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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 us on the PortAudio mailing list.
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 src/common/ and platform specific files from src/os/win/ (for Win32)
or src/os/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.de/329+M52087573ab0.html
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
You may also need to adjust your include paths to support inclusion of
header files from the above directories.
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 compilers 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;
}
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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 */

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// ****************************************************************************
// 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 */

3965
src/hostapi/asio/pa_asio.cpp Normal file

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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 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).
Additionally, the user may define a "channel mapping" by calling
paSetupMacCoreChannelMap() on their stream info structure before opening
the stream with it. See below for creating a channel map.
Known issues:
- Buffering: No buffering beyond that provided by core audio is provided
except where absolutely needed for the implementation to work. This may cause
issues with large framesPerBuffer settings and it also means that no additional
latency will be provided even if a large latency setting is selected.
- Latency: Latency settings are generally ignored. They may be used as a
hint for buffer size in paHostFramesPerBufferUnspecified, or the value may
be used in cases where additional buffering is needed, such as doing input and
output on seperate devices. 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. In the case of paHostFramesPerBuffer, an
appropriate framesPerBuffer value will be used that guarantees minimum
requested latency if that's possible.
- Timing info. It reports on stream time, but I'm probably doing something
wrong since patest_sine_time often reports negative latency numbers. Also,
there are currently issues with some devices whehn plugging/unplugging
devices.
- 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: should work fine.
- multichannel: It has been tested successfully on multichannel hardware
from MOTU: traveler and 896HD. Also Presonus firepod and others. It is
believed to work with all Core Audio devices, including virtual devices
such as soundflower.
- 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 rate conversion quality
as "documented" here:
http://lists.apple.com/archives/coreaudio-api/2004/Jan/msg00141.html
- x86/Universal Binary: Universal binaries can be build.
Creating a channel map:
How to create the map array - Text taken From AUHAL.rtfd :
[3] Channel Maps
Clients can tell the AUHAL units which channels of the device they are interested in. For example, the client may be processing stereo data, but outputting to a six-channel device. This is done by using the kAudioOutputUnitProperty_ChannelMap property. To use this property:
For Output:
Create an array of SInt32 that is the size of the number of channels of the device (Get the Format of the AUHAL's output Element == 0)
Initialize each of the array's values to -1 (-1 indicates that that channel is NOT to be presented in the conversion.)
Next, for each channel of your app's output, set:
channelMapArray[deviceOutputChannel] = desiredAppOutputChannel.
For example: we have a 6 channel output device and our application has a stereo source it wants to provide to the device. Suppose we want that stereo source to go to the 3rd and 4th channels of the device. The channel map would look like this: { -1, -1, 0, 1, -1, -1 }
Where the formats are:
Input Element == 0: 2 channels (- client format - settable)
Output Element == 0: 6 channels (- device format - NOT settable)
So channel 2 (zero-based) of the device will take the first channel of output and channel 3 will take the second channel of output. (This translates to the 3rd and 4th plugs of the 6 output plugs of the device of course!)
For Input:
Create an array of SInt32 that is the size of the number of channels of the format you require for input. Get (or Set in this case as needed) the AUHAL's output Element == 1.
Next, for each channel of input you require, set:
channelMapArray[desiredAppInputChannel] = deviceOutputChannel;
For example: we have a 6 channel input device from which we wish to receive stereo input from the 3rd and 4th channels. The channel map looks like this: { 2, 3 }
Where the formats are:
Input Element == 0: 2 channels (- device format - NOT settable)
Output Element == 0: 6 channels (- client format - settable)
----------------------------------------
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.

