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137
portaudio-v19/pa_asio/ASIO-README.txt
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ASIO-README.txt
This document contains information to help you compile PortAudio with
ASIO support. If you find any omissions or errors in this document
please notify Ross Bencina <rossb@audiomulch.com>.
Building PortAudio with ASIO support
------------------------------------
To build PortAudio with ASIO support you need to compile and link with
pa_asio.c, and files from the ASIO SDK (see below), along with the common
files from pa_common/ and platform specific files from pa_win/ (for Win32)
or pa_mac/ (for Macintosh).
If you are compiling with a non-Microsoft compiler on windows, also
compile and link with iasiothiscallresolver.cpp (see below for
an explanation).
For some platforms (MingW, possibly Mac), you may simply
be able to type:
./configure --with-host_os=mingw --with-winapi=asio [--with-asiodir=/usr/local/asiosdk2]
make
./configure --with-host_os=darwin --with-winapi=asio [--with-asiodir=/usr/local/asiosdk2]
make
and life will be good.
Obtaining the ASIO SDK
----------------------
In order to build PortAudio with ASIO support, you need to download
the ASIO SDK (version 2.0) from Steinberg. Steinberg makes the ASIO
SDK available to anyone free of charge, however they do not permit its
source code to be distributed.
NOTE: In some cases the ASIO SDK may require patching, see below
for further details.
http://www.steinberg.net/en/ps/support/3rdparty/asio_sdk/
If the above link is broken search Google for:
"download steinberg ASIO SDK"
Building the ASIO SDK on Macintosh
----------------------------------
To build the ASIO SDK on Macintosh you need to compile and link with the
following files from the ASIO SDK:
host/asiodrivers.cpp
host/mac/asioshlib.cpp
host/mac/codefragements.cpp
Building the ASIO SDK on Windows
--------------------------------
To build the ASIO SDK on Windows you need to compile and link with the
following files from the ASIO SDK:
asio_sdk\common\asio.cpp
asio_sdk\host\asiodrivers.cpp
asio_sdk\host\pc\asiolist.cpp
You may also need to adjust your include paths to support inclusion of
header files from the above directories.
The ASIO SDK depends on the following COM API functions:
CoInitialize, CoUninitialize, CoCreateInstance, CLSIDFromString
For compilation with MinGW you will need to link with -lole32, for
Borland link with Import32.lib.
Non-Microsoft (MSVC) Compilers on Windows including Borland and GCC
-------------------------------------------------------------------
Steinberg did not specify a calling convention in the IASIO interface
definition. This causes the Microsoft compiler to use the proprietary
thiscall convention which is not compatible with other compilers, such
as compilers from Borland (BCC and C++Builder) and GNU (gcc).
Steinberg's ASIO SDK will compile but crash on initialization if
compiled with a non-Microsoft compiler on Windows.
PortAudio solves this problem using the iasiothiscallresolver library
which is included in the distribution. When building ASIO support for
non-Microsoft compilers, be sure to compile and link with
iasiothiscallresolver.cpp. Note that iasiothiscallresolver includes
conditional directives which cause it to have no effect if it is
compiled with a Microsoft compiler, or on the Macintosh.
If you use configure and make (see above), this should be handled
automatically for you.
For further information about the IASIO thiscall problem see this page:
http://www.audiomulch.com/~rossb/code/calliasio
Macintosh ASIO SDK Bug Patch
----------------------------
There is a bug in the ASIO SDK that causes the Macintosh version to
often fail during initialization. Below is a patch that you can apply.
In codefragments.cpp replace getFrontProcessDirectory function with
the following one (GetFrontProcess replaced by GetCurrentProcess).
bool CodeFragments::getFrontProcessDirectory(void *specs)
{
FSSpec *fss = (FSSpec *)specs;
ProcessInfoRec pif;
ProcessSerialNumber psn;
memset(&psn,0,(long)sizeof(ProcessSerialNumber));
// if(GetFrontProcess(&psn) == noErr) // wrong !!!
if(GetCurrentProcess(&psn) == noErr) // correct !!!
{
pif.processName = 0;
pif.processAppSpec = fss;
pif.processInfoLength = sizeof(ProcessInfoRec);
if(GetProcessInformation(&psn, &pif) == noErr)
return true;
}
return false;
}
---

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portaudio-v19/pa_asio/iasiothiscallresolver.cpp
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/*
IASIOThiscallResolver.cpp see the comments in iasiothiscallresolver.h for
the top level description - this comment describes the technical details of
the implementation.
The latest version of this file is available from:
http://www.audiomulch.com/~rossb/code/calliasio
please email comments to Ross Bencina <rossb@audiomulch.com>
BACKGROUND
The IASIO interface declared in the Steinberg ASIO 2 SDK declares
functions with no explicit calling convention. This causes MSVC++ to default
to using the thiscall convention, which is a proprietary convention not
implemented by some non-microsoft compilers - notably borland BCC,
C++Builder, and gcc. MSVC++ is the defacto standard compiler used by
Steinberg. As a result of this situation, the ASIO sdk will compile with
any compiler, however attempting to execute the compiled code will cause a
crash due to different default calling conventions on non-Microsoft
compilers.
IASIOThiscallResolver solves the problem by providing an adapter class that
delegates to the IASIO interface using the correct calling convention
(thiscall). Due to the lack of support for thiscall in the Borland and GCC
compilers, the calls have been implemented in assembly language.
A number of macros are defined for thiscall function calls with different
numbers of parameters, with and without return values - it may be possible
to modify the format of these macros to make them work with other inline
assemblers.
THISCALL DEFINITION
A number of definitions of the thiscall calling convention are floating
around the internet. The following definition has been validated against
output from the MSVC++ compiler:
For non-vararg functions, thiscall works as follows: the object (this)
pointer is passed in ECX. All arguments are passed on the stack in
right to left order. The return value is placed in EAX. The callee
clears the passed arguments from the stack.
FINDING FUNCTION POINTERS FROM AN IASIO POINTER
The first field of a COM object is a pointer to its vtble. Thus a pointer
to an object implementing the IASIO interface also points to a pointer to
that object's vtbl. The vtble is a table of function pointers for all of
the virtual functions exposed by the implemented interfaces.
If we consider a variable declared as a pointer to IASO:
IASIO *theAsioDriver
theAsioDriver points to:
object implementing IASIO
{
IASIOvtbl *vtbl
other data
}
in other words, theAsioDriver points to a pointer to an IASIOvtbl
vtbl points to a table of function pointers:
IASIOvtbl ( interface IASIO : public IUnknown )
{
(IUnknown functions)
0 virtual HRESULT STDMETHODCALLTYPE (*QueryInterface)(REFIID riid, void **ppv) = 0;
4 virtual ULONG STDMETHODCALLTYPE (*AddRef)() = 0;
8 virtual ULONG STDMETHODCALLTYPE (*Release)() = 0;
(IASIO functions)
12 virtual ASIOBool (*init)(void *sysHandle) = 0;
16 virtual void (*getDriverName)(char *name) = 0;
20 virtual long (*getDriverVersion)() = 0;
24 virtual void (*getErrorMessage)(char *string) = 0;
28 virtual ASIOError (*start)() = 0;
32 virtual ASIOError (*stop)() = 0;
36 virtual ASIOError (*getChannels)(long *numInputChannels, long *numOutputChannels) = 0;
40 virtual ASIOError (*getLatencies)(long *inputLatency, long *outputLatency) = 0;
44 virtual ASIOError (*getBufferSize)(long *minSize, long *maxSize,
long *preferredSize, long *granularity) = 0;
48 virtual ASIOError (*canSampleRate)(ASIOSampleRate sampleRate) = 0;
52 virtual ASIOError (*getSampleRate)(ASIOSampleRate *sampleRate) = 0;
56 virtual ASIOError (*setSampleRate)(ASIOSampleRate sampleRate) = 0;
60 virtual ASIOError (*getClockSources)(ASIOClockSource *clocks, long *numSources) = 0;
64 virtual ASIOError (*setClockSource)(long reference) = 0;
68 virtual ASIOError (*getSamplePosition)(ASIOSamples *sPos, ASIOTimeStamp *tStamp) = 0;
72 virtual ASIOError (*getChannelInfo)(ASIOChannelInfo *info) = 0;
76 virtual ASIOError (*createBuffers)(ASIOBufferInfo *bufferInfos, long numChannels,
long bufferSize, ASIOCallbacks *callbacks) = 0;
80 virtual ASIOError (*disposeBuffers)() = 0;
84 virtual ASIOError (*controlPanel)() = 0;
88 virtual ASIOError (*future)(long selector,void *opt) = 0;
92 virtual ASIOError (*outputReady)() = 0;
};
The numbers in the left column show the byte offset of each function ptr
from the beginning of the vtbl. These numbers are used in the code below
to select different functions.
In order to find the address of a particular function, theAsioDriver
must first be dereferenced to find the value of the vtbl pointer:
mov eax, theAsioDriver
mov edx, [theAsioDriver] // edx now points to vtbl[0]
Then an offset must be added to the vtbl pointer to select a
particular function, for example vtbl+44 points to the slot containing
a pointer to the getBufferSize function.
Finally vtbl+x must be dereferenced to obtain the value of the function
pointer stored in that address:
call [edx+44] // call the function pointed to by
// the value in the getBufferSize field of the vtbl
SEE ALSO
Martin Fay's OpenASIO DLL at http://www.martinfay.com solves the same
problem by providing a new COM interface which wraps IASIO with an
interface that uses portable calling conventions. OpenASIO must be compiled
with MSVC, and requires that you ship the OpenASIO DLL with your
application.
ACKNOWLEDGEMENTS
Ross Bencina: worked out the thiscall details above, wrote the original
Borland asm macros, and a patch for asio.cpp (which is no longer needed).
Thanks to Martin Fay for introducing me to the issues discussed here,
and to Rene G. Ceballos for assisting with asm dumps from MSVC++.
Antti Silvast: converted the original calliasio to work with gcc and NASM
by implementing the asm code in a separate file.
Fraser Adams: modified the original calliasio containing the Borland inline
asm to add inline asm for gcc i.e. Intel syntax for Borland and AT&T syntax
for gcc. This seems a neater approach for gcc than to have a separate .asm
file and it means that we only need one version of the thiscall patch.
Fraser Adams: rewrote the original calliasio patch in the form of the
IASIOThiscallResolver class in order to avoid modifications to files from
the Steinberg SDK, which may have had potential licence issues.
Andrew Baldwin: contributed fixes for compatibility problems with more
recent versions of the gcc assembler.
*/
// We only need IASIOThiscallResolver at all if we are on Win32. For other
// platforms we simply bypass the IASIOThiscallResolver definition to allow us
// to be safely #include'd whatever the platform to keep client code portable
#if defined(WIN32) || defined(_WIN32) || defined(__WIN32__)
// If microsoft compiler we can call IASIO directly so IASIOThiscallResolver
// is not used.
#if !defined(_MSC_VER)
#include <new>
#include <assert.h>
// We have a mechanism in iasiothiscallresolver.h to ensure that asio.h is
// #include'd before it in client code, we do NOT want to do this test here.
#define iasiothiscallresolver_sourcefile 1
#include "iasiothiscallresolver.h"
#undef iasiothiscallresolver_sourcefile
// iasiothiscallresolver.h redefines ASIOInit for clients, but we don't want
// this macro defined in this translation unit.
#undef ASIOInit
// theAsioDriver is a global pointer to the current IASIO instance which the
// ASIO SDK uses to perform all actions on the IASIO interface. We substitute
// our own forwarding interface into this pointer.
extern IASIO* theAsioDriver;
// The following macros define the inline assembler for BORLAND first then gcc
#if defined(__BCPLUSPLUS__) || defined(__BORLANDC__)
#define CALL_THISCALL_0( resultName, thisPtr, funcOffset )\
void *this_ = (thisPtr); \
__asm { \
mov ecx, this_ ; \
mov eax, [ecx] ; \
call [eax+funcOffset] ; \
mov resultName, eax ; \
}
#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 )\
void *this_ = (thisPtr); \
__asm { \
mov eax, param1 ; \
push eax ; \
mov ecx, this_ ; \
mov eax, [ecx] ; \
call [eax+funcOffset] ; \
}
#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 )\
void *this_ = (thisPtr); \
__asm { \
mov eax, param1 ; \
push eax ; \
mov ecx, this_ ; \
mov eax, [ecx] ; \
call [eax+funcOffset] ; \
mov resultName, eax ; \
}
#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 )\
void *this_ = (thisPtr); \
void *doubleParamPtr_ (&param1); \
__asm { \
mov eax, doubleParamPtr_ ; \
push [eax+4] ; \
push [eax] ; \
mov ecx, this_ ; \
mov eax, [ecx] ; \
call [eax+funcOffset] ; \
mov resultName, eax ; \
}
#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 )\
void *this_ = (thisPtr); \
__asm { \
mov eax, param2 ; \
push eax ; \
mov eax, param1 ; \
push eax ; \
mov ecx, this_ ; \
mov eax, [ecx] ; \
call [eax+funcOffset] ; \
mov resultName, eax ; \
}
#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\
void *this_ = (thisPtr); \
__asm { \
mov eax, param4 ; \
push eax ; \
mov eax, param3 ; \
push eax ; \
mov eax, param2 ; \
push eax ; \
mov eax, param1 ; \
push eax ; \
mov ecx, this_ ; \
mov eax, [ecx] ; \
call [eax+funcOffset] ; \
mov resultName, eax ; \
}
#elif defined(__GNUC__)
#define CALL_THISCALL_0( resultName, thisPtr, funcOffset ) \
__asm__ __volatile__ ("movl (%1), %%edx\n\t" \
"call *"#funcOffset"(%%edx)\n\t" \
:"=a"(resultName) /* Output Operands */ \
:"c"(thisPtr) /* Input Operands */ \
); \
#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 ) \
__asm__ __volatile__ ("pushl %0\n\t" \
"movl (%1), %%edx\n\t" \
"call *"#funcOffset"(%%edx)\n\t" \
: /* Output Operands */ \
:"r"(param1), /* Input Operands */ \
"c"(thisPtr) \
); \
#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 ) \
__asm__ __volatile__ ("pushl %1\n\t" \
"movl (%2), %%edx\n\t" \
"call *"#funcOffset"(%%edx)\n\t" \
:"=a"(resultName) /* Output Operands */ \
:"r"(param1), /* Input Operands */ \
"c"(thisPtr) \
); \
#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 ) \
__asm__ __volatile__ ("pushl 4(%1)\n\t" \
"pushl (%1)\n\t" \
"movl (%2), %%edx\n\t" \
"call *"#funcOffset"(%%edx);\n\t" \
:"=a"(resultName) /* Output Operands */ \
:"a"(&param1), /* Input Operands */ \
/* Note: Using "r" above instead of "a" fails */ \
/* when using GCC 3.3.3, and maybe later versions*/\
"c"(thisPtr) \
); \
#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 ) \
__asm__ __volatile__ ("pushl %1\n\t" \
"pushl %2\n\t" \
"movl (%3), %%edx\n\t" \
"call *"#funcOffset"(%%edx)\n\t" \
:"=a"(resultName) /* Output Operands */ \
:"r"(param2), /* Input Operands */ \
"r"(param1), \
"c"(thisPtr) \
); \
#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\
__asm__ __volatile__ ("pushl %1\n\t" \
"pushl %2\n\t" \
"pushl %3\n\t" \
"pushl %4\n\t" \
"movl (%5), %%edx\n\t" \
"call *"#funcOffset"(%%edx)\n\t" \
:"=a"(resultName) /* Output Operands */ \
:"r"(param4), /* Input Operands */ \
"r"(param3), \
"r"(param2), \
"r"(param1), \
"c"(thisPtr) \
); \
#endif
// Our static singleton instance.
IASIOThiscallResolver IASIOThiscallResolver::instance;
// Constructor called to initialize static Singleton instance above. Note that
// it is important not to clear that_ incase it has already been set by the call
// to placement new in ASIOInit().
IASIOThiscallResolver::IASIOThiscallResolver()
{
}
// Constructor called from ASIOInit() below
IASIOThiscallResolver::IASIOThiscallResolver(IASIO* that)
: that_( that )
{
}
// Implement IUnknown methods as assert(false). IASIOThiscallResolver is not
// really a COM object, just a wrapper which will work with the ASIO SDK.
// If you wanted to use ASIO without the SDK you might want to implement COM
// aggregation in these methods.
HRESULT STDMETHODCALLTYPE IASIOThiscallResolver::QueryInterface(REFIID riid, void **ppv)
{
(void)riid; // suppress unused variable warning
assert( false ); // this function should never be called by the ASIO SDK.
*ppv = NULL;
return E_NOINTERFACE;
}
ULONG STDMETHODCALLTYPE IASIOThiscallResolver::AddRef()
{
assert( false ); // this function should never be called by the ASIO SDK.
return 1;
}
ULONG STDMETHODCALLTYPE IASIOThiscallResolver::Release()
{
assert( false ); // this function should never be called by the ASIO SDK.
return 1;
}
// Implement the IASIO interface methods by performing the vptr manipulation
// described above then delegating to the real implementation.
ASIOBool IASIOThiscallResolver::init(void *sysHandle)
{
ASIOBool result;
CALL_THISCALL_1( result, that_, 12, sysHandle );
return result;
}
void IASIOThiscallResolver::getDriverName(char *name)
{
CALL_VOID_THISCALL_1( that_, 16, name );
}
long IASIOThiscallResolver::getDriverVersion()
{
ASIOBool result;
CALL_THISCALL_0( result, that_, 20 );
return result;
}
void IASIOThiscallResolver::getErrorMessage(char *string)
{
CALL_VOID_THISCALL_1( that_, 24, string );
}
ASIOError IASIOThiscallResolver::start()
{
ASIOBool result;
CALL_THISCALL_0( result, that_, 28 );
return result;
}
ASIOError IASIOThiscallResolver::stop()
{
ASIOBool result;
CALL_THISCALL_0( result, that_, 32 );
return result;
}
ASIOError IASIOThiscallResolver::getChannels(long *numInputChannels, long *numOutputChannels)
{
ASIOBool result;
CALL_THISCALL_2( result, that_, 36, numInputChannels, numOutputChannels );
return result;
}
ASIOError IASIOThiscallResolver::getLatencies(long *inputLatency, long *outputLatency)
{
ASIOBool result;
CALL_THISCALL_2( result, that_, 40, inputLatency, outputLatency );
return result;
}
ASIOError IASIOThiscallResolver::getBufferSize(long *minSize, long *maxSize,
long *preferredSize, long *granularity)
{
ASIOBool result;
CALL_THISCALL_4( result, that_, 44, minSize, maxSize, preferredSize, granularity );
return result;
}
ASIOError IASIOThiscallResolver::canSampleRate(ASIOSampleRate sampleRate)
{
ASIOBool result;
CALL_THISCALL_1_DOUBLE( result, that_, 48, sampleRate );
return result;
}
ASIOError IASIOThiscallResolver::getSampleRate(ASIOSampleRate *sampleRate)
{
ASIOBool result;
CALL_THISCALL_1( result, that_, 52, sampleRate );
return result;
}
ASIOError IASIOThiscallResolver::setSampleRate(ASIOSampleRate sampleRate)
{
ASIOBool result;
CALL_THISCALL_1_DOUBLE( result, that_, 56, sampleRate );
return result;
}
ASIOError IASIOThiscallResolver::getClockSources(ASIOClockSource *clocks, long *numSources)
{
ASIOBool result;
CALL_THISCALL_2( result, that_, 60, clocks, numSources );
return result;
}
ASIOError IASIOThiscallResolver::setClockSource(long reference)
{
ASIOBool result;
CALL_THISCALL_1( result, that_, 64, reference );
return result;
}
ASIOError IASIOThiscallResolver::getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp)
{
ASIOBool result;
CALL_THISCALL_2( result, that_, 68, sPos, tStamp );
return result;
}
ASIOError IASIOThiscallResolver::getChannelInfo(ASIOChannelInfo *info)
{
ASIOBool result;
CALL_THISCALL_1( result, that_, 72, info );
return result;
}
ASIOError IASIOThiscallResolver::createBuffers(ASIOBufferInfo *bufferInfos,
long numChannels, long bufferSize, ASIOCallbacks *callbacks)
{
ASIOBool result;
CALL_THISCALL_4( result, that_, 76, bufferInfos, numChannels, bufferSize, callbacks );
return result;
}
ASIOError IASIOThiscallResolver::disposeBuffers()
{
ASIOBool result;
CALL_THISCALL_0( result, that_, 80 );
return result;
}
ASIOError IASIOThiscallResolver::controlPanel()
{
ASIOBool result;
CALL_THISCALL_0( result, that_, 84 );
return result;
}
ASIOError IASIOThiscallResolver::future(long selector,void *opt)
{
ASIOBool result;
CALL_THISCALL_2( result, that_, 88, selector, opt );
return result;
}
ASIOError IASIOThiscallResolver::outputReady()
{
ASIOBool result;
CALL_THISCALL_0( result, that_, 92 );
return result;
}
// Implement our substitute ASIOInit() method
ASIOError IASIOThiscallResolver::ASIOInit(ASIODriverInfo *info)
{
// To ensure that our instance's vptr is correctly constructed, even if
// ASIOInit is called prior to main(), we explicitly call its constructor
// (potentially over the top of an existing instance). Note that this is
// pretty ugly, and is only safe because IASIOThiscallResolver has no
// destructor and contains no objects with destructors.
new((void*)&instance) IASIOThiscallResolver( theAsioDriver );
// Interpose between ASIO client code and the real driver.
theAsioDriver = &instance;
// Note that we never need to switch theAsioDriver back to point to the
// real driver because theAsioDriver is reset to zero in ASIOExit().
// Delegate to the real ASIOInit
return ::ASIOInit(info);
}
#endif /* !defined(_MSC_VER) */
#endif /* Win32 */

