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3rdparty/iaxclient/lib/portaudio/src/hostapi/asio/ASIO-README.txt

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+ASIO-README.txt
+
+This document contains information to help you compile PortAudio with 
+ASIO support. If you find any omissions or errors in this document 
+please notify Ross Bencina <rossb@audiomulch.com>.
+
+
+Building PortAudio with ASIO support
+------------------------------------
+
+To build PortAudio with ASIO support you need to compile and link with
+pa_asio.c, and files from the ASIO SDK (see below), along with the common 
+files from pa_common/ and platform specific files from pa_win/ (for Win32) 
+or pa_mac/ (for Macintosh).
+
+If you are compiling with a non-Microsoft compiler on windows, also 
+compile and link with iasiothiscallresolver.cpp (see below for 
+an explanation).
+
+For some platforms (MingW, possibly Mac), you may simply
+be able to type:
+
+./configure --with-host_os=mingw --with-winapi=asio [--with-asiodir=/usr/local/asiosdk2]
+make
+
+./configure --with-host_os=darwin --with-winapi=asio [--with-asiodir=/usr/local/asiosdk2]
+make
+
+and life will be good.
+
+
+Obtaining the ASIO SDK
+----------------------
+
+In order to build PortAudio with ASIO support, you need to download 
+the ASIO SDK (version 2.0) from Steinberg. Steinberg makes the ASIO 
+SDK available to anyone free of charge, however they do not permit its 
+source code to be distributed.
+
+NOTE: In some cases the ASIO SDK may require patching, see below 
+for further details.
+
+http://www.steinberg.net/en/ps/support/3rdparty/asio_sdk/
+
+If the above link is broken search Google for:
+"download steinberg ASIO SDK"
+
+
+
+Building the ASIO SDK on Macintosh
+----------------------------------
+
+To build the ASIO SDK on Macintosh you need to compile and link with the 
+following files from the ASIO SDK:
+
+host/asiodrivers.cpp 
+host/mac/asioshlib.cpp 
+host/mac/codefragements.cpp
+
+
+
+Building the ASIO SDK on Windows
+--------------------------------
+
+To build the ASIO SDK on Windows you need to compile and link with the 
+following files from the ASIO SDK:
+
+asio_sdk\common\asio.cpp
+asio_sdk\host\asiodrivers.cpp
+asio_sdk\host\pc\asiolist.cpp
+
+You may also need to adjust your include paths to support inclusion of 
+header files from the above directories.
+
+The ASIO SDK depends on the following COM API functions: 
+CoInitialize, CoUninitialize, CoCreateInstance, CLSIDFromString
+For compilation with MinGW you will need to link with -lole32, for
+Borland link with Import32.lib.
+
+
+
+Non-Microsoft (MSVC) Compilers on Windows including Borland and GCC
+-------------------------------------------------------------------
+
+Steinberg did not specify a calling convention in the IASIO interface 
+definition. This causes the Microsoft compiler to use the proprietary 
+thiscall convention which is not compatible with other compilers, such 
+as compilers from Borland (BCC and C++Builder) and GNU (gcc). 
+Steinberg's ASIO SDK will compile but crash on initialization if 
+compiled with a non-Microsoft compiler on Windows.
+
+PortAudio solves this problem using the iasiothiscallresolver library 
+which is included in the distribution. When building ASIO support for
+non-Microsoft compilers, be sure to compile and link with
+iasiothiscallresolver.cpp. Note that iasiothiscallresolver includes
+conditional directives which cause it to have no effect if it is
+compiled with a Microsoft compiler, or on the Macintosh.
+
+If you use configure and make (see above), this should be handled
+automatically for you.
+
+For further information about the IASIO thiscall problem see this page:
+http://www.audiomulch.com/~rossb/code/calliasio
+
+
+
+Macintosh ASIO SDK Bug Patch
+----------------------------
+
+There is a bug in the ASIO SDK that causes the Macintosh version to 
+often fail during initialization. Below is a patch that you can apply.
+
+In codefragments.cpp replace getFrontProcessDirectory function with 
+the following one (GetFrontProcess replaced by GetCurrentProcess).
+
+
+bool CodeFragments::getFrontProcessDirectory(void *specs)
+{
+	FSSpec *fss = (FSSpec *)specs;
+	ProcessInfoRec pif;
+	ProcessSerialNumber psn;
+
+	memset(&psn,0,(long)sizeof(ProcessSerialNumber));
+	//  if(GetFrontProcess(&psn) == noErr)  // wrong !!!
+	if(GetCurrentProcess(&psn) == noErr)  // correct !!!
+	{
+		pif.processName = 0;
+		pif.processAppSpec = fss;
+		pif.processInfoLength = sizeof(ProcessInfoRec);
+		if(GetProcessInformation(&psn, &pif) == noErr)
+				return true;
+	}
+	return false;
+}
+
+
+---

3rdparty/iaxclient/lib/portaudio/src/hostapi/asio/iasiothiscallresolver.cpp

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+/*
+	IASIOThiscallResolver.cpp see the comments in iasiothiscallresolver.h for
+    the top level description - this comment describes the technical details of
+    the implementation.
+
+    The latest version of this file is available from:
+    http://www.audiomulch.com/~rossb/code/calliasio
+
+    please email comments to Ross Bencina <rossb@audiomulch.com>
+
+    BACKGROUND
+
+    The IASIO interface declared in the Steinberg ASIO 2 SDK declares
+    functions with no explicit calling convention. This causes MSVC++ to default
+    to using the thiscall convention, which is a proprietary convention not
+    implemented by some non-microsoft compilers - notably borland BCC,
+    C++Builder, and gcc. MSVC++ is the defacto standard compiler used by
+    Steinberg. As a result of this situation, the ASIO sdk will compile with
+    any compiler, however attempting to execute the compiled code will cause a
+    crash due to different default calling conventions on non-Microsoft
+    compilers.
+
+    IASIOThiscallResolver solves the problem by providing an adapter class that
+    delegates to the IASIO interface using the correct calling convention
+    (thiscall). Due to the lack of support for thiscall in the Borland and GCC
+    compilers, the calls have been implemented in assembly language.
+
+    A number of macros are defined for thiscall function calls with different
+    numbers of parameters, with and without return values - it may be possible
+    to modify the format of these macros to make them work with other inline
+    assemblers.
+
+
+    THISCALL DEFINITION
+
+    A number of definitions of the thiscall calling convention are floating
+    around the internet. The following definition has been validated against
+    output from the MSVC++ compiler:
+
+    For non-vararg functions, thiscall works as follows: the object (this)
+    pointer is passed in ECX. All arguments are passed on the stack in
+    right to left order. The return value is placed in EAX. The callee
+    clears the passed arguments from the stack.
+
+
+    FINDING FUNCTION POINTERS FROM AN IASIO POINTER
+
+    The first field of a COM object is a pointer to its vtble. Thus a pointer
+    to an object implementing the IASIO interface also points to a pointer to
+    that object's vtbl. The vtble is a table of function pointers for all of
+    the virtual functions exposed by the implemented interfaces.
+
+    If we consider a variable declared as a pointer to IASO:
+
+    IASIO *theAsioDriver
+
+    theAsioDriver points to:
+
+    object implementing IASIO
+    {
+        IASIOvtbl *vtbl
+        other data
+    }
+
+    in other words, theAsioDriver points to a pointer to an IASIOvtbl
+
+    vtbl points to a table of function pointers:
+
+    IASIOvtbl ( interface IASIO : public IUnknown )
+    {
+    (IUnknown functions)
+    0   virtual HRESULT STDMETHODCALLTYPE (*QueryInterface)(REFIID riid, void **ppv) = 0;
+    4   virtual ULONG STDMETHODCALLTYPE (*AddRef)() = 0;
+    8   virtual ULONG STDMETHODCALLTYPE (*Release)() = 0;      
+
+    (IASIO functions)
+    12	virtual ASIOBool (*init)(void *sysHandle) = 0;
+    16	virtual void (*getDriverName)(char *name) = 0;
+    20	virtual long (*getDriverVersion)() = 0;
+    24	virtual void (*getErrorMessage)(char *string) = 0;
+    28	virtual ASIOError (*start)() = 0;
+    32	virtual ASIOError (*stop)() = 0;
+    36	virtual ASIOError (*getChannels)(long *numInputChannels, long *numOutputChannels) = 0;
+    40	virtual ASIOError (*getLatencies)(long *inputLatency, long *outputLatency) = 0;
+    44	virtual ASIOError (*getBufferSize)(long *minSize, long *maxSize,
+            long *preferredSize, long *granularity) = 0;
+    48	virtual ASIOError (*canSampleRate)(ASIOSampleRate sampleRate) = 0;
+    52	virtual ASIOError (*getSampleRate)(ASIOSampleRate *sampleRate) = 0;
+    56	virtual ASIOError (*setSampleRate)(ASIOSampleRate sampleRate) = 0;
+    60	virtual ASIOError (*getClockSources)(ASIOClockSource *clocks, long *numSources) = 0;
+    64	virtual ASIOError (*setClockSource)(long reference) = 0;
+    68	virtual ASIOError (*getSamplePosition)(ASIOSamples *sPos, ASIOTimeStamp *tStamp) = 0;
+    72	virtual ASIOError (*getChannelInfo)(ASIOChannelInfo *info) = 0;
+    76	virtual ASIOError (*createBuffers)(ASIOBufferInfo *bufferInfos, long numChannels,
+            long bufferSize, ASIOCallbacks *callbacks) = 0;
+    80	virtual ASIOError (*disposeBuffers)() = 0;
+    84	virtual ASIOError (*controlPanel)() = 0;
+    88	virtual ASIOError (*future)(long selector,void *opt) = 0;
+    92	virtual ASIOError (*outputReady)() = 0;
+    };
+
+    The numbers in the left column show the byte offset of each function ptr
+    from the beginning of the vtbl. These numbers are used in the code below
+    to select different functions.
+
+    In order to find the address of a particular function, theAsioDriver
+    must first be dereferenced to find the value of the vtbl pointer:
+
+    mov     eax, theAsioDriver
+    mov     edx, [theAsioDriver]  // edx now points to vtbl[0]
+
+    Then an offset must be added to the vtbl pointer to select a
+    particular function, for example vtbl+44 points to the slot containing
+    a pointer to the getBufferSize function.
+
+    Finally vtbl+x must be dereferenced to obtain the value of the function
+    pointer stored in that address:
+
+    call    [edx+44]    // call the function pointed to by
+                        // the value in the getBufferSize field of the vtbl
+
+
+    SEE ALSO
+
+    Martin Fay's OpenASIO DLL at http://www.martinfay.com solves the same
+    problem by providing a new COM interface which wraps IASIO with an
+    interface that uses portable calling conventions. OpenASIO must be compiled
+    with MSVC, and requires that you ship the OpenASIO DLL with your
+    application.
+
+    
+    ACKNOWLEDGEMENTS
+
+    Ross Bencina: worked out the thiscall details above, wrote the original
+    Borland asm macros, and a patch for asio.cpp (which is no longer needed).
+    Thanks to Martin Fay for introducing me to the issues discussed here,
+    and to Rene G. Ceballos for assisting with asm dumps from MSVC++.
+
+    Antti Silvast: converted the original calliasio to work with gcc and NASM
+    by implementing the asm code in a separate file.
+
+	Fraser Adams: modified the original calliasio containing the Borland inline
+    asm to add inline asm for gcc i.e. Intel syntax for Borland and AT&T syntax
+    for gcc. This seems a neater approach for gcc than to have a separate .asm
+    file and it means that we only need one version of the thiscall patch.
+
+    Fraser Adams: rewrote the original calliasio patch in the form of the
+    IASIOThiscallResolver class in order to avoid modifications to files from
+    the Steinberg SDK, which may have had potential licence issues.
+
+    Andrew Baldwin: contributed fixes for compatibility problems with more
+    recent versions of the gcc assembler.
+*/
+
+
+// We only need IASIOThiscallResolver at all if we are on Win32. For other
+// platforms we simply bypass the IASIOThiscallResolver definition to allow us
+// to be safely #include'd whatever the platform to keep client code portable
+#if defined(WIN32) || defined(_WIN32) || defined(__WIN32__)
+
+
+// If microsoft compiler we can call IASIO directly so IASIOThiscallResolver
+// is not used.
+#if !defined(_MSC_VER)
+
+
+#include <new>
+#include <assert.h>
+
+// We have a mechanism in iasiothiscallresolver.h to ensure that asio.h is
+// #include'd before it in client code, we do NOT want to do this test here.
+#define iasiothiscallresolver_sourcefile 1
+#include "iasiothiscallresolver.h"
+#undef iasiothiscallresolver_sourcefile
+
+// iasiothiscallresolver.h redefines ASIOInit for clients, but we don't want
+// this macro defined in this translation unit.
+#undef ASIOInit
+
+
+// theAsioDriver is a global pointer to the current IASIO instance which the
+// ASIO SDK uses to perform all actions on the IASIO interface. We substitute
+// our own forwarding interface into this pointer.
+extern IASIO* theAsioDriver;
+
+
+// The following macros define the inline assembler for BORLAND first then gcc
+
+#if defined(__BCPLUSPLUS__) || defined(__BORLANDC__)          
+
+
+#define CALL_THISCALL_0( resultName, thisPtr, funcOffset )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     eax, param1           ;                                     \
+        push    eax                   ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+    }
+
+
+#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     eax, param1           ;                                     \
+        push    eax                   ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 )\
+    void *this_ = (thisPtr);                                                \
+    void *doubleParamPtr_ (&param1);                                        \
+    __asm {                                                                 \
+        mov     eax, doubleParamPtr_  ;                                     \
+        push    [eax+4]               ;                                     \
+        push    [eax]                 ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     eax, param2           ;                                     \
+        push    eax                   ;                                     \
+        mov     eax, param1           ;                                     \
+        push    eax                   ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     eax, param4           ;                                     \
+        push    eax                   ;                                     \
+        mov     eax, param3           ;                                     \
+        push    eax                   ;                                     \
+        mov     eax, param2           ;                                     \
+        push    eax                   ;                                     \
+        mov     eax, param1           ;                                     \
+        push    eax                   ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#elif defined(__GNUC__)
+
+
+#define CALL_THISCALL_0( resultName, thisPtr, funcOffset )                  \
+    __asm__ __volatile__ ("movl (%1), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"c"(thisPtr)     /* Input Operands */            \
+                         );                                                 \
+
+
+#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 )                 \
+    __asm__ __volatile__ ("pushl %0\n\t"                                    \
+                          "movl (%1), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :                 /* Output Operands */           \
+                          :"r"(param1),     /* Input Operands */            \
+                           "c"(thisPtr)                                     \
+                         );                                                 \
+
+
+#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 )          \
+    __asm__ __volatile__ ("pushl %1\n\t"                                    \
+                          "movl (%2), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"r"(param1),     /* Input Operands */            \
+                           "c"(thisPtr)                                     \
+                          );                                                \
+
+
+#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 )   \
+    __asm__ __volatile__ ("pushl 4(%1)\n\t"                                 \
+                          "pushl (%1)\n\t"                                  \
+                          "movl (%2), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx);\n\t"                 \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"a"(&param1),    /* Input Operands */            \
+                           /* Note: Using "r" above instead of "a" fails */ \
+                           /* when using GCC 3.3.3, and maybe later versions*/\
+                           "c"(thisPtr)                                     \
+                          );                                                \
+
+
+#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 )  \
+    __asm__ __volatile__ ("pushl %1\n\t"                                    \
+                          "pushl %2\n\t"                                    \
+                          "movl (%3), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"r"(param2),     /* Input Operands */            \
+                           "r"(param1),                                     \
+                           "c"(thisPtr)                                     \
+                          );                                                \
+
+
+#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\
+    __asm__ __volatile__ ("pushl %1\n\t"                                    \
+                          "pushl %2\n\t"                                    \
+                          "pushl %3\n\t"                                    \
+                          "pushl %4\n\t"                                    \
+                          "movl (%5), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"r"(param4),     /* Input Operands  */           \
+                           "r"(param3),                                     \
+                           "r"(param2),                                     \
+                           "r"(param1),                                     \
+                           "c"(thisPtr)                                     \
+                          );                                                \
+
+#endif
+
+
+
+// Our static singleton instance.
+IASIOThiscallResolver IASIOThiscallResolver::instance;
+
+// Constructor called to initialize static Singleton instance above. Note that
+// it is important not to clear that_ incase it has already been set by the call
+// to placement new in ASIOInit().
+IASIOThiscallResolver::IASIOThiscallResolver()
+{
+}
+
+// Constructor called from ASIOInit() below
+IASIOThiscallResolver::IASIOThiscallResolver(IASIO* that)
+: that_( that )
+{
+}
+
+// Implement IUnknown methods as assert(false). IASIOThiscallResolver is not
+// really a COM object, just a wrapper which will work with the ASIO SDK.
+// If you wanted to use ASIO without the SDK you might want to implement COM
+// aggregation in these methods.
+HRESULT STDMETHODCALLTYPE IASIOThiscallResolver::QueryInterface(REFIID riid, void **ppv)
+{
+    (void)riid;     // suppress unused variable warning
+
+    assert( false ); // this function should never be called by the ASIO SDK.
+
+    *ppv = NULL;
+    return E_NOINTERFACE;
+}
+
+ULONG STDMETHODCALLTYPE IASIOThiscallResolver::AddRef()
+{
+    assert( false ); // this function should never be called by the ASIO SDK.
+
+    return 1;
+}
+
+ULONG STDMETHODCALLTYPE IASIOThiscallResolver::Release()
+{
+    assert( false ); // this function should never be called by the ASIO SDK.
+    
+    return 1;
+}
+
+
+// Implement the IASIO interface methods by performing the vptr manipulation
+// described above then delegating to the real implementation.
+ASIOBool IASIOThiscallResolver::init(void *sysHandle)
+{
+    ASIOBool result;
+    CALL_THISCALL_1( result, that_, 12, sysHandle );
+    return result;
+}
+
+void IASIOThiscallResolver::getDriverName(char *name)
+{
+    CALL_VOID_THISCALL_1( that_, 16, name );
+}
+
+long IASIOThiscallResolver::getDriverVersion()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 20 );
+    return result;
+}
+
+void IASIOThiscallResolver::getErrorMessage(char *string)
+{
+     CALL_VOID_THISCALL_1( that_, 24, string );
+}
+
+ASIOError IASIOThiscallResolver::start()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 28 );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::stop()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 32 );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getChannels(long *numInputChannels, long *numOutputChannels)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 36, numInputChannels, numOutputChannels );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getLatencies(long *inputLatency, long *outputLatency)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 40, inputLatency, outputLatency );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getBufferSize(long *minSize, long *maxSize,
+        long *preferredSize, long *granularity)
+{
+    ASIOBool result;
+    CALL_THISCALL_4( result, that_, 44, minSize, maxSize, preferredSize, granularity );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::canSampleRate(ASIOSampleRate sampleRate)
+{
+    ASIOBool result;
+    CALL_THISCALL_1_DOUBLE( result, that_, 48, sampleRate );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getSampleRate(ASIOSampleRate *sampleRate)
+{
+    ASIOBool result;
+    CALL_THISCALL_1( result, that_, 52, sampleRate );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::setSampleRate(ASIOSampleRate sampleRate)
+{    
+    ASIOBool result;
+    CALL_THISCALL_1_DOUBLE( result, that_, 56, sampleRate );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getClockSources(ASIOClockSource *clocks, long *numSources)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 60, clocks, numSources );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::setClockSource(long reference)
+{
+    ASIOBool result;
+    CALL_THISCALL_1( result, that_, 64, reference );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 68, sPos, tStamp );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getChannelInfo(ASIOChannelInfo *info)
+{
+    ASIOBool result;
+    CALL_THISCALL_1( result, that_, 72, info );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::createBuffers(ASIOBufferInfo *bufferInfos,
+        long numChannels, long bufferSize, ASIOCallbacks *callbacks)
+{
+    ASIOBool result;
+    CALL_THISCALL_4( result, that_, 76, bufferInfos, numChannels, bufferSize, callbacks );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::disposeBuffers()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 80 );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::controlPanel()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 84 );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::future(long selector,void *opt)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 88, selector, opt );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::outputReady()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 92 );
+    return result;
+}
+
+
+// Implement our substitute ASIOInit() method
+ASIOError IASIOThiscallResolver::ASIOInit(ASIODriverInfo *info)
+{
+    // To ensure that our instance's vptr is correctly constructed, even if
+    // ASIOInit is called prior to main(), we explicitly call its constructor
+    // (potentially over the top of an existing instance). Note that this is
+    // pretty ugly, and is only safe because IASIOThiscallResolver has no
+    // destructor and contains no objects with destructors.
+    new((void*)&instance) IASIOThiscallResolver( theAsioDriver );
+
+    // Interpose between ASIO client code and the real driver.
+    theAsioDriver = &instance;
+
+    // Note that we never need to switch theAsioDriver back to point to the
+    // real driver because theAsioDriver is reset to zero in ASIOExit().
+
+    // Delegate to the real ASIOInit
+	return ::ASIOInit(info);
+}
+
+
+#endif /* !defined(_MSC_VER) */
+
+#endif /* Win32 */
+