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/*
* Implementation of the PortAudio API for Apple AUHAL
*
* PortAudio Portable Real-Time Audio Library
* Latest Version at: http://www.portaudio.com
*
* 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.
*
* 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.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire PortAudio license; however,
* the PortAudio community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
/**
@file
@ingroup hostapi_src
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>
#ifdef MOSX_USE_NON_ATOMIC_FLAG_BITS
# define OSAtomicOr32( a, b ) ( (*(b)) |= (a) )
# define OSAtomicAnd32( a, b ) ( (*(b)) &= (a) )
#else
# include <libkern/OSAtomic.h>
#endif
/*
* 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;
}
}
/*
* Same as computeSampleSizeFromFormat, except that if
* the size is not a power of two, it returns the next power of two up
*/
static size_t computeSampleSizeFromFormatPow2( PaSampleFormat format )
{
switch( format ) {
case paFloat32: return 4;
case paInt32: return 4;
case paInt24: return 4;
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;
OSStatus err;
/* 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->ringBufferFrames = ringBufferSize;
blio->inputSampleFormat = inputSampleFormat;
blio->inputSampleSizeActual = computeSampleSizeFromFormat(inputSampleFormat);
blio->inputSampleSizePow2 = computeSampleSizeFromFormatPow2(inputSampleFormat);
blio->outputSampleFormat = outputSampleFormat;
blio->outputSampleSizeActual = computeSampleSizeFromFormat(outputSampleFormat);
blio->outputSampleSizePow2 = computeSampleSizeFromFormatPow2(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->inputSampleSizePow2*inChan );
if( !data )
{
result = paInsufficientMemory;
goto error;
}
err = PaUtil_InitializeRingBuffer(
&blio->inputRingBuffer,
1, ringBufferSize*blio->inputSampleSizePow2*inChan,
data );
assert( !err );
}
if( outChan ) {
data = calloc( ringBufferSize, blio->outputSampleSizePow2*outChan );
if( !data )
{
result = paInsufficientMemory;
goto error;
}
err = PaUtil_InitializeRingBuffer(
&blio->outputRingBuffer,
1, ringBufferSize*blio->outputSampleSizePow2*outChan,
data );
assert( !err );
}
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 ) {
PaUtil_FlushRingBuffer( &blio->outputRingBuffer );
bzero( blio->outputRingBuffer.buffer,
blio->outputRingBuffer.bufferSize );
/* Advance buffer */
PaUtil_AdvanceRingBufferWriteIndex( &blio->outputRingBuffer, blio->ringBufferFrames*blio->outputSampleSizeActual*blio->outChan );
//PaUtil_AdvanceRingBufferWriteIndex( &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 ) {
PaUtil_FlushRingBuffer( &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;
long read;
long written;
/* set flags returned by OS: */
OSAtomicOr32( statusFlags, &blio->statusFlags ) ;
/* --- Handle Input Buffer --- */
if( blio->inChan ) {
avail = PaUtil_GetRingBufferWriteAvailable( &blio->inputRingBuffer );
/* check for underflow */
if( avail < frameCount * blio->inputSampleSizeActual * blio->inChan )
OSAtomicOr32( paInputOverflow, &blio->statusFlags );
toRead = MIN( avail, frameCount * blio->inputSampleSizeActual * blio->inChan );
/* copy the data */
/*printf( "reading %d\n", toRead );*/
read = PaUtil_WriteRingBuffer( &blio->inputRingBuffer, input, toRead );
assert( toRead == read );
#ifdef PA_MAC__BLIO_MUTEX
/* Priority inversion. See notes below. */
blioSetIsInputEmpty( blio, false );
#endif
}
/* --- Handle Output Buffer --- */
if( blio->outChan ) {
avail = PaUtil_GetRingBufferReadAvailable( &blio->outputRingBuffer );
/* check for underflow */
if( avail < frameCount * blio->outputSampleSizeActual * blio->outChan )
OSAtomicOr32( paOutputUnderflow, &blio->statusFlags );
toWrite = MIN( avail, frameCount * blio->outputSampleSizeActual * blio->outChan );
if( toWrite != frameCount * blio->outputSampleSizeActual * blio->outChan )
bzero( ((char *)output)+toWrite,
frameCount * blio->outputSampleSizeActual * blio->outChan - toWrite );
/* copy the data */
/*printf( "writing %d\n", toWrite );*/
written = PaUtil_ReadRingBuffer( &blio->outputRingBuffer, output, toWrite );
assert( toWrite == written );
#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 = PaUtil_GetRingBufferReadAvailable( &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->inputSampleSizeActual * blio->inChan );
toRead -= toRead % blio->inputSampleSizeActual * blio->inChan ;
PaUtil_ReadRingBuffer( &blio->inputRingBuffer, (void *)cbuf, toRead );
cbuf += toRead;
frames -= toRead / ( blio->inputSampleSizeActual * 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( PaUtil_GetRingBufferReadAvailable( &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 ) {
OSAtomicAnd32( (uint32_t)(~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 = PaUtil_GetRingBufferWriteAvailable( &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->outputSampleSizeActual * blio->outChan );
toWrite -= toWrite % blio->outputSampleSizeActual * blio->outChan ;
PaUtil_WriteRingBuffer( &blio->outputRingBuffer, (void *)cbuf, toWrite );
cbuf += toWrite;
frames -= toWrite / ( blio->outputSampleSizeActual * 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( PaUtil_GetRingBufferWriteAvailable( &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 ) {
OSAtomicAnd32( (uint32_t)(~paOutputUnderflow), &blio->statusFlags );
ret = paOutputUnderflowed;
}
return ret;
}
/*
*
*/
void waitUntilBlioWriteBufferIsFlushed( PaMacBlio *blio )
{
if( blio->outputRingBuffer.buffer ) {
long avail = PaUtil_GetRingBufferWriteAvailable( &blio->outputRingBuffer );
while( avail != blio->outputRingBuffer.bufferSize ) {
if( avail == 0 )
Pa_Sleep( PA_MAC_BLIO_BUSY_WAIT_SLEEP_INTERVAL );
avail = PaUtil_GetRingBufferWriteAvailable( &blio->outputRingBuffer );
}
}
}
signed long GetStreamReadAvailable( PaStream* stream )
{
PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
VVDBUG(("GetStreamReadAvailable()\n"));
return PaUtil_GetRingBufferReadAvailable( &blio->inputRingBuffer )
/ ( blio->outputSampleSizeActual * blio->outChan );
}
signed long GetStreamWriteAvailable( PaStream* stream )
{
PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
VVDBUG(("GetStreamWriteAvailable()\n"));
return PaUtil_GetRingBufferWriteAvailable( &blio->outputRingBuffer )
/ ( blio->outputSampleSizeActual * blio->outChan );
}

