAdder/DspAdder.cpp	2015-04-05 22:23:36 +0000
@@ -0,0 +1,76 @@
+/************************************************************************
+DSPatch - Cross-Platform, Object-Oriented, Flow-Based Programming Library
+Copyright (c) 2012-2015 Marcus Tomlinson
+
+This file is part of DSPatch.
+
+GNU Lesser General Public License Usage
+This file may be used under the terms of the GNU Lesser General Public
+License version 3.0 as published by the Free Software Foundation and
+appearing in the file LGPLv3.txt included in the packaging of this
+file. Please review the following information to ensure the GNU Lesser
+General Public License version 3.0 requirements will be met:
+http://www.gnu.org/copyleft/lgpl.html.
+
+Other Usage
+Alternatively, this file may be used in accordance with the terms and
+conditions contained in a signed written agreement between you and
+Marcus Tomlinson.
+
+DSPatch is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+************************************************************************/
+
+#include <DspAdder.h>
+
+//=================================================================================================
+
+DspAdder::DspAdder()
+{
+    // add 2 inputs
+    AddInput_("Input1");
+    AddInput_("Input2");
+
+    // add 1 output
+    AddOutput_("Output1");
+}
+
+//-------------------------------------------------------------------------------------------------
+
+DspAdder::~DspAdder()
+{
+}
+
+//=================================================================================================
+
+void DspAdder::Process_(DspSignalBus& inputs, DspSignalBus& outputs)
+{
+    // get input values from inputs bus (GetValue() returns true if successful)
+    if (!inputs.GetValue(0, _stream1))
+    {
+        _stream1.assign(_stream1.size(), 0);  // clear buffer if no input received
+    }
+    // do the same to the 2nd input buffer
+    if (!inputs.GetValue(1, _stream2))
+    {
+        _stream2.assign(_stream2.size(), 0);
+    }
+
+    // ensure that the 2 input buffer sizes match
+    if (_stream1.size() == _stream2.size())
+    {
+        for (size_t i = 0; i < _stream1.size(); i++)
+        {
+            _stream1[i] += _stream2[i];  // perform addition element-by-element
+        }
+        outputs.SetValue(0, _stream1);  // set output 1
+    }
+    // if input sizes don't match
+    else
+    {
+        outputs.ClearValue(0);  // clear the output
+    }
+}
+
+//=================================================================================================
 
=== added file 'DspAdder/DspAdder.h'
--- DspAdder/DspAdder.h	1970-01-01 00:00:00 +0000
+++ DspAdder/DspAdder.h	2015-04-05 22:23:36 +0000
@@ -0,0 +1,80 @@
+/************************************************************************
+DSPatch - Cross-Platform, Object-Oriented, Flow-Based Programming Library
+Copyright (c) 2012-2015 Marcus Tomlinson
+
+This file is part of DSPatch.
+
+GNU Lesser General Public License Usage
+This file may be used under the terms of the GNU Lesser General Public
+License version 3.0 as published by the Free Software Foundation and
+appearing in the file LGPLv3.txt included in the packaging of this
+file. Please review the following information to ensure the GNU Lesser
+General Public License version 3.0 requirements will be met:
+http://www.gnu.org/copyleft/lgpl.html.
+
+Other Usage
+Alternatively, this file may be used in accordance with the terms and
+conditions contained in a signed written agreement between you and
+Marcus Tomlinson.
+
+DSPatch is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+************************************************************************/
+
+#ifndef DSPADDER_H
+#define DSPADDER_H
+
+#include <DSPatch.h>
+
+//=================================================================================================
+/// Example DspComponent: Adder
+
+/** This component has 2 inputs and 1 output. The component receives 2 floating-point buffers into
+it's 2 inputs, adds each buffer element of the 1st buffer to the corresponding element of the 2nd
+buffer, then stores the resultant buffer into a 3rd buffer. This resultant buffer is then passed to
+output 1 of the component output bus. */
+
+class DspAdder : public DspComponent
+{
+public:
+    //! Component constructor
+    /*! When a component is constructed, it's input and output buses must be configured. This is
+    achieved by making calls to the base class protected methods: "AddInput_()" and "AddOutput_().