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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 */

2958
portaudio-v19/pa_asio/pa_asio.cpp
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122
portaudio-v19/pa_asio/pa_asio.h
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#ifndef PA_ASIO_H
#define PA_ASIO_H
/*
* $Id$
* PortAudio Portable Real-Time Audio Library
* ASIO specific extensions
*
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
/** @file
@brief ASIO-specific PortAudio API extension header file.
*/
#include "portaudio.h"
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
/** Retrieve legal latency settings for the specificed device, in samples.
@param device The global index of the device about which the query is being made.
@param minLatency A pointer to the location which will recieve the minimum latency value.
@param maxLatency A pointer to the location which will recieve the maximum latency value.
@param preferredLatency A pointer to the location which will recieve the preferred latency value.
@param granularity A pointer to the location which will recieve the granularity. This value
determines which values between minLatency and maxLatency are available. ie the step size,
if granularity is -1 then available latency settings are powers of two.
@see ASIOGetBufferSize in the ASIO SDK.
@todo This function should have a better name, any suggestions?
*/
PaError PaAsio_GetAvailableLatencyValues( PaDeviceIndex device,
long *minLatency, long *maxLatency, long *preferredLatency, long *granularity );
/** Display the ASIO control panel for the specified device.
@param device The global index of the device whose control panel is to be displayed.
@param systemSpecific On Windows, the calling application's main window handle,
on Macintosh this value should be zero.
*/
PaError PaAsio_ShowControlPanel( PaDeviceIndex device, void* systemSpecific );
/** Retrieve a pointer to a string containing the name of the specified
input channel. The string is valid until Pa_Terminate is called.
The string will be no longer than 32 characters including the null terminator.
*/
PaError PaAsio_GetInputChannelName( PaDeviceIndex device, int channelIndex,
const char** channelName );
/** Retrieve a pointer to a string containing the name of the specified
input channel. The string is valid until Pa_Terminate is called.
The string will be no longer than 32 characters including the null terminator.
*/
PaError PaAsio_GetOutputChannelName( PaDeviceIndex device, int channelIndex,
const char** channelName );
#define paAsioUseChannelSelectors (0x01)
typedef struct PaAsioStreamInfo{
unsigned long size; /**< sizeof(PaAsioStreamInfo) */
PaHostApiTypeId hostApiType; /**< paASIO */
unsigned long version; /**< 1 */
unsigned long flags;
/* Support for opening only specific channels of an ASIO device.
If the paAsioUseChannelSelectors flag is set, channelSelectors is a
pointer to an array of integers specifying the device channels to use.
When used, the length of the channelSelectors array must match the
corresponding channelCount parameter to Pa_OpenStream() otherwise a
crash may result.
The values in the selectors array must specify channels within the
range of supported channels for the device or paInvalidChannelCount will
result.
*/
int *channelSelectors;
}PaAsioStreamInfo;
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* PA_ASIO_H */

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portaudio-v19/pa_mac/pa_mac_hostapis.c
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@ -1,79 +0,0 @@
/*
* $Id$
* Portable Audio I/O Library Windows initialization table
*
* Based on the Open Source API proposed by Ross Bencina
* Copyright (c) 1999-2002 Ross Bencina, Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/** @file
Mac OS host API initialization function table.
*/
#include "pa_hostapi.h"
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
PaError PaSkeleton_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
PaError PaMacSm_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
PaError PaJack_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
PaError PaMacAsio_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
#ifdef __cplusplus
}
#endif /* __cplusplus */
PaUtilHostApiInitializer *paHostApiInitializers[] =
{
#ifdef PA_USE_COREAUDIO
PaMacCore_Initialize,
#endif
#ifdef PA_USE_SM
PaMacSm_Initialize,
#endif
#ifdef PA_USE_JACK
PaJack_Initialize,
#endif
#ifdef PA_USE_ASIO
PaMacAsio_Initialize,
#endif
PaSkeleton_Initialize, /* just for testing */
0 /* NULL terminated array */
};
int paDefaultHostApiIndex = 0;

60
portaudio-v19/pa_tests/README.txt
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@ -1,60 +0,0 @@
This directory contains various programs to test PortAudio. The files
named patest_* are tests, the files named debug_* are just scratch
files that may or may not work.
All following tests are up to date with the V19 API. They should all compile
(without any warnings on GCC 3.3). Note that this does not necissarily mean that
the tests pass, just that they compile.
x- paqa_devs.c
x- paqa_errs.c (needs reviewing)
x- patest1.c
x- patest_buffer.c
x- patest_callbackstop.c
x- patest_clip.c (last test fails, dither doesn't currently force clip in V19)
x- patest_dither.c
x- patest_hang.c
x- patest_latency.c
x- patest_leftright.c
x- patest_longsine.c
x- patest_many.c
x- patest_maxsines.c
o- patest_mono.c
x- patest_multi_sine.c
x- patest_pink.c
x- patest_prime.c
x- patest_read_record.c
x- patest_record.c
x- patest_ringmix.c
x- patest_saw.c
x- patest_sine.c
x- patest_sine8.c
x- patest_sine_formats.c
x- patest_sine_time.c
x- patest_start_stop.c
x- patest_stop.c
x- patest_sync.c
x- patest_toomanysines.c
o- patest_two_rates.c
x- patest_underflow.c
x- patest_wire.c
x- patest_write_sine.c
x- pa_devs.c
x- pa_fuzz.c
x- pa_minlat.c
The debug_ files are still in V18 format and may need some V19 adaption.
Feel free to fix them, most simply require adjusting to the new API.
o- pa_tests/debug_convert.c
o- pa_tests/debug_dither_calc.c
o- pa_tests/debug_dual.c
o- pa_tests/debug_multi_in.c
o- pa_tests/debug_multi_out.c
o- pa_tests/debug_record.c
o- pa_tests/debug_record_reuse.c
o- pa_tests/debug_sine.c
o- pa_tests/debug_sine_amp.c
o- pa_tests/debug_sine_formats.c
o- pa_tests/debug_srate.c
o- pa_tests/debug_test1.c

131
portaudio-v19/pa_tests/debug_convert.c
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@ -1,131 +0,0 @@
/*
* $Id$
* Convert tagged values.
*
* Author: Phil Burk <philburk@softsynth.com>
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID())
//#define OUTPUT_DEVICE (11)
#define NUM_SECONDS (8)
#define SLEEP_DUR (800)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (256)
#define NUM_BUFFERS (0)
typedef struct
{
unsigned int framesToGo;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
short *out = (short*)outputBuffer;
int i;
int finished = 0;
(void) outTime; /* Prevent unused variable warnings. */
(void) inputBuffer;
if( data->framesToGo < framesPerBuffer ) finished = 1;
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = 0x0000 + i; /* left */
*out++ = 0x1000 + i; /* right */
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *stream;
PaError err;
paTestData data;
int i;
int totalSamps;
printf("PortAudio Test: output debug values\n" );
data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */
printf("totalSamps = %d\n", totalSamps );
err = Pa_Initialize();
if( err != paNoError ) goto error;
printf("PortAudio Test: output device = %d\n", OUTPUT_DEVICE );
err = Pa_OpenStream(
&stream,
paNoDevice,
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
OUTPUT_DEVICE,
2, /* stereo output */
paInt16, /* 32 bit floating point output */
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
NUM_BUFFERS, /* number of buffers, if zero then use default minimum */
paClipOff|paDitherOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Is callback being called?\n");
for( i=0; i<((NUM_SECONDS+1)*1000); i+=SLEEP_DUR )
{
printf("data.framesToGo = %d\n", data.framesToGo ); fflush(stdout);
Pa_Sleep( SLEEP_DUR );
}
/* Stop sound until ENTER hit. */
printf("Call Pa_StopStream()\n");
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

55
portaudio-v19/pa_tests/debug_dither_calc.c
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@ -1,55 +0,0 @@
/*
* $Id$
* Test Dither calculations.
*
* Author: Phil Burk http://www.softsynth.com
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#include "pa_host.h"
/*******************************************************************/
int main(void);
int main(void)
{
long max,min;
int i;
for( i=0; i<10000; i++ )
{
long dither = PaConvert_TriangularDither();
// printf("dither = 0x%08X\n", dither );
if( dither < min ) min = dither;
else if( dither > max ) max = dither;
}
printf("min = 0x%08X = %d, max = 0x%08X = %d\n", min, min, max, max );
}

183
portaudio-v19/pa_tests/debug_dual.c
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@ -1,183 +0,0 @@
/*
* $Id$
* debug_dual.c
* Try to open TWO streams on separate cards.
* Play a sine sweep using the Portable Audio api for several seconds.
* Hacked test for debugging PA.
*
* Author: Phil Burk <philburk@softsynth.com>
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define DEV_ID_1 (13)
#define DEV_ID_2 (15)
#define NUM_SECONDS (8)
#define SLEEP_DUR (800)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (256)
#if 0
#define MIN_LATENCY_MSEC (200)
#define NUM_BUFFERS ((MIN_LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000))
#else
#define NUM_BUFFERS (0)
#endif
#define MIN_FREQ (100.0f)
#define MAX_FREQ (4000.0f)
#define FREQ_SCALAR (1.00002f)
#define CalcPhaseIncrement(freq) (freq/SAMPLE_RATE)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (400)
typedef struct
{
float sine[TABLE_SIZE + 1]; // add one for guard point for interpolation
float phase_increment;
float left_phase;
float right_phase;
}
paTestData;
/* Convert phase between and 1.0 to sine value
* using linear interpolation.
*/
float LookupSine( paTestData *data, float phase );
float LookupSine( paTestData *data, float phase )
{
float fIndex = phase*TABLE_SIZE;
int index = (int) fIndex;
float fract = fIndex - index;
float lo = data->sine[index];
float hi = data->sine[index+1];
float val = lo + fract*(hi-lo);
return val;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
int finished = 0;
(void) outTime; /* Prevent unused variable warnings. */
(void) inputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = LookupSine(data, data->left_phase); /* left */
*out++ = LookupSine(data, data->right_phase); /* right */
data->left_phase += data->phase_increment;
if( data->left_phase >= 1.0f ) data->left_phase -= 1.0f;
data->right_phase += (data->phase_increment * 1.5f); /* fifth above */
if( data->right_phase >= 1.0f ) data->right_phase -= 1.0f;
/* sweep frequency then start over. */
data->phase_increment *= FREQ_SCALAR;
if( data->phase_increment > CalcPhaseIncrement(MAX_FREQ) ) data->phase_increment = CalcPhaseIncrement(MIN_FREQ);
}
return 0;
}
PaError TestStart( PortAudioStream **streamPtr, PaDeviceID devID,
paTestData *data );
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *stream1, *stream2;
PaError err;
paTestData DATA1, DATA2;
printf("PortAudio Test: DUAL sine sweep. ask for %d buffers\n", NUM_BUFFERS );
err = Pa_Initialize();
if( err != paNoError ) goto error;
err = TestStart( &stream1, DEV_ID_1, &DATA1 );
if( err != paNoError ) goto error;
err = TestStart( &stream2, DEV_ID_2, &DATA2 );
if( err != paNoError ) goto error;
printf("Hit ENTER\n");
getchar();
err = Pa_StopStream( stream1 );
if( err != paNoError ) goto error;
err = Pa_StopStream( stream2 );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}
PaError TestStart( PortAudioStream **streamPtr, PaDeviceID devID, paTestData *data )
{
PortAudioStream *stream;
PaError err;
int i;
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data->sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data->sine[TABLE_SIZE] = data->sine[0]; // set guard point
data->left_phase = data->right_phase = 0.0;
data->phase_increment = CalcPhaseIncrement(MIN_FREQ);
printf("PortAudio Test: output device = %d\n", devID );
err = Pa_OpenStream(
&stream,
paNoDevice,
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
devID,
2, /* stereo output */
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
NUM_BUFFERS, /* number of buffers, if zero then use default minimum */
paClipOff|paDitherOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
*streamPtr = stream;
return 0;
error:
return err;
}

179
portaudio-v19/pa_tests/debug_multi_in.c
View File

@ -1,179 +0,0 @@
/*
* $Id$
* debug_multi_in.c
* Pass output from each of multiple channels
* to a stereo output using the Portable Audio api.
* Hacked test for debugging PA.
*
* Author: Phil Burk http://www.softsynth.com
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include <string.h>
#include "portaudio.h"
//#define INPUT_DEVICE_NAME ("EWS88 MT Interleaved Rec")
#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID())
//#define OUTPUT_DEVICE (18)
#define SAMPLE_RATE (22050)
#define FRAMES_PER_BUFFER (256)
#define MIN_LATENCY_MSEC (400)
#define NUM_BUFFERS ((MIN_LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000))
#ifndef M_PI
#define M_PI (3.14159265)
#endif
typedef struct
{
int liveChannel;
int numChannels;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
float *in = (float*)inputBuffer;
int i;
int finished = 0;
(void) outTime; /* Prevent unused variable warnings. */
(void) inputBuffer;
if( in == NULL ) return 0;
for( i=0; i<(int)framesPerBuffer; i++ )
{
/* Copy one channel of input to output. */
*out++ = in[data->liveChannel];
*out++ = in[data->liveChannel];
in += data->numChannels;
}
return 0;
}
/*******************************************************************/
int PaFindDeviceByName( const char *name )
{
int i;
int numDevices;
const PaDeviceInfo *pdi;
int len = strlen( name );
PaDeviceID result = paNoDevice;
numDevices = Pa_CountDevices();
for( i=0; i<numDevices; i++ )
{
pdi = Pa_GetDeviceInfo( i );
if( strncmp( name, pdi->name, len ) == 0 )
{
result = i;
break;
}
}
return result;
}
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *stream;
PaError err;
paTestData data;
int i;
PaDeviceID inputDevice;
const PaDeviceInfo *pdi;
printf("PortAudio Test: input signal from each channel. %d buffers\n", NUM_BUFFERS );
data.liveChannel = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
#ifdef INPUT_DEVICE_NAME
printf("Try to use device: %s\n", INPUT_DEVICE_NAME );
inputDevice = PaFindDeviceByName(INPUT_DEVICE_NAME);
if( inputDevice == paNoDevice )
{
printf("Could not find %s. Using default instead.\n", INPUT_DEVICE_NAME );
inputDevice = Pa_GetDefaultInputDeviceID();
}
#else
printf("Using default input device.\n");
inputDevice = Pa_GetDefaultInputDeviceID();
#endif
pdi = Pa_GetDeviceInfo( inputDevice );
if( pdi == NULL )
{
printf("Could not get device info!\n");
goto error;
}
data.numChannels = pdi->maxInputChannels;
printf("Input Device name is %s\n", pdi->name );
printf("Input Device has %d channels.\n", pdi->maxInputChannels);
err = Pa_OpenStream(
&stream,
inputDevice,
pdi->maxInputChannels,
paFloat32, /* 32 bit floating point input */
NULL,
OUTPUT_DEVICE,
2,
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
NUM_BUFFERS, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
data.liveChannel = 0;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
for( i=0; i<data.numChannels; i++ )
{
data.liveChannel = i;
printf("Channel %d being sent to output. Hit ENTER for next channel.", i );
fflush(stdout);
getchar();
}
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

144
portaudio-v19/pa_tests/debug_multi_out.c
View File

@ -1,144 +0,0 @@
/*
* $Id$
* debug_multi_out.c
* Play a different sine wave on each channels,
* using the Portable Audio api.
*
* Author: Phil Burk http://www.softsynth.com
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID())
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (256)
#define FREQ_INCR (300.0 / SAMPLE_RATE)
#define MAX_CHANNELS (64)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
typedef struct
{
int numChannels;
double phases[MAX_CHANNELS];
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
int frameIndex, channelIndex;
int finished = 0;
(void) outTime; /* Prevent unused variable warnings. */
(void) inputBuffer;
for( frameIndex=0; frameIndex<(int)framesPerBuffer; frameIndex++ )
{
for( channelIndex=0; channelIndex<data->numChannels; channelIndex++ )
{
/* Output sine wave on every channel. */
*out++ = (float) sin(data->phases[channelIndex]);
/* Play each channel at a higher frequency. */
data->phases[channelIndex] += FREQ_INCR * (4 + channelIndex);
if( data->phases[channelIndex] >= (2.0 * M_PI) ) data->phases[channelIndex] -= (2.0 * M_PI);
}
}
return 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *stream;
PaError err;
const PaDeviceInfo *pdi;
paTestData data = {0};
printf("PortAudio Test: output sine wave on each channel.\n" );
err = Pa_Initialize();
if( err != paNoError ) goto error;
pdi = Pa_GetDeviceInfo( OUTPUT_DEVICE );
data.numChannels = pdi->maxOutputChannels;
if( data.numChannels > MAX_CHANNELS ) data.numChannels = MAX_CHANNELS;
printf("Number of Channels = %d\n", data.numChannels );
err = Pa_OpenStream(
&stream,
paNoDevice, /* default input device */
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
OUTPUT_DEVICE,
data.numChannels,
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
0, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Hit ENTER to stop sound.\n");
fflush(stdout);
getchar();
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
Pa_CloseStream( stream );
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