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/*
* Internal blocking interfaces for PortAudio Apple AUHAL implementation
*
* PortAudio Portable Real-Time Audio Library
* Latest Version at: http://www.portaudio.com
*
* 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.
*
* 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.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire PortAudio license; however,
* the PortAudio community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
/**
@file
@ingroup hostapi_src
*/
#ifndef PA_MAC_CORE_BLOCKING_H_
#define PA_MAC_CORE_BLOCKING_H_
#include "pa_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 {
PaUtilRingBuffer inputRingBuffer;
PaUtilRingBuffer outputRingBuffer;
size_t ringBufferFrames;
PaSampleFormat inputSampleFormat;
size_t inputSampleSizeActual;
size_t inputSampleSizePow2;
PaSampleFormat outputSampleFormat;
size_t outputSampleSizeActual;
size_t outputSampleSizePow2;
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_*/

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/*
* Internal interfaces for PortAudio Apple AUHAL implementation
*
* PortAudio Portable Real-Time Audio Library
* Latest Version at: http://www.portaudio.com
*
* 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.
*
* 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.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire PortAudio license; however,
* the PortAudio community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
/**
@file pa_mac_core
@ingroup hostapi_src
@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 "pa_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. */
PaUtilRingBuffer 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;
/* FIXME: instead of volatile, these should be properly memory barriered */
volatile PaStreamCallbackFlags xrunFlags;
volatile bool isTimeSet;
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
because the user has called StopStream.
This state is just used internally;
externally it is indistinguishable from
ACTIVE.*/
ACTIVE = 3 /* The stream is active and running. */
} state;
double sampleRate;
//these may be different from the stream sample rate due to SR conversion:
double outDeviceSampleRate;
double inDeviceSampleRate;
}
PaMacCoreStream;
#endif /* PA_MAC_CORE_INTERNAL_H__ */

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/*
* $Id$
* 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.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire PortAudio license; however,
* the PortAudio community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
#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;
}