+    These methods must be called once per input / output required. IO signal names are optional
+    (Component IO can be referenced by either string ID or index) and can be assigned to each
+    input / output by supplying the desired string ID as an argument to the respective AddInput_()
+    / AddOutput_() method call.*/
+
+    DspAdder();
+    ~DspAdder();
+
+protected:
+    //! Virtual process method inherited from DspComponent
+    /*! The Process_() method is called from the DSPatch engine when a new set of component input
+    signals are ready for processing. The Process() method has 2 arguments: the input bus and the
+    output bus. This method's purpose is to pull its required inputs out of the input bus, process
+    these inputs, and populate the output bus with the results (see DspSignalBus). */
+
+    virtual void Process_(DspSignalBus& inputs, DspSignalBus& outputs);
+
+private:
+    std::vector<float> _stream1;
+    std::vector<float> _stream2;
+};
+
+//=================================================================================================
+
+class DspAdderPlugin : public DspPlugin
+{
+    DspComponent* Create(std::map<std::string, DspParameter>&) const
+    {
+        return new DspAdder();
+    }
+};
+
+EXPORT_DSPPLUGIN(DspAdderPlugin)
+
+//=================================================================================================
+
+#endif  // DSPADDER_H
 
=== added directory 'DspAudioDevice'
=== added file 'DspAudioDevice/CMakeLists.txt'
--- DspAudioDevice/CMakeLists.txt	1970-01-01 00:00:00 +0000
+++ DspAudioDevice/CMakeLists.txt	2015-04-05 22:23:36 +0000
@@ -0,0 +1,61 @@
+project(DspAudioDevice)
+
+file(GLOB srcs *.cpp rtaudio/*.cpp)
+file(GLOB hdrs *.h rtaudio/*.h)
+
+include_directories(
+    ${CMAKE_CURRENT_SOURCE_DIR}
+    ${CMAKE_CURRENT_SOURCE_DIR}/rtaudio
+)
+
+add_library(
+    ${PROJECT_NAME} SHARED
+    ${srcs}
+    ${hdrs}
+)
+
+target_link_libraries(
+    ${PROJECT_NAME}
+    DSPatch
+)
+
+# Definition for RtAudio Windows, using direct sound
+if(${CMAKE_SYSTEM_NAME} MATCHES "Windows")
+    add_definitions(-DEXAMPLE_PLUGIN_FILE="${CMAKE_CURRENT_BINARY_DIR}/oscillator-plugin/$<CONFIGURATION>/DspOscillator.dll")
+    add_definitions(-D__WINDOWS_WASAPI__)
+
+    add_custom_command(
+        TARGET ${PROJECT_NAME} POST_BUILD
+        COMMAND ${CMAKE_COMMAND} -E copy_if_different
+        ${CMAKE_BINARY_DIR}/$<CONFIGURATION>/DSPatch.dll
+        ${CMAKE_CURRENT_BINARY_DIR}/$<CONFIGURATION>
+    )
+endif(${CMAKE_SYSTEM_NAME} MATCHES "Windows")
+
+# Definition for RtAudio Linux, using ALSA
+if(${CMAKE_SYSTEM_NAME} MATCHES "Linux")
+    add_definitions(-DEXAMPLE_PLUGIN_FILE="${CMAKE_CURRENT_BINARY_DIR}/oscillator-plugin/libDspOscillator.so")
+    add_definitions(-D__LINUX_ALSA__)
+
+    find_library(ASOUND asound)
+    if(NOT ASOUND)
+        message(FATAL_ERROR "ALSA not found (Ensure that libasound2-dev is installed)")
+    endif()
+
+    target_link_libraries(
+        ${PROJECT_NAME}
+        ${ASOUND}
+    )
+endif(${CMAKE_SYSTEM_NAME} MATCHES "Linux")
+
+# Definition for RtAudio Mac OSX, using Core Audio
+if(${CMAKE_SYSTEM_NAME} MATCHES "Darwin")
+    add_definitions(-DEXAMPLE_PLUGIN_FILE="${CMAKE_CURRENT_BINARY_DIR}/oscillator-plugin/$<CONFIGURATION>/libDspOscillator.dylib")
+    add_definitions(-D__MACOSX_CORE__)
+
+    target_link_libraries(
+        ${PROJECT_NAME}
+        "-framework CoreAudio"
+        "-framework CoreFoundation"
+    )
+endif(${CMAKE_SYSTEM_NAME} MATCHES "Darwin")
 
=== added file 'DspAudioDevice/DspAudioDevice.cpp'
--- DspAudioDevice/DspAudioDevice.cpp	1970-01-01 00:00:00 +0000
+++ DspAudioDevice/DspAudioDevice.cpp	2015-04-05 22:23:36 +0000
@@ -0,0 +1,427 @@
+/************************************************************************
+DSPatch - Cross-Platform, Object-Oriented, Flow-Based Programming Library
+Copyright (c) 2012-2015 Marcus Tomlinson
+
+This file is part of DSPatch.
+
+GNU Lesser General Public License Usage
+This file may be used under the terms of the GNU Lesser General Public
+License version 3.0 as published by the Free Software Foundation and
+appearing in the file LGPLv3.txt included in the packaging of this
+file. Please review the following information to ensure the GNU Lesser
+General Public License version 3.0 requirements will be met:
+http://www.gnu.org/copyleft/lgpl.html.