3rdparty/iaxclient/lib/portaudio/src/hostapi/asio/iasiothiscallresolver.h

@@ -0,0 +1,197 @@
+// ****************************************************************************
+// File:			IASIOThiscallResolver.h
+// Description:     The IASIOThiscallResolver class implements the IASIO
+//					interface and acts as a proxy to the real IASIO interface by
+//                  calling through its vptr table using the thiscall calling
+//                  convention. To put it another way, we interpose
+//                  IASIOThiscallResolver between ASIO SDK code and the driver.
+//                  This is necessary because most non-Microsoft compilers don't
+//                  implement the thiscall calling convention used by IASIO.
+//
+//					iasiothiscallresolver.cpp contains the background of this
+//					problem plus a technical description of the vptr
+//                  manipulations.
+//
+//					In order to use this mechanism one simply has to add
+//					iasiothiscallresolver.cpp to the list of files to compile
+//                  and #include <iasiothiscallresolver.h>
+//
+//					Note that this #include must come after the other ASIO SDK
+//                  #includes, for example:
+//
+//					#include <windows.h>
+//					#include <asiosys.h>
+//					#include <asio.h>
+//					#include <asiodrivers.h>
+//					#include <iasiothiscallresolver.h>
+//
+//					Actually the important thing is to #include
+//                  <iasiothiscallresolver.h> after <asio.h>. We have
+//                  incorporated a test to enforce this ordering.
+//
+//					The code transparently takes care of the interposition by
+//                  using macro substitution to intercept calls to ASIOInit()
+//                  and ASIOExit(). We save the original ASIO global
+//                  "theAsioDriver" in our "that" variable, and then set
+//                  "theAsioDriver" to equal our IASIOThiscallResolver instance.
+//
+// 					Whilst this method of resolving the thiscall problem requires
+//					the addition of #include <iasiothiscallresolver.h> to client
+//                  code it has the advantage that it does not break the terms
+//                  of the ASIO licence by publishing it. We are NOT modifying
+//                  any Steinberg code here, we are merely implementing the IASIO
+//					interface in the same way that we would need to do if we
+//					wished to provide an open source ASIO driver.
+//
+//					For compilation with MinGW -lole32 needs to be added to the
+//                  linker options. For BORLAND, linking with Import32.lib is
+//                  sufficient.
+//
+//					The dependencies are with: CoInitialize, CoUninitialize,
+//					CoCreateInstance, CLSIDFromString - used by asiolist.cpp
+//					and are required on Windows whether ThiscallResolver is used
+//					or not.
+//
+//					Searching for the above strings in the root library path
+//					of your compiler should enable the correct libraries to be
+//					identified if they aren't immediately obvious.
+//
+//                  Note that the current implementation of IASIOThiscallResolver
+//                  is not COM compliant - it does not correctly implement the
+//                  IUnknown interface. Implementing it is not necessary because
+//                  it is not called by parts of the ASIO SDK which call through
+//                  theAsioDriver ptr. The IUnknown methods are implemented as
+//                  assert(false) to ensure that the code fails if they are
+//                  ever called.
+// Restrictions:	None. Public Domain & Open Source distribute freely
+//					You may use IASIOThiscallResolver commercially as well as
+//                  privately.
+//					You the user assume the responsibility for the use of the
+//					files, binary or text, and there is no guarantee or warranty,
+//					expressed or implied, including but not limited to the
+//					implied warranties of merchantability and fitness for a
+//					particular purpose. You assume all responsibility and agree
+//					to hold no entity, copyright holder or distributors liable
+//					for any loss of data or inaccurate representations of data
+//					as a result of using IASIOThiscallResolver.
+// Version:         1.4 Added separate macro CALL_THISCALL_1_DOUBLE from
+//                  Andrew Baldwin, and volatile for whole gcc asm blocks,
+//                  both for compatibility with newer gcc versions. Cleaned up
+//                  Borland asm to use one less register.
+//                  1.3 Switched to including assert.h for better compatibility.
+//                  Wrapped entire .h and .cpp contents with a check for
+//                  _MSC_VER to provide better compatibility with MS compilers.
+//                  Changed Singleton implementation to use static instance
+//                  instead of freestore allocated instance. Removed ASIOExit
+//                  macro as it is no longer needed.
+//                  1.2 Removed semicolons from ASIOInit and ASIOExit macros to
+//                  allow them to be embedded in expressions (if statements).
+//                  Cleaned up some comments. Removed combase.c dependency (it
+//                  doesn't compile with BCB anyway) by stubbing IUnknown.
+//                  1.1 Incorporated comments from Ross Bencina including things
+//					such as changing name from ThiscallResolver to
+//					IASIOThiscallResolver, tidying up the constructor, fixing
+//					a bug in IASIOThiscallResolver::ASIOExit() and improving
+//					portability through the use of conditional compilation
+//					1.0 Initial working version.
+// Created:			6/09/2003
+// Authors:         Fraser Adams
+//                  Ross Bencina
+//                  Rene G. Ceballos
+//                  Martin Fay
+//                  Antti Silvast
+//                  Andrew Baldwin
+//
+// ****************************************************************************
+
+
+#ifndef included_iasiothiscallresolver_h
+#define included_iasiothiscallresolver_h
+
+// We only need IASIOThiscallResolver at all if we are on Win32. For other
+// platforms we simply bypass the IASIOThiscallResolver definition to allow us
+// to be safely #include'd whatever the platform to keep client code portable
+#if defined(WIN32) || defined(_WIN32) || defined(__WIN32__)
+
+
+// If microsoft compiler we can call IASIO directly so IASIOThiscallResolver
+// is not used.
+#if !defined(_MSC_VER)
+
+
+// The following is in order to ensure that this header is only included after
+// the other ASIO headers (except for the case of iasiothiscallresolver.cpp).
+// We need to do this because IASIOThiscallResolver works by eclipsing the
+// original definition of ASIOInit() with a macro (see below).
+#if !defined(iasiothiscallresolver_sourcefile)
+	#if !defined(__ASIO_H)
+	#error iasiothiscallresolver.h must be included AFTER asio.h
+	#endif
+#endif
+
+#include <windows.h>
+#include <asiodrvr.h> /* From ASIO SDK */
+
+
+class IASIOThiscallResolver : public IASIO {
+private:
+	IASIO* that_; // Points to the real IASIO
+
+	static IASIOThiscallResolver instance; // Singleton instance
+
+	// Constructors - declared private so construction is limited to
+    // our Singleton instance
+    IASIOThiscallResolver();
+	IASIOThiscallResolver(IASIO* that);
+public:
+
+    // Methods from the IUnknown interface. We don't fully implement IUnknown
+    // because the ASIO SDK never calls these methods through theAsioDriver ptr.
+    // These methods are implemented as assert(false).
+    virtual HRESULT STDMETHODCALLTYPE QueryInterface(REFIID riid, void **ppv);
+    virtual ULONG STDMETHODCALLTYPE AddRef();
+    virtual ULONG STDMETHODCALLTYPE Release();
+
+    // Methods from the IASIO interface, implemented as forwarning calls to that.
+	virtual ASIOBool init(void *sysHandle);
+	virtual void getDriverName(char *name);
+	virtual long getDriverVersion();
+	virtual void getErrorMessage(char *string);
+	virtual ASIOError start();
+	virtual ASIOError stop();
+	virtual ASIOError getChannels(long *numInputChannels, long *numOutputChannels);
+	virtual ASIOError getLatencies(long *inputLatency, long *outputLatency);
+	virtual ASIOError getBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity);
+	virtual ASIOError canSampleRate(ASIOSampleRate sampleRate);
+	virtual ASIOError getSampleRate(ASIOSampleRate *sampleRate);
+	virtual ASIOError setSampleRate(ASIOSampleRate sampleRate);
+	virtual ASIOError getClockSources(ASIOClockSource *clocks, long *numSources);
+	virtual ASIOError setClockSource(long reference);
+	virtual ASIOError getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp);
+	virtual ASIOError getChannelInfo(ASIOChannelInfo *info);
+	virtual ASIOError createBuffers(ASIOBufferInfo *bufferInfos, long numChannels, long bufferSize, ASIOCallbacks *callbacks);
+	virtual ASIOError disposeBuffers();
+	virtual ASIOError controlPanel();
+	virtual ASIOError future(long selector,void *opt);
+	virtual ASIOError outputReady();
+
+    // Class method, see ASIOInit() macro below.
+    static ASIOError ASIOInit(ASIODriverInfo *info); // Delegates to ::ASIOInit
+};
+
+
+// Replace calls to ASIOInit with our interposing version.
+// This macro enables us to perform thiscall resolution simply by #including
+// <iasiothiscallresolver.h> after the asio #includes (this file _must_ be
+// included _after_ the asio #includes)
+
+#define ASIOInit(name) IASIOThiscallResolver::ASIOInit((name))
+
+
+#endif /* !defined(_MSC_VER) */
+
+#endif /* Win32 */
+
+#endif /* included_iasiothiscallresolver_h */
+
+

3rdparty/iaxclient/lib/portaudio/src/hostapi/coreaudio/notes.txt

@@ -0,0 +1,145 @@
+Notes on status of CoreAudio Implementation of PortAudio
+
+Document Last Updated December 9, 2005
+
+There are currently two implementations of PortAudio for Mac Core Audio.
+
+The original is in pa_mac_core_old.c, and the newer, default implementation
+is in pa_mac_core.c.
+Only pa_mac_core.c is currently developed and supported as it uses apple's
+current core audio technology. To select use the old implementation, replace
+pa_mac_core.c with pa_mac_core_old.c (eg. "cp pa_mac_core_auhal.c
+pa_mac_core.c"), then run configure and make as usual.
+
+----------------------------------------
+
+Notes on Original implementation:
+
+by Phil Burk and Darren Gibbs
+
+Last updated March 20, 2002
+
+WHAT WORKS
+
+Output with very low latency, <10 msec.
+Half duplex input or output.
+Full duplex on the same CoreAudio device.
+The paFLoat32, paInt16, paInt8, paUInt8 sample formats.
+Pa_GetCPULoad()
+Pa_StreamTime()
+
+KNOWN BUGS OR LIMITATIONS
+
+We do not yet support simultaneous input and output on different 
+devices. Note that some CoreAudio devices like the Roland UH30 look 
+like one device but are actually two different CoreAudio devices. The 
+Built-In audio is typically one CoreAudio device.
+
+Mono doesn't work.
+
+DEVICE MAPPING
+
+CoreAudio devices can support both input and output. But the sample 
+rates supported may be different. So we have map one or two PortAudio 
+device to each CoreAudio device depending on whether it supports 
+input, output or both.
+
+When we query devices, we first get a list of CoreAudio devices. Then 
+we scan the list and add a PortAudio device for each CoreAudio device 
+that supports input. Then we make a scan for output devices.
+
+-------------------------------------------
+
+Notes on Newer/Default AUHAL implementation:
+
+by Bjorn Roche
+
+Last Updated December 9, 2005
+
+Principle of Operation:
+
+This implementation uses AUHAL for audio I/O. To some extent, it also
+operates at the "HAL" Layer, though this behavior can be limited by
+platform specific flags (see pa_mac_core.h for details). The default
+settings should be reasonable: they don't change the SR of the device and
+don't cause interruptions if other devices are using the device.
+
+Major Software Elements Used: Apple's HAL AUs provide output SR
+conversion transparently, however, only on output, so this
+implementation uses AudioConverters to convert the sample rate on input.
+A PortAudio ring buffer is used to buffer input when sample rate
+conversion is required or when separate audio units are used for duplex
+IO. Finally, a PortAudio buffer processor is used to convert formats and
+provide additional buffers if needed. Internally, interleaved floating
+point data streams are used exclusively - the audio unit converts from
+the audio hardware's native format to interleaved float PCM and
+PortAudio's Buffer processor is used for conversion to user formats.
+
+Simplex Input: Simplex input uses a single callback. If sample rate
+conversion is required, a ring buffer and AudioConverter are used as
+well.
+
+Simplex output: Simplex output uses a single callback. No ring buffer or
+audio converter is used because AUHAL does its own output SR conversion.
+
+Duplex, one device (no SR conversion): When one device is used, a single
+callback is used. This achieves very low latency.
+
+Duplex, separate devices or SR conversion: When SR conversion is
+required, data must be buffered before it is converted and data is not
+always available at the same times on input and output, so SR conversion
+requires the same treatment as separate devices. The input callback
+reads data and puts it in the ring buffer. The output callback reads the
+data off the ring buffer, into an audio converter and finally to the
+buffer processor.
+
+Platform Specific Options:
+
+By using the flags in pa_mac_core.h, the user may specify several options.
+For example, the user can specify the sample-rate conversion quality, and
+the extent to which PA will attempt to "play nice" and to what extent it
+will interrupt other apps to improve performance. For example, if 44100 Hz
+sample rate is requested but the device is set at 48000 Hz, PA can either
+change the device for optimal playback ("Pro" mode), which may interrupt
+other programs playing back audio, or simple use a sample-rate coversion,
+which allows for friendlier sharing of the device ("Play Nice" mode).
+
+
+Known issues:
+
+- Latency: Latency settings are ignored in most cases. Exceptions are when
+doing I/O between different devices and as a hint for selecting a realtively
+low or relatively high latency in conjunction with
+paHostFramesPerBufferUnspecified. Latency settings are always automatically
+bound to "safe" values, however, so setting extreme values here should not be
+an issue.
+
+- Buffer Size: paHostFramesPerBufferUnspecified and specific host buffer sizes
+are supported. paHostFramesPerBufferUnspecified works best in "pro" mode,
+where the buffer size and sample rate of the audio device is most likely
+to match the expected values.
+
+- Timing info. It reports on stream time, but I'm probably doing something
+wrong since patest_sine_time often reports negative latency numbers.
+
+- xrun detection: The only xrun detection performed is when reading
+and writing the ring buffer. There is probably more that can be done.
+
+- abort/stop issues: stopping a stream is always a complete operation,
+but latency should be low enough to make the lack of a separate abort
+unnecessary. Apple clarifies its AudioOutputUnitStop() call here:
+http://lists.apple.com/archives/coreaudio-api/2005/Dec/msg00055.html
+
+- blocking interface: Not implemented.
+
+- multichannel: It has been tested successfully on multichannel hardware
+from MOTU: traveler and 896HD.
+
+- sample rate conversion quality: By default, SR conversion is the maximum
+available. This can be tweaked using flags pa_mac_core.h. Note that the AU
+render quyality property is used to set the sample rat conversion quality
+as "documented" here:
+http://lists.apple.com/archives/coreaudio-api/2004/Jan/msg00141.html
+
+- x86: I haven't tested it on an x86 Mac myself, but users have reported 
+being able to comiple and run it.