339
portaudio-v19/pa_tests/debug_record.c
View File

@ -1,339 +0,0 @@
/*
* $Id$
* debug_record.c
* Record input into an array.
* Save array to a file.
* Based on patest_record.c but with various ugly debug hacks thrown in.
*
* Author: Phil Burk http://www.softsynth.com
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <memory.h>
#include "portaudio.h"
#define SAMPLE_RATE (22050)
#define NUM_SECONDS (10)
#define SLEEP_DUR_MSEC (200)
#define FRAMES_PER_BUFFER (1<<10)
#define NUM_REC_BUFS (0)
#if 1
#define PA_SAMPLE_TYPE paFloat32
typedef float SAMPLE;
#else
#define PA_SAMPLE_TYPE paInt16
typedef short SAMPLE;
#endif
typedef struct
{
long frameIndex; /* Index into sample array. */
long maxFrameIndex;
long samplesPerFrame;
long numSamples;
SAMPLE *recordedSamples;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int recordCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
SAMPLE *rptr = (SAMPLE*)inputBuffer;
SAMPLE *wptr = &data->recordedSamples[data->frameIndex * data->samplesPerFrame];
long framesToCalc;
unsigned long i;
int finished;
unsigned long framesLeft = data->maxFrameIndex - data->frameIndex;
(void) outputBuffer; /* Prevent unused variable warnings. */
(void) outTime;
if( framesLeft < framesPerBuffer )
{
framesToCalc = framesLeft;
finished = 1;
}
else
{
framesToCalc = framesPerBuffer;
finished = 0;
}
if( inputBuffer == NULL )
{
for( i=0; i<framesToCalc; i++ )
{
*wptr++ = 0; /* left */
*wptr++ = 0; /* right */
}
}
else
{
for( i=0; i<framesToCalc; i++ )
{
*wptr++ = *rptr++; /* left */
*wptr++ = *rptr++; /* right */
}
}
data->frameIndex += framesToCalc;
return finished;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int playCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
SAMPLE *rptr = &data->recordedSamples[data->frameIndex * data->samplesPerFrame];
SAMPLE *wptr = (SAMPLE*)outputBuffer;
unsigned long i;
int finished;
unsigned int framesLeft = data->maxFrameIndex - data->frameIndex;
if( outputBuffer == NULL ) return 0;
(void) inputBuffer; /* Prevent unused variable warnings. */
(void) outTime;
if( framesLeft < framesPerBuffer )
{
/* final buffer... */
for( i=0; i<framesLeft; i++ )
{
*wptr++ = *rptr++; /* left */
*wptr++ = *rptr++; /* right */
}
for( ; i<framesPerBuffer; i++ )
{
*wptr++ = 0; /* left */
*wptr++ = 0; /* right */
}
data->frameIndex += framesLeft;
finished = 1;
}
else
{
for( i=0; i<framesPerBuffer; i++ )
{
*wptr++ = *rptr++; /* left */
*wptr++ = *rptr++; /* right */
}
data->frameIndex += framesPerBuffer;
finished = 0;
}
return finished;
}
/****************************************************************/
PaError TestRecording( paTestData *dataPtr )
{
PortAudioStream *stream;
PaError err;
int i;
/* Record some audio. */
err = Pa_OpenStream(
&stream,
Pa_GetDefaultInputDeviceID(),
dataPtr->samplesPerFrame, /* stereo input */
PA_SAMPLE_TYPE,
NULL,
paNoDevice,
0,
PA_SAMPLE_TYPE,
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
NUM_REC_BUFS, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
recordCallback,
dataPtr );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Now recording!\n"); fflush(stdout);
for( i=0; i<(NUM_SECONDS*1000/SLEEP_DUR_MSEC); i++ )
{
if( Pa_StreamActive( stream ) <= 0)
{
printf("Stream inactive!\n");
break;
}
if( dataPtr->maxFrameIndex <= dataPtr->frameIndex )
{
printf("Buffer recording complete.\n");
break;
}
Pa_Sleep(100);
printf("index = %d\n", dataPtr->frameIndex ); fflush(stdout);
}
printf("Finished loop. Close stream.\n"); fflush(stdout);
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
printf("Done.\n"); fflush(stdout);
{
SAMPLE max = 0;
SAMPLE posVal;
int i;
for( i=0; i<dataPtr->numSamples; i++ )
{
posVal = dataPtr->recordedSamples[i];
if( posVal < 0 ) posVal = -posVal;
if( posVal > max ) max = posVal;
}
printf("Largest recorded sample = %d\n", max );
}
/* Write recorded data to a file. */
#if 0
{
FILE *fid;
fid = fopen("recorded.raw", "wb");
if( fid == NULL )
{
printf("Could not open file.");
}
else
{
fwrite( dataPtr->recordedSamples, dataPtr->samplesPerFrame * sizeof(SAMPLE), totalFrames, fid );
fclose( fid );
printf("Wrote data to 'recorded.raw'\n");
}
}
#endif
error:
return err;
}
/****************************************************************/
PaError TestPlayback( paTestData *dataPtr )
{
PortAudioStream *stream;
PaError err;
int i;
/* Playback recorded data. */
dataPtr->frameIndex = 0;
printf("Begin playback.\n"); fflush(stdout);
err = Pa_OpenStream(
&stream,
paNoDevice,
0, /* NO input */
PA_SAMPLE_TYPE,
NULL,
Pa_GetDefaultOutputDeviceID(),
dataPtr->samplesPerFrame, /* stereo output */
PA_SAMPLE_TYPE,
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
0, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
playCallback,
dataPtr );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Waiting for playback to finish.\n"); fflush(stdout);
for( i=0; i<(NUM_SECONDS*1000/SLEEP_DUR_MSEC); i++ )
{
Pa_Sleep(100);
printf("index = %d\n", dataPtr->frameIndex );
}
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
error:
return err;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError err;
paTestData data;
long totalFrames;
long numBytes;
long i;
printf("patest_record.c\n"); fflush(stdout);
data.frameIndex = 0;
data.samplesPerFrame = 2;
data.maxFrameIndex = totalFrames = NUM_SECONDS*SAMPLE_RATE;
printf("totalFrames = %d\n", totalFrames ); fflush(stdout);
data.numSamples = totalFrames * data.samplesPerFrame;
numBytes = data.numSamples * sizeof(SAMPLE);
data.recordedSamples = (SAMPLE *) malloc( numBytes );
if( data.recordedSamples == NULL )
{
printf("Could not allocate record array.\n");
exit(1);
}
for( i=0; i<data.numSamples; i++ ) data.recordedSamples[i] = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
for( i=0; i<2; i++ )
{
err = TestRecording( &data );
if( err != paNoError ) goto error;
err = TestPlayback( &data );
if( err != paNoError ) goto error;
}
free( data.recordedSamples );
Pa_Terminate();
return 0;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
if( err == paHostError )
{
fprintf( stderr, "Host Error number: %d\n", Pa_GetHostError() );
}
return -1;
}

351
portaudio-v19/pa_tests/debug_record_reuse.c
View File

@ -1,351 +0,0 @@
/*
* $Id$
* debug_record_reuse.c
* Record input into an array.
* Save array to a file.
* Based on patest_record.c but with various ugly debug hacks thrown in.
* Loop twice and reuse same streams.
*
* Author: Phil Burk http://www.softsynth.com
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <memory.h>
#include "portaudio.h"
#define SAMPLE_RATE (22050)
#define NUM_SECONDS (4)
#define SLEEP_DUR_MSEC (200)
#define FRAMES_PER_BUFFER (256)
#define NUM_REC_BUFS (0)
#if 1
#define PA_SAMPLE_TYPE paFloat32
typedef float SAMPLE;
#else
#define PA_SAMPLE_TYPE paInt16
typedef short SAMPLE;
#endif
typedef struct
{
long frameIndex; /* Index into sample array. */
long maxFrameIndex;
long samplesPerFrame;
long numSamples;
PortAudioStream *outputStream;
PortAudioStream *inputStream;
SAMPLE *recordedSamples;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int recordCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
SAMPLE *rptr = (SAMPLE*)inputBuffer;
SAMPLE *wptr = &data->recordedSamples[data->frameIndex * data->samplesPerFrame];
long framesToCalc;
unsigned long i;
int finished;
unsigned long framesLeft = data->maxFrameIndex - data->frameIndex;
(void) outputBuffer; /* Prevent unused variable warnings. */
(void) outTime;
if( framesLeft < framesPerBuffer )
{
framesToCalc = framesLeft;
finished = 1;
}
else
{
framesToCalc = framesPerBuffer;
finished = 0;
}
if( inputBuffer == NULL )
{
for( i=0; i<framesToCalc; i++ )
{
*wptr++ = 0; /* left */
*wptr++ = 0; /* right */
}
}
else
{
for( i=0; i<framesToCalc; i++ )
{
*wptr++ = *rptr++; /* left */
*wptr++ = *rptr++; /* right */
}
}
data->frameIndex += framesToCalc;
return finished;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int playCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
SAMPLE *rptr = &data->recordedSamples[data->frameIndex * data->samplesPerFrame];
SAMPLE *wptr = (SAMPLE*)outputBuffer;
unsigned long i;
int finished;
unsigned int framesLeft = data->maxFrameIndex - data->frameIndex;
if( outputBuffer == NULL ) return 0;
(void) inputBuffer; /* Prevent unused variable warnings. */
(void) outTime;
if( framesLeft < framesPerBuffer )
{
/* final buffer... */
for( i=0; i<framesLeft; i++ )
{
*wptr++ = *rptr++; /* left */
*wptr++ = *rptr++; /* right */
}
for( ; i<framesPerBuffer; i++ )
{
*wptr++ = 0; /* left */
*wptr++ = 0; /* right */
}
data->frameIndex += framesLeft;
finished = 1;
}
else
{
for( i=0; i<framesPerBuffer; i++ )
{
*wptr++ = *rptr++; /* left */
*wptr++ = *rptr++; /* right */
}
data->frameIndex += framesPerBuffer;
finished = 0;
}
return finished;
}
/****************************************************************/
PaError TestRecording( paTestData *dataPtr )
{
PaError err;
int i;
int lastIndex = 0;
/* Open input stream if not already open. */
if( dataPtr->inputStream == NULL )
{
/* Record some audio. */
err = Pa_OpenStream(
&dataPtr->inputStream,
Pa_GetDefaultInputDeviceID(),
dataPtr->samplesPerFrame, /* stereo input */
PA_SAMPLE_TYPE,
NULL,
paNoDevice,
0,
PA_SAMPLE_TYPE,
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
NUM_REC_BUFS, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
recordCallback,
dataPtr );
if( err != paNoError ) goto error;
}
dataPtr->frameIndex = 0;
err = Pa_StartStream( dataPtr->inputStream );
if( err != paNoError ) goto error;
printf("Now recording!\n"); fflush(stdout);
for( i=0; i<(NUM_SECONDS*1000/SLEEP_DUR_MSEC); i++ )
{
int frameIndex, delta;
Pa_Sleep(SLEEP_DUR_MSEC);
frameIndex = dataPtr->frameIndex;
if( Pa_StreamActive( dataPtr->inputStream ) <= 0)
{
printf("Stream inactive!\n");
break;
}
if( dataPtr->maxFrameIndex <= frameIndex )
{
printf("Buffer recording complete.\n");
break;
}
delta = frameIndex - lastIndex;
lastIndex = frameIndex;
printf("index = %d, delta = %d\n", frameIndex, delta ); fflush(stdout);
}
err = Pa_StopStream( dataPtr->inputStream );
if( err != paNoError ) goto error;
printf("Done.\n"); fflush(stdout);
error:
return err;
}
/****************************************************************/
PaError TestPlayback( paTestData *dataPtr )
{
PaError err;
int i;
int lastIndex = 0;
/* Playback recorded data. */
dataPtr->frameIndex = 0;
printf("Begin playback.\n"); fflush(stdout);
/* Open output stream if not already open. */
if( dataPtr->outputStream == NULL )
{
err = Pa_OpenStream(
&dataPtr->outputStream,
paNoDevice,
0, /* NO input */
PA_SAMPLE_TYPE,
NULL,
Pa_GetDefaultOutputDeviceID(),
dataPtr->samplesPerFrame, /* stereo output */
PA_SAMPLE_TYPE,
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
0, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
playCallback,
dataPtr );
if( err != paNoError ) goto error;
}
err = Pa_StartStream( dataPtr->outputStream );
if( err != paNoError ) goto error;
printf("Waiting for playback to finish.\n"); fflush(stdout);
for( i=0; i<(NUM_SECONDS*1000/SLEEP_DUR_MSEC); i++ )
{
int frameIndex, delta;
Pa_Sleep(SLEEP_DUR_MSEC);
frameIndex = dataPtr->frameIndex;
delta = frameIndex - lastIndex;
lastIndex = frameIndex;
printf("index = %d, delta = %d\n", frameIndex, delta ); fflush(stdout);
}
err = Pa_StopStream( dataPtr->outputStream );
if( err != paNoError ) goto error;
error:
return err;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError err;
paTestData data = { 0 };
long totalFrames;
long numBytes;
long i;
printf("patest_record.c\n"); fflush(stdout);
/* Set up test data structure and sample array. */
data.frameIndex = 0;
data.samplesPerFrame = 2;
data.maxFrameIndex = totalFrames = NUM_SECONDS*SAMPLE_RATE;
printf("totalFrames = %d\n", totalFrames ); fflush(stdout);
data.numSamples = totalFrames * data.samplesPerFrame;
numBytes = data.numSamples * sizeof(SAMPLE);
data.recordedSamples = (SAMPLE *) malloc( numBytes );
if( data.recordedSamples == NULL )
{
printf("Could not allocate record array.\n");
exit(1);
}
for( i=0; i<data.numSamples; i++ ) data.recordedSamples[i] = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
/* Record and playback multiple times. */
for( i=0; i<2; i++ )
{
err = TestRecording( &data );
if( err != paNoError ) goto error;
err = TestPlayback( &data );
if( err != paNoError ) goto error;
}
/* Clean up. */
err = Pa_CloseStream( data.inputStream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( data.outputStream );
if( err != paNoError ) goto error;
if( err != paNoError ) goto error;
free( data.recordedSamples );
Pa_Terminate();
printf("Test complete.\n"); fflush(stdout);
return 0;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
if( err == paHostError )
{
fprintf( stderr, "Host Error number: %d\n", Pa_GetHostError() );
}
return -1;
}

192
portaudio-v19/pa_tests/debug_sine.c
View File

@ -1,192 +0,0 @@
/*
* $Id$
* debug_sine.c
* Play a sine sweep using the Portable Audio api for several seconds.
* Hacked test for debugging PA.
*
* Author: Phil Burk <philburk@softsynth.com>
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID())
//#define OUTPUT_DEVICE (11)
#define NUM_SECONDS (8)
#define SLEEP_DUR (800)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (256)
#if 0
#define MIN_LATENCY_MSEC (200)
#define NUM_BUFFERS ((MIN_LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000))
#else
#define NUM_BUFFERS (0)
#endif
#define MIN_FREQ (100.0f)
#define MAX_FREQ (4000.0f)
#define FREQ_SCALAR (1.00002f)
#define CalcPhaseIncrement(freq) (freq/SAMPLE_RATE)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (400)
typedef struct
{
float sine[TABLE_SIZE + 1]; // add one for guard point for interpolation
float phase_increment;
float left_phase;
float right_phase;
unsigned int framesToGo;
}
paTestData;
/* Convert phase between and 1.0 to sine value
* using linear interpolation.
*/
float LookupSine( paTestData *data, float phase );
float LookupSine( paTestData *data, float phase )
{
float fIndex = phase*TABLE_SIZE;
int index = (int) fIndex;
float fract = fIndex - index;
float lo = data->sine[index];
float hi = data->sine[index+1];
float val = lo + fract*(hi-lo);
return val;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
int framesToCalc;
int i;
int finished = 0;
(void) outTime; /* Prevent unused variable warnings. */
(void) inputBuffer;
if( data->framesToGo < framesPerBuffer )
{
framesToCalc = data->framesToGo;
data->framesToGo = 0;
finished = 1;
}
else
{
framesToCalc = framesPerBuffer;
data->framesToGo -= framesPerBuffer;
}
for( i=0; i<framesToCalc; i++ )
{
*out++ = LookupSine(data, data->left_phase); /* left */
*out++ = LookupSine(data, data->right_phase); /* right */
data->left_phase += data->phase_increment;
if( data->left_phase >= 1.0f ) data->left_phase -= 1.0f;
data->right_phase += (data->phase_increment * 1.5f); /* fifth above */
if( data->right_phase >= 1.0f ) data->right_phase -= 1.0f;
/* sweep frequency then start over. */
data->phase_increment *= FREQ_SCALAR;
if( data->phase_increment > CalcPhaseIncrement(MAX_FREQ) ) data->phase_increment = CalcPhaseIncrement(MIN_FREQ);
}
/* zero remainder of final buffer */
for( ; i<(int)framesPerBuffer; i++ )
{
*out++ = 0; /* left */
*out++ = 0; /* right */
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *stream;
PaError err;
paTestData data;
int i;
int totalSamps;
printf("PortAudio Test: output sine sweep. ask for %d buffers\n", NUM_BUFFERS );
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data.sine[TABLE_SIZE] = data.sine[0]; // set guard point
data.left_phase = data.right_phase = 0.0;
data.phase_increment = CalcPhaseIncrement(MIN_FREQ);
data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */
printf("totalSamps = %d\n", totalSamps );
err = Pa_Initialize();
if( err != paNoError ) goto error;
printf("PortAudio Test: output device = %d\n", OUTPUT_DEVICE );
err = Pa_OpenStream(
&stream,
paNoDevice,
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
OUTPUT_DEVICE,
2, /* stereo output */
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
NUM_BUFFERS, /* number of buffers, if zero then use default minimum */
paClipOff|paDitherOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Is callback being called?\n");
for( i=0; i<((NUM_SECONDS+1)*1000); i+=SLEEP_DUR )
{
printf("data.framesToGo = %d\n", data.framesToGo ); fflush(stdout);
Pa_Sleep( SLEEP_DUR );
}
/* Stop sound until ENTER hit. */
printf("Call Pa_StopStream()\n");
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

157
portaudio-v19/pa_tests/debug_sine_amp.c
View File

@ -1,157 +0,0 @@
/*
* $Id$
* Play a different sine wave on each channels,
* using the Portable Audio api.
* Allos amplitude to be set interactively.
*
* Author: Phil Burk http://www.softsynth.com
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID())
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (256)
#define FREQ_INCR (300.0 / SAMPLE_RATE)
#define MAX_CHANNELS (64)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
typedef struct
{
int numChannels;
double phases[MAX_CHANNELS];
float amplitude;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
int frameIndex, channelIndex;
(void) outTime; /* Prevent unused variable warnings. */
(void) inputBuffer;
for( frameIndex=0; frameIndex<(int)framesPerBuffer; frameIndex++ )
{
for( channelIndex=0; channelIndex<data->numChannels; channelIndex++ )
{
/* Output sine wave on every channel. */
*out++ = (float) ( data->amplitude * sin(data->phases[channelIndex]) );
/* Play each channel at a higher frequency. */
data->phases[channelIndex] += FREQ_INCR * (4 + channelIndex);
if( data->phases[channelIndex] >= (2.0 * M_PI) ) data->phases[channelIndex] -= (2.0 * M_PI);
}
}
return 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
char pad[256];
PortAudioStream *stream;
PaError err;
const PaDeviceInfo *pdi;
paTestData data = {0};
printf("PortAudio Test: output sine wave on each channel.\n" );
err = Pa_Initialize();
if( err != paNoError ) goto error;
pdi = Pa_GetDeviceInfo( OUTPUT_DEVICE );
data.numChannels = pdi->maxOutputChannels;
if( data.numChannels > MAX_CHANNELS ) data.numChannels = MAX_CHANNELS;
printf("Number of Channels = %d\n", data.numChannels );
data.amplitude = 1.0;
err = Pa_OpenStream(
&stream,
paNoDevice, /* default input device */
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
OUTPUT_DEVICE,
data.numChannels,
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
0, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
do
{
printf("Current amplitude = %f\n", data.amplitude );
printf("Enter new amplitude or 'q' to quit.\n");
fflush(stdout);
gets( pad );
if( pad[0] != 'q' )
{
// I tried to use atof but it seems to be broken on Mac OS X 10.1
float amp;
sscanf( pad, "%f", &amp );
data.amplitude = amp;
}
} while( pad[0] != 'q' );
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
Pa_CloseStream( stream );
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