+
+Other Usage
+Alternatively, this file may be used in accordance with the terms and
+conditions contained in a signed written agreement between you and
+Marcus Tomlinson.
+
+DSPatch is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+************************************************************************/
+
+#include <DspAudioDevice.h>
+
+#include <RtAudio.h>
+
+#include <iostream>
+#include <string.h>
+#include <cstdlib>
+
+//=================================================================================================
+
+struct RtAudioMembers
+{
+    std::vector<RtAudio::DeviceInfo> deviceList;
+
+    RtAudio audioStream;
+    RtAudio::StreamParameters outputParams;
+    RtAudio::StreamParameters inputParams;
+};
+
+//=================================================================================================
+
+DspAudioDevice::DspAudioDevice()
+    : _rtAudio(new RtAudioMembers())
+    , _gotWaitReady(false)
+    , _gotSyncReady(true)
+{
+    _outputChannels.resize(8);
+    for (int i = 0; i < 8; i++)
+    {
+        AddInput_();
+    }
+
+    AddInput_("Sample Rate");
+
+    _inputChannels.resize(8);
+    for (int i = 0; i < 8; i++)
+    {
+        AddOutput_();
+    }
+
+    std::vector<std::string> deviceNameList;
+
+    for (unsigned int i = 0; i < _rtAudio->audioStream.getDeviceCount(); i++)
+    {
+        _rtAudio->deviceList.push_back(_rtAudio->audioStream.getDeviceInfo(i));
+        deviceNameList.push_back(_rtAudio->audioStream.getDeviceInfo(i).name);
+    }
+
+    pDeviceList = AddParameter_("deviceList", DspParameter(DspParameter::List, deviceNameList));
+    pIsStreaming = AddParameter_("isStreaming", DspParameter(DspParameter::Bool, false));
+    pBufferSize = AddParameter_("bufferSize", DspParameter(DspParameter::Int, 256));
+    pSampleRate = AddParameter_("sampleRate", DspParameter(DspParameter::Int, 44100));
+
+    SetDevice(_rtAudio->audioStream.getDefaultOutputDevice());
+    SetBufferSize(GetBufferSize());
+    SetSampleRate(GetSampleRate());
+}
+
+//-------------------------------------------------------------------------------------------------
+
+DspAudioDevice::~DspAudioDevice()
+{
+    _StopStream();
+
+    delete _rtAudio;
+}
+
+//-------------------------------------------------------------------------------------------------
+
+bool DspAudioDevice::SetDevice(int deviceIndex)
+{
+    if (deviceIndex >= 0 && deviceIndex < GetDeviceCount())
+    {
+        _StopStream();
+
+        SetParameter_(pDeviceList, DspParameter(DspParameter::Int, deviceIndex));
+
+        _rtAudio->inputParams.nChannels = _rtAudio->deviceList[deviceIndex].inputChannels;
+        _rtAudio->inputParams.deviceId = deviceIndex;
+
+        _rtAudio->outputParams.nChannels = _rtAudio->deviceList[deviceIndex].outputChannels;
+        _rtAudio->outputParams.deviceId = deviceIndex;
+
+        _StartStream();
+
+        return true;
+    }
+
+    return false;
+}
+
+//-------------------------------------------------------------------------------------------------
+
+std::string DspAudioDevice::GetDeviceName(int deviceIndex) const
+{
+    if (deviceIndex >= 0 && deviceIndex < GetDeviceCount())
+    {
+        return _rtAudio->deviceList[deviceIndex].name;
+    }
+
+    return "";
+}
+
+//-------------------------------------------------------------------------------------------------
+
+int DspAudioDevice::GetDeviceInputCount(int deviceIndex) const
+{
+    return _rtAudio->deviceList[deviceIndex].inputChannels;
+}
+
+//-------------------------------------------------------------------------------------------------
+
+int DspAudioDevice::GetDeviceOutputCount(int deviceIndex) const
+{
+    return _rtAudio->deviceList[deviceIndex].outputChannels;
+}
+
+//-------------------------------------------------------------------------------------------------
+
+int DspAudioDevice::GetCurrentDevice() const
+{
+    return *GetParameter_(pDeviceList)->GetInt();
+}
+
+//-------------------------------------------------------------------------------------------------
+
+int DspAudioDevice::GetDeviceCount() const
+{
+    return GetParameter_(pDeviceList)->GetList()->size();
+}
+
+//-------------------------------------------------------------------------------------------------
+
+void DspAudioDevice::SetBufferSize(int bufferSize)
+{
+    _StopStream();
+
+    SetParameter_(pBufferSize, DspParameter(DspParameter::Int, bufferSize));
+    for (size_t i = 0; i < _inputChannels.size(); i++)
+    {
+        _inputChannels[i].resize(bufferSize);
+    }
+
+    _StartStream();