3rdparty/iaxclient/lib/portaudio/src/hostapi/coreaudio/pa_mac_core.c

@@ -0,0 +1,907 @@
+/*
+ * $Id: pa_mac_core_old.c,v 1.1 2006/06/10 21:30:56 dmazzoni Exp $
+ * pa_mac_core.c
+ * Implementation of PortAudio for Mac OS X CoreAudio       
+ *                                                                                         
+ * PortAudio Portable Real-Time Audio Library
+ * Latest Version at: http://www.portaudio.com
+ *
+ * Authors: Ross Bencina and Phil Burk
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ *
+ */
+
+#include <CoreAudio/CoreAudio.h>
+#include <AudioToolbox/AudioToolbox.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <math.h>
+#include <assert.h>
+
+#include "portaudio.h"
+#include "pa_trace.h"
+#include "pa_util.h"
+#include "pa_allocation.h"
+#include "pa_hostapi.h"
+#include "pa_stream.h"
+#include "pa_cpuload.h"
+#include "pa_process.h"
+
+// =====  constants  =====
+
+// =====  structs  =====
+#pragma mark structs
+
+// PaMacCoreHostApiRepresentation - host api datastructure specific to this implementation
+typedef struct PaMacCore_HAR
+{
+    PaUtilHostApiRepresentation inheritedHostApiRep;
+    PaUtilStreamInterface callbackStreamInterface;
+    PaUtilStreamInterface blockingStreamInterface;
+    
+    PaUtilAllocationGroup *allocations;
+    AudioDeviceID *macCoreDeviceIds;
+}
+PaMacCoreHostApiRepresentation;
+
+typedef struct PaMacCore_DI
+{
+    PaDeviceInfo inheritedDeviceInfo;
+}
+PaMacCoreDeviceInfo;
+
+// PaMacCoreStream - a stream data structure specifically for this implementation
+typedef struct PaMacCore_S
+{
+    PaUtilStreamRepresentation streamRepresentation;
+    PaUtilCpuLoadMeasurer cpuLoadMeasurer;
+    PaUtilBufferProcessor bufferProcessor;
+    
+    int primeStreamUsingCallback;
+    
+    AudioDeviceID inputDevice;
+    AudioDeviceID outputDevice;
+    
+    // Processing thread management --------------
+//    HANDLE abortEvent;
+//    HANDLE processingThread;
+//    DWORD processingThreadId;
+    
+    char throttleProcessingThreadOnOverload; // 0 -> don't throtte, non-0 -> throttle
+    int processingThreadPriority;
+    int highThreadPriority;
+    int throttledThreadPriority;
+    unsigned long throttledSleepMsecs;
+    
+    int isStopped;
+    volatile int isActive;
+    volatile int stopProcessing; // stop thread once existing buffers have been returned
+    volatile int abortProcessing; // stop thread immediately
+    
+//    DWORD allBuffersDurationMs; // used to calculate timeouts
+}
+PaMacCoreStream;
+
+// Data needed by the CoreAudio callback functions
+typedef struct PaMacCore_CD
+{
+    PaMacCoreStream *stream;
+    PaStreamCallback *callback;
+    void *userData;
+    PaUtilConverter *inputConverter;
+    PaUtilConverter *outputConverter;
+    void *inputBuffer;
+    void *outputBuffer;
+    int inputChannelCount;
+    int outputChannelCount;
+    PaSampleFormat inputSampleFormat;
+    PaSampleFormat outputSampleFormat;
+    PaUtilTriangularDitherGenerator *ditherGenerator;
+}
+PaMacClientData;
+
+// =====  CoreAudio-PortAudio bridge functions =====
+#pragma mark CoreAudio-PortAudio bridge functions
+
+// Maps CoreAudio OSStatus codes to PortAudio PaError codes
+static PaError conv_err(OSStatus error)
+{
+    PaError result;
+    
+    switch (error) {
+        case kAudioHardwareNoError:
+            result = paNoError; break;
+        case kAudioHardwareNotRunningError:
+            result = paInternalError; break;
+        case kAudioHardwareUnspecifiedError:
+            result = paInternalError; break;
+        case kAudioHardwareUnknownPropertyError:
+            result = paInternalError; break;
+        case kAudioHardwareBadPropertySizeError:
+            result = paInternalError; break;
+        case kAudioHardwareIllegalOperationError:
+            result = paInternalError; break;
+        case kAudioHardwareBadDeviceError:
+            result = paInvalidDevice; break;
+        case kAudioHardwareBadStreamError:
+            result = paBadStreamPtr; break;
+        case kAudioHardwareUnsupportedOperationError:
+            result = paInternalError; break;
+        case kAudioDeviceUnsupportedFormatError:
+            result = paSampleFormatNotSupported; break;
+        case kAudioDevicePermissionsError:
+            result = paDeviceUnavailable; break;
+        default:
+            result = paInternalError;
+    }
+    
+    return result;
+}
+
+/* This function is unused
+static AudioStreamBasicDescription *InitializeStreamDescription(const PaStreamParameters *parameters, double sampleRate)
+{
+    struct AudioStreamBasicDescription *streamDescription = PaUtil_AllocateMemory(sizeof(AudioStreamBasicDescription));
+    streamDescription->mSampleRate = sampleRate;
+    streamDescription->mFormatID = kAudioFormatLinearPCM;
+    streamDescription->mFormatFlags = 0;
+    streamDescription->mFramesPerPacket = 1;
+    
+    if (parameters->sampleFormat & paNonInterleaved) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsNonInterleaved;
+        streamDescription->mChannelsPerFrame = 1;
+        streamDescription->mBytesPerFrame = Pa_GetSampleSize(parameters->sampleFormat);
+        streamDescription->mBytesPerPacket = Pa_GetSampleSize(parameters->sampleFormat);
+    }
+    else {
+        streamDescription->mChannelsPerFrame = parameters->channelCount;
+    }
+    
+    streamDescription->mBytesPerFrame = Pa_GetSampleSize(parameters->sampleFormat) * streamDescription->mChannelsPerFrame;
+    streamDescription->mBytesPerPacket = streamDescription->mBytesPerFrame * streamDescription->mFramesPerPacket;
+    
+    if (parameters->sampleFormat & paFloat32) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsFloat;
+        streamDescription->mBitsPerChannel = 32;
+    }
+    else if (parameters->sampleFormat & paInt32) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
+        streamDescription->mBitsPerChannel = 32;
+    }
+    else if (parameters->sampleFormat & paInt24) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
+        streamDescription->mBitsPerChannel = 24;
+    }
+    else if (parameters->sampleFormat & paInt16) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
+        streamDescription->mBitsPerChannel = 16;
+    }
+    else if (parameters->sampleFormat & paInt8) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
+        streamDescription->mBitsPerChannel = 8;
+    }    
+    else if (parameters->sampleFormat & paInt32) {
+        streamDescription->mBitsPerChannel = 8;
+    }
+    
+    return streamDescription;
+}
+*/
+
+static PaStreamCallbackTimeInfo *InitializeTimeInfo(const AudioTimeStamp* now, const AudioTimeStamp* inputTime, const AudioTimeStamp* outputTime)
+{
+    PaStreamCallbackTimeInfo *timeInfo = PaUtil_AllocateMemory(sizeof(PaStreamCallbackTimeInfo));
+    
+    timeInfo->inputBufferAdcTime = inputTime->mSampleTime;
+    timeInfo->currentTime = now->mSampleTime;
+    timeInfo->outputBufferDacTime = outputTime->mSampleTime;
+    
+    return timeInfo;
+}
+
+// =====  support functions  =====
+#pragma mark support functions
+
+static void CleanUp(PaMacCoreHostApiRepresentation *macCoreHostApi)
+{
+    if( macCoreHostApi->allocations )
+    {
+        PaUtil_FreeAllAllocations( macCoreHostApi->allocations );
+        PaUtil_DestroyAllocationGroup( macCoreHostApi->allocations );
+    }
+    
+    PaUtil_FreeMemory( macCoreHostApi );    
+}
+
+static PaError GetChannelInfo(PaDeviceInfo *deviceInfo, AudioDeviceID macCoreDeviceId, int isInput)
+{
+    UInt32 propSize;
+    PaError err = paNoError;
+    UInt32 i;
+    int numChannels = 0;
+    AudioBufferList *buflist;
+
+    err = conv_err(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, NULL));
+    buflist = PaUtil_AllocateMemory(propSize);
+    err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, buflist));
+    if (!err) {
+        for (i = 0; i < buflist->mNumberBuffers; ++i) {
+            numChannels += buflist->mBuffers[i].mNumberChannels;
+        }
+		
+		if (isInput)
+			deviceInfo->maxInputChannels = numChannels;
+		else
+			deviceInfo->maxOutputChannels = numChannels;
+		
+        int frameLatency;
+        propSize = sizeof(UInt32);
+        err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyLatency, &propSize, &frameLatency));
+        if (!err) {
+            double secondLatency = frameLatency / deviceInfo->defaultSampleRate;
+            if (isInput) {
+                deviceInfo->defaultLowInputLatency = secondLatency;
+                deviceInfo->defaultHighInputLatency = secondLatency;
+            }
+            else {
+                deviceInfo->defaultLowOutputLatency = secondLatency;
+                deviceInfo->defaultHighOutputLatency = secondLatency;
+            }
+        }
+    }
+    PaUtil_FreeMemory(buflist);
+    
+    return err;
+}
+
+static PaError InitializeDeviceInfo(PaMacCoreDeviceInfo *macCoreDeviceInfo,  AudioDeviceID macCoreDeviceId, PaHostApiIndex hostApiIndex )
+{
+    PaDeviceInfo *deviceInfo = &macCoreDeviceInfo->inheritedDeviceInfo;
+    deviceInfo->structVersion = 2;
+    deviceInfo->hostApi = hostApiIndex;
+    
+    PaError err = paNoError;
+    UInt32 propSize;
+
+    err = conv_err(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, NULL));
+    // FIXME: this allocation should be part of the allocations group
+    char *name = PaUtil_AllocateMemory(propSize);
+    err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, name));
+    if (!err) {
+        deviceInfo->name = name;
+    }
+    
+    Float64 sampleRate;
+    propSize = sizeof(Float64);
+    err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyNominalSampleRate, &propSize, &sampleRate));
+    if (!err) {
+        deviceInfo->defaultSampleRate = sampleRate;
+    }
+
+
+    // Get channel info
+    err = GetChannelInfo(deviceInfo, macCoreDeviceId, 1);
+    err = GetChannelInfo(deviceInfo, macCoreDeviceId, 0);
+
+    return err;
+}
+
+static PaError InitializeDeviceInfos( PaMacCoreHostApiRepresentation *macCoreHostApi, PaHostApiIndex hostApiIndex )
+{
+    PaError result = paNoError;
+    PaUtilHostApiRepresentation *hostApi;
+    PaMacCoreDeviceInfo *deviceInfoArray;
+
+    // initialise device counts and default devices under the assumption that there are no devices. These values are incremented below if and when devices are successfully initialized.
+    hostApi = &macCoreHostApi->inheritedHostApiRep;
+    hostApi->info.deviceCount = 0;
+    hostApi->info.defaultInputDevice = paNoDevice;
+    hostApi->info.defaultOutputDevice = paNoDevice;
+    
+    UInt32 propsize;
+    AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &propsize, NULL);
+    int numDevices = propsize / sizeof(AudioDeviceID);
+    hostApi->info.deviceCount = numDevices;
+    if (numDevices > 0) {
+        hostApi->deviceInfos = (PaDeviceInfo**)PaUtil_GroupAllocateMemory(
+                                            macCoreHostApi->allocations, sizeof(PaDeviceInfo*) * numDevices );
+        if( !hostApi->deviceInfos )
+        {
+            return paInsufficientMemory;
+        }
+
+        // allocate all device info structs in a contiguous block
+        deviceInfoArray = (PaMacCoreDeviceInfo*)PaUtil_GroupAllocateMemory(
+                                macCoreHostApi->allocations, sizeof(PaMacCoreDeviceInfo) * numDevices );
+        if( !deviceInfoArray )
+        {
+            return paInsufficientMemory;
+        }
+        
+        macCoreHostApi->macCoreDeviceIds = PaUtil_GroupAllocateMemory(macCoreHostApi->allocations, propsize);
+        AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &propsize, macCoreHostApi->macCoreDeviceIds);
+
+        AudioDeviceID defaultInputDevice, defaultOutputDevice;
+        propsize = sizeof(AudioDeviceID);
+        AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propsize, &defaultInputDevice);
+        AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &propsize, &defaultOutputDevice);
+        
+        UInt32 i;
+        for (i = 0; i < numDevices; ++i) {
+            if (macCoreHostApi->macCoreDeviceIds[i] == defaultInputDevice) {
+                hostApi->info.defaultInputDevice = i;
+            }
+            if (macCoreHostApi->macCoreDeviceIds[i] == defaultOutputDevice) {
+                hostApi->info.defaultOutputDevice = i;
+            }
+            InitializeDeviceInfo(&deviceInfoArray[i], macCoreHostApi->macCoreDeviceIds[i], hostApiIndex);
+            hostApi->deviceInfos[i] = &(deviceInfoArray[i].inheritedDeviceInfo);      
+        }
+    }
+
+    return result;
+}
+
+static OSStatus CheckFormat(AudioDeviceID macCoreDeviceId, const PaStreamParameters *parameters, double sampleRate, int isInput)
+{
+    UInt32 propSize = sizeof(AudioStreamBasicDescription);
+    AudioStreamBasicDescription *streamDescription = PaUtil_AllocateMemory(propSize);
+
+    streamDescription->mSampleRate = sampleRate;
+    streamDescription->mFormatID = 0;
+    streamDescription->mFormatFlags = 0;
+    streamDescription->mBytesPerPacket = 0;
+    streamDescription->mFramesPerPacket = 0;
+    streamDescription->mBytesPerFrame = 0;
+    streamDescription->mChannelsPerFrame = 0;
+    streamDescription->mBitsPerChannel = 0;
+    streamDescription->mReserved = 0;
+
+    OSStatus result = AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamFormatSupported, &propSize, streamDescription);
+    PaUtil_FreeMemory(streamDescription);
+    return result;
+}
+
+static OSStatus CopyInputData(PaMacClientData* destination, const AudioBufferList *source, unsigned long frameCount)
+{
+    int frameSpacing, channelSpacing;
+    if (destination->inputSampleFormat & paNonInterleaved) {
+        frameSpacing = 1;
+        channelSpacing = destination->inputChannelCount;
+    }
+    else {
+        frameSpacing = destination->inputChannelCount;
+        channelSpacing = 1;
+    }
+    
+    AudioBuffer const *inputBuffer = &source->mBuffers[0];
+    void *coreAudioBuffer = inputBuffer->mData;
+    void *portAudioBuffer = destination->inputBuffer;
+    UInt32 i, streamNumber, streamChannel;
+    for (i = streamNumber = streamChannel = 0; i < destination->inputChannelCount; ++i, ++streamChannel) {
+        if (streamChannel >= inputBuffer->mNumberChannels) {
+            ++streamNumber;
+            inputBuffer = &source->mBuffers[streamNumber];
+            coreAudioBuffer = inputBuffer->mData;
+            streamChannel = 0;
+        }
+        destination->inputConverter(portAudioBuffer, frameSpacing, coreAudioBuffer, inputBuffer->mNumberChannels, frameCount, destination->ditherGenerator);
+        coreAudioBuffer += sizeof(Float32);
+        portAudioBuffer += Pa_GetSampleSize(destination->inputSampleFormat) * channelSpacing;
+    }
+    return noErr;
+}
+
+static OSStatus CopyOutputData(AudioBufferList* destination, PaMacClientData *source, unsigned long frameCount)
+{
+    int frameSpacing, channelSpacing;
+    if (source->outputSampleFormat & paNonInterleaved) {
+        frameSpacing = 1;
+        channelSpacing = source->outputChannelCount;
+    }
+    else {
+        frameSpacing = source->outputChannelCount;
+        channelSpacing = 1;
+    }
+    
+    AudioBuffer *outputBuffer = &destination->mBuffers[0];
+    void *coreAudioBuffer = outputBuffer->mData;
+    void *portAudioBuffer = source->outputBuffer;
+    UInt32 i, streamNumber, streamChannel;
+    for (i = streamNumber = streamChannel = 0; i < source->outputChannelCount; ++i, ++streamChannel) {
+        if (streamChannel >= outputBuffer->mNumberChannels) {
+            ++streamNumber;
+            outputBuffer = &destination->mBuffers[streamNumber];
+            coreAudioBuffer = outputBuffer->mData;
+            streamChannel = 0;
+        }
+        source->outputConverter(coreAudioBuffer, outputBuffer->mNumberChannels, portAudioBuffer, frameSpacing, frameCount, NULL);
+        coreAudioBuffer += sizeof(Float32);
+        portAudioBuffer += Pa_GetSampleSize(source->outputSampleFormat) * channelSpacing;
+    }
+    return noErr;
+}
+
+static OSStatus AudioIOProc( AudioDeviceID inDevice,
+                      const AudioTimeStamp* inNow,
+                      const AudioBufferList* inInputData,
+                      const AudioTimeStamp* inInputTime,
+                      AudioBufferList* outOutputData, 
+                      const AudioTimeStamp* inOutputTime,
+                      void* inClientData)
+{
+    PaMacClientData *clientData = (PaMacClientData *)inClientData;
+    PaStreamCallbackTimeInfo *timeInfo = InitializeTimeInfo(inNow, inInputTime, inOutputTime);
+    
+    PaUtil_BeginCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer );
+    
+    AudioBuffer *outputBuffer = &outOutputData->mBuffers[0];
+    unsigned long frameCount = outputBuffer->mDataByteSize / (outputBuffer->mNumberChannels * sizeof(Float32));
+
+    if (clientData->inputBuffer) {
+        CopyInputData(clientData, inInputData, frameCount);
+    }
+    PaStreamCallbackResult result = clientData->callback(clientData->inputBuffer, clientData->outputBuffer, frameCount, timeInfo, paNoFlag, clientData->userData);