202
portaudio-v19/pa_tests/debug_sine_formats.c
View File

@ -1,202 +0,0 @@
/*
* $Id$
* patest_sine_formats.c
* Play a sine wave using the Portable Audio api for several seconds.
* Test various data formats.
*
* Author: Phil Burk
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (10)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (256)
#define LEFT_FREQ (SAMPLE_RATE/512.0) /* So we hit 1.0 */
#define RIGHT_FREQ (500.0)
#define AMPLITUDE (1.0)
/* Select ONE format for testing. */
#define TEST_UINT8 (1)
#define TEST_INT8 (0)
#define TEST_INT16 (0)
#define TEST_FLOAT32 (0)
#if TEST_UINT8
#define TEST_FORMAT paUInt8
typedef unsigned char SAMPLE_t;
#define SAMPLE_ZERO (0x80)
#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(127.0 * (x)))
#define FORMAT_NAME "Unsigned 8 Bit"
#elif TEST_INT8
#define TEST_FORMAT paInt8
typedef char SAMPLE_t;
#define SAMPLE_ZERO (0)
#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(127.0 * (x)))
#define FORMAT_NAME "Signed 8 Bit"
#elif TEST_INT16
#define TEST_FORMAT paInt16
typedef short SAMPLE_t;
#define SAMPLE_ZERO (0)
#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(32767 * (x)))
#define FORMAT_NAME "Signed 16 Bit"
#elif TEST_FLOAT32
#define TEST_FORMAT paFloat32
typedef float SAMPLE_t;
#define SAMPLE_ZERO (0.0)
#define DOUBLE_TO_SAMPLE(x) ((SAMPLE_t)(x))
#define FORMAT_NAME "Float 32 Bit"
#endif
#ifndef M_PI
#define M_PI (3.14159265)
#endif
typedef struct
{
double left_phase;
double right_phase;
unsigned int framesToGo;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
SAMPLE_t *out = (SAMPLE_t *)outputBuffer;
SAMPLE_t sample;
int i;
int framesToCalc;
int finished = 0;
(void) outTime; /* Prevent unused variable warnings. */
(void) inputBuffer;
if( data->framesToGo < framesPerBuffer )
{
framesToCalc = data->framesToGo;
data->framesToGo = 0;
finished = 1;
}
else
{
framesToCalc = framesPerBuffer;
data->framesToGo -= framesPerBuffer;
}
for( i=0; i<framesToCalc; i++ )
{
data->left_phase += (LEFT_FREQ / SAMPLE_RATE);
if( data->left_phase > 1.0) data->left_phase -= 1.0;
sample = DOUBLE_TO_SAMPLE( AMPLITUDE * sin( (data->left_phase * M_PI * 2. ))); /**/
*out++ = sample;
/* *out++ = sample; /**/
/* *out++ = 0; /**/
data->right_phase += (RIGHT_FREQ / SAMPLE_RATE);
if( data->right_phase > 1.0) data->right_phase -= 1.0;
*out++ = DOUBLE_TO_SAMPLE( AMPLITUDE * sin( (data->right_phase * M_PI * 2. ))); /**/
/* *out++ = 0; /* */
}
/* zero remainder of final buffer */
for( ; i<(int)framesPerBuffer; i++ )
{
*out++ = SAMPLE_ZERO; /* left */
*out++ = SAMPLE_ZERO; /* right */
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *stream;
PaError err;
paTestData data;
int totalSamps;
printf("PortAudio Test: output " FORMAT_NAME "\n");
data.left_phase = data.right_phase = 0.0;
data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */
err = Pa_Initialize();
if( err != paNoError ) goto error;
err = Pa_OpenStream(
&stream,
paNoDevice,/* default input device */
0, /* no input */
TEST_FORMAT,
NULL,
Pa_GetDefaultOutputDeviceID(), /* default output device */
2, /* stereo output */
TEST_FORMAT,
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
0, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Waiting %d seconds for sound to finish.\n", NUM_SECONDS );
while( Pa_StreamActive( stream ) ) Pa_Sleep(10);
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("PortAudio Test Finished: " FORMAT_NAME "\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

265
portaudio-v19/pa_tests/debug_srate.c
View File

@ -1,265 +0,0 @@
/*
* $Id$
* debug_record_reuse.c
* Record input into an array.
* Save array to a file.
* Based on patest_record.c but with various ugly debug hacks thrown in.
* Loop twice and reuse same streams.
*
* Author: Phil Burk http://www.softsynth.com
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <memory.h>
#include "portaudio.h"
#define EWS88MT_12_REC (1)
#define EWS88MT_12_PLAY (10)
#define SBLIVE_REC (2)
#define SBLIVE_PLAY (11)
#if 0
#define INPUT_DEVICE_ID Pa_GetDefaultInputDeviceID()
#define OUTPUT_DEVICE_ID Pa_GetDefaultOutputDeviceID()
#else
#define INPUT_DEVICE_ID (EWS88MT_12_REC)
#define OUTPUT_DEVICE_ID (SBLIVE_PLAY)
#endif
#define INPUT_SAMPLE_RATE (22050.0)
#define OUTPUT_SAMPLE_RATE (22050.0)
#define NUM_SECONDS (4)
#define SLEEP_DUR_MSEC (1000)
#define FRAMES_PER_BUFFER (64)
#define NUM_REC_BUFS (0)
#define SAMPLES_PER_FRAME (2)
#define PA_SAMPLE_TYPE paInt16
typedef short SAMPLE;
typedef struct
{
long frameIndex; /* Index into sample array. */
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int recordCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData *) userData;
(void) outputBuffer; /* Prevent unused variable warnings. */
(void) outTime;
if( inputBuffer != NULL )
{
data->frameIndex += framesPerBuffer;
}
return 0;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int playCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData *) userData;
(void) inputBuffer; /* Prevent unused variable warnings. */
(void) outTime;
if( outputBuffer != NULL )
{
data->frameIndex += framesPerBuffer;
}
return 0;
}
/****************************************************************/
PaError MeasureStreamRate( PortAudioStream *stream, paTestData *dataPtr, double *ratePtr )
{
PaError err;
int i;
int totalFrames = 0;
int totalMSec = 0;
dataPtr->frameIndex = 0;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
for( i=0; i<(NUM_SECONDS*1000/SLEEP_DUR_MSEC); i++ )
{
int delta, endIndex;
int startIndex = dataPtr->frameIndex;
Pa_Sleep(SLEEP_DUR_MSEC);
endIndex = dataPtr->frameIndex;
delta = endIndex - startIndex;
totalFrames += delta;
totalMSec += SLEEP_DUR_MSEC;
printf("index = %d, delta = %d\n", endIndex, delta ); fflush(stdout);
}
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
*ratePtr = (totalFrames * 1000.0) / totalMSec;
error:
return err;
}
void ReportRate( double measuredRate, double expectedRate )
{
double error;
error = (measuredRate - expectedRate) / expectedRate;
error = (error < 0 ) ? -error : error;
printf("Measured rate = %6.1f, expected rate = %6.1f\n",
measuredRate, expectedRate );
if( error > 0.1 )
{
printf("ERROR: unexpected rate! --------------------- ERROR!\n");
}
else
{
printf("SUCCESS: rate within tolerance!\n");
}
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError err;
paTestData data = { 0 };
long i;
double rate;
const PaDeviceInfo *pdi;
PortAudioStream *outputStream;
PortAudioStream *inputStream;
err = Pa_Initialize();
if( err != paNoError ) goto error;
pdi = Pa_GetDeviceInfo( INPUT_DEVICE_ID );
printf("Input device = %s\n", pdi->name );
pdi = Pa_GetDeviceInfo( OUTPUT_DEVICE_ID );
printf("Output device = %s\n", pdi->name );
/* Open input stream. */
err = Pa_OpenStream(
&inputStream,
INPUT_DEVICE_ID,
SAMPLES_PER_FRAME, /* stereo input */
PA_SAMPLE_TYPE,
NULL,
paNoDevice,
0,
PA_SAMPLE_TYPE,
NULL,
INPUT_SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
NUM_REC_BUFS, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
recordCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_OpenStream(
&outputStream,
paNoDevice,
0, /* NO input */
PA_SAMPLE_TYPE,
NULL,
OUTPUT_DEVICE_ID,
SAMPLES_PER_FRAME, /* stereo output */
PA_SAMPLE_TYPE,
NULL,
OUTPUT_SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
0, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
playCallback,
&data );
if( err != paNoError ) goto error;
/* Record and playback multiple times. */
for( i=0; i<2; i++ )
{
printf("Measuring INPUT ------------------------- \n");
err = MeasureStreamRate( inputStream, &data, &rate );
if( err != paNoError ) goto error;
ReportRate( rate, INPUT_SAMPLE_RATE );
printf("Measuring OUTPUT ------------------------- \n");
err = MeasureStreamRate( outputStream, &data, &rate );
if( err != paNoError ) goto error;
ReportRate( rate, OUTPUT_SAMPLE_RATE );
}
/* Clean up. */
err = Pa_CloseStream( inputStream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( outputStream );
if( err != paNoError ) goto error;
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test complete.\n"); fflush(stdout);
return 0;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
if( err == paHostError )
{
fprintf( stderr, "Host Error number: %d\n", Pa_GetHostError() );
}
return -1;
}

114
portaudio-v19/pa_tests/debug_test1.c
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@ -1,114 +0,0 @@
/*
* $Id$
patest1.c
Ring modulate the audio input with a 441hz sine wave for 20 seconds
using the Portable Audio api
Author: Ross Bencina <rossb@audiomulch.com>
Modifications:
April 5th, 2001 - PLB - Check for NULL inputBuffer.
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#ifndef M_PI
#define M_PI (3.14159265)
#endif
typedef struct
{
float sine[100];
int phase;
int sampsToGo;
}
patest1data;
static int patest1Callback( void *inputBuffer, void *outputBuffer,
unsigned long bufferFrames,
PaTimestamp outTime, void *userData )
{
patest1data *data = (patest1data*)userData;
float *in = (float*)inputBuffer;
float *out = (float*)outputBuffer;
int framesToCalc = bufferFrames;
unsigned long i;
int finished = 0;
if(inputBuffer == NULL) return 0;
if( data->sampsToGo < bufferFrames )
{
finished = 1;
}
for( i=0; i<bufferFrames; i++ )
{
*out++ = *in++;
*out++ = *in++;
if( data->phase >= 100 )
data->phase = 0;
}
data->sampsToGo -= bufferFrames;
/* zero remainder of final buffer if not already done */
for( ; i<bufferFrames; i++ )
{
*out++ = 0; /* left */
*out++ = 0; /* right */
}
return finished;
}
int main(int argc, char* argv[]);
int main(int argc, char* argv[])
{
PaStream *stream;
PaError err;
patest1data data;
int i;
int inputDevice = Pa_GetDefaultInputDeviceID();
int outputDevice = Pa_GetDefaultOutputDeviceID();
/* initialise sinusoidal wavetable */
for( i=0; i<100; i++ )
data.sine[i] = sin( ((double)i/100.) * M_PI * 2. );
data.phase = 0;
data.sampsToGo = 44100 * 4; // 20 seconds
/* initialise portaudio subsytem */
Pa_Initialize();
err = Pa_OpenStream(
&stream,
inputDevice,
2, /* stereo input */
paFloat32, /* 32 bit floating point input */
NULL,
outputDevice,
2, /* stereo output */
paFloat32, /* 32 bit floating point output */
NULL,
44100.,
// 22050, /* half second buffers */
// 4, /* four buffers */
512, /* half second buffers */
0, /* four buffers */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patest1Callback,
&data );
if( err == paNoError )
{
err = Pa_StartStream( stream );
// printf( "Press any key to end.\n" );
// getc( stdin ); //wait for input before exiting
// Pa_AbortStream( stream );
printf( "Waiting for stream to complete...\n" );
while( Pa_StreamActive( stream ) )
Pa_Sleep(1000); /* sleep until playback has finished */
err = Pa_CloseStream( stream );
}
else
{
fprintf( stderr, "An error occured while opening the portaudio stream\n" );
if( err == paHostError )
fprintf( stderr, "Host error number: %d\n", Pa_GetHostError() );
else
fprintf( stderr, "Error number: %d\n", err );
}
Pa_Terminate();
printf( "bye\n" );
return 0;
}

230
portaudio-v19/pa_tests/pa_devs.c
View File

@ -1,230 +0,0 @@
/** @file pa_devs.c
@brief List available devices, including device information.
@author Phil Burk http://www.softsynth.com
@note Define PA_NO_ASIO to compile this code on Windows without
ASIO support.
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#ifdef WIN32
#ifndef PA_NO_ASIO
#include "pa_asio.h"
#endif
#endif
/*******************************************************************/
static void PrintSupportedStandardSampleRates(
const PaStreamParameters *inputParameters,
const PaStreamParameters *outputParameters )
{
static double standardSampleRates[] = {
8000.0, 9600.0, 11025.0, 12000.0, 16000.0, 22050.0, 24000.0, 32000.0,
44100.0, 48000.0, 88200.0, 96000.0, 192000.0, -1 /* negative terminated list */
};
int i, printCount;
PaError err;
printCount = 0;
for( i=0; standardSampleRates[i] > 0; i++ )
{
err = Pa_IsFormatSupported( inputParameters, outputParameters, standardSampleRates[i] );
if( err == paFormatIsSupported )
{
if( printCount == 0 )
{
printf( "\t%8.2f", standardSampleRates[i] );
printCount = 1;
}
else if( printCount == 4 )
{
printf( ",\n\t%8.2f", standardSampleRates[i] );
printCount = 1;
}
else
{
printf( ", %8.2f", standardSampleRates[i] );
++printCount;
}
}
}
if( !printCount )
printf( "None\n" );
else
printf( "\n" );
}
/*******************************************************************/
int main(void);
int main(void)
{
int i, numDevices, defaultDisplayed;
const PaDeviceInfo *deviceInfo;
PaStreamParameters inputParameters, outputParameters;
PaError err;
Pa_Initialize();
printf( "PortAudio version number = %d\nPortAudio version text = '%s'\n",
Pa_GetVersion(), Pa_GetVersionText() );
numDevices = Pa_GetDeviceCount();
if( numDevices < 0 )
{
printf( "ERROR: Pa_CountDevices returned 0x%x\n", numDevices );
err = numDevices;
goto error;
}
printf( "Number of devices = %d\n", numDevices );
for( i=0; i<numDevices; i++ )
{
deviceInfo = Pa_GetDeviceInfo( i );
printf( "--------------------------------------- device #%d\n", i );
/* Mark global and API specific default devices */
defaultDisplayed = 0;
if( i == Pa_GetDefaultInputDevice() )
{
printf( "[ Default Input" );
defaultDisplayed = 1;
}
else if( i == Pa_GetHostApiInfo( deviceInfo->hostApi )->defaultInputDevice )
{
const PaHostApiInfo *hostInfo = Pa_GetHostApiInfo( deviceInfo->hostApi );
printf( "[ Default %s Input", hostInfo->name );
defaultDisplayed = 1;
}
if( i == Pa_GetDefaultOutputDevice() )
{
printf( (defaultDisplayed ? "," : "[") );
printf( " Default Output" );
defaultDisplayed = 1;
}
else if( i == Pa_GetHostApiInfo( deviceInfo->hostApi )->defaultOutputDevice )
{
const PaHostApiInfo *hostInfo = Pa_GetHostApiInfo( deviceInfo->hostApi );
printf( (defaultDisplayed ? "," : "[") );
printf( " Default %s Output", hostInfo->name );
defaultDisplayed = 1;
}
if( defaultDisplayed )
printf( " ]\n" );
/* print device info fields */
printf( "Name = %s\n", deviceInfo->name );
printf( "Host API = %s\n", Pa_GetHostApiInfo( deviceInfo->hostApi )->name );
printf( "Max inputs = %d", deviceInfo->maxInputChannels );
printf( ", Max outputs = %d\n", deviceInfo->maxOutputChannels );
printf( "Default low input latency = %8.3f\n", deviceInfo->defaultLowInputLatency );
printf( "Default low output latency = %8.3f\n", deviceInfo->defaultLowOutputLatency );
printf( "Default high input latency = %8.3f\n", deviceInfo->defaultHighInputLatency );
printf( "Default high output latency = %8.3f\n", deviceInfo->defaultHighOutputLatency );
#ifdef WIN32
#ifndef PA_NO_ASIO
/* ASIO specific latency information */
if( Pa_GetHostApiInfo( deviceInfo->hostApi )->type == paASIO ){
long minLatency, maxLatency, preferredLatency, granularity;
err = PaAsio_GetAvailableLatencyValues( i,
&minLatency, &maxLatency, &preferredLatency, &granularity );
printf( "ASIO minimum buffer size = %ld\n", minLatency );
printf( "ASIO maximum buffer size = %ld\n", maxLatency );
printf( "ASIO preferred buffer size = %ld\n", preferredLatency );
if( granularity == -1 )
printf( "ASIO buffer granularity = power of 2\n" );
else
printf( "ASIO buffer granularity = %ld\n", granularity );
}
#endif /* !PA_NO_ASIO */
#endif /* WIN32 */
printf( "Default sample rate = %8.2f\n", deviceInfo->defaultSampleRate );
/* poll for standard sample rates */
inputParameters.device = i;
inputParameters.channelCount = deviceInfo->maxInputChannels;
inputParameters.sampleFormat = paInt16;
inputParameters.suggestedLatency = 0; /* ignored by Pa_IsFormatSupported() */
inputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.device = i;
outputParameters.channelCount = deviceInfo->maxOutputChannels;
outputParameters.sampleFormat = paInt16;
outputParameters.suggestedLatency = 0; /* ignored by Pa_IsFormatSupported() */
outputParameters.hostApiSpecificStreamInfo = NULL;
if( inputParameters.channelCount > 0 )
{
printf("Supported standard sample rates\n for half-duplex 16 bit %d channel input = \n",
inputParameters.channelCount );
PrintSupportedStandardSampleRates( &inputParameters, NULL );
}
if( outputParameters.channelCount > 0 )
{
printf("Supported standard sample rates\n for half-duplex 16 bit %d channel output = \n",
outputParameters.channelCount );
PrintSupportedStandardSampleRates( NULL, &outputParameters );
}
if( inputParameters.channelCount > 0 && outputParameters.channelCount > 0 )
{
printf("Supported standard sample rates\n for full-duplex 16 bit %d channel input, %d channel output = \n",
inputParameters.channelCount, outputParameters.channelCount );
PrintSupportedStandardSampleRates( &inputParameters, &outputParameters );
}
}
Pa_Terminate();
printf("----------------------------------------------\n");
return 0;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

168
portaudio-v19/pa_tests/pa_fuzz.c
View File

@ -1,168 +0,0 @@
/** @file pa_fuzz.c
@brief Distort input like a fuzz boz.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
/*
** Note that many of the older ISA sound cards on PCs do NOT support
** full duplex audio (simultaneous record and playback).
** And some only support full duplex at lower sample rates.
*/
#define SAMPLE_RATE (44100)
#define PA_SAMPLE_TYPE paFloat32
#define FRAMES_PER_BUFFER (64)
typedef float SAMPLE;
float CubicAmplifier( float input );
static int fuzzCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData );
/* Non-linear amplifier with soft distortion curve. */
float CubicAmplifier( float input )
{
float output, temp;
if( input < 0.0 )
{
temp = input + 1.0f;
output = (temp * temp * temp) - 1.0f;
}
else
{
temp = input - 1.0f;
output = (temp * temp * temp) + 1.0f;
}
return output;
}
#define FUZZ(x) CubicAmplifier(CubicAmplifier(CubicAmplifier(CubicAmplifier(x))))
static int gNumNoInputs = 0;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int fuzzCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
SAMPLE *out = (SAMPLE*)outputBuffer;
const SAMPLE *in = (const SAMPLE*)inputBuffer;
unsigned int i;
(void) timeInfo; /* Prevent unused variable warnings. */
(void) statusFlags;
(void) userData;
if( inputBuffer == NULL )
{
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = 0; /* left - silent */
*out++ = 0; /* right - silent */
}
gNumNoInputs += 1;
}
else
{
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = FUZZ(*in++); /* left - distorted */
*out++ = *in++; /* right - clean */
}
}
return paContinue;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters inputParameters, outputParameters;
PaStream *stream;
PaError err;
err = Pa_Initialize();
if( err != paNoError ) goto error;
inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
inputParameters.channelCount = 2; /* stereo input */
inputParameters.sampleFormat = PA_SAMPLE_TYPE;
inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = PA_SAMPLE_TYPE;
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
&inputParameters,
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
0, /* paClipOff, */ /* we won't output out of range samples so don't bother clipping them */
fuzzCallback,
NULL );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Hit ENTER to stop program.\n");
getchar();
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
printf("Finished. gNumNoInputs = %d\n", gNumNoInputs );
Pa_Terminate();
return 0;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return -1;
}