+}
+
+//-------------------------------------------------------------------------------------------------
+
+void DspAudioDevice::SetSampleRate(int sampleRate)
+{
+    _StopStream();
+    SetParameter_(pSampleRate, DspParameter(DspParameter::Int, sampleRate));
+    _StartStream();
+}
+
+//-------------------------------------------------------------------------------------------------
+
+bool DspAudioDevice::IsStreaming() const
+{
+    return *GetParameter_(pIsStreaming)->GetBool();
+}
+
+//-------------------------------------------------------------------------------------------------
+
+int DspAudioDevice::GetBufferSize() const
+{
+    return *GetParameter_(pBufferSize)->GetInt();
+}
+
+//-------------------------------------------------------------------------------------------------
+
+int DspAudioDevice::GetSampleRate() const
+{
+    return *GetParameter_(pSampleRate)->GetInt();
+}
+
+//=================================================================================================
+
+void DspAudioDevice::Process_(DspSignalBus& inputs, DspSignalBus& outputs)
+{
+    // Wait until the sound card is ready for the next set of buffers
+    // ==============================================================
+    _syncMutex.Lock();
+    if (!_gotSyncReady)  // if haven't already got the release
+    {
+        _syncCondt.Wait(_syncMutex);  // wait for sync
+    }
+    _gotSyncReady = false;  // reset the release flag
+    _syncMutex.Unlock();
+
+    // Synchronise sample rate with the "Sample Rate" input feed
+    // =========================================================
+    int sampleRate;
+    if (inputs.GetValue("Sample Rate", sampleRate))
+    {
+        if (sampleRate != GetSampleRate())
+        {
+            SetSampleRate(sampleRate);
+        }
+    }
+
+    // Synchronise buffer size with the size of incoming buffers
+    // =========================================================
+    if (inputs.GetValue(0, _outputChannels[0]))
+    {
+        if (GetBufferSize() != (int)_outputChannels[0].size() && _outputChannels[0].size() != 0)
+        {
+            SetBufferSize(_outputChannels[0].size());
+        }
+    }
+
+    // Retrieve incoming component buffers for the sound card to output
+    // ================================================================
+    for (size_t i = 0; i < _outputChannels.size(); i++)
+    {
+        if (!inputs.GetValue(i, _outputChannels[i]))
+        {
+            _outputChannels[i].assign(_outputChannels[i].size(), 0);
+        }
+    }
+
+    // Retrieve incoming sound card buffers for the component to output
+    // ================================================================
+    for (size_t i = 0; i < _inputChannels.size(); i++)
+    {
+        outputs.SetValue(i, _inputChannels[i]);
+    }
+
+    // Inform the sound card that buffers are now ready
+    // ================================================
+    _buffersMutex.Lock();
+    _gotWaitReady = true;  // set release flag
+    _waitCondt.WakeAll();  // release sync
+    _buffersMutex.Unlock();
+}
+
+//-------------------------------------------------------------------------------------------------
+
+bool DspAudioDevice::ParameterUpdating_(int index, DspParameter const& param)
+{
+    if (index == pDeviceList)
+    {
+        return SetDevice(*param.GetInt());
+    }
+    else if (index == pBufferSize)
+    {
+        SetBufferSize(*param.GetInt());
+        return true;
+    }
+    else if (index == pSampleRate)
+    {
+        SetSampleRate(*param.GetInt());
+        return true;
+    }
+
+    return false;
+}
+
+//=================================================================================================
+
+void DspAudioDevice::_SetIsStreaming(bool isStreaming)
+{
+    SetParameter_(pIsStreaming, DspParameter(DspParameter::Bool, isStreaming));
+}
+
+//-------------------------------------------------------------------------------------------------
+
+void DspAudioDevice::_WaitForBuffer()
+{
+    _buffersMutex.Lock();
+    if (!_gotWaitReady)  // if haven't already got the release
+    {
+        _waitCondt.Wait(_buffersMutex);  // wait for sync
+    }
+    _gotWaitReady = false;  // reset the release flag
+    _buffersMutex.Unlock();
+}
+
+//-------------------------------------------------------------------------------------------------
+
+void DspAudioDevice::_SyncBuffer()
+{
+    _syncMutex.Lock();
+    _gotSyncReady = true;  // set release flag
+    _syncCondt.WakeAll();  // release sync
+    _syncMutex.Unlock();
+}
+