+    if (clientData->outputBuffer) {
+        CopyOutputData(outOutputData, clientData, frameCount);
+    }
+
+    PaUtil_EndCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer, frameCount );
+    
+    if (result == paComplete || result == paAbort) {
+        Pa_StopStream(clientData->stream);
+    }
+
+    PaUtil_FreeMemory( timeInfo );
+    return noErr;
+}
+
+// This is not for input-only streams, this is for streams where the input device is different from the output device
+static OSStatus AudioInputProc( AudioDeviceID inDevice,
+                         const AudioTimeStamp* inNow,
+                         const AudioBufferList* inInputData,
+                         const AudioTimeStamp* inInputTime,
+                         AudioBufferList* outOutputData, 
+                         const AudioTimeStamp* inOutputTime,
+                         void* inClientData)
+{
+    PaMacClientData *clientData = (PaMacClientData *)inClientData;
+    PaStreamCallbackTimeInfo *timeInfo = InitializeTimeInfo(inNow, inInputTime, inOutputTime);
+
+    PaUtil_BeginCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer );
+
+    AudioBuffer const *inputBuffer = &inInputData->mBuffers[0];
+    unsigned long frameCount = inputBuffer->mDataByteSize / (inputBuffer->mNumberChannels * sizeof(Float32));
+
+    CopyInputData(clientData, inInputData, frameCount);
+    PaStreamCallbackResult result = clientData->callback(clientData->inputBuffer, clientData->outputBuffer, frameCount, timeInfo, paNoFlag, clientData->userData);
+    
+    PaUtil_EndCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer, frameCount );
+    if( result == paComplete || result == paAbort )
+       Pa_StopStream(clientData->stream);
+    PaUtil_FreeMemory( timeInfo );
+    return noErr;
+}
+
+// This is not for output-only streams, this is for streams where the input device is different from the output device
+static OSStatus AudioOutputProc( AudioDeviceID inDevice,
+                          const AudioTimeStamp* inNow,
+                          const AudioBufferList* inInputData,
+                          const AudioTimeStamp* inInputTime,
+                          AudioBufferList* outOutputData, 
+                          const AudioTimeStamp* inOutputTime,
+                          void* inClientData)
+{
+    PaMacClientData *clientData = (PaMacClientData *)inClientData;
+    //PaStreamCallbackTimeInfo *timeInfo = InitializeTimeInfo(inNow, inInputTime, inOutputTime);
+
+    PaUtil_BeginCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer );
+
+    AudioBuffer *outputBuffer = &outOutputData->mBuffers[0];
+    unsigned long frameCount = outputBuffer->mDataByteSize / (outputBuffer->mNumberChannels * sizeof(Float32));
+
+    //clientData->callback(NULL, clientData->outputBuffer, frameCount, timeInfo, paNoFlag, clientData->userData);
+
+    CopyOutputData(outOutputData, clientData, frameCount);
+
+    PaUtil_EndCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer, frameCount );
+    return noErr;
+}
+
+static PaError SetSampleRate(AudioDeviceID device, double sampleRate, int isInput)
+{
+    PaError result = paNoError;
+    
+    double actualSampleRate;
+    UInt32 propSize = sizeof(double);
+    result = conv_err(AudioDeviceSetProperty(device, NULL, 0, isInput, kAudioDevicePropertyNominalSampleRate, propSize, &sampleRate));
+    
+    result = conv_err(AudioDeviceGetProperty(device, 0, isInput, kAudioDevicePropertyNominalSampleRate, &propSize, &actualSampleRate));
+    
+    if (result == paNoError && actualSampleRate != sampleRate) {
+        result = paInvalidSampleRate;
+    }
+    
+    return result;    
+}
+
+static PaError SetFramesPerBuffer(AudioDeviceID device, unsigned long framesPerBuffer, int isInput)
+{
+    PaError result = paNoError;
+    UInt32 preferredFramesPerBuffer = framesPerBuffer;
+    //    while (preferredFramesPerBuffer > UINT32_MAX) {
+    //        preferredFramesPerBuffer /= 2;
+    //    }
+    
+    UInt32 actualFramesPerBuffer;
+    UInt32 propSize = sizeof(UInt32);
+    result = conv_err(AudioDeviceSetProperty(device, NULL, 0, isInput, kAudioDevicePropertyBufferFrameSize, propSize, &preferredFramesPerBuffer));
+    
+    result = conv_err(AudioDeviceGetProperty(device, 0, isInput, kAudioDevicePropertyBufferFrameSize, &propSize, &actualFramesPerBuffer));
+    
+    if (result != paNoError) {
+        // do nothing
+    }
+    else if (actualFramesPerBuffer > framesPerBuffer) {
+        result = paBufferTooSmall;
+    }
+    else if (actualFramesPerBuffer < framesPerBuffer) {
+        result = paBufferTooBig;
+    }
+    
+    return result;    
+}
+    
+static PaError SetUpUnidirectionalStream(AudioDeviceID device, double sampleRate, unsigned long framesPerBuffer, int isInput)
+{
+    PaError err = paNoError;
+    err = SetSampleRate(device, sampleRate, isInput);
+    if( err == paNoError )
+        err = SetFramesPerBuffer(device, framesPerBuffer, isInput);
+    return err;
+}
+
+// =====  PortAudio functions  =====
+#pragma mark PortAudio functions
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+    
+    PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
+    
+#ifdef __cplusplus
+}
+#endif // __cplusplus
+
+static void Terminate( struct PaUtilHostApiRepresentation *hostApi )
+{
+    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation*)hostApi;
+    
+    CleanUp(macCoreHostApi);
+}
+
+static PaError IsFormatSupported( struct PaUtilHostApiRepresentation *hostApi,
+                                  const PaStreamParameters *inputParameters,
+                                  const PaStreamParameters *outputParameters,
+                                  double sampleRate )
+{
+    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation*)hostApi;
+    PaDeviceInfo *deviceInfo;
+    
+    PaError result = paNoError;
+    if (inputParameters) {
+        deviceInfo = macCoreHostApi->inheritedHostApiRep.deviceInfos[inputParameters->device];
+        if (inputParameters->channelCount > deviceInfo->maxInputChannels)
+            result = paInvalidChannelCount;
+        else if (CheckFormat(macCoreHostApi->macCoreDeviceIds[inputParameters->device], inputParameters, sampleRate, 1) != kAudioHardwareNoError) {
+            result = paInvalidSampleRate;
+        }
+    }
+    if (outputParameters && result == paNoError) {
+        deviceInfo = macCoreHostApi->inheritedHostApiRep.deviceInfos[outputParameters->device];
+        if (outputParameters->channelCount > deviceInfo->maxOutputChannels)
+            result = paInvalidChannelCount;
+        else if (CheckFormat(macCoreHostApi->macCoreDeviceIds[outputParameters->device], outputParameters, sampleRate, 0) != kAudioHardwareNoError) {
+            result = paInvalidSampleRate;
+        }
+    }
+
+    return result;
+}
+
+static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
+                           PaStream** s,
+                           const PaStreamParameters *inputParameters,
+                           const PaStreamParameters *outputParameters,
+                           double sampleRate,
+                           unsigned long framesPerBuffer,
+                           PaStreamFlags streamFlags,
+                           PaStreamCallback *streamCallback,
+                           void *userData )
+{
+    PaError err = paNoError;
+    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation *)hostApi;
+    PaMacCoreStream *stream = PaUtil_AllocateMemory(sizeof(PaMacCoreStream));
+    stream->isActive = 0;
+    stream->isStopped = 1;
+    stream->inputDevice = kAudioDeviceUnknown;
+    stream->outputDevice = kAudioDeviceUnknown;
+    
+    PaUtil_InitializeStreamRepresentation( &stream->streamRepresentation,
+                                           ( (streamCallback)
+                                             ? &macCoreHostApi->callbackStreamInterface
+                                             : &macCoreHostApi->blockingStreamInterface ),
+                                           streamCallback, userData );
+    PaUtil_InitializeCpuLoadMeasurer( &stream->cpuLoadMeasurer, sampleRate );
+    
+    *s = (PaStream*)stream;
+    PaMacClientData *clientData = PaUtil_AllocateMemory(sizeof(PaMacClientData));
+    clientData->stream = stream;
+    clientData->callback = streamCallback;
+    clientData->userData = userData;
+    clientData->inputBuffer = 0;
+    clientData->outputBuffer = 0;
+    clientData->ditherGenerator = PaUtil_AllocateMemory(sizeof(PaUtilTriangularDitherGenerator));
+    PaUtil_InitializeTriangularDitherState(clientData->ditherGenerator);
+    
+    if (inputParameters != NULL) {
+        stream->inputDevice = macCoreHostApi->macCoreDeviceIds[inputParameters->device];
+        clientData->inputConverter = PaUtil_SelectConverter(paFloat32, inputParameters->sampleFormat, streamFlags);
+        clientData->inputBuffer = PaUtil_AllocateMemory(Pa_GetSampleSize(inputParameters->sampleFormat) * framesPerBuffer * inputParameters->channelCount);
+        clientData->inputChannelCount = inputParameters->channelCount;
+        clientData->inputSampleFormat = inputParameters->sampleFormat;
+        err = SetUpUnidirectionalStream(stream->inputDevice, sampleRate, framesPerBuffer, 1);
+    }
+    
+    if (err == paNoError && outputParameters != NULL) {
+        stream->outputDevice = macCoreHostApi->macCoreDeviceIds[outputParameters->device];
+        clientData->outputConverter = PaUtil_SelectConverter(outputParameters->sampleFormat, paFloat32, streamFlags);
+        clientData->outputBuffer = PaUtil_AllocateMemory(Pa_GetSampleSize(outputParameters->sampleFormat) * framesPerBuffer * outputParameters->channelCount);
+        clientData->outputChannelCount = outputParameters->channelCount;
+        clientData->outputSampleFormat = outputParameters->sampleFormat;
+        err = SetUpUnidirectionalStream(stream->outputDevice, sampleRate, framesPerBuffer, 0);
+    }
+
+    if (inputParameters == NULL || outputParameters == NULL || stream->inputDevice == stream->outputDevice) {
+        AudioDeviceID device = (inputParameters == NULL) ? stream->outputDevice : stream->inputDevice;
+
+        AudioDeviceAddIOProc(device, AudioIOProc, clientData);
+    }
+    else {
+        // using different devices for input and output
+        AudioDeviceAddIOProc(stream->inputDevice, AudioInputProc, clientData);
+        AudioDeviceAddIOProc(stream->outputDevice, AudioOutputProc, clientData);
+    }
+    
+    return err;
+}
+
+// Note: When CloseStream() is called, the multi-api layer ensures that the stream has already been stopped or aborted.
+static PaError CloseStream( PaStream* s )
+{
+    PaError err = paNoError;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+
+    PaUtil_ResetCpuLoadMeasurer( &stream->cpuLoadMeasurer );
+
+    if (stream->inputDevice != kAudioDeviceUnknown) {
+        if (stream->outputDevice == kAudioDeviceUnknown || stream->outputDevice == stream->inputDevice) {
+            err = conv_err(AudioDeviceRemoveIOProc(stream->inputDevice, AudioIOProc));
+        }
+        else {
+            err = conv_err(AudioDeviceRemoveIOProc(stream->inputDevice, AudioInputProc));
+            err = conv_err(AudioDeviceRemoveIOProc(stream->outputDevice, AudioOutputProc));
+        }
+    }
+    else {
+        err = conv_err(AudioDeviceRemoveIOProc(stream->outputDevice, AudioIOProc));
+    }
+    
+    return err;
+}
+
+
+static PaError StartStream( PaStream *s )
+{
+    PaError err = paNoError;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+
+    if (stream->inputDevice != kAudioDeviceUnknown) {
+        if (stream->outputDevice == kAudioDeviceUnknown || stream->outputDevice == stream->inputDevice) {
+            err = conv_err(AudioDeviceStart(stream->inputDevice, AudioIOProc));
+        }
+        else {
+            err = conv_err(AudioDeviceStart(stream->inputDevice, AudioInputProc));
+            err = conv_err(AudioDeviceStart(stream->outputDevice, AudioOutputProc));
+        }
+    }
+    else {
+        err = conv_err(AudioDeviceStart(stream->outputDevice, AudioIOProc));
+    }
+    
+    stream->isActive = 1;
+    stream->isStopped = 0;
+    return err;
+}
+
+static PaError AbortStream( PaStream *s )
+{
+    PaError err = paNoError;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    
+    if (stream->inputDevice != kAudioDeviceUnknown) {
+        if (stream->outputDevice == kAudioDeviceUnknown || stream->outputDevice == stream->inputDevice) {
+            err = conv_err(AudioDeviceStop(stream->inputDevice, AudioIOProc));
+        }
+        else {
+            err = conv_err(AudioDeviceStop(stream->inputDevice, AudioInputProc));
+            err = conv_err(AudioDeviceStop(stream->outputDevice, AudioOutputProc));
+        }
+    }
+    else {
+        err = conv_err(AudioDeviceStop(stream->outputDevice, AudioIOProc));
+    }
+    
+    stream->isActive = 0;
+    stream->isStopped = 1;
+    return err;
+}    
+
+static PaError StopStream( PaStream *s )
+{
+    // TODO: this should be nicer than abort
+    return AbortStream(s);
+}
+
+static PaError IsStreamStopped( PaStream *s )
+{
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    
+    return stream->isStopped;
+}
+
+
+static PaError IsStreamActive( PaStream *s )
+{
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+
+    return stream->isActive;
+}
+
+
+static PaTime GetStreamTime( PaStream *s )
+{
+    OSStatus err;
+    PaTime result;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+
+    AudioTimeStamp *timeStamp = PaUtil_AllocateMemory(sizeof(AudioTimeStamp));
+    if (stream->inputDevice != kAudioDeviceUnknown) {
+        err = AudioDeviceGetCurrentTime(stream->inputDevice, timeStamp);
+    }
+    else {
+        err = AudioDeviceGetCurrentTime(stream->outputDevice, timeStamp);
+    }
+    
+    result = err ? 0 : timeStamp->mSampleTime;
+    PaUtil_FreeMemory(timeStamp);
+
+    return result;
+}
+
+
+static double GetStreamCpuLoad( PaStream* s )
+{
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    
+    return PaUtil_GetCpuLoad( &stream->cpuLoadMeasurer );
+}
+
+
+// As separate stream interfaces are used for blocking and callback streams, the following functions can be guaranteed to only be called for blocking streams.
+
+static PaError ReadStream( PaStream* s,
+                           void *buffer,
+                           unsigned long frames )
+{
+    return paInternalError;
+}
+
+
+static PaError WriteStream( PaStream* s,
+                            const void *buffer,
+                            unsigned long frames )
+{
+    return paInternalError;
+}
+
+
+static signed long GetStreamReadAvailable( PaStream* s )
+{
+    return paInternalError;
+}
+
+
+static signed long GetStreamWriteAvailable( PaStream* s )
+{
+    return paInternalError;
+}
+
+// HostAPI-specific initialization function
+PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex hostApiIndex )
+{
+    PaError result = paNoError;
+    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation *)PaUtil_AllocateMemory( sizeof(PaMacCoreHostApiRepresentation) );
+    if( !macCoreHostApi )
+    {
+        result = paInsufficientMemory;
+        goto error;
+    }
+    
+    macCoreHostApi->allocations = PaUtil_CreateAllocationGroup();
+    if( !macCoreHostApi->allocations )
+    {
+        result = paInsufficientMemory;
+        goto error;
+    }
+    
+    *hostApi = &macCoreHostApi->inheritedHostApiRep;
+    (*hostApi)->info.structVersion = 1;
+    (*hostApi)->info.type = paCoreAudio;
+    (*hostApi)->info.name = "CoreAudio";
+
+    result = InitializeDeviceInfos(macCoreHostApi, hostApiIndex);
+    if (result != paNoError) {
+        goto error;
+    }
+    
+    // Set up the proper callbacks to this HostApi's functions
+    (*hostApi)->Terminate = Terminate;
+    (*hostApi)->OpenStream = OpenStream;
+    (*hostApi)->IsFormatSupported = IsFormatSupported;
+    
+    PaUtil_InitializeStreamInterface( &macCoreHostApi->callbackStreamInterface, CloseStream, StartStream,
+                                      StopStream, AbortStream, IsStreamStopped, IsStreamActive,
+                                      GetStreamTime, GetStreamCpuLoad,
+                                      PaUtil_DummyRead, PaUtil_DummyWrite,
+                                      PaUtil_DummyGetReadAvailable, PaUtil_DummyGetWriteAvailable );
+    
+    PaUtil_InitializeStreamInterface( &macCoreHostApi->blockingStreamInterface, CloseStream, StartStream,
+                                      StopStream, AbortStream, IsStreamStopped, IsStreamActive,
+                                      GetStreamTime, PaUtil_DummyGetCpuLoad,
+                                      ReadStream, WriteStream, GetStreamReadAvailable, GetStreamWriteAvailable );
+    
+    return result;
+    
+error:
+        if( macCoreHostApi ) {
+            CleanUp(macCoreHostApi);
+        }
+    
+    return result;
+}