187
portaudio-v19/pa_tests/pa_minlat.c
View File

@ -1,187 +0,0 @@
/** @file pa_minlat.c
@brief Experiment with different numbers of buffers to determine the
minimum latency for a computer.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <string.h>
#include "portaudio.h"
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TWOPI (M_PI * 2.0)
#define DEFAULT_BUFFER_SIZE (32)
typedef struct
{
double left_phase;
double right_phase;
}
paTestData;
/* Very simple synthesis routine to generate two sine waves. */
static int paminlatCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned int i;
double left_phaseInc = 0.02;
double right_phaseInc = 0.06;
double left_phase = data->left_phase;
double right_phase = data->right_phase;
for( i=0; i<framesPerBuffer; i++ )
{
left_phase += left_phaseInc;
if( left_phase > TWOPI ) left_phase -= TWOPI;
*out++ = (float) sin( left_phase );
right_phase += right_phaseInc;
if( right_phase > TWOPI ) right_phase -= TWOPI;
*out++ = (float) sin( right_phase );
}
data->left_phase = left_phase;
data->right_phase = right_phase;
return 0;
}
int main( int argc, char **argv );
int main( int argc, char **argv )
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
int go;
int outLatency = 0;
int minLatency = DEFAULT_BUFFER_SIZE * 2;
int framesPerBuffer;
double sampleRate = 44100.0;
char str[256];
printf("pa_minlat - Determine minimum latency for your computer.\n");
printf(" usage: pa_minlat {userBufferSize}\n");
printf(" for example: pa_minlat 64\n");
printf("Adjust your stereo until you hear a smooth tone in each speaker.\n");
printf("Then try to find the smallest number of frames that still sounds smooth.\n");
printf("Note that the sound will stop momentarily when you change the number of buffers.\n");
/* Get bufferSize from command line. */
framesPerBuffer = ( argc > 1 ) ? atol( argv[1] ) : DEFAULT_BUFFER_SIZE;
printf("Frames per buffer = %d\n", framesPerBuffer );
data.left_phase = data.right_phase = 0.0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outLatency = sampleRate * 200.0 / 1000.0; /* 200 msec. */
/* Try different numBuffers in a loop. */
go = 1;
while( go )
{
outputParameters.device = Pa_GetDefaultOutputDevice(); /* Default output device. */
outputParameters.channelCount = 2; /* Stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output. */
outputParameters.suggestedLatency = (double)outLatency / sampleRate; /* In seconds. */
outputParameters.hostApiSpecificStreamInfo = NULL;
printf("Latency = %d frames = %6.1f msec.\n", outLatency, outputParameters.suggestedLatency * 1000.0 );
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
sampleRate,
framesPerBuffer,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
paminlatCallback,
&data );
if( err != paNoError ) goto error;
if( stream == NULL ) goto error;
/* Start audio. */
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
/* Ask user for a new nlatency. */
printf("\nMove windows around to see if the sound glitches.\n");
printf("Latency now %d, enter new number of frames, or 'q' to quit: ", outLatency );
fgets( str, 256, stdin );
{
/* Get rid of newline */
size_t l = strlen( str ) - 1;
if( str[ l ] == '\n')
str[ l ] = '\0';
}
if( str[0] == 'q' ) go = 0;
else
{
outLatency = atol( str );
if( outLatency < minLatency )
{
printf( "Latency below minimum of %d! Set to minimum!!!\n", minLatency );
outLatency = minLatency;
}
}
/* Stop sound until ENTER hit. */
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
}
printf("A good setting for latency would be somewhat higher than\n");
printf("the minimum latency that worked.\n");
printf("PortAudio: Test finished.\n");
Pa_Terminate();
return 0;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return 1;
}

347
portaudio-v19/pa_tests/paqa_devs.c
View File

@ -1,347 +0,0 @@
/** @file paqa_devs.c
@brief Self Testing Quality Assurance app for PortAudio
Try to open each device and run through all the
possible configurations.
@author Phil Burk http://www.softsynth.com
Pieter adapted to V19 API. Test now relies heavily on
Pa_IsFormatSupported(). Uses same 'standard' sample rates
as in test pa_devs.c.
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#include "pa_trace.h"
/****************************************** Definitions ***********/
#define MODE_INPUT (0)
#define MODE_OUTPUT (1)
typedef struct PaQaData
{
unsigned long framesLeft;
int numChannels;
int bytesPerSample;
int mode;
short sawPhase;
PaSampleFormat format;
}
PaQaData;
/****************************************** Prototypes ***********/
static void TestDevices( int mode );
static void TestFormats( int mode, PaDeviceIndex deviceID, double sampleRate,
int numChannels );
static int TestAdvance( int mode, PaDeviceIndex deviceID, double sampleRate,
int numChannels, PaSampleFormat format );
static int QaCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData );
/****************************************** Globals ***********/
static int gNumPassed = 0;
static int gNumFailed = 0;
/****************************************** Macros ***********/
/* Print ERROR if it fails. Tally success or failure. */
/* Odd do-while wrapper seems to be needed for some compilers. */
#define EXPECT(_exp) \
do \
{ \
if ((_exp)) {\
/* printf("SUCCESS for %s\n", #_exp ); */ \
gNumPassed++; \
} \
else { \
printf("ERROR - 0x%x - %s for %s\n", result, \
((result == 0) ? "-" : Pa_GetErrorText(result)), \
#_exp ); \
gNumFailed++; \
goto error; \
} \
} while(0)
/*******************************************************************/
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int QaCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
unsigned long i;
short phase;
PaQaData *data = (PaQaData *) userData;
(void) inputBuffer;
/* Play simle sawtooth wave. */
if( data->mode == MODE_OUTPUT )
{
phase = data->sawPhase;
switch( data->format )
{
case paFloat32:
{
float *out = (float *) outputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
phase += 0x123;
*out++ = (float) (phase * (1.0 / 32768.0));
if( data->numChannels == 2 )
{
*out++ = (float) (phase * (1.0 / 32768.0));
}
}
}
break;
case paInt32:
{
int *out = (int *) outputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
phase += 0x123;
*out++ = ((int) phase ) << 16;
if( data->numChannels == 2 )
{
*out++ = ((int) phase ) << 16;
}
}
}
break;
case paInt16:
{
short *out = (short *) outputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
phase += 0x123;
*out++ = phase;
if( data->numChannels == 2 )
{
*out++ = phase;
}
}
}
break;
default:
{
unsigned char *out = (unsigned char *) outputBuffer;
unsigned long numBytes = framesPerBuffer * data->numChannels * data->bytesPerSample;
for( i=0; i<numBytes; i++ )
{
*out++ = 0;
}
}
break;
}
data->sawPhase = phase;
}
/* Are we through yet? */
if( data->framesLeft > framesPerBuffer )
{
PaUtil_AddTraceMessage("QaCallback: running. framesLeft", data->framesLeft );
data->framesLeft -= framesPerBuffer;
return 0;
}
else
{
PaUtil_AddTraceMessage("QaCallback: DONE! framesLeft", data->framesLeft );
data->framesLeft = 0;
return 1;
}
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError result;
EXPECT( ((result=Pa_Initialize()) == 0) );
printf("Test OUTPUT ---------------\n");
TestDevices( MODE_OUTPUT );
printf("Test INPUT ---------------\n");
TestDevices( MODE_INPUT );
error:
Pa_Terminate();
printf("QA Report: %d passed, %d failed.\n", gNumPassed, gNumFailed );
return 0;
}
/*******************************************************************
* Try each output device, through its full range of capabilities. */
static void TestDevices( int mode )
{
int id, jc, i;
int maxChannels;
const PaDeviceInfo *pdi;
static double standardSampleRates[] = { 8000.0, 9600.0, 11025.0, 12000.0,
16000.0, 22050.0, 24000.0,
32000.0, 44100.0, 48000.0,
88200.0, 96000.0,
-1.0 }; /* Negative terminated list. */
int numDevices = Pa_GetDeviceCount();
for( id=0; id<numDevices; id++ ) /* Iterate through all devices. */
{
pdi = Pa_GetDeviceInfo( id );
/* Try 1 to maxChannels on each device. */
maxChannels = (( mode == MODE_INPUT ) ? pdi->maxInputChannels : pdi->maxOutputChannels);
for( jc=1; jc<=maxChannels; jc++ )
{
printf("Name = %s\n", pdi->name );
/* Try each standard sample rate. */
for( i=0; standardSampleRates[i] > 0; i++ )
{
TestFormats( mode, (PaDeviceIndex)id, standardSampleRates[i], jc );
}
}
}
}
/*******************************************************************/
static void TestFormats( int mode, PaDeviceIndex deviceID, double sampleRate,
int numChannels )
{
TestAdvance( mode, deviceID, sampleRate, numChannels, paFloat32 );
TestAdvance( mode, deviceID, sampleRate, numChannels, paInt16 );
TestAdvance( mode, deviceID, sampleRate, numChannels, paInt32 );
/* TestAdvance( mode, deviceID, sampleRate, numChannels, paInt24 ); */
}
/*******************************************************************/
static int TestAdvance( int mode, PaDeviceIndex deviceID, double sampleRate,
int numChannels, PaSampleFormat format )
{
PaStreamParameters inputParameters, outputParameters, *ipp, *opp;
PaStream *stream = NULL;
PaError result = paNoError;
PaQaData myData;
#define FRAMES_PER_BUFFER (64)
/* Setup data for synthesis thread. */
myData.framesLeft = (unsigned long) (sampleRate * 100); /* 100 seconds */
myData.numChannels = numChannels;
myData.mode = mode;
myData.format = format;
switch( format )
{
case paFloat32:
case paInt32:
case paInt24:
myData.bytesPerSample = 4;
break;
/* case paPackedInt24:
myData.bytesPerSample = 3;
break; */
default:
myData.bytesPerSample = 2;
break;
}
if( mode == MODE_INPUT )
{
inputParameters.device = deviceID;
inputParameters.channelCount = numChannels;
inputParameters.sampleFormat = format;
inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
ipp = &inputParameters;
}
else
ipp = NULL;
if( mode == MODE_OUTPUT ) /* Pa_GetDeviceInfo(paNoDevice) COREDUMPS!!! */
{
outputParameters.device = deviceID;
outputParameters.channelCount = numChannels;
outputParameters.sampleFormat = format;
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
opp = &outputParameters;
}
else
opp = NULL;
if(paFormatIsSupported == Pa_IsFormatSupported( ipp, opp, sampleRate ))
{
printf("------ TestAdvance: %s, device = %d, rate = %g, numChannels = %d, format = %lu -------\n",
( mode == MODE_INPUT ) ? "INPUT" : "OUTPUT",
deviceID, sampleRate, numChannels, (unsigned long)format);
EXPECT( ((result = Pa_OpenStream( &stream,
ipp,
opp,
sampleRate,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
QaCallback,
&myData ) ) == 0) );
if( stream )
{
PaTime oldStamp, newStamp;
unsigned long oldFrames;
int minDelay = ( mode == MODE_INPUT ) ? 1000 : 400;
/* Was:
int minNumBuffers = Pa_GetMinNumBuffers( FRAMES_PER_BUFFER, sampleRate );
int msec = (int) ((minNumBuffers * 3 * 1000.0 * FRAMES_PER_BUFFER) / sampleRate);
*/
int msec = (int)( 3.0 *
(( mode == MODE_INPUT ) ? inputParameters.suggestedLatency : outputParameters.suggestedLatency ));
if( msec < minDelay ) msec = minDelay;
printf("msec = %d\n", msec); /**/
EXPECT( ((result=Pa_StartStream( stream )) == 0) );
/* Check to make sure PortAudio is advancing timeStamp. */
oldStamp = Pa_GetStreamTime(stream);
Pa_Sleep(msec);
newStamp = Pa_GetStreamTime(stream);
printf("oldStamp = %g,newStamp = %g\n", oldStamp, newStamp ); /**/
EXPECT( (oldStamp < newStamp) );
/* Check to make sure callback is decrementing framesLeft. */
oldFrames = myData.framesLeft;
Pa_Sleep(msec);
printf("oldFrames = %lu, myData.framesLeft = %lu\n", oldFrames, myData.framesLeft ); /**/
EXPECT( (oldFrames > myData.framesLeft) );
EXPECT( ((result=Pa_CloseStream( stream )) == 0) );
stream = NULL;
}
}
error:
if( stream != NULL ) Pa_CloseStream( stream );
return result;
}

380
portaudio-v19/pa_tests/paqa_errs.c
View File

@ -1,380 +0,0 @@
/** @file paqa_errs.c
@brief Self Testing Quality Assurance app for PortAudio
Do lots of bad things to test error reporting.
@author Phil Burk http://www.softsynth.com
Pieter Suurmond adapted to V19 API.
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
/*--------- Definitions ---------*/
#define MODE_INPUT (0)
#define MODE_OUTPUT (1)
#define FRAMES_PER_BUFFER (64)
#define SAMPLE_RATE (44100.0)
typedef struct PaQaData
{
unsigned long framesLeft;
int numChannels;
int bytesPerSample;
int mode;
}
PaQaData;
static int gNumPassed = 0; /* Two globals */
static int gNumFailed = 0;
/*------------------- Macros ------------------------------*/
/* Print ERROR if it fails. Tally success or failure. Odd */
/* do-while wrapper seems to be needed for some compilers. */
#define EXPECT(_exp) \
do \
{ \
if ((_exp)) {\
gNumPassed++; \
} \
else { \
printf("\nERROR - 0x%x - %s for %s\n", result, Pa_GetErrorText(result), #_exp ); \
gNumFailed++; \
goto error; \
} \
} while(0)
#define HOPEFOR(_exp) \
do \
{ \
if ((_exp)) {\
gNumPassed++; \
} \
else { \
printf("\nERROR - 0x%x - %s for %s\n", result, Pa_GetErrorText(result), #_exp ); \
gNumFailed++; \
} \
} while(0)
/*-------------------------------------------------------------------------*/
/* This routine will be called by the PortAudio engine when audio is needed.
It may be called at interrupt level on some machines so don't do anything
that could mess up the system like calling malloc() or free().
*/
static int QaCallback( const void* inputBuffer,
void* outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void* userData )
{
unsigned long i;
unsigned char* out = (unsigned char *) outputBuffer;
PaQaData* data = (PaQaData *) userData;
(void)inputBuffer; /* Prevent "unused variable" warnings. */
/* Zero out buffer so we don't hear terrible noise. */
if( data->mode == MODE_OUTPUT )
{
unsigned long numBytes = framesPerBuffer * data->numChannels * data->bytesPerSample;
for( i=0; i<numBytes; i++ )
{
*out++ = 0;
}
}
/* Are we through yet? */
if( data->framesLeft > framesPerBuffer )
{
data->framesLeft -= framesPerBuffer;
return 0;
}
else
{
data->framesLeft = 0;
return 1;
}
}
static PaDeviceIndex FindInputOnlyDevice(void)
{
PaDeviceIndex result = Pa_GetDefaultInputDevice();
if( result != paNoDevice && Pa_GetDeviceInfo(result)->maxOutputChannels == 0 )
return result;
for( result = 0; result < Pa_GetDeviceCount(); ++result )
{
if( Pa_GetDeviceInfo(result)->maxOutputChannels == 0 )
return result;
}
return paNoDevice;
}
static PaDeviceIndex FindOutputOnlyDevice(void)
{
PaDeviceIndex result = Pa_GetDefaultOutputDevice();
if( result != paNoDevice && Pa_GetDeviceInfo(result)->maxInputChannels == 0 )
return result;
for( result = 0; result < Pa_GetDeviceCount(); ++result )
{
if( Pa_GetDeviceInfo(result)->maxInputChannels == 0 )
return result;
}
return paNoDevice;
}
/*-------------------------------------------------------------------------------------------------*/
static int TestBadOpens( void )
{
PaStream* stream = NULL;
PaError result;
PaQaData myData;
PaStreamParameters ipp, opp;
/* Setup data for synthesis thread. */
myData.framesLeft = (unsigned long) (SAMPLE_RATE * 100); /* 100 seconds */
myData.numChannels = 1;
myData.mode = MODE_OUTPUT;
/*----------------------------- No devices specified: */
ipp.device = opp.device = paNoDevice;
ipp.channelCount = opp.channelCount = 0; /* Also no channels. */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
/* Take the low latency of the default device for all subsequent tests. */
ipp.suggestedLatency = Pa_GetDeviceInfo(Pa_GetDefaultInputDevice())->defaultLowInputLatency;
opp.suggestedLatency = Pa_GetDeviceInfo(Pa_GetDefaultOutputDevice())->defaultLowOutputLatency;
HOPEFOR(((result = Pa_OpenStream(&stream, &ipp, &opp,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidDevice));
/*----------------------------- No devices specified #2: */
HOPEFOR(((result = Pa_OpenStream(&stream, NULL, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidDevice));
/*----------------------------- Out of range input device specified: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = Pa_GetDeviceCount(); /* And no output device, and no channels. */
opp.channelCount = 0; opp.device = paNoDevice;
HOPEFOR(((result = Pa_OpenStream(&stream, &ipp, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidDevice));
/*----------------------------- Out of range output device specified: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = paNoDevice; /* And no input device, and no channels. */
opp.channelCount = 0; opp.device = Pa_GetDeviceCount();
HOPEFOR(((result = Pa_OpenStream(&stream, NULL, &opp,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidDevice));
/*----------------------------- Zero input channels: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = Pa_GetDefaultInputDevice();
opp.channelCount = 0; opp.device = paNoDevice; /* And no output device, and no output channels. */
HOPEFOR(((result = Pa_OpenStream(&stream, &ipp, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidChannelCount));
/*----------------------------- Zero output channels: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = paNoDevice; /* And no input device, and no input channels. */
opp.channelCount = 0; opp.device = Pa_GetDefaultOutputDevice();
HOPEFOR(((result = Pa_OpenStream(&stream, NULL, &opp,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidChannelCount));
/*----------------------------- Nonzero input and output channels but no output device: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 2; ipp.device = Pa_GetDefaultInputDevice(); /* Both stereo. */
opp.channelCount = 2; opp.device = paNoDevice;
HOPEFOR(((result = Pa_OpenStream(&stream, &ipp, &opp,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidDevice));
/*----------------------------- Nonzero input and output channels but no input device: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 2; ipp.device = paNoDevice;
opp.channelCount = 2; opp.device = Pa_GetDefaultOutputDevice();
HOPEFOR(((result = Pa_OpenStream(&stream, &ipp, &opp,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidDevice));
/*----------------------------- NULL stream pointer: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = paNoDevice; /* Output is more likely than input. */
opp.channelCount = 2; opp.device = Pa_GetDefaultOutputDevice(); /* Only 2 output channels. */
HOPEFOR(((result = Pa_OpenStream(NULL, &ipp, &opp,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paBadStreamPtr));
/*----------------------------- Low sample rate: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = paNoDevice;
opp.channelCount = 2; opp.device = Pa_GetDefaultOutputDevice();
HOPEFOR(((result = Pa_OpenStream(&stream, NULL, &opp,
1.0, FRAMES_PER_BUFFER, /* 1 cycle per second (1 Hz) is too low. */
paClipOff, QaCallback, &myData )) == paInvalidSampleRate));
/*----------------------------- High sample rate: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = paNoDevice;
opp.channelCount = 2; opp.device = Pa_GetDefaultOutputDevice();
HOPEFOR(((result = Pa_OpenStream(&stream, NULL, &opp,
10000000.0, FRAMES_PER_BUFFER, /* 10^6 cycles per second (10 MHz) is too high. */
paClipOff, QaCallback, &myData )) == paInvalidSampleRate));
/*----------------------------- NULL callback: */
/* NULL callback is valid in V19 -- it means use blocking read/write stream
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = paNoDevice;
opp.channelCount = 2; opp.device = Pa_GetDefaultOutputDevice();
HOPEFOR(((result = Pa_OpenStream(&stream, NULL, &opp,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff,
NULL,
&myData )) == paNullCallback));
*/
/*----------------------------- Bad flag: */
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = paNoDevice;
opp.channelCount = 2; opp.device = Pa_GetDefaultOutputDevice();
HOPEFOR(((result = Pa_OpenStream(&stream, NULL, &opp,
SAMPLE_RATE, FRAMES_PER_BUFFER,
255, /* Is 8 maybe legal V19 API? */
QaCallback, &myData )) == paInvalidFlag));
/*----------------------------- using input device as output device: */
if( FindInputOnlyDevice() != paNoDevice )
{
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 0; ipp.device = paNoDevice; /* And no input device, and no channels. */
opp.channelCount = 2; opp.device = FindInputOnlyDevice();
HOPEFOR(((result = Pa_OpenStream(&stream, NULL, &opp,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidChannelCount));
}
/*----------------------------- using output device as input device: */
if( FindOutputOnlyDevice() != paNoDevice )
{
ipp.hostApiSpecificStreamInfo = opp.hostApiSpecificStreamInfo = NULL;
ipp.sampleFormat = opp.sampleFormat = paFloat32;
ipp.channelCount = 2; ipp.device = FindOutputOnlyDevice();
opp.channelCount = 0; opp.device = paNoDevice; /* And no output device, and no channels. */
HOPEFOR(((result = Pa_OpenStream(&stream, &ipp, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paInvalidChannelCount));
}
if( stream != NULL ) Pa_CloseStream( stream );
return result;
}
/*-----------------------------------------------------------------------------------------*/
static int TestBadActions( void )
{
PaStream* stream = NULL;
PaError result;
PaQaData myData;
PaStreamParameters opp;
/* Setup data for synthesis thread. */
myData.framesLeft = (unsigned long)(SAMPLE_RATE * 100); /* 100 seconds */
myData.numChannels = 1;
myData.mode = MODE_OUTPUT;
opp.device = Pa_GetDefaultOutputDevice(); /* Default output. */
opp.channelCount = 2; /* Stereo output. */
opp.hostApiSpecificStreamInfo = NULL;
opp.sampleFormat = paFloat32;
opp.suggestedLatency = Pa_GetDeviceInfo(opp.device)->defaultLowOutputLatency;
HOPEFOR(((result = Pa_OpenStream(&stream, NULL, /* Take NULL as input parame- */
&opp, /* ters, meaning try only output. */
SAMPLE_RATE, FRAMES_PER_BUFFER,
paClipOff, QaCallback, &myData )) == paNoError));
HOPEFOR(((result = Pa_StartStream(NULL)) == paBadStreamPtr));
HOPEFOR(((result = Pa_StopStream(NULL)) == paBadStreamPtr));
HOPEFOR(((result = Pa_IsStreamStopped(NULL)) == paBadStreamPtr));
HOPEFOR(((result = Pa_IsStreamActive(NULL)) == paBadStreamPtr));
HOPEFOR(((result = Pa_CloseStream(NULL)) == paBadStreamPtr));
HOPEFOR(((result = Pa_SetStreamFinishedCallback(NULL, NULL)) == paBadStreamPtr));
HOPEFOR(((result = !Pa_GetStreamInfo(NULL))));
HOPEFOR(((result = Pa_GetStreamTime(NULL)) == 0.0));
HOPEFOR(((result = Pa_GetStreamCpuLoad(NULL)) == 0.0));
HOPEFOR(((result = Pa_ReadStream(NULL, NULL, 0)) == paBadStreamPtr));
HOPEFOR(((result = Pa_WriteStream(NULL, NULL, 0)) == paBadStreamPtr));
/** @todo test Pa_GetStreamReadAvailable and Pa_GetStreamWriteAvailable */
if (stream != NULL) Pa_CloseStream(stream);
return result;
}
/*---------------------------------------------------------------------*/
int main(void);
int main(void)
{
PaError result;
EXPECT(((result = Pa_Initialize()) == paNoError));
TestBadOpens();
TestBadActions();
error:
Pa_Terminate();
printf("QA Report: %d passed, %d failed.\n", gNumPassed, gNumFailed);
return 0;
}