+//-------------------------------------------------------------------------------------------------
+
+void DspAudioDevice::_StopStream()
+{
+    _SetIsStreaming(false);
+
+    _buffersMutex.Lock();
+    _gotWaitReady = true;  // set release flag
+    _waitCondt.WakeAll();  // release sync
+    _buffersMutex.Unlock();
+
+    if (_rtAudio->audioStream.isStreamOpen())
+    {
+        _rtAudio->audioStream.closeStream();
+    }
+}
+
+//-------------------------------------------------------------------------------------------------
+
+void DspAudioDevice::_StartStream()
+{
+    RtAudio::StreamParameters* inputParams = NULL;
+    RtAudio::StreamParameters* outputParams = NULL;
+
+    if (_rtAudio->inputParams.nChannels != 0)
+    {
+        inputParams = &_rtAudio->inputParams;
+    }
+
+    if (_rtAudio->outputParams.nChannels != 0)
+    {
+        outputParams = &_rtAudio->outputParams;
+    }
+
+    RtAudio::StreamOptions options;
+    options.flags |= RTAUDIO_SCHEDULE_REALTIME;
+    options.flags |= RTAUDIO_NONINTERLEAVED;
+
+    _rtAudio->audioStream.openStream(outputParams,
+                                     inputParams,
+                                     RTAUDIO_FLOAT32,
+                                     GetSampleRate(),
+                                     (unsigned int*)const_cast<int*>(GetParameter_(pBufferSize)->GetInt()),
+                                     &_StaticCallback,
+                                     this,
+                                     &options);
+
+    _rtAudio->audioStream.startStream();
+    
+    while (!_rtAudio->audioStream.isStreamOpen())
+    {
+        DspThread::MsSleep(10);
+    }
+
+    _SetIsStreaming(true);
+}
+
+//-------------------------------------------------------------------------------------------------
+
+int DspAudioDevice::_StaticCallback(
+    void* outputBuffer, void* inputBuffer, unsigned int, double, unsigned int, void* userData)
+{
+    return (reinterpret_cast<DspAudioDevice*>(userData))->_DynamicCallback(inputBuffer, outputBuffer);
+}
+
+//-------------------------------------------------------------------------------------------------
+
+int DspAudioDevice::_DynamicCallback(void* inputBuffer, void* outputBuffer)
+{
+    _WaitForBuffer();
+
+    if (IsStreaming())
+    {
+        float* floatOutput = (float*)outputBuffer;
+        float* floatInput = (float*)inputBuffer;
+
+        if (outputBuffer != NULL)
+        {
+            for (size_t i = 0; i < _outputChannels.size(); i++)
+            {
+                if (_rtAudio->deviceList[GetCurrentDevice()].outputChannels >= (i + 1))
+                {
+                    for (size_t j = 0; j < _outputChannels[i].size(); j++)
+                    {
+                        *floatOutput++ = _outputChannels[i][j];
+                    }
+                }
+            }
+        }
+
+        if (inputBuffer != NULL)
+        {
+            for (size_t i = 0; i < _inputChannels.size(); i++)
+            {
+                if (_rtAudio->deviceList[GetCurrentDevice()].inputChannels >= (i + 1))
+                {
+                    for (size_t j = 0; j < _inputChannels[i].size(); j++)
+                    {
+                        _inputChannels[i][j] = *floatInput++;
+                    }
+                }
+            }
+        }
+    }
+    else
+    {
+        _SyncBuffer();
+        return 1;
+    }
+
+    _SyncBuffer();
+    return 0;
+}
+
+//=================================================================================================
 
=== added file 'DspAudioDevice/DspAudioDevice.h'
--- DspAudioDevice/DspAudioDevice.h	1970-01-01 00:00:00 +0000
+++ DspAudioDevice/DspAudioDevice.h	2015-04-05 22:23:36 +0000
@@ -0,0 +1,109 @@
+/************************************************************************
+DSPatch - Cross-Platform, Object-Oriented, Flow-Based Programming Library
+Copyright (c) 2012-2015 Marcus Tomlinson
+
+This file is part of DSPatch.
+
+GNU Lesser General Public License Usage
+This file may be used under the terms of the GNU Lesser General Public
+License version 3.0 as published by the Free Software Foundation and
+appearing in the file LGPLv3.txt included in the packaging of this
+file. Please review the following information to ensure the GNU Lesser
+General Public License version 3.0 requirements will be met:
+http://www.gnu.org/copyleft/lgpl.html.
+
+Other Usage
+Alternatively, this file may be used in accordance with the terms and
+conditions contained in a signed written agreement between you and
+Marcus Tomlinson.