3rdparty/iaxclient/lib/portaudio/src/hostapi/coreaudio/pa_mac_core_blocking.c

@@ -0,0 +1,512 @@
+/* This file contains the implementation
+ * required for blocking I/O. It is separated from pa_mac_core.c simply to ease
+ * development. */
+
+#include "pa_mac_core_blocking.h"
+#include "pa_mac_core_internal.h"
+#include <assert.h>
+//#include <libkern/OSAtomic.h>
+#include <CoreServices/CoreServices.h>
+
+/*
+ * This fnuction determines the size of a particular sample format.
+ * if the format is not recognized, this returns zero.
+ */
+static size_t computeSampleSizeFromFormat( PaSampleFormat format )
+{
+   switch( format ) {
+   case paFloat32: return 4;
+   case paInt32: return 4;
+   case paInt24: return 3;
+   case paInt16: return 2;
+   case paInt8: case paUInt8: return 1;
+   default: return 0;
+   }
+}
+
+
+/*
+ * Functions for initializing, resetting, and destroying BLIO structures.
+ *
+ */
+
+/* This should be called with the relevant info when initializing a stream for
+   callback. */
+PaError initializeBlioRingBuffers(
+                                       PaMacBlio *blio,
+                                       PaSampleFormat inputSampleFormat,
+                                       PaSampleFormat outputSampleFormat,
+                                       size_t framesPerBuffer,
+                                       long ringBufferSize,
+                                       int inChan,
+                                       int outChan )
+{
+   void *data;
+   int result;
+
+   /* zeroify things */
+   bzero( blio, sizeof( PaMacBlio ) );
+   /* this is redundant, but the buffers are used to check
+      if the bufffers have been initialized, so we do it explicitly. */
+   blio->inputRingBuffer.buffer = NULL;
+   blio->outputRingBuffer.buffer = NULL;
+
+   /* initialize simple data */
+   blio->inputSampleFormat = inputSampleFormat;
+   blio->inputSampleSize = computeSampleSizeFromFormat(inputSampleFormat);
+   blio->outputSampleFormat = outputSampleFormat;
+   blio->outputSampleSize = computeSampleSizeFromFormat(outputSampleFormat);
+   blio->framesPerBuffer = framesPerBuffer;
+   blio->inChan = inChan;
+   blio->outChan = outChan;
+   blio->statusFlags = 0;
+   blio->errors = paNoError;
+#ifdef PA_MAC_BLIO_MUTEX
+   blio->isInputEmpty = false;
+   blio->isOutputFull = false;
+#endif
+
+   /* setup ring buffers */
+#ifdef PA_MAC_BLIO_MUTEX
+   result = PaMacCore_SetUnixError( pthread_mutex_init(&(blio->inputMutex),NULL), 0 );
+   if( result )
+      goto error;
+   result = UNIX_ERR( pthread_cond_init( &(blio->inputCond), NULL ) );
+   if( result )
+      goto error;
+   result = UNIX_ERR( pthread_mutex_init(&(blio->outputMutex),NULL) );
+   if( result )
+      goto error;
+   result = UNIX_ERR( pthread_cond_init( &(blio->outputCond), NULL ) );
+#endif
+   if( inChan ) {
+      data = calloc( ringBufferSize, blio->inputSampleSize );
+      if( !data )
+      {
+         result = paInsufficientMemory;
+         goto error;
+      }
+
+      assert( 0 == RingBuffer_Init(
+            &blio->inputRingBuffer,
+            ringBufferSize*blio->inputSampleSize,
+            data ) );
+   }
+   if( outChan ) {
+      data = calloc( ringBufferSize, blio->outputSampleSize );
+      if( !data )
+      {
+         result = paInsufficientMemory;
+         goto error;
+      }
+
+      assert( 0 == RingBuffer_Init(
+            &blio->outputRingBuffer,
+            ringBufferSize*blio->outputSampleSize,
+            data ) );
+   }
+
+   result = resetBlioRingBuffers( blio );
+   if( result )
+      goto error;
+
+   return 0;
+
+ error:
+   destroyBlioRingBuffers( blio );
+   return result;
+}
+
+#ifdef PA_MAC_BLIO_MUTEX
+PaError blioSetIsInputEmpty( PaMacBlio *blio, bool isEmpty )
+{
+   PaError result = paNoError;
+   if( isEmpty == blio->isInputEmpty )
+      goto done;
+
+   /* we need to update the value. Here's what we do:
+    * - Lock the mutex, so noone else can write.
+    * - update the value.
+    * - unlock.
+    * - broadcast to all listeners.
+    */
+   result = UNIX_ERR( pthread_mutex_lock( &blio->inputMutex ) );
+   if( result )
+      goto done;
+   blio->isInputEmpty = isEmpty;
+   result = UNIX_ERR( pthread_mutex_unlock( &blio->inputMutex ) );
+   if( result )
+      goto done;
+   result = UNIX_ERR( pthread_cond_broadcast( &blio->inputCond ) );
+   if( result )
+      goto done;
+
+ done:
+   return result;
+}
+PaError blioSetIsOutputFull( PaMacBlio *blio, bool isFull )
+{
+   PaError result = paNoError;
+   if( isFull == blio->isOutputFull )
+      goto done;
+
+   /* we need to update the value. Here's what we do:
+    * - Lock the mutex, so noone else can write.
+    * - update the value.
+    * - unlock.
+    * - broadcast to all listeners.
+    */
+   result = UNIX_ERR( pthread_mutex_lock( &blio->outputMutex ) );
+   if( result )
+      goto done;
+   blio->isOutputFull = isFull;
+   result = UNIX_ERR( pthread_mutex_unlock( &blio->outputMutex ) );
+   if( result )
+      goto done;
+   result = UNIX_ERR( pthread_cond_broadcast( &blio->outputCond ) );
+   if( result )
+      goto done;
+
+ done:
+   return result;
+}
+#endif
+
+/* This should be called after stopping or aborting the stream, so that on next
+   start, the buffers will be ready. */
+PaError resetBlioRingBuffers( PaMacBlio *blio )
+{
+#ifdef PA_MAC__BLIO_MUTEX
+   int result;
+#endif
+   blio->statusFlags = 0;
+   if( blio->outputRingBuffer.buffer ) {
+      RingBuffer_Flush( &blio->outputRingBuffer );
+      bzero( blio->outputRingBuffer.buffer,
+             blio->outputRingBuffer.bufferSize );
+      /* Advance buffer */
+      RingBuffer_AdvanceWriteIndex( &blio->outputRingBuffer, blio->outputRingBuffer.bufferSize );
+
+      /* Update isOutputFull. */
+#ifdef PA_MAC__BLIO_MUTEX
+      result = blioSetIsOutputFull( blio, toAdvance == blio->outputRingBuffer.bufferSize );
+      if( result )
+         goto error;
+#endif
+/*
+      printf( "------%d\n" ,  blio->framesPerBuffer );
+      printf( "------%d\n" ,  blio->outChan );
+      printf( "------%d\n" ,  blio->outputSampleSize );
+      printf( "------%d\n" ,  blio->framesPerBuffer*blio->outChan*blio->outputSampleSize );
+*/
+   }
+   if( blio->inputRingBuffer.buffer ) {
+      RingBuffer_Flush( &blio->inputRingBuffer );
+      bzero( blio->inputRingBuffer.buffer,
+             blio->inputRingBuffer.bufferSize );
+      /* Update isInputEmpty. */
+#ifdef PA_MAC__BLIO_MUTEX
+      result = blioSetIsInputEmpty( blio, true );
+      if( result )
+         goto error;
+#endif
+   }
+   return paNoError;
+#ifdef PA_MAC__BLIO_MUTEX
+ error:
+   return result;
+#endif
+}
+
+/*This should be called when you are done with the blio. It can safely be called
+  multiple times if there are no exceptions. */
+PaError destroyBlioRingBuffers( PaMacBlio *blio )
+{
+   PaError result = paNoError;
+   if( blio->inputRingBuffer.buffer ) {
+      free( blio->inputRingBuffer.buffer );
+#ifdef PA_MAC__BLIO_MUTEX
+      result = UNIX_ERR( pthread_mutex_destroy( & blio->inputMutex ) );
+      if( result ) return result;
+      result = UNIX_ERR( pthread_cond_destroy( & blio->inputCond ) );
+      if( result ) return result;
+#endif
+   }
+   blio->inputRingBuffer.buffer = NULL;
+   if( blio->outputRingBuffer.buffer ) {
+      free( blio->outputRingBuffer.buffer );
+#ifdef PA_MAC__BLIO_MUTEX
+      result = UNIX_ERR( pthread_mutex_destroy( & blio->outputMutex ) );
+      if( result ) return result;
+      result = UNIX_ERR( pthread_cond_destroy( & blio->outputCond ) );
+      if( result ) return result;
+#endif
+   }
+   blio->outputRingBuffer.buffer = NULL;
+
+   return result;
+}
+
+/*
+ * this is the BlioCallback function. It expects to recieve a PaMacBlio Object
+ * pointer as userData.
+ *
+ */
+int BlioCallback( const void *input, void *output, unsigned long frameCount,
+	const PaStreamCallbackTimeInfo* timeInfo,
+        PaStreamCallbackFlags statusFlags,
+        void *userData )
+{
+   PaMacBlio *blio = (PaMacBlio*)userData;
+   long avail;
+   long toRead;
+   long toWrite;
+
+   /* set flags returned by OS: */
+   // Mihai: replacing OSAtomic32 with BitOrAtomic 
+   //OSAtomicOr32( statusFlags, &blio->statusFlags ) ;
+   BitOrAtomic( statusFlags, &blio->statusFlags );
+   
+   /* --- Handle Input Buffer --- */
+   if( blio->inChan ) {
+      avail = RingBuffer_GetWriteAvailable( &blio->inputRingBuffer );
+
+      /* check for underflow */
+      if( avail < frameCount * blio->inputSampleSize * blio->inChan )
+      	 // Mihai: replacing OSAtomic32 with BitOrAtomic
+         //OSAtomicOr32( paInputOverflow, &blio->statusFlags );
+         BitOrAtomic( paInputOverflow, &blio->statusFlags );
+
+      toRead = MIN( avail, frameCount * blio->inputSampleSize * blio->inChan );
+
+      /* copy the data */
+      /*printf( "reading %d\n", toRead );*/
+      assert( toRead == RingBuffer_Write( &blio->inputRingBuffer, input, toRead ) );
+#ifdef PA_MAC__BLIO_MUTEX
+      /* Priority inversion. See notes below. */
+      blioSetIsInputEmpty( blio, false );
+#endif
+   }
+
+
+   /* --- Handle Output Buffer --- */
+   if( blio->outChan ) {
+      avail = RingBuffer_GetReadAvailable( &blio->outputRingBuffer );
+
+      /* check for underflow */
+      if( avail < frameCount * blio->outputSampleSize * blio->outChan )
+         //Mihai: replacing OSAtomic32 with BitOrAtomic
+         //OSAtomicOr32( paOutputUnderflow, &blio->statusFlags );
+         BitOrAtomic( paOutputUnderflow, &blio->statusFlags );
+
+      toWrite = MIN( avail, frameCount * blio->outputSampleSize * blio->outChan );
+
+      if( toWrite != frameCount * blio->outputSampleSize * blio->outChan )
+         bzero( ((char *)output)+toWrite,
+                frameCount * blio->outputSampleSize * blio->outChan - toWrite );
+      /* copy the data */
+      /*printf( "writing %d\n", toWrite );*/
+      assert( toWrite == RingBuffer_Read( &blio->outputRingBuffer, output, toWrite ) );
+#ifdef PA_MAC__BLIO_MUTEX
+      /* We have a priority inversion here. However, we will only have to
+         wait if this was true and is now false, which means we've got
+         some room in the buffer.
+         Hopefully problems will be minimized. */
+      blioSetIsOutputFull( blio, false );
+#endif
+   }
+
+   return paContinue;
+}
+
+PaError ReadStream( PaStream* stream,
+                           void *buffer,
+                           unsigned long frames )
+{
+    PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
+    char *cbuf = (char *) buffer;
+    PaError ret = paNoError;
+    VVDBUG(("ReadStream()\n"));
+
+    while( frames > 0 ) {
+       long avail;
+       long toRead;
+       do {
+          avail = RingBuffer_GetReadAvailable( &blio->inputRingBuffer );
+/*
+          printf( "Read Buffer is %%%g full: %ld of %ld.\n",
+                  100 * (float)avail / (float) blio->inputRingBuffer.bufferSize,
+                  avail, blio->inputRingBuffer.bufferSize );
+*/
+          if( avail == 0 ) {
+#ifdef PA_MAC_BLIO_MUTEX
+             /**block when empty*/
+             ret = UNIX_ERR( pthread_mutex_lock( &blio->inputMutex ) );
+             if( ret )
+                return ret;
+             while( blio->isInputEmpty ) {
+                ret = UNIX_ERR( pthread_cond_wait( &blio->inputCond, &blio->inputMutex ) );
+                if( ret )
+                   return ret;
+             }
+             ret = UNIX_ERR( pthread_mutex_unlock( &blio->inputMutex ) );
+             if( ret )
+                return ret;
+#else
+             Pa_Sleep( PA_MAC_BLIO_BUSY_WAIT_SLEEP_INTERVAL );
+#endif
+          }
+       } while( avail == 0 );
+       toRead = MIN( avail, frames * blio->inputSampleSize * blio->inChan );
+       toRead -= toRead % blio->inputSampleSize * blio->inChan ;
+       RingBuffer_Read( &blio->inputRingBuffer, (void *)cbuf, toRead );
+       cbuf += toRead;
+       frames -= toRead / ( blio->inputSampleSize * blio->inChan );
+
+       if( toRead == avail ) {
+#ifdef PA_MAC_BLIO_MUTEX
+          /* we just emptied the buffer, so we need to mark it as empty. */
+          ret = blioSetIsInputEmpty( blio, true );
+          if( ret )
+             return ret;
+          /* of course, in the meantime, the callback may have put some sats
+             in, so
+             so check for that, too, to avoid a race condition. */
+          if( RingBuffer_GetReadAvailable( &blio->inputRingBuffer ) ) {
+             blioSetIsInputEmpty( blio, false );
+             if( ret )
+                return ret;
+          }
+#endif
+       }
+    }
+
+    /*   Report either paNoError or paInputOverflowed. */
+    /*   may also want to report other errors, but this is non-standard. */
+    ret = blio->statusFlags & paInputOverflow;
+
+    /* report underflow only once: */
+    if( ret ) {
+       //Mihai: replacing OSAtomicAnd32 with BitAndAtomic
+       //OSAtomicAnd32( ~paInputOverflow, &blio->statusFlags );
+       BitAndAtomic( ~paInputOverflow, &blio->statusFlags );
+       ret = paInputOverflowed;
+    }
+
+    return ret;
+}
+
+
+PaError WriteStream( PaStream* stream,
+                            const void *buffer,
+                            unsigned long frames )
+{
+    PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
+    char *cbuf = (char *) buffer;
+    PaError ret = paNoError;
+    VVDBUG(("WriteStream()\n"));
+
+    while( frames > 0 ) {
+       long avail = 0;
+       long toWrite;
+
+       do {
+          avail = RingBuffer_GetWriteAvailable( &blio->outputRingBuffer );
+/*
+          printf( "Write Buffer is %%%g full: %ld of %ld.\n",
+                  100 - 100 * (float)avail / (float) blio->outputRingBuffer.bufferSize,
+                  avail, blio->outputRingBuffer.bufferSize );
+*/
+          if( avail == 0 ) {
+#ifdef PA_MAC_BLIO_MUTEX
+             /*block while full*/
+             ret = UNIX_ERR( pthread_mutex_lock( &blio->outputMutex ) );
+             if( ret )
+                return ret;
+             while( blio->isOutputFull ) {
+                ret = UNIX_ERR( pthread_cond_wait( &blio->outputCond, &blio->outputMutex ) );
+                if( ret )
+                   return ret;
+             }
+             ret = UNIX_ERR( pthread_mutex_unlock( &blio->outputMutex ) );
+             if( ret )
+                return ret;
+#else
+             Pa_Sleep( PA_MAC_BLIO_BUSY_WAIT_SLEEP_INTERVAL );
+#endif
+          }
+       } while( avail == 0 );
+
+       toWrite = MIN( avail, frames * blio->outputSampleSize * blio->outChan );
+       toWrite -= toWrite % blio->outputSampleSize * blio->outChan ;
+       RingBuffer_Write( &blio->outputRingBuffer, (void *)cbuf, toWrite );
+       cbuf += toWrite;
+       frames -= toWrite / ( blio->outputSampleSize * blio->outChan );
+
+#ifdef PA_MAC_BLIO_MUTEX
+       if( toWrite == avail ) {
+          /* we just filled up the buffer, so we need to mark it as filled. */
+          ret = blioSetIsOutputFull( blio, true );
+          if( ret )
+             return ret;
+          /* of course, in the meantime, we may have emptied the buffer, so
+             so check for that, too, to avoid a race condition. */
+          if( RingBuffer_GetWriteAvailable( &blio->outputRingBuffer ) ) {
+             blioSetIsOutputFull( blio, false );
+             if( ret )
+                return ret;
+          }
+       }
+#endif
+    }
+
+    /*   Report either paNoError or paOutputUnderflowed. */
+    /*   may also want to report other errors, but this is non-standard. */
+    ret = blio->statusFlags & paOutputUnderflow;
+
+    /* report underflow only once: */
+    if( ret ) {
+    	//Mihai: replacing OSAtomicAnd32 with BitAndAtomic
+      	//OSAtomicAnd32( ~paOutputUnderflow, &blio->statusFlags );
+      	BitAndAtomic( ~paOutputUnderflow, &blio->statusFlags );
+      	ret = paOutputUnderflowed;
+    }
+
+    return ret;
+}
+
+/*
+ *
+ */
+void waitUntilBlioWriteBufferIsFlushed( PaMacBlio *blio )
+{
+    if( blio->outputRingBuffer.buffer ) {
+       long avail = RingBuffer_GetWriteAvailable( &blio->outputRingBuffer );
+       while( avail != blio->outputRingBuffer.bufferSize ) {
+          if( avail == 0 )
+             Pa_Sleep( PA_MAC_BLIO_BUSY_WAIT_SLEEP_INTERVAL );
+          avail = RingBuffer_GetWriteAvailable( &blio->outputRingBuffer );
+       }
+    }
+}
+
+
+signed long GetStreamReadAvailable( PaStream* stream )
+{
+    PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
+    VVDBUG(("GetStreamReadAvailable()\n"));
+
+    return RingBuffer_GetReadAvailable( &blio->inputRingBuffer )
+                         / ( blio->outputSampleSize * blio->outChan );
+}
+
+
+signed long GetStreamWriteAvailable( PaStream* stream )
+{
+    PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
+    VVDBUG(("GetStreamWriteAvailable()\n"));
+
+    return RingBuffer_GetWriteAvailable( &blio->outputRingBuffer )
+                         / ( blio->outputSampleSize * blio->outChan );
+}
+

3rdparty/iaxclient/lib/portaudio/src/hostapi/coreaudio/pa_mac_core_blocking.h

@@ -0,0 +1,76 @@
+
+#ifndef PA_MAC_CORE_BLOCKING_H_
+#define PA_MAC_CORE_BLOCKING_H_
+
+#include "ringbuffer.h"
+#include "portaudio.h"
+#include "pa_mac_core_utilities.h"
+
+/*
+ * Number of miliseconds to busy wait whil waiting for data in blocking calls.
+ */
+#define PA_MAC_BLIO_BUSY_WAIT_SLEEP_INTERVAL (5)
+/*
+ * Define exactly one of these blocking methods
+ * PA_MAC_BLIO_MUTEX is not actively maintained.
+ */
+#define PA_MAC_BLIO_BUSY_WAIT
+/*
+#define PA_MAC_BLIO_MUTEX
+*/
+
+typedef struct {
+    RingBuffer inputRingBuffer;
+    RingBuffer outputRingBuffer;
+    PaSampleFormat inputSampleFormat;
+    size_t inputSampleSize;
+    PaSampleFormat outputSampleFormat;
+    size_t outputSampleSize;
+
+    size_t framesPerBuffer;
+
+    int inChan;
+    int outChan;
+
+    //PaStreamCallbackFlags statusFlags;
+    uint32_t statusFlags;
+    PaError errors;
+
+    /* Here we handle blocking, using condition variables. */
+#ifdef  PA_MAC_BLIO_MUTEX
+    volatile bool isInputEmpty;
+    pthread_mutex_t inputMutex;
+    pthread_cond_t inputCond;
+
+    volatile bool isOutputFull;
+    pthread_mutex_t outputMutex;
+    pthread_cond_t outputCond;
+#endif
+}
+PaMacBlio;
+
+/*
+ * These functions operate on condition and related variables.
+ */
+
+PaError initializeBlioRingBuffers(
+                                       PaMacBlio *blio,
+                                       PaSampleFormat inputSampleFormat,
+                                       PaSampleFormat outputSampleFormat,
+                                       size_t framesPerBuffer,
+                                       long ringBufferSize,
+                                       int inChan,
+                                       int outChan );
+PaError destroyBlioRingBuffers( PaMacBlio *blio );
+PaError resetBlioRingBuffers( PaMacBlio *blio );
+
+int BlioCallback(
+        const void *input, void *output,
+        unsigned long frameCount,
+        const PaStreamCallbackTimeInfo* timeInfo,
+        PaStreamCallbackFlags statusFlags,
+        void *userData );
+
+void waitUntilBlioWriteBufferIsFlushed( PaMacBlio *blio );
+
+#endif /*PA_MAC_CORE_BLOCKING_H_*/

3rdparty/iaxclient/lib/portaudio/src/hostapi/coreaudio/pa_mac_core_internal.h

@@ -0,0 +1,151 @@
+/*
+ * This is the AUHAL implementation of portaudio.
+ *
+ * Written by Bjorn Roche of XO Audio LLC, from PA skeleton code.
+ * Portions copied from code by Dominic Mazzoni (who wrote a HAL implementation)
+ *
+ * Dominic's code was based on code by Phil Burk, Darren Gibbs,
+ * Gord Peters, Stephane Letz, and Greg Pfiel.
+ *
+ * Bjorn Roche and XO Audio LLC reserve no rights to this code.
+ * The maintainers of PortAudio may redistribute and modify the code and
+ * licenses as they deam appropriate.
+ *
+ * The following people also deserve acknowledgements:
+ *
+ * Olivier Tristan for feedback and testing
+ * Glenn Zelniker and Z-Systems engineering for sponsoring the Blocking I/O
+ * interface.
+ * 
+ *
+ * Based on the Open Source API proposed by Ross Bencina
+ * Copyright (c) 1999-2002 Ross Bencina, Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/**
+ @file pa_mac_core
+ @author Bjorn Roche
+ @brief AUHAL implementation of PortAudio
+*/
+
+#ifndef PA_MAC_CORE_INTERNAL_H__
+#define PA_MAC_CORE_INTERNAL_H__
+
+#include <AudioUnit/AudioUnit.h>
+#include <AudioToolbox/AudioToolbox.h>
+
+
+#include "portaudio.h"
+#include "pa_util.h"
+#include "pa_hostapi.h"
+#include "pa_stream.h"
+#include "pa_allocation.h"
+#include "pa_cpuload.h"
+#include "pa_process.h"
+#include "ringbuffer.h"
+
+#include "pa_mac_core_blocking.h"
+
+/* function prototypes */
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#define RING_BUFFER_ADVANCE_DENOMINATOR (4)
+
+PaError ReadStream( PaStream* stream, void *buffer, unsigned long frames );
+PaError WriteStream( PaStream* stream, const void *buffer, unsigned long frames );
+signed long GetStreamReadAvailable( PaStream* stream );
+signed long GetStreamWriteAvailable( PaStream* stream );
+/* PaMacAUHAL - host api datastructure specific to this implementation */
+typedef struct
+{
+    PaUtilHostApiRepresentation inheritedHostApiRep;
+    PaUtilStreamInterface callbackStreamInterface;
+    PaUtilStreamInterface blockingStreamInterface;
+
+    PaUtilAllocationGroup *allocations;
+
+    /* implementation specific data goes here */
+    long devCount;
+    AudioDeviceID *devIds; /*array of all audio devices*/
+    AudioDeviceID defaultIn;
+    AudioDeviceID defaultOut;
+}
+PaMacAUHAL;
+
+
+
+/* stream data structure specifically for this implementation */
+typedef struct PaMacCoreStream
+{
+    PaUtilStreamRepresentation streamRepresentation;
+    PaUtilCpuLoadMeasurer cpuLoadMeasurer;
+    PaUtilBufferProcessor bufferProcessor;
+
+    /* implementation specific data goes here */
+    bool bufferProcessorIsInitialized;
+    AudioUnit inputUnit;
+    AudioUnit outputUnit;
+    AudioDeviceID inputDevice;
+    AudioDeviceID outputDevice;
+    size_t userInChan;
+    size_t userOutChan;
+    size_t inputFramesPerBuffer;
+    size_t outputFramesPerBuffer;
+    PaMacBlio blio;
+    /* We use this ring buffer when input and out devs are different. */
+    RingBuffer inputRingBuffer;
+    /* We may need to do SR conversion on input. */
+    AudioConverterRef inputSRConverter;
+    /* We need to preallocate an inputBuffer for reading data. */
+    AudioBufferList inputAudioBufferList;
+    AudioTimeStamp startTime;
+    volatile PaStreamCallbackFlags xrunFlags;
+    volatile enum {
+       STOPPED          = 0, /* playback is completely stopped,
+                                and the user has called StopStream(). */
+       CALLBACK_STOPPED = 1, /* callback has requested stop,
+                                but user has not yet called StopStream(). */
+       STOPPING         = 2, /* The stream is in the process of closing.
+                                This state is just used internally;
+                                externally it is indistinguishable from
+                                ACTIVE.*/
+       ACTIVE           = 3  /* The stream is active and running. */
+    } state;
+    double sampleRate;
+}
+PaMacCoreStream;
+
+#endif /* PA_MAC_CORE_INTERNAL_H__ */