192
portaudio-v19/pa_tests/patest1.c
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@ -1,192 +0,0 @@
/** @file patest1.c
@brief Ring modulate the audio input with a sine wave for 20 seconds.
@author Ross Bencina <rossb@audiomulch.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define SAMPLE_RATE (44100)
typedef struct
{
float sine[100];
int phase;
int sampsToGo;
}
patest1data;
static int patest1Callback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
patest1data *data = (patest1data*)userData;
float *in = (float*)inputBuffer;
float *out = (float*)outputBuffer;
int framesToCalc = framesPerBuffer;
unsigned long i = 0;
int finished;
if( data->sampsToGo < framesPerBuffer )
{
framesToCalc = data->sampsToGo;
finished = paComplete;
}
else
{
finished = paContinue;
}
for( ; i<framesToCalc; i++ )
{
*out++ = *in++ * data->sine[data->phase]; /* left */
*out++ = *in++ * data->sine[data->phase++]; /* right */
if( data->phase >= 100 )
data->phase = 0;
}
data->sampsToGo -= framesToCalc;
/* zero remainder of final buffer if not already done */
for( ; i<framesPerBuffer; i++ )
{
*out++ = 0; /* left */
*out++ = 0; /* right */
}
return finished;
}
int main(int argc, char* argv[]);
int main(int argc, char* argv[])
{
PaStream *stream;
PaError err;
patest1data data;
int i;
PaStreamParameters inputParameters, outputParameters;
const PaHostErrorInfo* herr;
printf("patest1.c\n"); fflush(stdout);
printf("Ring modulate input for 20 seconds.\n"); fflush(stdout);
/* initialise sinusoidal wavetable */
for( i=0; i<100; i++ )
data.sine[i] = sin( ((double)i/100.) * M_PI * 2. );
data.phase = 0;
data.sampsToGo = SAMPLE_RATE * 20; /* 20 seconds. */
/* initialise portaudio subsytem */
err = Pa_Initialize();
inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
inputParameters.channelCount = 2; /* stereo input */
inputParameters.sampleFormat = paFloat32; /* 32 bit floating point input */
inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
&inputParameters,
&outputParameters,
(double)SAMPLE_RATE, /* Samplerate in Hertz. */
512, /* Small buffers */
paClipOff, /* We won't output out of range samples so don't bother clipping them. */
patest1Callback,
&data );
if( err != paNoError ) goto done;
err = Pa_StartStream( stream );
if( err != paNoError ) goto done;
printf( "Press any key to end.\n" ); fflush(stdout);
getc( stdin ); /* wait for input before exiting */
err = Pa_AbortStream( stream );
if( err != paNoError ) goto done;
printf( "Waiting for stream to complete...\n" );
/* sleep until playback has finished */
while( ( err = Pa_IsStreamActive( stream ) ) == 1 ) Pa_Sleep(1000);
if( err < 0 ) goto done;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto done;
done:
Pa_Terminate();
if( err != paNoError )
{
fprintf( stderr, "An error occured while using portaudio\n" );
if( err == paUnanticipatedHostError )
{
fprintf( stderr, " unanticipated host error.\n");
herr = Pa_GetLastHostErrorInfo();
if (herr)
{
fprintf( stderr, " Error number: %ld\n", herr->errorCode );
if (herr->errorText)
fprintf( stderr, " Error text: %s\n", herr->errorText );
}
else
fprintf( stderr, " Pa_GetLastHostErrorInfo() failed!\n" );
}
else
{
fprintf( stderr, " Error number: %d\n", err );
fprintf( stderr, " Error text: %s\n", Pa_GetErrorText( err ) );
}
err = 1; /* Always return 0 or 1, but no other return codes. */
}
printf( "bye\n" );
return err;
}

180
portaudio-v19/pa_tests/patest_buffer.c
View File

@ -1,180 +0,0 @@
/** @file patest_buffer.c
@brief Test opening streams with different buffer sizes.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (1)
#define SAMPLE_RATE (44100)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
#define BUFFER_TABLE 9
long buffer_table[] = {200,256,500,512,600, 723, 1000, 1024, 2345};
typedef struct
{
short sine[TABLE_SIZE];
int left_phase;
int right_phase;
unsigned int sampsToGo;
}
paTestData;
PaError TestOnce( int buffersize );
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patest1Callback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
short *out = (short*)outputBuffer;
unsigned int i;
int finished = 0;
(void) inputBuffer; /* Prevent "unused variable" warnings. */
if( data->sampsToGo < framesPerBuffer )
{
/* final buffer... */
for( i=0; i<data->sampsToGo; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
/* zero remainder of final buffer */
for( ; i<framesPerBuffer; i++ )
{
*out++ = 0; /* left */
*out++ = 0; /* right */
}
finished = 1;
}
else
{
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
data->sampsToGo -= framesPerBuffer;
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
int i;
PaError err;
printf("Test opening streams with different buffer sizes\n\n");
for (i = 0 ; i < BUFFER_TABLE; i++)
{
printf("Buffer size %ld\n", buffer_table[i]);
err = TestOnce(buffer_table[i]);
if( err < 0 ) return 0;
}
return 0;
}
PaError TestOnce( int buffersize )
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
int i;
int totalSamps;
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (short) (32767.0 * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ));
}
data.left_phase = data.right_phase = 0;
data.sampsToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paInt16; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
buffersize, /* frames per buffer */
(paClipOff | paDitherOff),
patest1Callback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Waiting for sound to finish.\n");
Pa_Sleep(1000);
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
return paNoError;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

221
portaudio-v19/pa_tests/patest_callbackstop.c
View File

@ -1,221 +0,0 @@
/** @file patest_callbackstop.c
@brief Test the paComplete callback result code.
@author Ross Bencina <rossb@audiomulch.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (5)
#define NUM_LOOPS (4)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (67)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
float sine[TABLE_SIZE];
int phase;
unsigned long generatedFramesCount;
volatile int callbackReturnedPaComplete;
volatile int callbackInvokedAfterReturningPaComplete;
}
TestData;
/*
This routine will be called by the PortAudio stream when audio is needed.
It may be called at interrupt level on some machines so don't do anything
that could mess up the system like calling malloc() or free().
*/
static int TestCallback( const void *input, void *output,
unsigned long frameCount,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
TestData *data = (TestData*)userData;
float *out = (float*)output;
unsigned long i;
float x;
(void) input; /* Prevent unused variable warnings. */
(void) timeInfo;
(void) statusFlags;
if( data->callbackReturnedPaComplete )
data->callbackInvokedAfterReturningPaComplete = 1;
for( i=0; i<frameCount; i++ )
{
/* generate tone */
x = data->sine[ data->phase++ ];
if( data->phase >= TABLE_SIZE )
data->phase -= TABLE_SIZE;
*out++ = x; /* left */
*out++ = x; /* right */
}
data->generatedFramesCount += frameCount;
if( data->generatedFramesCount >= (NUM_SECONDS * SAMPLE_RATE) )
{
data->callbackReturnedPaComplete = 1;
return paComplete;
}
else
{
return paContinue;
}
}
/*----------------------------------------------------------------------------*/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
TestData data;
int i, j;
printf( "PortAudio Test: output sine wave. SR = %d, BufSize = %d\n",
SAMPLE_RATE, FRAMES_PER_BUFFER );
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice();
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* output will be in-range, so no need to clip */
TestCallback,
&data );
if( err != paNoError ) goto error;
printf("Repeating test %d times.\n", NUM_LOOPS );
for( i=0; i < NUM_LOOPS; ++i )
{
data.phase = 0;
data.generatedFramesCount = 0;
data.callbackReturnedPaComplete = 0;
data.callbackInvokedAfterReturningPaComplete = 0;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Play for %d seconds.\n", NUM_SECONDS );
/* wait for the callback to complete generating NUM_SECONDS of tone */
do
{
Pa_Sleep( 500 );
}
while( !data.callbackReturnedPaComplete );
printf( "Callback returned paComplete.\n" );
printf( "Waiting for buffers to finish playing...\n" );
/* wait for stream to become inactive,
or for a timeout of approximately NUM_SECONDS
*/
j = 0;
while( (err = Pa_IsStreamActive( stream )) == 1 && j < NUM_SECONDS * 2 )
{
printf(".\n" );
Pa_Sleep( 500 );
++j;
}
if( err < 0 )
{
goto error;
}
else if( err == 1 )
{
printf( "TEST FAILED: Timed out waiting for buffers to finish playing.\n" );
}
else
{
printf("Buffers finished.\n" );
}
if( data.callbackInvokedAfterReturningPaComplete )
{
printf( "TEST FAILED: Callback was invoked after returning paComplete.\n" );
}
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
}
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

178
portaudio-v19/pa_tests/patest_clip.c
View File

@ -1,178 +0,0 @@
/** @file patest_clip.c
@brief Play a sine wave for several seconds at an amplitude
that would require clipping.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (4)
#define SAMPLE_RATE (44100)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct paTestData
{
float sine[TABLE_SIZE];
float amplitude;
int left_phase;
int right_phase;
}
paTestData;
PaError PlaySine( paTestData *data, unsigned long flags, float amplitude );
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int sineCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
float amplitude = data->amplitude;
unsigned int i;
(void) inputBuffer; /* Prevent "unused variable" warnings. */
(void) timeInfo;
(void) statusFlags;
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = amplitude * data->sine[data->left_phase]; /* left */
*out++ = amplitude * data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
return 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError err;
paTestData data;
int i;
printf("PortAudio Test: output sine wave with and without clipping.\n");
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
printf("\nHalf amplitude. Should sound like sine wave.\n"); fflush(stdout);
err = PlaySine( &data, paClipOff | paDitherOff, 0.5f );
if( err < 0 ) goto error;
printf("\nFull amplitude. Should sound like sine wave.\n"); fflush(stdout);
err = PlaySine( &data, paClipOff | paDitherOff, 0.999f );
if( err < 0 ) goto error;
printf("\nOver range with clipping and dithering turned OFF. Should sound very nasty.\n");
fflush(stdout);
err = PlaySine( &data, paClipOff | paDitherOff, 1.1f );
if( err < 0 ) goto error;
printf("\nOver range with clipping and dithering turned ON. Should sound smoother than previous.\n");
fflush(stdout);
err = PlaySine( &data, paNoFlag, 1.1f );
if( err < 0 ) goto error;
printf("\nOver range with paClipOff but dithering ON.\n"
"That forces clipping ON so it should sound the same as previous.\n");
fflush(stdout);
err = PlaySine( &data, paClipOff, 1.1f );
if( err < 0 ) goto error;
return 0;
error:
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return 1;
}
/*****************************************************************************/
PaError PlaySine( paTestData *data, unsigned long flags, float amplitude )
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
data->left_phase = data->right_phase = 0;
data->amplitude = amplitude;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
1024,
flags,
sineCallback,
data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
Pa_Sleep( NUM_SECONDS * 1000 );
printf("CPULoad = %8.6f\n", Pa_GetStreamCpuLoad( stream ) );
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
return paNoError;
error:
return err;
}

178
portaudio-v19/pa_tests/patest_dither.c
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@ -1,178 +0,0 @@
/** @file patest_dither.c
@brief Attempt to hear difference between dithered and non-dithered signal.
This only has an effect if the native format is 16 bit.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (5)
#define SAMPLE_RATE (44100)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct paTestData
{
float sine[TABLE_SIZE];
float amplitude;
int left_phase;
int right_phase;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int sineCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo *timeInfo,
PaStreamCallbackFlags statusFlags, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
float amplitude = data->amplitude;
unsigned int i;
(void) inputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = amplitude * data->sine[data->left_phase]; /* left */
*out++ = amplitude * data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
return 0;
}
/*****************************************************************************/
/*
V18 version did not call Pa_Terminate() if Pa_Initialize() failed.
This V19 version ALWAYS calls Pa_Terminate(). PS.
*/
PaError PlaySine( paTestData *data, PaStreamFlags flags, float amplitude );
PaError PlaySine( paTestData *data, PaStreamFlags flags, float amplitude )
{
PaStream* stream;
PaStreamParameters outputParameters;
PaError err;
data->left_phase = data->right_phase = 0;
data->amplitude = amplitude;
err = Pa_Initialize();
if (err != paNoError)
goto done;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output. */
/* When you change this, also */
/* adapt the callback routine! */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )
->defaultLowOutputLatency; /* Low latency. */
err = Pa_OpenStream( &stream,
NULL, /* No input. */
&outputParameters,
SAMPLE_RATE,
1024, /* frames per buffer */
flags,
sineCallback,
(void*)data );
if (err != paNoError)
goto done;
err = Pa_StartStream( stream );
if (err != paNoError)
goto done;
Pa_Sleep( NUM_SECONDS * 1000 );
printf("CPULoad = %8.6f\n", Pa_GetStreamCpuLoad(stream));
err = Pa_CloseStream( stream );
done:
Pa_Sleep( 250 ); /* Just a small silence. */
Pa_Terminate();
return err;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError err;
paTestData DATA;
int i;
float amplitude = 4.0 / (1<<15);
printf("PortAudio Test: output EXTREMELY QUIET sine wave with and without dithering.\n");
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
DATA.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
printf("\nNo treatment..\n"); fflush(stdout);
err = PlaySine( &DATA, paClipOff | paDitherOff, amplitude );
if( err < 0 ) goto done;
printf("\nClip..\n");
fflush(stdout);
err = PlaySine( &DATA, paDitherOff, amplitude );
if( err < 0 ) goto done;
printf("\nClip and Dither..\n");
fflush(stdout);
err = PlaySine( &DATA, paNoFlag, amplitude );
done:
if (err)
{
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
err = 1; /* Though PlaySine() already called Pa_Terminate(), */
} /* we may still call Pa_GetErrorText(). */
else
printf("\n(Don't forget to turn the VOLUME DOWN after listening so carefully.)\n");
return err; /* 0 or 1. */
}

152
portaudio-v19/pa_tests/patest_hang.c
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@ -1,152 +0,0 @@
/** @file patest_hang.c
@brief Play a sine then hang audio callback to test watchdog.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (1024)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TWOPI (M_PI * 2.0)
typedef struct paTestData
{
int sleepFor;
double phase;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
int finished = 0;
double phaseInc = 0.02;
double phase = data->phase;
(void) inputBuffer; /* Prevent unused argument warning. */
for( i=0; i<framesPerBuffer; i++ )
{
phase += phaseInc;
if( phase > TWOPI ) phase -= TWOPI;
/* This is not a very efficient way to calc sines. */
*out++ = (float) sin( phase ); /* mono */
}
if( data->sleepFor > 0 )
{
Pa_Sleep( data->sleepFor );
}
data->phase = phase;
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStream* stream;
PaStreamParameters outputParameters;
PaError err;
int i;
paTestData data = {0};
printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n",
SAMPLE_RATE, FRAMES_PER_BUFFER );
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* Default output device. */
outputParameters.channelCount = 1; /* Mono output. */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point. */
outputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.suggestedLatency = Pa_GetDeviceInfo(outputParameters.device)
->defaultLowOutputLatency;
err = Pa_OpenStream(&stream,
NULL, /* No input. */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* No out of range samples. */
patestCallback,
&data);
if (err != paNoError) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
/* Gradually increase sleep time. */
/* Was: for( i=0; i<10000; i+= 1000 ) */
for(i=0; i <= 1000; i += 100)
{
printf("Sleep for %d milliseconds in audio callback.\n", i );
data.sleepFor = i;
Pa_Sleep( ((i<1000) ? 1000 : i) );
}
printf("Suffer for 10 seconds.\n");
Pa_Sleep( 10000 );
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

224
portaudio-v19/pa_tests/patest_in_overflow.c
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@ -1,224 +0,0 @@
/** @file patest_in_overflow.c
@brief Count input overflows (using paInputOverflow flag) under
overloaded and normal conditions.
This test uses the same method to overload the stream as does
patest_out_underflow.c -- it generates sine waves until the cpu load
exceeds a certain level. However this test is only concerned with
input and so doesn't ouput any sound.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2004 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define MAX_SINES (500)
#define MAX_LOAD (1.2)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (512)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TWOPI (M_PI * 2.0)
typedef struct paTestData
{
int sineCount;
double phases[MAX_SINES];
int countOverflows;
int inputOverflowCount;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float out; /* variable to hold dummy output */
unsigned long i;
int j;
int finished = paContinue;
(void) timeInfo; /* Prevent unused variable warning. */
(void) inputBuffer; /* Prevent unused variable warning. */
(void) outputBuffer; /* Prevent unused variable warning. */
if( data->countOverflows && (statusFlags & paInputOverflow) )
data->inputOverflowCount++;
for( i=0; i<framesPerBuffer; i++ )
{
float output = 0.0;
double phaseInc = 0.02;
double phase;
for( j=0; j<data->sineCount; j++ )
{
/* Advance phase of next oscillator. */
phase = data->phases[j];
phase += phaseInc;
if( phase > TWOPI ) phase -= TWOPI;
phaseInc *= 1.02;
if( phaseInc > 0.5 ) phaseInc *= 0.5;
/* This is not a very efficient way to calc sines. */
output += (float) sin( phase );
data->phases[j] = phase;
}
/* this is an input-only stream so we don't actually use the output */
out = (float) (output / data->sineCount);
(void) out; /* suppress unused variable warning*/
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters inputParameters;
PaStream *stream;
PaError err;
int safeSineCount, stressedSineCount;
int safeOverflowCount, stressedOverflowCount;
paTestData data = {0};
double load;
printf("PortAudio Test: input only, no sound output. Load callback by performing calculations, count input overflows. SR = %d, BufSize = %d. MAX_LOAD = %f\n",
SAMPLE_RATE, FRAMES_PER_BUFFER, (float)MAX_LOAD );
err = Pa_Initialize();
if( err != paNoError ) goto error;
inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
inputParameters.channelCount = 1; /* mono output */
inputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
&inputParameters,
NULL, /* no output */
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Establishing load conditions...\n" );
/* Determine number of sines required to get to 50% */
do
{
data.sineCount++;
Pa_Sleep( 100 );
load = Pa_GetStreamCpuLoad( stream );
printf("sineCount = %d, CPU load = %f\n", data.sineCount, load );
}
while( load < 0.5 && data.sineCount < (MAX_SINES-1));
safeSineCount = data.sineCount;
/* Calculate target stress value then ramp up to that level*/
stressedSineCount = (int) (2.0 * data.sineCount * MAX_LOAD );
if( stressedSineCount > MAX_SINES )
stressedSineCount = MAX_SINES;
for( ; data.sineCount < stressedSineCount; data.sineCount++ )
{
Pa_Sleep( 100 );
load = Pa_GetStreamCpuLoad( stream );
printf("STRESSING: sineCount = %d, CPU load = %f\n", data.sineCount, load );
}
printf("Counting overflows for 5 seconds.\n");
data.countOverflows = 1;
Pa_Sleep( 5000 );
stressedOverflowCount = data.inputOverflowCount;
data.countOverflows = 0;
data.sineCount = safeSineCount;
printf("Resuming safe load...\n");
Pa_Sleep( 1500 );
data.inputOverflowCount = 0;
Pa_Sleep( 1500 );
load = Pa_GetStreamCpuLoad( stream );
printf("sineCount = %d, CPU load = %f\n", data.sineCount, load );
printf("Counting overflows for 5 seconds.\n");
data.countOverflows = 1;
Pa_Sleep( 5000 );
safeOverflowCount = data.inputOverflowCount;
printf("Stop stream.\n");
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
if( stressedOverflowCount == 0 )
printf("Test failed, no input overflows detected under stress.\n");
else if( safeOverflowCount != 0 )
printf("Test failed, %d unexpected overflows detected under safe load.\n", safeOverflowCount);
else
printf("Test passed, %d expected input overflows detected under stress, 0 unexpected overflows detected under safe load.\n", stressedOverflowCount );
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