View file

@ -0,0 +1,738 @@
/*
* Helper and utility functions for pa_mac_core.c (Apple AUHAL implementation)
*
* PortAudio Portable Real-Time Audio Library
* Latest Version at: http://www.portaudio.com
*
* 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.
*
* 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.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire PortAudio license; however,
* the PortAudio community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
/**
@file
@ingroup hostapi_src
*/
#include "pa_mac_core_utilities.h"
#include "pa_mac_core_internal.h"
#include <libkern/OSAtomic.h>
#include <strings.h>
#include <pthread.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";
char str[20];
// see if it appears to be a 4-char-code
*(UInt32 *)(str + 1) = CFSwapInt32HostToBig(error);
if (isprint(str[1]) && isprint(str[2]) && isprint(str[3]) && isprint(str[4]))
{
str[0] = str[5] = '\'';
str[6] = '\0';
} else {
// no, format it as an integer
sprintf(str, "%d", (int)error);
}
DBUG(("%s on line %d: err='%s', msg=%s\n", errorType, line, str, 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,
UInt32 requestedFramesPerBuffer,
UInt32 *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;
}
/**********************
*
* XRun stuff
*
**********************/
struct PaMacXRunListNode_s {
PaMacCoreStream *stream;
struct PaMacXRunListNode_s *next;
} ;
typedef struct PaMacXRunListNode_s PaMacXRunListNode;
/** Always empty, so that it can always be the one returned by
addToXRunListenerList. note that it's not a pointer. */
static PaMacXRunListNode firstXRunListNode;
static int xRunListSize;
static pthread_mutex_t xrunMutex;
OSStatus xrunCallback(
AudioDeviceID inDevice,
UInt32 inChannel,
Boolean isInput,
AudioDevicePropertyID inPropertyID,
void* inClientData)
{
PaMacXRunListNode *node = (PaMacXRunListNode *) inClientData;
int ret = pthread_mutex_trylock( &xrunMutex ) ;
if( ret == 0 ) {
node = node->next ; //skip the first node
for( ; node; node=node->next ) {
PaMacCoreStream *stream = node->stream;
if( stream->state != ACTIVE )
continue; //if the stream isn't active, we don't care if the device is dropping
if( isInput ) {
if( stream->inputDevice == inDevice )
OSAtomicOr32( paInputOverflow, (uint32_t *)&(stream->xrunFlags) );
} else {
if( stream->outputDevice == inDevice )
OSAtomicOr32( paOutputUnderflow, (uint32_t *)&(stream->xrunFlags) );
}
}
pthread_mutex_unlock( &xrunMutex );
}
return 0;
}
int initializeXRunListenerList()
{
xRunListSize = 0;
bzero( (void *) &firstXRunListNode, sizeof(firstXRunListNode) );
return pthread_mutex_init( &xrunMutex, NULL );
}
int destroyXRunListenerList()
{
PaMacXRunListNode *node;
node = firstXRunListNode.next;
while( node ) {
PaMacXRunListNode *tmp = node;
node = node->next;
free( tmp );
}
xRunListSize = 0;
return pthread_mutex_destroy( &xrunMutex );
}
void *addToXRunListenerList( void *stream )
{
pthread_mutex_lock( &xrunMutex );
PaMacXRunListNode *newNode;
// setup new node:
newNode = (PaMacXRunListNode *) malloc( sizeof( PaMacXRunListNode ) );
newNode->stream = (PaMacCoreStream *) stream;
newNode->next = firstXRunListNode.next;
// insert:
firstXRunListNode.next = newNode;
pthread_mutex_unlock( &xrunMutex );
return &firstXRunListNode;
}
int removeFromXRunListenerList( void *stream )
{
pthread_mutex_lock( &xrunMutex );
PaMacXRunListNode *node, *prev;
prev = &firstXRunListNode;
node = firstXRunListNode.next;
while( node ) {
if( node->stream == stream ) {
//found it:
--xRunListSize;
prev->next = node->next;
free( node );
pthread_mutex_unlock( &xrunMutex );
return xRunListSize;
}
prev = prev->next;
node = node->next;
}
pthread_mutex_unlock( &xrunMutex );
// failure
return xRunListSize;
}

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/*
* Helper and utility functions for pa_mac_core.c (Apple AUHAL implementation)
*
* PortAudio Portable Real-Time Audio Library
* Latest Version at: http://www.portaudio.com
*
* 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.
*
* 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.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire PortAudio license; however,
* the PortAudio community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
/**
@file
@ingroup hostapi_src
*/
#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,
UInt32 requestedFramesPerBuffer,
UInt32 *actualFramesPerBuffer );
/*********************
*
* xrun handling
*
*********************/
OSStatus xrunCallback(
AudioDeviceID inDevice,
UInt32 inChannel,
Boolean isInput,
AudioDevicePropertyID inPropertyID,
void* inClientData ) ;
/** returns zero on success or a unix style error code. */
int initializeXRunListenerList();
/** returns zero on success or a unix style error code. */
int destroyXRunListenerList();
/**Returns the list, so that it can be passed to CorAudio.*/
void *addToXRunListenerList( void *stream );
/**Returns the number of Listeners in the list remaining.*/
int removeFromXRunListenerList( void *stream );
#endif /* PA_MAC_CORE_UTILITIES_H__*/

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/*
* 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.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire PortAudio license; however,
* the PortAudio community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
/**
@file
@ingroup hostapi_src
*/
#include "pa_win_ds_dynlink.h"
PaWinDsDSoundEntryPoints paWinDsDSoundEntryPoints = { 0, 0, 0, 0, 0, 0, 0 };
static HRESULT WINAPI DummyDllGetClassObject(REFCLSID rclsid, REFIID riid, LPVOID *ppv)
{
(void)rclsid; /* unused parameter */
(void)riid; /* unused parameter */
(void)ppv; /* unused parameter */
return CLASS_E_CLASSNOTAVAILABLE;
}
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.DllGetClassObject =
(HRESULT (WINAPI *)(REFCLSID, REFIID , LPVOID *))
GetProcAddress( paWinDsDSoundEntryPoints.hInstance_, "DllGetClassObject" );
if( paWinDsDSoundEntryPoints.DllGetClassObject == NULL )
paWinDsDSoundEntryPoints.DllGetClassObject = DummyDllGetClassObject;
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;
}
}

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/*
* 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.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire PortAudio license; however,
* the PortAudio community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
/**
@file
@ingroup hostapi_src
*/
#ifndef INCLUDED_PA_DSOUND_DYNLINK_H
#define INCLUDED_PA_DSOUND_DYNLINK_H
/* 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 *DllGetClassObject)(REFCLSID , REFIID , LPVOID *);
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 */

1761
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src/hostapi/oss/recplay.c Normal file
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/*
* 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 );
}

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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).

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