3rdparty/iaxclient/lib/portaudio/src/hostapi/coreaudio/pa_mac_core_utilities.c

@@ -0,0 +1,565 @@
+/*
+ *
+ * pa_mac_core_utilities.c
+ *
+ * utilitiy functions for pa_mac_core.c
+ *
+ * This functions are more like helper functions than part of the core,
+ *  so I moved them to a separate file so the core code would be cleaner.
+ *
+ * by Bjorn Roche.
+ */
+
+#include "pa_mac_core_utilities.h"
+
+PaError PaMacCore_SetUnixError( int err, int line )
+{
+   PaError ret;
+   const char *errorText;
+
+   if( err == 0 )
+   {
+      return paNoError;
+   }
+
+   ret = paNoError;
+   errorText = strerror( err );
+
+   /** Map Unix error to PaError. Pretty much the only one that maps
+       is ENOMEM. */
+   if( err == ENOMEM )
+      ret = paInsufficientMemory;
+   else
+      ret = paInternalError;
+
+   DBUG(("%d on line %d: msg='%s'\n", err, line, errorText));
+   PaUtil_SetLastHostErrorInfo( paCoreAudio, err, errorText );
+
+   return ret;
+}
+
+/*
+ * Translates MacOS generated errors into PaErrors
+ */
+PaError PaMacCore_SetError(OSStatus error, int line, int isError)
+{
+    /*FIXME: still need to handle possible ComponentResult values.*/
+    /*       unfortunately, they don't seem to be documented anywhere.*/
+    PaError result;
+    const char *errorType; 
+    const char *errorText;
+    
+    switch (error) {
+    case kAudioHardwareNoError:
+        return paNoError;
+    case kAudioHardwareNotRunningError:
+        errorText = "Audio Hardware Not Running";
+        result = paInternalError; break;
+    case kAudioHardwareUnspecifiedError: 
+        errorText = "Unspecified Audio Hardware Error";
+        result = paInternalError; break;
+    case kAudioHardwareUnknownPropertyError:
+        errorText = "Audio Hardware: Unknown Property";
+        result = paInternalError; break;
+    case kAudioHardwareBadPropertySizeError:
+        errorText = "Audio Hardware: Bad Property Size";
+        result = paInternalError; break;
+    case kAudioHardwareIllegalOperationError: 
+        errorText = "Audio Hardware: Illegal Operation";
+        result = paInternalError; break;
+    case kAudioHardwareBadDeviceError:
+        errorText = "Audio Hardware: Bad Device";
+        result = paInvalidDevice; break;
+    case kAudioHardwareBadStreamError:
+        errorText = "Audio Hardware: BadStream";
+        result = paBadStreamPtr; break;
+    case kAudioHardwareUnsupportedOperationError:
+        errorText = "Audio Hardware: Unsupported Operation";
+        result = paInternalError; break;
+    case kAudioDeviceUnsupportedFormatError:
+        errorText = "Audio Device: Unsupported Format";
+        result = paSampleFormatNotSupported; break;
+    case kAudioDevicePermissionsError:
+        errorText = "Audio Device: Permissions Error";
+        result = paDeviceUnavailable; break;
+    /* Audio Unit Errors: http://developer.apple.com/documentation/MusicAudio/Reference/CoreAudio/audio_units/chapter_5_section_3.html */
+    case kAudioUnitErr_InvalidProperty:
+        errorText = "Audio Unit: Invalid Property";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidParameter:
+        errorText = "Audio Unit: Invalid Parameter";
+        result = paInternalError; break;
+    case kAudioUnitErr_NoConnection:
+        errorText = "Audio Unit: No Connection";
+        result = paInternalError; break;
+    case kAudioUnitErr_FailedInitialization:
+        errorText = "Audio Unit: Initialization Failed";
+        result = paInternalError; break;
+    case kAudioUnitErr_TooManyFramesToProcess:
+        errorText = "Audio Unit: Too Many Frames";
+        result = paInternalError; break;
+    case kAudioUnitErr_IllegalInstrument:
+        errorText = "Audio Unit: Illegal Instrument";
+        result = paInternalError; break;
+    case kAudioUnitErr_InstrumentTypeNotFound:
+        errorText = "Audio Unit: Instrument Type Not Found";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidFile:
+        errorText = "Audio Unit: Invalid File";
+        result = paInternalError; break;
+    case kAudioUnitErr_UnknownFileType:
+        errorText = "Audio Unit: Unknown File Type";
+        result = paInternalError; break;
+    case kAudioUnitErr_FileNotSpecified:
+        errorText = "Audio Unit: File Not Specified";
+        result = paInternalError; break;
+    case kAudioUnitErr_FormatNotSupported:
+        errorText = "Audio Unit: Format Not Supported";
+        result = paInternalError; break;
+    case kAudioUnitErr_Uninitialized:
+        errorText = "Audio Unit: Unitialized";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidScope:
+        errorText = "Audio Unit: Invalid Scope";
+        result = paInternalError; break;
+    case kAudioUnitErr_PropertyNotWritable:
+        errorText = "Audio Unit: PropertyNotWritable";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidPropertyValue:
+        errorText = "Audio Unit: Invalid Property Value";
+        result = paInternalError; break;
+    case kAudioUnitErr_PropertyNotInUse:
+        errorText = "Audio Unit: Property Not In Use";
+        result = paInternalError; break;
+    case kAudioUnitErr_Initialized:
+        errorText = "Audio Unit: Initialized";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidOfflineRender:
+        errorText = "Audio Unit: Invalid Offline Render";
+        result = paInternalError; break;
+    case kAudioUnitErr_Unauthorized:
+        errorText = "Audio Unit: Unauthorized";
+        result = paInternalError; break;
+    case kAudioUnitErr_CannotDoInCurrentContext:
+        errorText = "Audio Unit: cannot do in current context";
+        result = paInternalError; break;
+    default:
+        errorText = "Unknown Error";
+        result = paInternalError;
+    }
+
+    if (isError)
+        errorType = "Error";
+    else
+        errorType = "Warning";
+
+    if ((int)error < -99999 || (int)error > 99999)
+        DBUG(("%s on line %d: err='%4s', msg='%s'\n", errorType, line, (const char *)&error, errorText));
+    else
+        DBUG(("%s on line %d: err=%d, 0x%x, msg='%s'\n", errorType, line, (int)error, (unsigned)error, errorText));
+
+    PaUtil_SetLastHostErrorInfo( paCoreAudio, error, errorText );
+
+    return result;
+}
+
+/*
+ * This function computes an appropriate ring buffer size given
+ * a requested latency (in seconds), sample rate and framesPerBuffer.
+ *
+ * The returned ringBufferSize is computed using the following
+ * constraints:
+ *   - it must be at least 4.
+ *   - it must be at least 3x framesPerBuffer.
+ *   - it must be at least 2x the suggestedLatency.
+ *   - it must be a power of 2.
+ * This function attempts to compute the minimum such size.
+ *
+ * FEEDBACK: too liberal/conservative/another way?
+ */
+long computeRingBufferSize( const PaStreamParameters *inputParameters,
+                                   const PaStreamParameters *outputParameters,
+                                   long inputFramesPerBuffer,
+                                   long outputFramesPerBuffer,
+                                   double sampleRate )
+{
+   long ringSize;
+   int index;
+   int i;
+   double latencyTimesChannelCount ;
+   long framesPerBufferTimesChannelCount ;
+
+   VVDBUG(( "computeRingBufferSize()\n" ));
+
+   assert( inputParameters || outputParameters );
+
+   if( outputParameters && inputParameters )
+   {
+      latencyTimesChannelCount = MAX(
+           inputParameters->suggestedLatency * inputParameters->channelCount,
+           outputParameters->suggestedLatency * outputParameters->channelCount );
+      framesPerBufferTimesChannelCount = MAX(
+           inputFramesPerBuffer * inputParameters->channelCount,
+           outputFramesPerBuffer * outputParameters->channelCount );
+   } 
+   else if( outputParameters )
+   {
+      latencyTimesChannelCount
+                = outputParameters->suggestedLatency * outputParameters->channelCount;
+      framesPerBufferTimesChannelCount
+                = outputFramesPerBuffer * outputParameters->channelCount;
+   }
+   else /* we have inputParameters  */
+   {
+      latencyTimesChannelCount
+                = inputParameters->suggestedLatency * inputParameters->channelCount;
+      framesPerBufferTimesChannelCount
+                = inputFramesPerBuffer * inputParameters->channelCount;
+   }
+
+   ringSize = (long) ( latencyTimesChannelCount * sampleRate * 2 + .5);
+   VDBUG( ( "suggested latency * channelCount: %d\n", (int) (latencyTimesChannelCount*sampleRate) ) );
+   if( ringSize < framesPerBufferTimesChannelCount * 3 )
+      ringSize = framesPerBufferTimesChannelCount * 3 ;
+   VDBUG(("framesPerBuffer*channelCount:%d\n",(int)framesPerBufferTimesChannelCount));
+   VDBUG(("Ringbuffer size (1): %d\n", (int)ringSize ));
+
+   /* make sure it's at least 4 */
+   ringSize = MAX( ringSize, 4 );
+
+   /* round up to the next power of 2 */
+   index = -1;
+   for( i=0; i<sizeof(long)*8; ++i )
+      if( ringSize >> i & 0x01 )
+         index = i;
+   assert( index > 0 );
+   if( ringSize <= ( 0x01 << index ) )
+      ringSize = 0x01 << index ;
+   else
+      ringSize = 0x01 << ( index + 1 );
+
+   VDBUG(( "Final Ringbuffer size (2): %d\n", (int)ringSize ));
+   return ringSize;
+}
+
+
+/*
+ * Durring testing of core audio, I found that serious crashes could occur
+ * if properties such as sample rate were changed multiple times in rapid
+ * succession. The function below has some fancy logic to make sure that changes
+ * are acknowledged before another is requested. That seems to help a lot.
+ */
+
+OSStatus propertyProc(
+    AudioDeviceID inDevice, 
+    UInt32 inChannel, 
+    Boolean isInput, 
+    AudioDevicePropertyID inPropertyID, 
+    void* inClientData )
+{
+   MutexAndBool *mab = (MutexAndBool *) inClientData;
+   mab->once = TRUE;
+   pthread_mutex_unlock( &(mab->mutex) );
+   return noErr;
+}
+
+/* sets the value of the given property and waits for the change to 
+   be acknowledged, and returns the final value, which is not guaranteed
+   by this function to be the same as the desired value. Obviously, this
+   function can only be used for data whose input and output are the
+   same size and format, and their size and format are known in advance.*/
+PaError AudioDeviceSetPropertyNowAndWaitForChange(
+    AudioDeviceID inDevice,
+    UInt32 inChannel, 
+    Boolean isInput, 
+    AudioDevicePropertyID inPropertyID,
+    UInt32 inPropertyDataSize, 
+    const void *inPropertyData,
+    void *outPropertyData )
+{
+   OSStatus macErr;
+   int unixErr;
+   MutexAndBool mab;
+   UInt32 outPropertyDataSize = inPropertyDataSize;
+
+   /* First, see if it already has that value. If so, return. */
+   macErr = AudioDeviceGetProperty( inDevice, inChannel,
+                                 isInput, inPropertyID, 
+                                 &outPropertyDataSize, outPropertyData );
+   if( macErr )
+      goto failMac2;
+   if( inPropertyDataSize!=outPropertyDataSize )
+      return paInternalError;
+   if( 0==memcmp( outPropertyData, inPropertyData, outPropertyDataSize ) )
+      return paNoError;
+
+   /* setup and lock mutex */
+   mab.once = FALSE;
+   unixErr = pthread_mutex_init( &mab.mutex, NULL );
+   if( unixErr )
+      goto failUnix2;
+   unixErr = pthread_mutex_lock( &mab.mutex );
+   if( unixErr )
+      goto failUnix;
+
+   /* add property listener */
+   macErr = AudioDeviceAddPropertyListener( inDevice, inChannel, isInput,
+                                   inPropertyID, propertyProc,
+                                   &mab ); 
+   if( macErr )
+      goto failMac;
+   /* set property */
+   macErr  = AudioDeviceSetProperty( inDevice, NULL, inChannel,
+                                 isInput, inPropertyID,
+                                 inPropertyDataSize, inPropertyData );
+   if( macErr ) {
+      /* we couldn't set the property, so we'll just unlock the mutex
+         and move on. */
+      pthread_mutex_unlock( &mab.mutex );
+   }
+
+   /* wait for property to change */                      
+   unixErr = pthread_mutex_lock( &mab.mutex );
+   if( unixErr )
+      goto failUnix;
+
+   /* now read the property back out */
+   macErr = AudioDeviceGetProperty( inDevice, inChannel,
+                                 isInput, inPropertyID, 
+                                 &outPropertyDataSize, outPropertyData );
+   if( macErr )
+      goto failMac;
+   /* cleanup */
+   AudioDeviceRemovePropertyListener( inDevice, inChannel, isInput,
+                                      inPropertyID, propertyProc );
+   unixErr = pthread_mutex_unlock( &mab.mutex );
+   if( unixErr )
+      goto failUnix2;
+   unixErr = pthread_mutex_destroy( &mab.mutex );
+   if( unixErr )
+      goto failUnix2;
+
+   return paNoError;
+
+ failUnix:
+   pthread_mutex_destroy( &mab.mutex );
+   AudioDeviceRemovePropertyListener( inDevice, inChannel, isInput,
+                                      inPropertyID, propertyProc );
+
+ failUnix2:
+   DBUG( ("Error #%d while setting a device property: %s\n", unixErr, strerror( unixErr ) ) );
+   return paUnanticipatedHostError;
+
+ failMac:
+   pthread_mutex_destroy( &mab.mutex );
+   AudioDeviceRemovePropertyListener( inDevice, inChannel, isInput,
+                                      inPropertyID, propertyProc );
+ failMac2:
+   return ERR( macErr );
+}
+
+/*
+ * Sets the sample rate the HAL device.
+ * if requireExact: set the sample rate or fail.
+ *
+ * otherwise      : set the exact sample rate.
+ *             If that fails, check for available sample rates, and choose one
+ *             higher than the requested rate. If there isn't a higher one,
+ *             just use the highest available.
+ */
+PaError setBestSampleRateForDevice( const AudioDeviceID device,
+                                    const bool isOutput,
+                                    const bool requireExact,
+                                    const Float64 desiredSrate )
+{
+   /*FIXME: changing the sample rate is disruptive to other programs using the
+            device, so it might be good to offer a custom flag to not change the
+            sample rate and just do conversion. (in my casual tests, there is
+            no disruption unless the sample rate really does need to change) */
+   const bool isInput = isOutput ? 0 : 1;
+   Float64 srate;
+   UInt32 propsize = sizeof( Float64 );
+   OSErr err;
+   AudioValueRange *ranges;
+   int i=0;
+   Float64 max  = -1; /*the maximum rate available*/
+   Float64 best = -1; /*the lowest sample rate still greater than desired rate*/
+   VDBUG(("Setting sample rate for device %ld to %g.\n",device,(float)desiredSrate));
+
+   /* -- try setting the sample rate -- */
+   err = AudioDeviceSetPropertyNowAndWaitForChange(
+                                 device, 0, isInput,
+                                 kAudioDevicePropertyNominalSampleRate,
+                                 propsize, &desiredSrate, &srate );
+   if( err )
+      return err;
+
+   /* -- if the rate agrees, and we got no errors, we are done -- */
+   if( !err && srate == desiredSrate )
+      return paNoError;
+   /* -- we've failed if the rates disagree and we are setting input -- */
+   if( requireExact )
+      return paInvalidSampleRate;
+
+   /* -- generate a list of available sample rates -- */
+   err = AudioDeviceGetPropertyInfo( device, 0, isInput,
+                                kAudioDevicePropertyAvailableNominalSampleRates,
+                                &propsize, NULL );
+   if( err )
+      return ERR( err );
+   ranges = (AudioValueRange *)calloc( 1, propsize );
+   if( !ranges )
+      return paInsufficientMemory;
+   err = AudioDeviceGetProperty( device, 0, isInput,
+                                kAudioDevicePropertyAvailableNominalSampleRates,
+                                &propsize, ranges );
+   if( err )
+   {
+      free( ranges );
+      return ERR( err );
+   }
+   VDBUG(("Requested sample rate of %g was not available.\n", (float)desiredSrate));
+   VDBUG(("%lu Available Sample Rates are:\n",propsize/sizeof(AudioValueRange)));
+#ifdef MAC_CORE_VERBOSE_DEBUG
+   for( i=0; i<propsize/sizeof(AudioValueRange); ++i )
+      VDBUG( ("\t%g-%g\n",
+              (float) ranges[i].mMinimum,
+              (float) ranges[i].mMaximum ) );
+#endif
+   VDBUG(("-----\n"));
+   
+   /* -- now pick the best available sample rate -- */
+   for( i=0; i<propsize/sizeof(AudioValueRange); ++i )
+   {
+      if( ranges[i].mMaximum > max ) max = ranges[i].mMaximum;
+      if( ranges[i].mMinimum > desiredSrate ) {
+         if( best < 0 )
+            best = ranges[i].mMinimum;
+         else if( ranges[i].mMinimum < best )
+            best = ranges[i].mMinimum;
+      }
+   }
+   if( best < 0 )
+      best = max;
+   VDBUG( ("Maximum Rate %g. best is %g.\n", max, best ) );
+   free( ranges );
+
+   /* -- set the sample rate -- */
+   propsize = sizeof( best );
+   err = AudioDeviceSetPropertyNowAndWaitForChange(
+                                 device, 0, isInput,
+                                 kAudioDevicePropertyNominalSampleRate,
+                                 propsize, &best, &srate );
+   if( err )
+      return err;
+
+   if( err )
+      return ERR( err );
+   /* -- if the set rate matches, we are done -- */
+   if( srate == best )
+      return paNoError;
+
+   /* -- otherwise, something wierd happened: we didn't set the rate, and we got no errors. Just bail. */
+   return paInternalError;
+}
+
+
+/*
+   Attempts to set the requestedFramesPerBuffer. If it can't set the exact
+   value, it settles for something smaller if available. If nothing smaller
+   is available, it uses the smallest available size.
+   actualFramesPerBuffer will be set to the actual value on successful return.
+   OK to pass NULL to actualFramesPerBuffer.
+   The logic is very simmilar too setBestSampleRate only failure here is
+   not usually catastrophic.
+*/
+PaError setBestFramesPerBuffer( const AudioDeviceID device,
+                                       const bool isOutput,
+                                       unsigned long requestedFramesPerBuffer, 
+                                       unsigned long *actualFramesPerBuffer )
+{
+   UInt32 afpb;
+   const bool isInput = !isOutput;
+   UInt32 propsize = sizeof(UInt32);
+   OSErr err;
+   Float64 min  = -1; /*the min blocksize*/
+   Float64 best = -1; /*the best blocksize*/
+   int i=0;
+   AudioValueRange *ranges;
+
+   if( actualFramesPerBuffer == NULL )
+      actualFramesPerBuffer = &afpb;
+
+
+   /* -- try and set exact FPB -- */
+   err = AudioDeviceSetProperty( device, NULL, 0, isInput,
+                                 kAudioDevicePropertyBufferFrameSize,
+                                 propsize, &requestedFramesPerBuffer);
+   err = AudioDeviceGetProperty( device, 0, isInput,
+                           kAudioDevicePropertyBufferFrameSize,
+                           &propsize, actualFramesPerBuffer);
+   if( err )
+      return ERR( err );
+   if( *actualFramesPerBuffer == requestedFramesPerBuffer )
+      return paNoError; /* we are done */
+
+   /* -- fetch available block sizes -- */
+   err = AudioDeviceGetPropertyInfo( device, 0, isInput,
+                           kAudioDevicePropertyBufferSizeRange,
+                           &propsize, NULL );
+   if( err )
+      return ERR( err );
+   ranges = (AudioValueRange *)calloc( 1, propsize );
+   if( !ranges )
+      return paInsufficientMemory;
+   err = AudioDeviceGetProperty( device, 0, isInput,
+                                kAudioDevicePropertyBufferSizeRange,
+                                &propsize, ranges );
+   if( err )
+   {
+      free( ranges );
+      return ERR( err );
+   }
+   VDBUG(("Requested block size of %lu was not available.\n",
+          requestedFramesPerBuffer ));
+   VDBUG(("%lu Available block sizes are:\n",propsize/sizeof(AudioValueRange)));
+#ifdef MAC_CORE_VERBOSE_DEBUG
+   for( i=0; i<propsize/sizeof(AudioValueRange); ++i )
+      VDBUG( ("\t%g-%g\n",
+              (float) ranges[i].mMinimum,
+              (float) ranges[i].mMaximum ) );
+#endif
+   VDBUG(("-----\n"));
+   
+   /* --- now pick the best available framesPerBuffer -- */
+   for( i=0; i<propsize/sizeof(AudioValueRange); ++i )
+   {
+      if( min == -1 || ranges[i].mMinimum < min ) min = ranges[i].mMinimum;
+      if( ranges[i].mMaximum < requestedFramesPerBuffer ) {
+         if( best < 0 )
+            best = ranges[i].mMaximum;
+         else if( ranges[i].mMaximum > best )
+            best = ranges[i].mMaximum;
+      }
+   }
+   if( best == -1 )
+      best = min;
+   VDBUG( ("Minimum FPB  %g. best is %g.\n", min, best ) );
+   free( ranges );
+
+   /* --- set the buffer size (ignore errors) -- */
+   requestedFramesPerBuffer = (UInt32) best ;
+   propsize = sizeof( UInt32 );
+   err = AudioDeviceSetProperty( device, NULL, 0, isInput,
+                                 kAudioDevicePropertyBufferSize,
+                                 propsize, &requestedFramesPerBuffer );
+   /* --- read the property to check that it was set -- */
+   err = AudioDeviceGetProperty( device, 0, isInput,
+                                 kAudioDevicePropertyBufferSize,
+                                 &propsize, actualFramesPerBuffer );
+
+   if( err )
+      return ERR( err );
+
+   return paNoError;
+}