182
portaudio-v19/pa_tests/patest_latency.c
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@ -1,182 +0,0 @@
/** @file
@brief Hear the latency caused by big buffers.
Play a sine wave and change frequency based on letter input.
@author Phil Burk <philburk@softsynth.com>, and Darren Gibbs
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define OUTPUT_DEVICE (Pa_GetDefaultOutputDevice())
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (64)
#define MIN_FREQ (100.0f)
#define CalcPhaseIncrement(freq) ((freq)/SAMPLE_RATE)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (400)
typedef struct
{
float sine[TABLE_SIZE + 1]; /* add one for guard point for interpolation */
float phase_increment;
float left_phase;
float right_phase;
}
paTestData;
float LookupSine( paTestData *data, float phase );
/* Convert phase between and 1.0 to sine value
* using linear interpolation.
*/
float LookupSine( paTestData *data, float phase )
{
float fIndex = phase*TABLE_SIZE;
int index = (int) fIndex;
float fract = fIndex - index;
float lo = data->sine[index];
float hi = data->sine[index+1];
float val = lo + fract*(hi-lo);
return val;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
int i;
(void) inputBuffer; /* Prevent unused variable warning. */
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = LookupSine(data, data->left_phase); /* left */
*out++ = LookupSine(data, data->right_phase); /* right */
data->left_phase += data->phase_increment;
if( data->left_phase >= 1.0f ) data->left_phase -= 1.0f;
data->right_phase += (data->phase_increment * 1.5f); /* fifth above */
if( data->right_phase >= 1.0f ) data->right_phase -= 1.0f;
}
return 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStream *stream;
PaStreamParameters outputParameters;
PaError err;
paTestData data;
int i;
int done = 0;
printf("PortAudio Test: enter letter then hit ENTER.\n" );
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = 0.90f * (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data.sine[TABLE_SIZE] = data.sine[0]; /* set guard point. */
data.left_phase = data.right_phase = 0.0;
data.phase_increment = CalcPhaseIncrement(MIN_FREQ);
err = Pa_Initialize();
if( err != paNoError ) goto error;
printf("PortAudio Test: output device = %d\n", OUTPUT_DEVICE );
outputParameters.device = OUTPUT_DEVICE;
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
printf("Requested output latency = %.4f seconds.\n", outputParameters.suggestedLatency );
printf("%d frames per buffer.\n.", FRAMES_PER_BUFFER );
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff|paDitherOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Play ASCII keyboard. Hit 'q' to stop. (Use RETURN key on Mac)\n");
fflush(stdout);
while ( !done )
{
float freq;
int index;
char c;
do
{
c = getchar();
}
while( c < ' '); /* Strip white space and control chars. */
if( c == 'q' ) done = 1;
index = c % 26;
freq = MIN_FREQ + (index * 40.0);
data.phase_increment = CalcPhaseIncrement(freq);
}
printf("Call Pa_StopStream()\n");
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

173
portaudio-v19/pa_tests/patest_leftright.c
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@ -1,173 +0,0 @@
/** @file patest_leftright.c
@brief Play different tone sine waves that
alternate between left and right channel.
The low tone should be on the left channel.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (8)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (512)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
float sine[TABLE_SIZE];
int left_phase;
int right_phase;
int toggle;
int countDown;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer,
void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
int finished = 0;
/* Prevent unused variable warnings. */
(void) inputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
if( data->toggle )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = 0; /* right */
}
else
{
*out++ = 0; /* left */
*out++ = data->sine[data->right_phase]; /* right */
}
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
if( data->countDown < 0 )
{
data->countDown = SAMPLE_RATE;
data->toggle = !data->toggle;
}
data->countDown -= framesPerBuffer;
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStream *stream;
PaStreamParameters outputParameters;
PaError err;
paTestData data;
int i;
int timeout;
printf("Play different tone sine waves that alternate between left and right channel.\n");
printf("The low tone should be on the left channel.\n");
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data.left_phase = data.right_phase = data.toggle = 0;
data.countDown = SAMPLE_RATE;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream( &stream,
NULL, /* No input. */
&outputParameters, /* As above. */
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Play for several seconds.\n");
timeout = NUM_SECONDS * 4;
while( timeout > 0 )
{
Pa_Sleep( 300 ); /*(Irix very much likes sleeps <= 1000 ms.)*/
timeout -= 1;
}
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

140
portaudio-v19/pa_tests/patest_longsine.c
View File

@ -1,140 +0,0 @@
/** @file patest_longsine.c
@brief Play a sine wave until ENTER hit.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define SAMPLE_RATE (44100)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
float sine[TABLE_SIZE];
int left_phase;
int right_phase;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback(const void* inputBuffer,
void* outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void* userData)
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned int i;
(void) inputBuffer; /* Prevent unused argument warning. */
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
return 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
int i;
printf("PortAudio Test: output sine wave.\n");
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data.left_phase = data.right_phase = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream( &stream,
NULL, /* No input. */
&outputParameters, /* As above. */
SAMPLE_RATE,
256, /* Frames per buffer. */
paClipOff, /* No out of range samples expected. */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Hit ENTER to stop program.\n");
getchar();
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

198
portaudio-v19/pa_tests/patest_many.c
View File

@ -1,198 +0,0 @@
/** @file patest_many.c
@brief Start and stop the PortAudio Driver multiple times.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (1)
#define SAMPLE_RATE (44100)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
short sine[TABLE_SIZE];
int left_phase;
int right_phase;
unsigned int sampsToGo;
}
paTestData;
PaError TestOnce( void );
static int patest1Callback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData );
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patest1Callback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
short *out = (short*)outputBuffer;
unsigned int i;
int finished = 0;
(void) inputBuffer; /* Prevent "unused variable" warnings. */
if( data->sampsToGo < framesPerBuffer )
{
/* final buffer... */
for( i=0; i<data->sampsToGo; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
/* zero remainder of final buffer */
for( ; i<framesPerBuffer; i++ )
{
*out++ = 0; /* left */
*out++ = 0; /* right */
}
finished = 1;
}
else
{
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
data->sampsToGo -= framesPerBuffer;
}
return finished;
}
/*******************************************************************/
#ifdef MACINTOSH
int main(void);
int main(void)
{
int i;
PaError err;
int numLoops = 10;
printf("Loop %d times.\n", numLoops );
for( i=0; i<numLoops; i++ )
{
printf("Loop %d out of %d.\n", i+1, numLoops );
err = TestOnce();
if( err < 0 ) return 0;
}
}
#else
int main(int argc, char **argv);
int main(int argc, char **argv)
{
PaError err;
int i, numLoops = 10;
if( argc > 1 )
{
numLoops = atoi(argv[1]);
}
for( i=0; i<numLoops; i++ )
{
printf("Loop %d out of %d.\n", i+1, numLoops );
err = TestOnce();
if( err < 0 ) return 1;
}
printf("Test complete.\n");
return 0;
}
#endif
PaError TestOnce( void )
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
int i;
int totalSamps;
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (short) (32767.0 * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ));
}
data.left_phase = data.right_phase = 0;
data.sampsToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paInt16;
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
1024, /* frames per buffer */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patest1Callback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Waiting for sound to finish.\n");
Pa_Sleep(1000);
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
return paNoError;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

204
portaudio-v19/pa_tests/patest_maxsines.c
View File

@ -1,204 +0,0 @@
/** @file patest_maxsines.c
@brief How many sine waves can we calculate and play in less than 80% CPU Load.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define MAX_SINES (500)
#define MAX_USAGE (0.8)
#define SAMPLE_RATE (44100)
#define FREQ_TO_PHASE_INC(freq) (freq/(float)SAMPLE_RATE)
#define MIN_PHASE_INC FREQ_TO_PHASE_INC(200.0f)
#define MAX_PHASE_INC (MIN_PHASE_INC * (1 << 5))
#define FRAMES_PER_BUFFER (512)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TWOPI (M_PI * 2.0)
#define TABLE_SIZE (512)
typedef struct paTestData
{
int numSines;
float sine[TABLE_SIZE + 1]; /* add one for guard point for interpolation */
float phases[MAX_SINES];
}
paTestData;
/* Convert phase between and 1.0 to sine value
* using linear interpolation.
*/
float LookupSine( paTestData *data, float phase );
float LookupSine( paTestData *data, float phase )
{
float fIndex = phase*TABLE_SIZE;
int index = (int) fIndex;
float fract = fIndex - index;
float lo = data->sine[index];
float hi = data->sine[index+1];
float val = lo + fract*(hi-lo);
return val;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback(const void* inputBuffer,
void* outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void* userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
float outSample;
float scaler;
int numForScale;
unsigned long i;
int j;
int finished = 0;
(void) inputBuffer; /* Prevent unused argument warning. */
/* Determine amplitude scaling factor */
numForScale = data->numSines;
if( numForScale < 8 ) numForScale = 8; /* prevent pops at beginning */
scaler = 1.0f / numForScale;
for( i=0; i<framesPerBuffer; i++ )
{
float output = 0.0;
float phaseInc = MIN_PHASE_INC;
float phase;
for( j=0; j<data->numSines; j++ )
{
/* Advance phase of next oscillator. */
phase = data->phases[j];
phase += phaseInc;
if( phase >= 1.0 ) phase -= 1.0;
output += LookupSine(data, phase);
data->phases[j] = phase;
phaseInc *= 1.02f;
if( phaseInc > MAX_PHASE_INC ) phaseInc = MIN_PHASE_INC;
}
outSample = (float) (output * scaler);
*out++ = outSample; /* Left */
*out++ = outSample; /* Right */
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
int i;
PaStream* stream;
PaStreamParameters outputParameters;
PaError err;
paTestData data = {0};
double load;
printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data.sine[TABLE_SIZE] = data.sine[0]; /* set guard point */
err = Pa_Initialize();
if( err != paNoError )
goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* Default output device. */
outputParameters.channelCount = 2; /* Stereo output. */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output. */
outputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.suggestedLatency = Pa_GetDeviceInfo(outputParameters.device)
->defaultHighOutputLatency;
err = Pa_OpenStream(&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* No out of range samples should occur. */
patestCallback,
&data);
if( err != paNoError )
goto error;
err = Pa_StartStream( stream );
if( err != paNoError )
goto error;
/* Play an increasing number of sine waves until we hit MAX_USAGE */
do {
data.numSines++;
Pa_Sleep(200);
load = Pa_GetStreamCpuLoad(stream);
printf("numSines = %d, CPU load = %f\n", data.numSines, load );
fflush(stdout);
} while((load < MAX_USAGE) && (data.numSines < MAX_SINES));
Pa_Sleep(2000); /* Stay for 2 seconds around 80% CPU. */
err = Pa_StopStream( stream );
if( err != paNoError )
goto error;
err = Pa_CloseStream( stream );
if( err != paNoError )
goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

145
portaudio-v19/pa_tests/patest_mono.c
View File

@ -1,145 +0,0 @@
/*
* $Id$
* patest_sine.c
* Play a monophonic sine wave using the Portable Audio api for several seconds.
*
* Authors:
* Ross Bencina <rossb@audiomulch.com>
* Phil Burk <philburk@softsynth.com>
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (10)
#define SAMPLE_RATE (44100)
#define AMPLITUDE (0.8)
#define FRAMES_PER_BUFFER (64)
#define OUTPUT_DEVICE Pa_GetDefaultOutputDevice()
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
float sine[TABLE_SIZE];
int phase;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
int finished = 0;
/* avoid unused variable warnings */
(void) inputBuffer;
(void) timeInfo;
(void) statusFlags;
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->sine[data->phase]; /* left */
data->phase += 1;
if( data->phase >= TABLE_SIZE ) data->phase -= TABLE_SIZE;
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
int i;
printf("PortAudio Test: output MONO sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) (AMPLITUDE * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ));
}
data.phase = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = OUTPUT_DEVICE;
outputParameters.channelCount = 1; /* MONO output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Play for %d seconds.\n", NUM_SECONDS ); fflush(stdout);
Pa_Sleep( NUM_SECONDS * 1000 );
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

155
portaudio-v19/pa_tests/patest_mono_asio_channel_select.c
View File

@ -1,155 +0,0 @@
/*
* $Id$
* patest_sine.c
* Play a monophonic sine wave using the Portable Audio api for several seconds.
*
* Authors:
* Ross Bencina <rossb@audiomulch.com>
* Phil Burk <philburk@softsynth.com>
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#include "pa_asio.h"
#define NUM_SECONDS (10)
#define SAMPLE_RATE (44100)
#define AMPLITUDE (0.8)
#define FRAMES_PER_BUFFER (64)
#define OUTPUT_DEVICE Pa_GetDefaultOutputDevice()
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
float sine[TABLE_SIZE];
int phase;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
int finished = 0;
/* avoid unused variable warnings */
(void) inputBuffer;
(void) timeInfo;
(void) statusFlags;
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->sine[data->phase]; /* left */
data->phase += 1;
if( data->phase >= TABLE_SIZE ) data->phase -= TABLE_SIZE;
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaAsioStreamInfo asioOutputInfo;
PaStream *stream;
PaError err;
paTestData data;
int outputChannelSelectors[1];
int i;
printf("PortAudio Test: output MONO sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) (AMPLITUDE * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ));
}
data.phase = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = OUTPUT_DEVICE;
outputParameters.channelCount = 1; /* MONO output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
asioOutputInfo.size = sizeof(PaAsioStreamInfo);
asioOutputInfo.hostApiType = paASIO;
asioOutputInfo.version = 1;
asioOutputInfo.flags = paAsioUseChannelSelectors;
outputChannelSelectors[0] = 1; /* select the second (right) ASIO device channel */
asioOutputInfo.channelSelectors = outputChannelSelectors;
outputParameters.hostApiSpecificStreamInfo = &asioOutputInfo;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Play for %d seconds.\n", NUM_SECONDS ); fflush(stdout);
Pa_Sleep( NUM_SECONDS * 1000 );
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

194
portaudio-v19/pa_tests/patest_multi_sine.c
View File

@ -1,194 +0,0 @@
/** @file patest_multi_sine.c
@brief Play a different sine wave on each channel.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (256)
#define FREQ_INCR (300.0 / SAMPLE_RATE)
#define MAX_CHANNELS (64)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
typedef struct
{
short interleaved; /* Nonzero for interleaved / zero for non-interleaved. */
int numChannels; /* Actually used. */
double phases[MAX_CHANNELS]; /* Each channel gets its' own frequency. */
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback(const void* inputBuffer,
void* outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void* userData)
{
int frameIndex, channelIndex;
float** outputs = (float**)outputBuffer;
paTestData* data = (paTestData*)userData;
(void) inputBuffer; /* Prevent unused arg warning. */
if (data->interleaved)
{
float *out = (float*)outputBuffer; /* interleaved version */
for( frameIndex=0; frameIndex<(int)framesPerBuffer; frameIndex++ )
{
for( channelIndex=0; channelIndex<data->numChannels; channelIndex++ )
{
/* Output sine wave on every channel. */
*out++ = (float) sin(data->phases[channelIndex]);
/* Play each channel at a higher frequency. */
data->phases[channelIndex] += FREQ_INCR * (4 + channelIndex);
if( data->phases[channelIndex] >= (2.0 * M_PI) ) data->phases[channelIndex] -= (2.0 * M_PI);
}
}
}
else
{
for( frameIndex=0; frameIndex<(int)framesPerBuffer; frameIndex++ )
{
for( channelIndex=0; channelIndex<data->numChannels; channelIndex++ )
{
/* Output sine wave on every channel. */
outputs[channelIndex][frameIndex] = (float) sin(data->phases[channelIndex]);
/* Play each channel at a higher frequency. */
data->phases[channelIndex] += FREQ_INCR * (4 + channelIndex);
if( data->phases[channelIndex] >= (2.0 * M_PI) ) data->phases[channelIndex] -= (2.0 * M_PI);
}
}
}
return 0;
}
/*******************************************************************/
int test(short interleaved)
{
PaStream* stream;
PaStreamParameters outputParameters;
PaError err;
const PaDeviceInfo* pdi;
paTestData data;
short n;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* Default output device, max channels. */
pdi = Pa_GetDeviceInfo(outputParameters.device);
outputParameters.channelCount = pdi->maxOutputChannels;
if (outputParameters.channelCount > MAX_CHANNELS)
outputParameters.channelCount = MAX_CHANNELS;
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = pdi->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
data.interleaved = interleaved;
data.numChannels = outputParameters.channelCount;
for (n = 0; n < data.numChannels; n++)
data.phases[n] = 0.0; /* Phases wrap and maybe don't need initialisation. */
printf("%d ", data.numChannels);
if (interleaved)
printf("interleaved ");
else
{
printf(" non-interleaved ");
outputParameters.sampleFormat |= paNonInterleaved;
}
printf("channels.\n");
err = Pa_OpenStream(&stream,
NULL, /* No input. */
&outputParameters,
SAMPLE_RATE, /* Sample rate. */
FRAMES_PER_BUFFER, /* Frames per buffer. */
paClipOff, /* Samples never out of range, no clipping. */
patestCallback,
&data);
if (err == paNoError)
{
err = Pa_StartStream(stream);
if (err == paNoError)
{
printf("Hit ENTER to stop this test.\n");
getchar();
err = Pa_StopStream(stream);
}
Pa_CloseStream( stream );
}
return err;
}
/*******************************************************************/
int main(void)
{
PaError err;
printf("PortAudio Test: output sine wave on each channel.\n" );
err = Pa_Initialize();
if (err != paNoError)
goto done;
err = test(1); /* 1 means interleaved. */
if (err != paNoError)
goto done;
err = test(0); /* 0 means not interleaved. */
if (err != paNoError)
goto done;
printf("Test finished.\n");
done:
if (err)
{
fprintf(stderr, "An error occured while using the portaudio stream\n");
fprintf(stderr, "Error number: %d\n", err );
fprintf(stderr, "Error message: %s\n", Pa_GetErrorText(err));
}
Pa_Terminate();
return 0;
}