+
+DSPatch is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+************************************************************************/
+
+#ifndef DSPAUDIODEVICE_H
+#define DSPAUDIODEVICE_H
+
+#include <DSPatch.h>
+
+struct RtAudioMembers;
+
+//=================================================================================================
+
+class DspAudioDevice : public DspComponent
+{
+public:
+    int pDeviceList;   // List
+    int pIsStreaming;  // Bool
+    int pBufferSize;   // Int
+    int pSampleRate;   // Int
+
+    DspAudioDevice();
+    ~DspAudioDevice();
+
+    bool SetDevice(int deviceIndex);
+
+    std::string GetDeviceName(int deviceIndex) const;
+    int GetDeviceInputCount(int deviceIndex) const;
+    int GetDeviceOutputCount(int deviceIndex) const;
+    int GetCurrentDevice() const;
+    int GetDeviceCount() const;
+
+    void SetBufferSize(int bufferSize);
+    void SetSampleRate(int sampleRate);
+
+    bool IsStreaming() const;
+    int GetBufferSize() const;
+    int GetSampleRate() const;
+
+protected:
+    virtual void Process_(DspSignalBus& inputs, DspSignalBus& outputs);
+    virtual bool ParameterUpdating_(int index, DspParameter const& param);
+
+private:
+    std::vector< std::vector<float> > _outputChannels;
+    std::vector< std::vector<float> > _inputChannels;
+
+    RtAudioMembers* _rtAudio;
+
+    DspMutex _buffersMutex;
+    DspMutex _syncMutex;
+    DspWaitCondition _waitCondt;
+    DspWaitCondition _syncCondt;
+    bool _gotWaitReady;
+    bool _gotSyncReady;
+
+    void _SetIsStreaming(bool isStreaming);
+
+    void _WaitForBuffer();
+    void _SyncBuffer();
+
+    void _StopStream();
+    void _StartStream();
+
+    static int _StaticCallback(void* outputBuffer,
+                               void* inputBuffer,
+                               unsigned int nBufferFrames,
+                               double streamTime,
+                               unsigned int status,
+                               void* userData);
+
+    int _DynamicCallback(void* inputBuffer, void* outputBuffer);
+};
+
+//=================================================================================================
+
+class DspAudioDevicePlugin : public DspPlugin
+{
+    DspComponent* Create(std::map<std::string, DspParameter>&) const
+    {
+        return new DspAudioDevice();
+    }
+};
+
+EXPORT_DSPPLUGIN(DspAudioDevicePlugin)
+
+//=================================================================================================
+
+#endif  // DSPAUDIODEVICE_H
 
=== added directory 'DspAudioDevice/rtaudio'
=== added file 'DspAudioDevice/rtaudio/RtAudio.cpp'
--- DspAudioDevice/rtaudio/RtAudio.cpp	1970-01-01 00:00:00 +0000
+++ DspAudioDevice/rtaudio/RtAudio.cpp	2015-04-05 22:23:36 +0000
@@ -0,0 +1,10136 @@
+/************************************************************************/
+/*! \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) 2001-2014 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 4.1.1
+
+#include "RtAudio.h"
+#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <climits>
+
+// Static variable definitions.
+const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
+const unsigned int RtApi::SAMPLE_RATES[] = {
+  4000, 5512, 8000, 9600, 11025, 16000, 22050,
+  32000, 44100, 48000, 88200, 96000, 176400, 192000
+};
+
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || 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)
+#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_->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 )
+      info.sampleRates.push_back( (unsigned int) rangeList[i].mMinimum );
+    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] );
+    }
+  }
+
+  // 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 ( stream_.state == STREAM_RUNNING )
+      AudioDeviceStop( handle->id[0], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+    AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+#else
+    // deprecated in favor of AudioDeviceDestroyIOProcID()
+    AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+#endif
+  }
+
+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+    if ( stream_.state == STREAM_RUNNING )
+      AudioDeviceStop( handle->id[1], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+    AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+#else
+    // deprecated in favor of AudioDeviceDestroyIOProcID()
+    AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
+#endif
+  }
+
+  for ( int i=0; i<2; i++ ) {
+    if ( stream_.userBuffer[i] ) {
+      free( stream_.userBuffer[i] );
+      stream_.userBuffer[i] = 0;
+    }
+  }
+
+  if ( stream_.deviceBuffer ) {
+    free( stream_.deviceBuffer );
+    stream_.deviceBuffer = 0;
+  }
+
+  // Destroy pthread condition variable.
+  pthread_cond_destroy( &handle->condition );
+  delete handle;
+  stream_.apiHandle = 0;
+
+  stream_.mode = UNINITIALIZED;
+  stream_.state = STREAM_CLOSED;
+}
+
+void RtApiCore :: startStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiCore::startStream(): the stream is already running!";
+    error( 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.sampleRates.push_back( jack_get_sample_rate( client ) );
+
+  // Count the available ports containing the client name as device
+  // channels.  Jack "input ports" equal RtAudio output channels.
+  unsigned int nChannels = 0;
+  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
+  if ( ports ) {
+    while ( ports[ nChannels ] ) nChannels++;
+    free( ports );
+    info.outputChannels = nChannels;
+  }
+
+  // Jack "output ports" equal RtAudio input channels.