3rdparty/iaxclient/lib/portaudio/src/hostapi/coreaudio/pa_mac_core_utilities.h

@@ -0,0 +1,159 @@
+/*
+ *
+ * pa_mac_core_utilities.c
+ *
+ * utilitiy functions for pa_mac_core.c
+ *
+ * This functions are more like helper functions than part of the core,
+ *  so I moved them to a separate file so the core code would be cleaner.
+ *
+ * by Bjorn Roche.
+ */
+
+#ifndef PA_MAC_CORE_UTILITIES_H__
+#define PA_MAC_CORE_UTILITIES_H__
+
+#include <pthread.h>
+#include "portaudio.h"
+#include "pa_util.h"
+#include <AudioUnit/AudioUnit.h>
+#include <AudioToolbox/AudioToolbox.h>
+
+#ifndef MIN
+#define MIN(a, b)  (((a)<(b))?(a):(b))
+#endif
+
+#ifndef MAX
+#define MAX(a, b)  (((a)<(b))?(b):(a))
+#endif
+
+#define ERR(mac_error) PaMacCore_SetError(mac_error, __LINE__, 1 ) 
+#define WARNING(mac_error) PaMacCore_SetError(mac_error, __LINE__, 0 )
+
+
+/* Help keep track of AUHAL element numbers */
+#define INPUT_ELEMENT  (1)
+#define OUTPUT_ELEMENT (0)
+
+/* Normal level of debugging: fine for most apps that don't mind the occational warning being printf'ed */
+/*
+ */
+#define MAC_CORE_DEBUG
+#ifdef MAC_CORE_DEBUG
+# define DBUG(MSG) do { printf("||PaMacCore (AUHAL)|| "); printf MSG ; fflush(stdout); } while(0)
+#else
+# define DBUG(MSG)
+#endif
+
+/* Verbose Debugging: useful for developement */
+/*
+#define MAC_CORE_VERBOSE_DEBUG
+*/
+#ifdef MAC_CORE_VERBOSE_DEBUG
+# define VDBUG(MSG) do { printf("||PaMacCore (v )|| "); printf MSG ; fflush(stdout); } while(0)
+#else
+# define VDBUG(MSG)
+#endif
+
+/* Very Verbose Debugging: Traces every call. */
+/*
+#define MAC_CORE_VERY_VERBOSE_DEBUG
+ */
+#ifdef MAC_CORE_VERY_VERBOSE_DEBUG
+# define VVDBUG(MSG) do { printf("||PaMacCore (vv)|| "); printf MSG ; fflush(stdout); } while(0)
+#else
+# define VVDBUG(MSG)
+#endif
+
+
+
+
+
+#define UNIX_ERR(err) PaMacCore_SetUnixError( err, __LINE__ )
+
+PaError PaMacCore_SetUnixError( int err, int line );
+
+/*
+ * Translates MacOS generated errors into PaErrors
+ */
+PaError PaMacCore_SetError(OSStatus error, int line, int isError);
+
+/*
+ * This function computes an appropriate ring buffer size given
+ * a requested latency (in seconds), sample rate and framesPerBuffer.
+ *
+ * The returned ringBufferSize is computed using the following
+ * constraints:
+ *   - it must be at least 4.
+ *   - it must be at least 3x framesPerBuffer.
+ *   - it must be at least 2x the suggestedLatency.
+ *   - it must be a power of 2.
+ * This function attempts to compute the minimum such size.
+ *
+ */
+long computeRingBufferSize( const PaStreamParameters *inputParameters,
+                                   const PaStreamParameters *outputParameters,
+                                   long inputFramesPerBuffer,
+                                   long outputFramesPerBuffer,
+                                   double sampleRate );
+
+/*
+ * Durring testing of core audio, I found that serious crashes could occur
+ * if properties such as sample rate were changed multiple times in rapid
+ * succession. The function below has some fancy logic to make sure that changes
+ * are acknowledged before another is requested. That seems to help a lot.
+ */
+
+typedef struct {
+   bool once; /* I didn't end up using this. bdr */
+   pthread_mutex_t mutex;
+} MutexAndBool ;
+
+OSStatus propertyProc(
+    AudioDeviceID inDevice, 
+    UInt32 inChannel, 
+    Boolean isInput, 
+    AudioDevicePropertyID inPropertyID, 
+    void* inClientData );
+
+/* sets the value of the given property and waits for the change to 
+   be acknowledged, and returns the final value, which is not guaranteed
+   by this function to be the same as the desired value. Obviously, this
+   function can only be used for data whose input and output are the
+   same size and format, and their size and format are known in advance.*/
+PaError AudioDeviceSetPropertyNowAndWaitForChange(
+    AudioDeviceID inDevice,
+    UInt32 inChannel, 
+    Boolean isInput, 
+    AudioDevicePropertyID inPropertyID,
+    UInt32 inPropertyDataSize, 
+    const void *inPropertyData,
+    void *outPropertyData );
+
+/*
+ * Sets the sample rate the HAL device.
+ * if requireExact: set the sample rate or fail.
+ *
+ * otherwise      : set the exact sample rate.
+ *             If that fails, check for available sample rates, and choose one
+ *             higher than the requested rate. If there isn't a higher one,
+ *             just use the highest available.
+ */
+PaError setBestSampleRateForDevice( const AudioDeviceID device,
+                                    const bool isOutput,
+                                    const bool requireExact,
+                                    const Float64 desiredSrate );
+/*
+   Attempts to set the requestedFramesPerBuffer. If it can't set the exact
+   value, it settles for something smaller if available. If nothing smaller
+   is available, it uses the smallest available size.
+   actualFramesPerBuffer will be set to the actual value on successful return.
+   OK to pass NULL to actualFramesPerBuffer.
+   The logic is very simmilar too setBestSampleRate only failure here is
+   not usually catastrophic.
+*/
+PaError setBestFramesPerBuffer( const AudioDeviceID device,
+                                       const bool isOutput,
+                                       unsigned long requestedFramesPerBuffer, 
+                                       unsigned long *actualFramesPerBuffer );
+#endif /* PA_MAC_CORE_UTILITIES_H__*/

3rdparty/iaxclient/lib/portaudio/src/hostapi/coreaudio/ringbuffer.c

@@ -0,0 +1,273 @@
+/*
+ * $Id: ringbuffer.c,v 1.1 2006/06/10 21:30:56 dmazzoni Exp $
+ * ringbuffer.c
+ * Ring Buffer utility..
+ *
+ * Author: Phil Burk, http://www.softsynth.com
+ * modified for SMP safety on Mac OS X by Bjorn Roche
+ * also, alowed for const where possible
+ * Note that this is safe only for a single-thread reader and a
+ * single-thread writer.
+ *
+ * This program uses the PortAudio Portable Audio Library.
+ * For more information see: http://www.audiomulch.com/portaudio/
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ *
+ */
+#include <stdio.h>
+#include <stdlib.h>
+#include <math.h>
+#include "ringbuffer.h"
+#include <string.h>
+
+/*
+ * We can undefine this, to turn off memory barriers, but that
+ * is only useful if we know we don't need to be MP safe or
+ * we are interested in doing some kind of tests.
+ */
+#define MPSAFE
+
+/****************
+ * First, we'll define some memory barrier primitives based on the system.
+ * right now only OS X and FreeBSD are supported. In addition to providing
+ * memory barriers, these functions should ensure that data cached in registers
+ * is written out to cache where it can be snooped by other CPUs. (ie, the volatile
+ * keyword should not be required)
+ *
+ * the primitives that must be defined are:
+ *
+ * FullMemoryBarrier()
+ * ReadMemoryBarrier()
+ * WriteMemoryBarrier()
+ *
+ ****************/
+
+#if defined(__APPLE__) || defined(__FreeBSD__)
+//#   include <libkern/OSAtomic.h>
+    /* Here are the memory barrier functions. Mac OS X and FreeBSD only provide
+       full memory barriers, so the three types of barriers are the same.
+       The asm volatile may be redundant with the memory barrier, but
+       until I have proof of that, I'm leaving it. */
+/* Mihai: trying to compile without assembly, since gcc barfs on asm statements for some reason
+#   define FullMemoryBarrier()  do{ asm volatile("":::"memory"); OSMemoryBarrier(); }while(false)
+#   define ReadMemoryBarrier()  do{ asm volatile("":::"memory"); OSMemoryBarrier(); }while(false)
+#   define WriteMemoryBarrier() do{ asm volatile("":::"memory"); OSMemoryBarrier(); }while(false)
+
+#   define FullMemoryBarrier()  do{ OSMemoryBarrier(); }while(false)
+#   define ReadMemoryBarrier()  do{ OSMemoryBarrier(); }while(false)
+#   define WriteMemoryBarrier() do{ OSMemoryBarrier(); }while(false)
+*/
+
+/* 
+ * Mihai: OSMemoryBarrier is not available on 10.3.9 and I am not able to find a replacement
+ * As far as I can tell, not having synchronized memory acces will hurt only MP machines, and
+ * more specifically only weakly ordered memory models architectures (i.e. PowerPC).  In case
+ * this code runs on a MP PPC machine, worst I can see happening is a corruption of the sound
+ * buffer, which will lead to crappy sound, but not much more. 
+ * If anybody finds a 10.3.9 replacement for OSMemoryBarrier, I'd be glad to hear about it
+ */ 
+#define FullMemoryBarrier() 
+#define ReadMemoryBarrier()
+#define WriteMemoryBarrier()
+
+#else
+#   error Memory Barriers not defined on this system or system unknown
+#endif
+
+/***************************************************************************
+ * Initialize FIFO.
+ * numBytes must be power of 2, returns -1 if not.
+ */
+long RingBuffer_Init( RingBuffer *rbuf, long numBytes, void *dataPtr )
+{
+    if( ((numBytes-1) & numBytes) != 0) return -1; /* Not Power of two. */
+    rbuf->bufferSize = numBytes;
+    rbuf->buffer = (char *)dataPtr;
+    RingBuffer_Flush( rbuf );
+    rbuf->bigMask = (numBytes*2)-1;
+    rbuf->smallMask = (numBytes)-1;
+    return 0;
+}
+/***************************************************************************
+** Return number of bytes available for reading. */
+long RingBuffer_GetReadAvailable( RingBuffer *rbuf )
+{
+#ifdef MPSAFE
+    ReadMemoryBarrier();
+#endif
+    return ( (rbuf->writeIndex - rbuf->readIndex) & rbuf->bigMask );
+}
+/***************************************************************************
+** Return number of bytes available for writing. */
+long RingBuffer_GetWriteAvailable( RingBuffer *rbuf )
+{
+    /* Since we are calling RingBuffer_GetReadAvailable, we don't need an aditional MB */
+    return ( rbuf->bufferSize - RingBuffer_GetReadAvailable(rbuf));
+}
+
+/***************************************************************************
+** Clear buffer. Should only be called when buffer is NOT being read. */
+void RingBuffer_Flush( RingBuffer *rbuf )
+{
+    rbuf->writeIndex = rbuf->readIndex = 0;
+}
+
+/***************************************************************************
+** Get address of region(s) to which we can write data.
+** If the region is contiguous, size2 will be zero.
+** If non-contiguous, size2 will be the size of second region.
+** Returns room available to be written or numBytes, whichever is smaller.
+*/
+long RingBuffer_GetWriteRegions( RingBuffer *rbuf, long numBytes,
+                                 void **dataPtr1, long *sizePtr1,
+                                 void **dataPtr2, long *sizePtr2 )
+{
+    long   index;
+    long   available = RingBuffer_GetWriteAvailable( rbuf );
+    if( numBytes > available ) numBytes = available;
+    /* Check to see if write is not contiguous. */
+    index = rbuf->writeIndex & rbuf->smallMask;
+    if( (index + numBytes) > rbuf->bufferSize )
+    {
+        /* Write data in two blocks that wrap the buffer. */
+        long   firstHalf = rbuf->bufferSize - index;
+        *dataPtr1 = &rbuf->buffer[index];
+        *sizePtr1 = firstHalf;
+        *dataPtr2 = &rbuf->buffer[0];
+        *sizePtr2 = numBytes - firstHalf;
+    }
+    else
+    {
+        *dataPtr1 = &rbuf->buffer[index];
+        *sizePtr1 = numBytes;
+        *dataPtr2 = NULL;
+        *sizePtr2 = 0;
+    }
+    return numBytes;
+}
+
+
+/***************************************************************************
+*/
+long RingBuffer_AdvanceWriteIndex( RingBuffer *rbuf, long numBytes )
+{
+#ifdef MPSAFE
+    /* we need to ensure that previous writes are seen before we update the write index */
+    WriteMemoryBarrier();
+    return rbuf->writeIndex = (rbuf->writeIndex + numBytes) & rbuf->bigMask;
+#else
+    return rbuf->writeIndex = (rbuf->writeIndex + numBytes) & rbuf->bigMask;
+#endif
+}
+
+/***************************************************************************
+** Get address of region(s) from which we can read data.
+** If the region is contiguous, size2 will be zero.
+** If non-contiguous, size2 will be the size of second region.
+** Returns room available to be written or numBytes, whichever is smaller.
+*/
+long RingBuffer_GetReadRegions( RingBuffer *rbuf, long numBytes,
+                                void **dataPtr1, long *sizePtr1,
+                                void **dataPtr2, long *sizePtr2 )
+{
+    long   index;
+    long   available = RingBuffer_GetReadAvailable( rbuf );
+    if( numBytes > available ) numBytes = available;
+    /* Check to see if read is not contiguous. */
+    index = rbuf->readIndex & rbuf->smallMask;
+    if( (index + numBytes) > rbuf->bufferSize )
+    {
+        /* Write data in two blocks that wrap the buffer. */
+        long firstHalf = rbuf->bufferSize - index;
+        *dataPtr1 = &rbuf->buffer[index];
+        *sizePtr1 = firstHalf;
+        *dataPtr2 = &rbuf->buffer[0];
+        *sizePtr2 = numBytes - firstHalf;
+    }
+    else
+    {
+        *dataPtr1 = &rbuf->buffer[index];
+        *sizePtr1 = numBytes;
+        *dataPtr2 = NULL;
+        *sizePtr2 = 0;
+    }
+    return numBytes;
+}
+/***************************************************************************
+*/
+long RingBuffer_AdvanceReadIndex( RingBuffer *rbuf, long numBytes )
+{
+#ifdef MPSAFE
+    /* we need to ensure that previous writes are always seen before updating the index. */
+    WriteMemoryBarrier();
+    return rbuf->readIndex = (rbuf->readIndex + numBytes) & rbuf->bigMask;
+#else
+    return rbuf->readIndex = (rbuf->readIndex + numBytes) & rbuf->bigMask;
+#endif
+}
+
+/***************************************************************************
+** Return bytes written. */
+long RingBuffer_Write( RingBuffer *rbuf, const void *data, long numBytes )
+{
+    long size1, size2, numWritten;
+    void *data1, *data2;
+    numWritten = RingBuffer_GetWriteRegions( rbuf, numBytes, &data1, &size1, &data2, &size2 );
+    if( size2 > 0 )
+    {
+
+        memcpy( data1, data, size1 );
+        data = ((char *)data) + size1;
+        memcpy( data2, data, size2 );
+    }
+    else
+    {
+        memcpy( data1, data, size1 );
+    }
+    RingBuffer_AdvanceWriteIndex( rbuf, numWritten );
+    return numWritten;
+}
+
+/***************************************************************************
+** Return bytes read. */
+long RingBuffer_Read( RingBuffer *rbuf, void *data, long numBytes )
+{
+    long size1, size2, numRead;
+    void *data1, *data2;
+    numRead = RingBuffer_GetReadRegions( rbuf, numBytes, &data1, &size1, &data2, &size2 );
+    if( size2 > 0 )
+    {
+        memcpy( data, data1, size1 );
+        data = ((char *)data) + size1;
+        memcpy( data, data2, size2 );
+    }
+    else
+    {
+        memcpy( data, data1, size1 );
+    }
+    RingBuffer_AdvanceReadIndex( rbuf, numRead );
+    return numRead;
+}