+  nChannels = 0;
+  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
+  if ( ports ) {
+    while ( ports[ nChannels ] ) nChannels++;
+    free( ports );
+    info.inputChannels = nChannels;
+  }
+
+  if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
+    jack_client_close(client);
+    errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+    error( 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] );
+  }
+
+  // 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 )
+{
+  // For ASIO, a duplex stream MUST use the same driver.
+  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
+    errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
+    return FAILURE;
+  }
+
+  char driverName[32];
+  ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Only load the driver once for duplex stream.
+  if ( mode != INPUT || stream_.mode != OUTPUT ) {
+    // The getDeviceInfo() function will not work when a stream is open
+    // because ASIO does not allow multiple devices to run at the same
+    // time.  Thus, we'll probe the system before opening a stream and
+    // save the results for use by getDeviceInfo().
+    this->saveDeviceInfo();
+
+    if ( !drivers.loadDriver( driverName ) ) {
+      errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    result = ASIOInit( &driverInfo );
+    if ( result != ASE_OK ) {
+      errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+  }
+
+  // Check the device channel count.
+  long inputChannels, outputChannels;
+  result = ASIOGetChannels( &inputChannels, &outputChannels );
+  if ( result != ASE_OK ) {
+    drivers.removeCurrentDriver();
+    errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
+       ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
+    drivers.removeCurrentDriver();
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+  stream_.nDeviceChannels[mode] = channels;
+  stream_.nUserChannels[mode] = channels;
+  stream_.channelOffset[mode] = firstChannel;
+
+  // Verify the sample rate is supported.
+  result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
+  if ( result != ASE_OK ) {
+    drivers.removeCurrentDriver();
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Get the current sample rate
+  ASIOSampleRate currentRate;
+  result = ASIOGetSampleRate( &currentRate );
+  if ( result != ASE_OK ) {
+    drivers.removeCurrentDriver();
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Set the sample rate only if necessary
+  if ( currentRate != sampleRate ) {
+    result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+    if ( result != ASE_OK ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+  }
+
+  // Determine the driver data type.
+  ASIOChannelInfo channelInfo;
+  channelInfo.channel = 0;
+  if ( mode == OUTPUT ) channelInfo.isInput = false;
+  else channelInfo.isInput = true;
+  result = ASIOGetChannelInfo( &channelInfo );
+  if ( result != ASE_OK ) {
+    drivers.removeCurrentDriver();
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Assuming WINDOWS host is always little-endian.
+  stream_.doByteSwap[mode] = false;
+  stream_.userFormat = format;
+  stream_.deviceFormat[mode] = 0;
+  if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
+    stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+    if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
+  }
+  else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
+    stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+    if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
+  }
+  else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+    if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
+  }
+  else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+    if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
+  }
+  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 ) {
+    drivers.removeCurrentDriver();
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Set the buffer size.  For a duplex stream, this will end up
+  // setting the buffer size based on the input constraints, which
+  // should be ok.
+  long minSize, maxSize, preferSize, granularity;
+  result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
+  if ( result != ASE_OK ) {
+    drivers.removeCurrentDriver();
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+  else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+  else if ( granularity == -1 ) {
+    // Make sure bufferSize is a power of two.
+    int log2_of_min_size = 0;
+    int log2_of_max_size = 0;
+
+    for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
+      if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
+      if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
+    }
+
+    long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
+    int min_delta_num = log2_of_min_size;
+
+    for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
+      long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
+      if (current_delta < min_delta) {
+        min_delta = current_delta;
+        min_delta_num = i;
+      }
+    }
+
+    *bufferSize = ( (unsigned int)1 << min_delta_num );
+    if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+    else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+  }
+  else if ( granularity != 0 ) {
+    // Set to an even multiple of granularity, rounding up.
+    *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
+  }
+
+  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
+    drivers.removeCurrentDriver();
+    errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
+    return FAILURE;
+  }
+
+  stream_.bufferSize = *bufferSize;
+  stream_.nBuffers = 2;
+
+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+  else stream_.userInterleaved = true;
+
+  // ASIO always uses non-interleaved buffers.
+  stream_.deviceInterleaved[mode] = false;
+
+  // Allocate, if necessary, our AsioHandle structure for the stream.
+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+  if ( handle == 0 ) {
+    try {
+      handle = new AsioHandle;
+    }
+    catch ( std::bad_alloc& ) {
+      //if ( handle == NULL ) {    
+      drivers.removeCurrentDriver();
+      errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+      return FAILURE;
+    }
+    handle->bufferInfos = 0;
+
+    // Create a manual-reset event.
+    handle->condition = CreateEvent( NULL,   // no security
+                                     TRUE,   // manual-reset
+                                     FALSE,  // non-signaled initially
+                                     NULL ); // unnamed
+    stream_.apiHandle = (void *) handle;
+  }
+
+  // Create the ASIO internal buffers.  Since RtAudio sets up input
+  // and output separately, we'll have to dispose of previously
+  // created output buffers for a duplex stream.
+  long inputLatency, outputLatency;
+  if ( mode == INPUT && stream_.mode == OUTPUT ) {
+    ASIODisposeBuffers();
+    if ( handle->bufferInfos ) free( handle->bufferInfos );
+  }
+
+  // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
+  bool buffersAllocated = false;
+  unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+  handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
+  if ( handle->bufferInfos == NULL ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
+    errorText_ = errorStream_.str();
+    goto error;
+  }
+
+  ASIOBufferInfo *infos;
+  infos = handle->bufferInfos;
+  for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
+    infos->isInput = ASIOFalse;
+    infos->channelNum = i + stream_.channelOffset[0];
+    infos->buffers[0] = infos->buffers[1] = 0;
+  }
+  for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
+    infos->isInput = ASIOTrue;
+    infos->channelNum = i + stream_.channelOffset[1];
+    infos->buffers[0] = infos->buffers[1] = 0;
+  }
+
+  // Set up the ASIO callback structure and create the ASIO data buffers.
+  asioCallbacks.bufferSwitch = &bufferSwitch;
+  asioCallbacks.sampleRateDidChange = &sampleRateChanged;
+  asioCallbacks.asioMessage = &asioMessages;
+  asioCallbacks.bufferSwitchTimeInfo = NULL;
+  result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
+    errorText_ = errorStream_.str();
+    goto error;
+  }
+  buffersAllocated = true;
+
+  // Set flags for buffer conversion.
+  stream_.doConvertBuffer[mode] = false;
+  if ( stream_.userFormat != stream_.deviceFormat[mode] )
+    stream_.doConvertBuffer[mode] = true;
+  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+       stream_.nUserChannels[mode] > 1 )
+    stream_.doConvertBuffer[mode] = true;
+
+  // Allocate necessary internal buffers
+  unsigned long bufferBytes;
+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+  if ( stream_.userBuffer[mode] == NULL ) {
+    errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
+    goto error;
+  }
+
+  if ( stream_.doConvertBuffer[mode] ) {
+
+    bool makeBuffer = true;
+    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+    if ( mode == INPUT ) {
+      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+        if ( bufferBytes <= bytesOut ) makeBuffer = false;
+      }
+    }
+
+    if ( makeBuffer ) {
+      bufferBytes *= *bufferSize;
+      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+      if ( stream_.deviceBuffer == NULL ) {
+        errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
+        goto error;
+      }
+    }
+  }
+
+  stream_.sampleRate = sampleRate;
+  stream_.device[mode] = device;
+  stream_.state = STREAM_STOPPED;
+  asioCallbackInfo = &stream_.callbackInfo;
+  stream_.callbackInfo.object = (void *) this;
+  if ( stream_.mode == OUTPUT && mode == INPUT )
+    // We had already set up an output stream.
+    stream_.mode = DUPLEX;
+  else
+    stream_.mode = mode;
+
+  // Determine device latencies
+  result = ASIOGetLatencies( &inputLatency, &outputLatency );
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+    errorText_ = errorStream_.str();
+    error( 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 ( buffersAllocated )
+    ASIODisposeBuffers();
+  drivers.removeCurrentDriver();
+
+  if ( handle ) {
+    CloseHandle( handle->condition );
+    if ( handle->bufferInfos )
+      free( handle->bufferInfos );
+    delete handle;
+    stream_.apiHandle = 0;
+  }
+
+  for ( int i=0; i<2; i++ ) {
+    if ( stream_.userBuffer[i] ) {
+      free( stream_.userBuffer[i] );
+      stream_.userBuffer[i] = 0;
+    }
+  }
+
+  if ( stream_.deviceBuffer ) {
+    free( stream_.deviceBuffer );
+    stream_.deviceBuffer = 0;
+  }
+
+  return FAILURE;
+}
+
+void RtApiAsio :: closeStream()
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
+    error( 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 <[e-mail]>, 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() {
+    delete buffer_;
+  }
+
+  // sets the length of the internal ring buffer
+  void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
+    delete 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 ) ( 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::wstring deviceName;
+  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;
+  }
+
+  deviceName = defaultDeviceNameProp.pwszVal;
+  defaultDeviceName = std::string( deviceName.begin(), deviceName.end() );
+
+  // 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;
+  }
+
+  deviceName = deviceNameProp.pwszVal;
+  info.name = std::string( deviceName.begin(), deviceName.end() );
+
+  // 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] );
+  }
+
+  // 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;
+}
+