3rdparty/iaxclient/lib/portaudio/src/hostapi/coreaudio/ringbuffer.h

@@ -0,0 +1,105 @@
+#ifndef _RINGBUFFER_H
+#define _RINGBUFFER_H
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+/*
+ * $Id: ringbuffer.h,v 1.1 2006/06/10 21:30:56 dmazzoni Exp $
+ * ringbuffer.h
+ * Ring Buffer utility..
+ *
+ * Author: Phil Burk, http://www.softsynth.com
+ * modified for SMP safety on OS X by Bjorn Roche.
+ * also allowed for const where possible.
+ * Note that this is safe only for a single-thread reader
+ * and a single-thread writer.
+ *
+ * This program is distributed with the PortAudio Portable Audio Library.
+ * For more information see: http://www.audiomulch.com/portaudio/
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ *
+ */
+#include <stdio.h>
+#include <stdlib.h>
+#include <math.h>
+#include "ringbuffer.h"
+#include <string.h>
+
+typedef struct
+{
+    long   bufferSize; /* Number of bytes in FIFO. Power of 2. Set by RingBuffer_Init. */
+    long   writeIndex; /* Index of next writable byte. Set by RingBuffer_AdvanceWriteIndex. */
+    long   readIndex;  /* Index of next readable byte. Set by RingBuffer_AdvanceReadIndex. */
+    long   bigMask;    /* Used for wrapping indices with extra bit to distinguish full/empty. */
+    long   smallMask;  /* Used for fitting indices to buffer. */
+    char * buffer;
+}
+RingBuffer;
+/*
+ * Initialize Ring Buffer.
+ * numBytes must be power of 2, returns -1 if not.
+ */
+long RingBuffer_Init( RingBuffer *rbuf, long numBytes, void *dataPtr );
+
+/* Clear buffer. Should only be called when buffer is NOT being read. */
+void RingBuffer_Flush( RingBuffer *rbuf );
+
+/* Return number of bytes available for writing. */
+long RingBuffer_GetWriteAvailable( RingBuffer *rbuf );
+/* Return number of bytes available for read. */
+long RingBuffer_GetReadAvailable( RingBuffer *rbuf );
+/* Return bytes written. */
+long RingBuffer_Write( RingBuffer *rbuf, const void *data, long numBytes );
+/* Return bytes read. */
+long RingBuffer_Read( RingBuffer *rbuf, void *data, long numBytes );
+
+/* Get address of region(s) to which we can write data.
+** If the region is contiguous, size2 will be zero.
+** If non-contiguous, size2 will be the size of second region.
+** Returns room available to be written or numBytes, whichever is smaller.
+*/
+long RingBuffer_GetWriteRegions( RingBuffer *rbuf, long numBytes,
+                                 void **dataPtr1, long *sizePtr1,
+                                 void **dataPtr2, long *sizePtr2 );
+long RingBuffer_AdvanceWriteIndex( RingBuffer *rbuf, long numBytes );
+
+/* Get address of region(s) from which we can read data.
+** If the region is contiguous, size2 will be zero.
+** If non-contiguous, size2 will be the size of second region.
+** Returns room available to be written or numBytes, whichever is smaller.
+*/
+long RingBuffer_GetReadRegions( RingBuffer *rbuf, long numBytes,
+                                void **dataPtr1, long *sizePtr1,
+                                void **dataPtr2, long *sizePtr2 );
+
+long RingBuffer_AdvanceReadIndex( RingBuffer *rbuf, long numBytes );
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif /* _RINGBUFFER_H */

3rdparty/iaxclient/lib/portaudio/src/hostapi/dsound/pa_win_ds_dynlink.c

@@ -0,0 +1,121 @@
+#include "pa_win_ds_dynlink.h"
+
+
+PaWinDsDSoundEntryPoints paWinDsDSoundEntryPoints = { 0, 0, 0, 0, 0, 0, 0 };
+
+
+static HRESULT WINAPI DummyDirectSoundCreate(LPGUID lpcGuidDevice, LPDIRECTSOUND *ppDS, LPUNKNOWN pUnkOuter)
+{
+    (void)lpcGuidDevice; /* unused parameter */
+    (void)ppDS; /* unused parameter */
+    (void)pUnkOuter; /* unused parameter */
+    return E_NOTIMPL;
+}
+
+static HRESULT WINAPI DummyDirectSoundEnumerateW(LPDSENUMCALLBACKW lpDSEnumCallback, LPVOID lpContext)
+{
+    (void)lpDSEnumCallback; /* unused parameter */
+    (void)lpContext; /* unused parameter */
+    return E_NOTIMPL;
+}
+
+static HRESULT WINAPI DummyDirectSoundEnumerateA(LPDSENUMCALLBACKA lpDSEnumCallback, LPVOID lpContext)
+{
+    (void)lpDSEnumCallback; /* unused parameter */
+    (void)lpContext; /* unused parameter */
+    return E_NOTIMPL;
+}
+
+static HRESULT WINAPI DummyDirectSoundCaptureCreate(LPGUID lpcGUID, LPDIRECTSOUNDCAPTURE *lplpDSC, LPUNKNOWN pUnkOuter)
+{
+    (void)lpcGUID; /* unused parameter */
+    (void)lplpDSC; /* unused parameter */
+    (void)pUnkOuter; /* unused parameter */
+    return E_NOTIMPL;
+}
+
+static HRESULT WINAPI DummyDirectSoundCaptureEnumerateW(LPDSENUMCALLBACKW lpDSCEnumCallback, LPVOID lpContext)
+{
+    (void)lpDSCEnumCallback; /* unused parameter */
+    (void)lpContext; /* unused parameter */
+    return E_NOTIMPL;
+}
+
+static HRESULT WINAPI DummyDirectSoundCaptureEnumerateA(LPDSENUMCALLBACKA lpDSCEnumCallback, LPVOID lpContext)
+{
+    (void)lpDSCEnumCallback; /* unused parameter */
+    (void)lpContext; /* unused parameter */
+    return E_NOTIMPL;
+}
+
+
+void PaWinDs_InitializeDSoundEntryPoints(void)
+{
+    paWinDsDSoundEntryPoints.hInstance_ = LoadLibrary("dsound.dll");
+    if( paWinDsDSoundEntryPoints.hInstance_ != NULL )
+    {
+        paWinDsDSoundEntryPoints.DirectSoundCreate =
+                (HRESULT (WINAPI *)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN))
+                GetProcAddress( paWinDsDSoundEntryPoints.hInstance_, "DirectSoundCreate" );
+        if( paWinDsDSoundEntryPoints.DirectSoundCreate == NULL )
+            paWinDsDSoundEntryPoints.DirectSoundCreate = DummyDirectSoundCreate;
+
+        paWinDsDSoundEntryPoints.DirectSoundEnumerateW =
+                (HRESULT (WINAPI *)(LPDSENUMCALLBACKW, LPVOID))
+                GetProcAddress( paWinDsDSoundEntryPoints.hInstance_, "DirectSoundEnumerateW" );
+        if( paWinDsDSoundEntryPoints.DirectSoundEnumerateW == NULL )
+            paWinDsDSoundEntryPoints.DirectSoundEnumerateW = DummyDirectSoundEnumerateW;
+
+        paWinDsDSoundEntryPoints.DirectSoundEnumerateA =
+                (HRESULT (WINAPI *)(LPDSENUMCALLBACKA, LPVOID))
+                GetProcAddress( paWinDsDSoundEntryPoints.hInstance_, "DirectSoundEnumerateA" );
+        if( paWinDsDSoundEntryPoints.DirectSoundEnumerateA == NULL )
+            paWinDsDSoundEntryPoints.DirectSoundEnumerateA = DummyDirectSoundEnumerateA;
+
+        paWinDsDSoundEntryPoints.DirectSoundCaptureCreate =
+                (HRESULT (WINAPI *)(LPGUID, LPDIRECTSOUNDCAPTURE *, LPUNKNOWN))
+                GetProcAddress( paWinDsDSoundEntryPoints.hInstance_, "DirectSoundCaptureCreate" );
+        if( paWinDsDSoundEntryPoints.DirectSoundCaptureCreate == NULL )
+            paWinDsDSoundEntryPoints.DirectSoundCaptureCreate = DummyDirectSoundCaptureCreate;
+
+        paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateW =
+                (HRESULT (WINAPI *)(LPDSENUMCALLBACKW, LPVOID))
+                GetProcAddress( paWinDsDSoundEntryPoints.hInstance_, "DirectSoundCaptureEnumerateW" );
+        if( paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateW == NULL )
+            paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateW = DummyDirectSoundCaptureEnumerateW;
+
+        paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateA =
+                (HRESULT (WINAPI *)(LPDSENUMCALLBACKA, LPVOID))
+                GetProcAddress( paWinDsDSoundEntryPoints.hInstance_, "DirectSoundCaptureEnumerateA" );
+        if( paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateA == NULL )
+            paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateA = DummyDirectSoundCaptureEnumerateA;
+    }
+    else
+    {
+        /* initialize with dummy entry points to make live easy when ds isn't present */
+        paWinDsDSoundEntryPoints.DirectSoundCreate = DummyDirectSoundCreate;
+        paWinDsDSoundEntryPoints.DirectSoundEnumerateW = DummyDirectSoundEnumerateW;
+        paWinDsDSoundEntryPoints.DirectSoundEnumerateA = DummyDirectSoundEnumerateA;
+        paWinDsDSoundEntryPoints.DirectSoundCaptureCreate = DummyDirectSoundCaptureCreate;
+        paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateW = DummyDirectSoundCaptureEnumerateW;
+        paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateA = DummyDirectSoundCaptureEnumerateA;
+    }
+}
+
+
+void PaWinDs_TerminateDSoundEntryPoints(void)
+{
+    if( paWinDsDSoundEntryPoints.hInstance_ != NULL )
+    {
+        /* ensure that we crash reliably if the entry points arent initialised */
+        paWinDsDSoundEntryPoints.DirectSoundCreate = 0;
+        paWinDsDSoundEntryPoints.DirectSoundEnumerateW = 0;
+        paWinDsDSoundEntryPoints.DirectSoundEnumerateA = 0;
+        paWinDsDSoundEntryPoints.DirectSoundCaptureCreate = 0;
+        paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateW = 0;
+        paWinDsDSoundEntryPoints.DirectSoundCaptureEnumerateA = 0;
+
+        FreeLibrary( paWinDsDSoundEntryPoints.hInstance_ );
+        paWinDsDSoundEntryPoints.hInstance_ = NULL;
+    }
+}

3rdparty/iaxclient/lib/portaudio/src/hostapi/dsound/pa_win_ds_dynlink.h

@@ -0,0 +1,82 @@
+#ifndef INCLUDED_PA_DSOUND_DYNLINK_H
+#define INCLUDED_PA_DSOUND_DYNLINK_H
+/*
+ * Interface for dynamically loading directsound and providing a dummy
+ * implementation if it isn't present.
+ *
+ * Author: Ross Bencina (some portions Phil Burk & Robert Marsanyi
+ *
+ * For PortAudio Portable Real-Time Audio Library
+ * For more information see: http://www.portaudio.com
+ * Copyright (c) 1999-2006 Phil Burk, Robert Marsanyi and Ross Bencina
+ *
+ * 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
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version.
+ *
+ * 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.
+ *
+ */
+
+/* on Borland compilers, WIN32 doesn't seem to be defined by default, which
+    breaks DSound.h. Adding the define here fixes the problem. - rossb. */
+#ifdef __BORLANDC__
+#if !defined(WIN32)
+#define WIN32
+#endif
+#endif
+
+/*
+  We are only using DX3 in here, no need to polute the namespace - davidv
+*/
+#define DIRECTSOUND_VERSION 0x0300
+
+#include <DSound.h>
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+
+typedef struct
+{
+    HINSTANCE hInstance_;
+    
+    HRESULT (WINAPI *DirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
+    HRESULT (WINAPI *DirectSoundEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
+    HRESULT (WINAPI *DirectSoundEnumerateA)(LPDSENUMCALLBACKA, LPVOID);
+
+    HRESULT (WINAPI *DirectSoundCaptureCreate)(LPGUID, LPDIRECTSOUNDCAPTURE *, LPUNKNOWN);
+    HRESULT (WINAPI *DirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
+    HRESULT (WINAPI *DirectSoundCaptureEnumerateA)(LPDSENUMCALLBACKA, LPVOID);
+}PaWinDsDSoundEntryPoints;
+
+extern PaWinDsDSoundEntryPoints paWinDsDSoundEntryPoints;
+
+void PaWinDs_InitializeDSoundEntryPoints(void);
+void PaWinDs_TerminateDSoundEntryPoints(void);
+
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#endif /* INCLUDED_PA_DSOUND_DYNLINK_H */

3rdparty/iaxclient/lib/portaudio/src/hostapi/oss/recplay.c

@@ -0,0 +1,114 @@
+/*
+ * recplay.c
+ * Phil Burk
+ * Minimal record and playback test.
+ * 
+ */
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#ifndef __STDC__
+/* #include <getopt.h> */
+#endif /* __STDC__ */
+#include <fcntl.h>
+#ifdef __STDC__
+#include <string.h>
+#else /* __STDC__ */
+#include <strings.h>
+#endif /* __STDC__ */
+#include <sys/soundcard.h>
+
+#define NUM_BYTES   (64*1024)
+#define BLOCK_SIZE   (4*1024)
+
+#define AUDIO "/dev/dsp"
+
+char buffer[NUM_BYTES];
+
+int audioDev = 0;
+
+main (int argc, char *argv[])
+{
+    int   numLeft;
+    char *ptr;
+    int   num;
+    int   samplesize;
+
+    /********** RECORD ********************/
+    /* Open audio device. */
+    audioDev = open (AUDIO, O_RDONLY, 0);
+    if (audioDev == -1)
+    {
+        perror (AUDIO);
+        exit (-1);
+    }
+
+    /* Set to 16 bit samples. */
+    samplesize = 16;
+    ioctl(audioDev, SNDCTL_DSP_SAMPLESIZE, &samplesize);
+    if (samplesize != 16)
+    {
+        perror("Unable to set the sample size.");
+        exit(-1);
+    }
+
+    /* Record in blocks */
+    printf("Begin recording.\n");
+    numLeft = NUM_BYTES;
+    ptr = buffer;
+    while( numLeft >= BLOCK_SIZE )
+    {
+        if ( (num = read (audioDev, ptr, BLOCK_SIZE)) < 0 )
+        {
+            perror (AUDIO);
+            exit (-1);
+        }
+        else
+        {
+            printf("Read %d bytes\n", num);
+            ptr += num;
+            numLeft -= num;
+        }
+    }
+
+    close( audioDev );
+
+    /********** PLAYBACK ********************/
+    /* Open audio device for writing. */
+    audioDev = open (AUDIO, O_WRONLY, 0);
+    if (audioDev == -1)
+    {
+        perror (AUDIO);
+        exit (-1);
+    }
+
+    /* Set to 16 bit samples. */
+    samplesize = 16;
+    ioctl(audioDev, SNDCTL_DSP_SAMPLESIZE, &samplesize);
+    if (samplesize != 16)
+    {
+        perror("Unable to set the sample size.");
+        exit(-1);
+    }
+
+    /* Play in blocks */
+    printf("Begin playing.\n");
+    numLeft = NUM_BYTES;
+    ptr = buffer;
+    while( numLeft >= BLOCK_SIZE )
+    {
+        if ( (num = write (audioDev, ptr, BLOCK_SIZE)) < 0 )
+        {
+            perror (AUDIO);
+            exit (-1);
+        }
+        else
+        {
+            printf("Wrote %d bytes\n", num);
+            ptr += num;
+            numLeft -= num;
+        }
+    }
+
+    close( audioDev );
+}

3rdparty/iaxclient/lib/portaudio/src/hostapi/wdmks/readme.txt

@@ -0,0 +1,82 @@
+Notes about WDM-KS host API
+---------------------------
+
+Status history
+--------------
+10th November 2005:
+Made following changes:
+ * OpenStream: Try all PaSampleFormats internally if the the chosen
+     format is not supported natively.  This fixed several problems
+     with soundcards that soundcards that did not take kindly to
+     using 24-bit 3-byte formats.
+ * OpenStream: Make the minimum framesPerHostIBuffer (and framesPerHostOBuffer)
+     the default frameSize for the playback/recording pin.
+ * ProcessingThread: Added a switch to only call PaUtil_EndBufferProcessing
+     if the total input frames equals the total output frames
+
+5th September 2004:
+This is the first public version of the code. It should be considered
+an alpha release with zero guarantee not to crash on any particular 
+system. So far it has only been tested in the author's development
+environment, which means a Win2k/SP2 PIII laptop with integrated 
+SoundMAX driver and USB Tascam US-428 compiled with both MinGW
+(GCC 3.3) and MSVC++6 using the MS DirectX 9 SDK.
+It has been most widely tested with the MinGW build, with most of the
+test programs (particularly paqa_devs and paqa_errs) passing.
+There are some notable failures: patest_out_underflow and both of the
+blocking I/O tests (as blocking I/O is not implemented).
+At this point the code needs to be tested with a much wider variety 
+of configurations and feedback provided from testers regarding
+both working and failing cases.
+
+What is the WDM-KS host API?
+----------------------------
+PortAudio for Windows currently has 3 functional host implementations.
+MME uses the oldest Windows audio API which does not offer good
+play/record latency. 
+DirectX improves this, but still imposes a penalty
+of 10s of milliseconds due to the system mixing of streams from
+multiple applications. 
+ASIO offers very good latency, but requires special drivers which are
+not always available for cheaper audio hardware. Also, when ASIO 
+drivers are available, they are not always so robust because they 
+bypass all of the standardised Windows device driver architecture 
+and hit the hardware their own way.
+Alternatively there are a couple of free (but closed source) ASIO 
+implementations which connect to the lower level Windows 
+"Kernel Streaming" API, but again these require special installation 
+by the user, and can be limited in functionality or difficult to use. 
+
+This is where the PortAudio "WDM-KS" host implementation comes in.
+It directly connects PortAudio to the same Kernel Streaming API which
+those ASIO bridges use. This avoids the mixing penatly of DirectX, 
+giving at least as good latency as any ASIO driver, but it has the
+advantage of working with ANY Windows audio hardware which is available
+through the normal MME/DirectX routes without the user requiring 
+any additional device drivers to be installed, and allowing all 
+device selection to be done through the normal PortAudio API.
+
+Note that in general you should only be using this host API if your 
+application has a real requirement for very low latency audio (<20ms), 
+either because you are generating sounds in real-time based upon 
+user input, or you a processing recorded audio in real time.
+
+The only thing to be aware of is that using the KS interface will
+block that device from being used by the rest of system through
+the higher level APIs, or conversely, if the system is using
+a device, the KS API will not be able to use it. MS recommend that
+you should keep the device open only when your application has focus.
+In PortAudio terms, this means having a stream Open on a WDMKS device.
+
+Usage
+-----
+To add the WDMKS backend to your program which is already using 
+PortAudio, you must undefine PA_NO_WDMKS from your build file,
+and include the pa_win_wdmks\pa_win_wdmks.c into your build.
+The file should compile in both C and C++.
+You will need a DirectX SDK installed on your system for the
+ks.h and ksmedia.h header files.
+You will need to link to the system "setupapi" library.
+Note that if you use MinGW, you will get more warnings from 
+the DX header files when using GCC(C), and still a few warnings
+with G++(CPP).