217
portaudio-v19/pa_tests/patest_out_underflow.c
View File

@ -1,217 +0,0 @@
/** @file patest_out_underflow.c
@brief Count output underflows (using paOutputUnderflow flag)
under overloaded and normal conditions.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2004 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define MAX_SINES (500)
#define MAX_LOAD (1.2)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (512)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TWOPI (M_PI * 2.0)
typedef struct paTestData
{
int sineCount;
double phases[MAX_SINES];
int countUnderflows;
int outputUnderflowCount;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
int j;
int finished = paContinue;
(void) timeInfo; /* Prevent unused variable warning. */
(void) inputBuffer; /* Prevent unused variable warning. */
if( data->countUnderflows && (statusFlags & paOutputUnderflow) )
data->outputUnderflowCount++;
for( i=0; i<framesPerBuffer; i++ )
{
float output = 0.0;
double phaseInc = 0.02;
double phase;
for( j=0; j<data->sineCount; j++ )
{
/* Advance phase of next oscillator. */
phase = data->phases[j];
phase += phaseInc;
if( phase > TWOPI ) phase -= TWOPI;
phaseInc *= 1.02;
if( phaseInc > 0.5 ) phaseInc *= 0.5;
/* This is not a very efficient way to calc sines. */
output += (float) sin( phase );
data->phases[j] = phase;
}
*out++ = (float) (output / data->sineCount);
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
int safeSineCount, stressedSineCount;
int safeUnderflowCount, stressedUnderflowCount;
paTestData data = {0};
double load;
printf("PortAudio Test: output sine waves, count underflows. SR = %d, BufSize = %d. MAX_LOAD = %f\n",
SAMPLE_RATE, FRAMES_PER_BUFFER, (float)MAX_LOAD );
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 1; /* mono output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Establishing load conditions...\n" );
/* Determine number of sines required to get to 50% */
do
{
data.sineCount++;
Pa_Sleep( 100 );
load = Pa_GetStreamCpuLoad( stream );
printf("sineCount = %d, CPU load = %f\n", data.sineCount, load );
}
while( load < 0.5 && data.sineCount < (MAX_SINES-1));
safeSineCount = data.sineCount;
/* Calculate target stress value then ramp up to that level*/
stressedSineCount = (int) (2.0 * data.sineCount * MAX_LOAD );
if( stressedSineCount > MAX_SINES )
stressedSineCount = MAX_SINES;
for( ; data.sineCount < stressedSineCount; data.sineCount++ )
{
Pa_Sleep( 100 );
load = Pa_GetStreamCpuLoad( stream );
printf("STRESSING: sineCount = %d, CPU load = %f\n", data.sineCount, load );
}
printf("Counting underflows for 5 seconds.\n");
data.countUnderflows = 1;
Pa_Sleep( 5000 );
stressedUnderflowCount = data.outputUnderflowCount;
data.countUnderflows = 0;
data.sineCount = safeSineCount;
printf("Resuming safe load...\n");
Pa_Sleep( 1500 );
data.outputUnderflowCount = 0;
Pa_Sleep( 1500 );
load = Pa_GetStreamCpuLoad( stream );
printf("sineCount = %d, CPU load = %f\n", data.sineCount, load );
printf("Counting underflows for 5 seconds.\n");
data.countUnderflows = 1;
Pa_Sleep( 5000 );
safeUnderflowCount = data.outputUnderflowCount;
printf("Stop stream.\n");
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
if( stressedUnderflowCount == 0 )
printf("Test failed, no output underflows detected under stress.\n");
else if( safeUnderflowCount != 0 )
printf("Test failed, %d unexpected underflows detected under safe load.\n", safeUnderflowCount);
else
printf("Test passed, %d expected output underflows detected under stress, 0 unexpected underflows detected under safe load.\n", stressedUnderflowCount );
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

269
portaudio-v19/pa_tests/patest_pink.c
View File

@ -1,269 +0,0 @@
/** @file patest_pink.c
@brief Generate Pink Noise using Gardner method.
Optimization suggested by James McCartney uses a tree
to select which random value to replace.
<pre>
x x x x x x x x x x x x x x x x
x x x x x x x x
x x x x
x x
x
</pre>
Tree is generated by counting trailing zeros in an increasing index.
When the index is zero, no random number is selected.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define PINK_MAX_RANDOM_ROWS (30)
#define PINK_RANDOM_BITS (24)
#define PINK_RANDOM_SHIFT ((sizeof(long)*8)-PINK_RANDOM_BITS)
typedef struct
{
long pink_Rows[PINK_MAX_RANDOM_ROWS];
long pink_RunningSum; /* Used to optimize summing of generators. */
int pink_Index; /* Incremented each sample. */
int pink_IndexMask; /* Index wrapped by ANDing with this mask. */
float pink_Scalar; /* Used to scale within range of -1.0 to +1.0 */
}
PinkNoise;
/* Prototypes */
static unsigned long GenerateRandomNumber( void );
void InitializePinkNoise( PinkNoise *pink, int numRows );
float GeneratePinkNoise( PinkNoise *pink );
/************************************************************/
/* Calculate pseudo-random 32 bit number based on linear congruential method. */
static unsigned long GenerateRandomNumber( void )
{
/* Change this seed for different random sequences. */
static unsigned long randSeed = 22222;
randSeed = (randSeed * 196314165) + 907633515;
return randSeed;
}
/************************************************************/
/* Setup PinkNoise structure for N rows of generators. */
void InitializePinkNoise( PinkNoise *pink, int numRows )
{
int i;
long pmax;
pink->pink_Index = 0;
pink->pink_IndexMask = (1<<numRows) - 1;
/* Calculate maximum possible signed random value. Extra 1 for white noise always added. */
pmax = (numRows + 1) * (1<<(PINK_RANDOM_BITS-1));
pink->pink_Scalar = 1.0f / pmax;
/* Initialize rows. */
for( i=0; i<numRows; i++ ) pink->pink_Rows[i] = 0;
pink->pink_RunningSum = 0;
}
#define PINK_MEASURE
#ifdef PINK_MEASURE
float pinkMax = -999.0;
float pinkMin = 999.0;
#endif
/* Generate Pink noise values between -1.0 and +1.0 */
float GeneratePinkNoise( PinkNoise *pink )
{
long newRandom;
long sum;
float output;
/* Increment and mask index. */
pink->pink_Index = (pink->pink_Index + 1) & pink->pink_IndexMask;
/* If index is zero, don't update any random values. */
if( pink->pink_Index != 0 )
{
/* Determine how many trailing zeros in PinkIndex. */
/* This algorithm will hang if n==0 so test first. */
int numZeros = 0;
int n = pink->pink_Index;
while( (n & 1) == 0 )
{
n = n >> 1;
numZeros++;
}
/* Replace the indexed ROWS random value.
* Subtract and add back to RunningSum instead of adding all the random
* values together. Only one changes each time.
*/
pink->pink_RunningSum -= pink->pink_Rows[numZeros];
newRandom = ((long)GenerateRandomNumber()) >> PINK_RANDOM_SHIFT;
pink->pink_RunningSum += newRandom;
pink->pink_Rows[numZeros] = newRandom;
}
/* Add extra white noise value. */
newRandom = ((long)GenerateRandomNumber()) >> PINK_RANDOM_SHIFT;
sum = pink->pink_RunningSum + newRandom;
/* Scale to range of -1.0 to 0.9999. */
output = pink->pink_Scalar * sum;
#ifdef PINK_MEASURE
/* Check Min/Max */
if( output > pinkMax ) pinkMax = output;
else if( output < pinkMin ) pinkMin = output;
#endif
return output;
}
/*******************************************************************/
#define PINK_TEST
#ifdef PINK_TEST
/* Context for callback routine. */
typedef struct
{
PinkNoise leftPink;
PinkNoise rightPink;
unsigned int sampsToGo;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback(const void* inputBuffer,
void* outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void* userData)
{
int finished;
int i;
int numFrames;
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
(void) inputBuffer; /* Prevent "unused variable" warnings. */
/* Are we almost at end. */
if( data->sampsToGo < framesPerBuffer )
{
numFrames = data->sampsToGo;
finished = 1;
}
else
{
numFrames = framesPerBuffer;
finished = 0;
}
for( i=0; i<numFrames; i++ )
{
*out++ = GeneratePinkNoise( &data->leftPink );
*out++ = GeneratePinkNoise( &data->rightPink );
}
data->sampsToGo -= numFrames;
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStream* stream;
PaError err;
paTestData data;
PaStreamParameters outputParameters;
int totalSamps;
static const double SR = 44100.0;
static const int FPB = 2048; /* Frames per buffer: 46 ms buffers. */
/* Initialize two pink noise signals with different numbers of rows. */
InitializePinkNoise( &data.leftPink, 12 );
InitializePinkNoise( &data.rightPink, 16 );
/* Look at a few values. */
{
int i;
float pink;
for( i=0; i<20; i++ )
{
pink = GeneratePinkNoise( &data.leftPink );
printf("Pink = %f\n", pink );
}
}
data.sampsToGo = totalSamps = (int)(60.0 * SR); /* Play a whole minute. */
err = Pa_Initialize();
if( err != paNoError ) goto error;
/* Open a stereo PortAudio stream so we can hear the result. */
outputParameters.device = Pa_GetDefaultOutputDevice(); /* Take the default output device. */
outputParameters.channelCount = 2; /* Stereo output, most likely supported. */
outputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output. */
outputParameters.suggestedLatency =
Pa_GetDeviceInfo(outputParameters.device)->defaultLowOutputLatency;
err = Pa_OpenStream(&stream,
NULL, /* No input. */
&outputParameters,
SR, /* Sample rate. */
FPB, /* Frames per buffer. */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data);
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Stereo pink noise for one minute...\n");
while( ( err = Pa_IsStreamActive( stream ) ) == 1 ) Pa_Sleep(100);
if( err < 0 ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
#ifdef PINK_MEASURE
printf("Pink min = %f, max = %f\n", pinkMin, pinkMax );
#endif
Pa_Terminate();
return 0;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return 0;
}
#endif /* PINK_TEST */

223
portaudio-v19/pa_tests/patest_prime.c
View File

@ -1,223 +0,0 @@
/** @file patest_prime.c
@brief Test stream priming mode.
@author Ross Bencina http://www.audiomulch.com/~rossb
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#include "pa_util.h"
#define NUM_BEEPS (3)
#define SAMPLE_RATE (44100)
#define SAMPLE_PERIOD (1.0/44100.0)
#define FRAMES_PER_BUFFER (256)
#define BEEP_DURATION (400)
#define IDLE_DURATION (SAMPLE_RATE*2) /* 2 seconds */
#define SLEEP_MSEC (50)
#define STATE_BKG_IDLE (0)
#define STATE_BKG_BEEPING (1)
typedef struct
{
float leftPhase;
float rightPhase;
int state;
int beepCountdown;
int idleCountdown;
}
paTestData;
static void InitializeTestData( paTestData *testData )
{
testData->leftPhase = 0;
testData->rightPhase = 0;
testData->state = STATE_BKG_BEEPING;
testData->beepCountdown = BEEP_DURATION;
testData->idleCountdown = IDLE_DURATION;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo *timeInfo,
PaStreamCallbackFlags statusFlags, void *userData )
{
/* Cast data passed through stream to our structure. */
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned int i;
int result = paContinue;
/* supress unused parameter warnings */
(void) inputBuffer;
(void) timeInfo;
(void) statusFlags;
for( i=0; i<framesPerBuffer; i++ )
{
switch( data->state )
{
case STATE_BKG_IDLE:
*out++ = 0.0; /* left */
*out++ = 0.0; /* right */
--data->idleCountdown;
if( data->idleCountdown <= 0 ) result = paComplete;
break;
case STATE_BKG_BEEPING:
if( data->beepCountdown <= 0 )
{
data->state = STATE_BKG_IDLE;
*out++ = 0.0; /* left */
*out++ = 0.0; /* right */
}
else
{
/* Play sawtooth wave. */
*out++ = data->leftPhase; /* left */
*out++ = data->rightPhase; /* right */
/* Generate simple sawtooth phaser that ranges between -1.0 and 1.0. */
data->leftPhase += 0.01f;
/* When signal reaches top, drop back down. */
if( data->leftPhase >= 1.0f ) data->leftPhase -= 2.0f;
/* higher pitch so we can distinguish left and right. */
data->rightPhase += 0.03f;
if( data->rightPhase >= 1.0f ) data->rightPhase -= 2.0f;
}
--data->beepCountdown;
break;
}
}
return result;
}
/*******************************************************************/
static PaError DoTest( int flags )
{
PaStream *stream;
PaError err;
paTestData data;
PaStreamParameters outputParameters;
InitializeTestData( &data );
outputParameters.device = Pa_GetDefaultOutputDevice();
outputParameters.channelCount = 2;
outputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.sampleFormat = paFloat32;
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultHighOutputLatency;
/* Open an audio I/O stream. */
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
paClipOff | flags, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("hear \"BEEP\"\n" );
fflush(stdout);
while( ( err = Pa_IsStreamActive( stream ) ) == 1 ) Pa_Sleep(SLEEP_MSEC);
if( err < 0 ) goto error;
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
return err;
error:
return err;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError err = paNoError;
int i;
/* Initialize library before making any other calls. */
err = Pa_Initialize();
if( err != paNoError ) goto error;
printf("PortAudio Test: Testing stream playback with no priming.\n");
printf("PortAudio Test: you should see BEEP before you hear it.\n");
printf("BEEP %d times.\n", NUM_BEEPS );
for( i=0; i< NUM_BEEPS; ++i )
{
err = DoTest( 0 );
if( err != paNoError )
goto error;
}
printf("PortAudio Test: Testing stream playback with priming.\n");
printf("PortAudio Test: you should see BEEP around the same time you hear it.\n");
for( i=0; i< NUM_BEEPS; ++i )
{
err = DoTest( paPrimeOutputBuffersUsingStreamCallback );
if( err != paNoError )
goto error;
}
printf("Test finished.\n");
Pa_Terminate();
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

229
portaudio-v19/pa_tests/patest_read_record.c
View File

@ -1,229 +0,0 @@
/** @file patest_read_record.c
@brief Record input into an array; Save array to a file; Playback recorded
data. Implemented using the blocking API (Pa_ReadStream(), Pa_WriteStream() )
@author Phil Burk http://www.softsynth.com
@author Ross Bencina rossb@audiomulch.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include "portaudio.h"
/* #define SAMPLE_RATE (17932) // Test failure to open with this value. */
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (1024)
#define NUM_SECONDS (5)
#define NUM_CHANNELS (2)
/* #define DITHER_FLAG (paDitherOff) */
#define DITHER_FLAG (0) /**/
/* Select sample format. */
#if 1
#define PA_SAMPLE_TYPE paFloat32
typedef float SAMPLE;
#define SAMPLE_SILENCE (0.0f)
#define PRINTF_S_FORMAT "%.8f"
#elif 1
#define PA_SAMPLE_TYPE paInt16
typedef short SAMPLE;
#define SAMPLE_SILENCE (0)
#define PRINTF_S_FORMAT "%d"
#elif 0
#define PA_SAMPLE_TYPE paInt8
typedef char SAMPLE;
#define SAMPLE_SILENCE (0)
#define PRINTF_S_FORMAT "%d"
#else
#define PA_SAMPLE_TYPE paUInt8
typedef unsigned char SAMPLE;
#define SAMPLE_SILENCE (128)
#define PRINTF_S_FORMAT "%d"
#endif
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters inputParameters, outputParameters;
PaStream *stream;
PaError err;
SAMPLE *recordedSamples;
int i;
int totalFrames;
int numSamples;
int numBytes;
SAMPLE max, average, val;
printf("patest_read_record.c\n"); fflush(stdout);
totalFrames = NUM_SECONDS * SAMPLE_RATE; /* Record for a few seconds. */
numSamples = totalFrames * NUM_CHANNELS;
numBytes = numSamples * sizeof(SAMPLE);
recordedSamples = (SAMPLE *) malloc( numBytes );
if( recordedSamples == NULL )
{
printf("Could not allocate record array.\n");
exit(1);
}
for( i=0; i<numSamples; i++ ) recordedSamples[i] = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
inputParameters.channelCount = NUM_CHANNELS;
inputParameters.sampleFormat = PA_SAMPLE_TYPE;
inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
/* Record some audio. -------------------------------------------- */
err = Pa_OpenStream(
&stream,
&inputParameters,
NULL, /* &outputParameters, */
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
NULL, /* no callback, use blocking API */
NULL ); /* no callback, so no callback userData */
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Now recording!!\n"); fflush(stdout);
err = Pa_ReadStream( stream, recordedSamples, totalFrames );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
/* Measure maximum peak amplitude. */
max = 0;
average = 0;
for( i=0; i<numSamples; i++ )
{
val = recordedSamples[i];
if( val < 0 ) val = -val; /* ABS */
if( val > max )
{
max = val;
}
average += val;
}
average = average / numSamples;
printf("Sample max amplitude = "PRINTF_S_FORMAT"\n", max );
printf("Sample average = "PRINTF_S_FORMAT"\n", average );
/* Was as below. Better choose at compile time because this
keeps generating compiler-warnings:
if( PA_SAMPLE_TYPE == paFloat32 )
{
printf("sample max amplitude = %f\n", max );
printf("sample average = %f\n", average );
}
else
{
printf("sample max amplitude = %d\n", max );
printf("sample average = %d\n", average );
}
*/
/* Write recorded data to a file. */
#if 0
{
FILE *fid;
fid = fopen("recorded.raw", "wb");
if( fid == NULL )
{
printf("Could not open file.");
}
else
{
fwrite( recordedSamples, NUM_CHANNELS * sizeof(SAMPLE), totalFrames, fid );
fclose( fid );
printf("Wrote data to 'recorded.raw'\n");
}
}
#endif
/* Playback recorded data. -------------------------------------------- */
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = NUM_CHANNELS;
outputParameters.sampleFormat = PA_SAMPLE_TYPE;
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
printf("Begin playback.\n"); fflush(stdout);
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
NULL, /* no callback, use blocking API */
NULL ); /* no callback, so no callback userData */
if( err != paNoError ) goto error;
if( stream )
{
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Waiting for playback to finish.\n"); fflush(stdout);
err = Pa_WriteStream( stream, recordedSamples, totalFrames );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
printf("Done.\n"); fflush(stdout);
}
free( recordedSamples );
Pa_Terminate();
return 0;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return -1;
}

337
portaudio-v19/pa_tests/patest_record.c
View File

@ -1,337 +0,0 @@
/** @file patest_record.c
@brief Record input into an array; Save array to a file; Playback recorded data.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include "portaudio.h"
/* #define SAMPLE_RATE (17932) // Test failure to open with this value. */
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (1024)
#define NUM_SECONDS (5)
#define NUM_CHANNELS (2)
/* #define DITHER_FLAG (paDitherOff) */
#define DITHER_FLAG (0) /**/
/* Select sample format. */
#if 1
#define PA_SAMPLE_TYPE paFloat32
typedef float SAMPLE;
#define SAMPLE_SILENCE (0.0f)
#define PRINTF_S_FORMAT "%.8f"
#elif 1
#define PA_SAMPLE_TYPE paInt16
typedef short SAMPLE;
#define SAMPLE_SILENCE (0)
#define PRINTF_S_FORMAT "%d"
#elif 0
#define PA_SAMPLE_TYPE paInt8
typedef char SAMPLE;
#define SAMPLE_SILENCE (0)
#define PRINTF_S_FORMAT "%d"
#else
#define PA_SAMPLE_TYPE paUInt8
typedef unsigned char SAMPLE;
#define SAMPLE_SILENCE (128)
#define PRINTF_S_FORMAT "%d"
#endif
typedef struct
{
int frameIndex; /* Index into sample array. */
int maxFrameIndex;
SAMPLE *recordedSamples;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int recordCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
const SAMPLE *rptr = (const SAMPLE*)inputBuffer;
SAMPLE *wptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS];
long framesToCalc;
long i;
int finished;
unsigned long framesLeft = data->maxFrameIndex - data->frameIndex;
(void) outputBuffer; /* Prevent unused variable warnings. */
(void) timeInfo;
(void) statusFlags;
(void) userData;
if( framesLeft < framesPerBuffer )
{
framesToCalc = framesLeft;
finished = paComplete;
}
else
{
framesToCalc = framesPerBuffer;
finished = paContinue;
}
if( inputBuffer == NULL )
{
for( i=0; i<framesToCalc; i++ )
{
*wptr++ = SAMPLE_SILENCE; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = SAMPLE_SILENCE; /* right */
}
}
else
{
for( i=0; i<framesToCalc; i++ )
{
*wptr++ = *rptr++; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */
}
}
data->frameIndex += framesToCalc;
return finished;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int playCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
SAMPLE *rptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS];
SAMPLE *wptr = (SAMPLE*)outputBuffer;
unsigned int i;
int finished;
unsigned int framesLeft = data->maxFrameIndex - data->frameIndex;
(void) inputBuffer; /* Prevent unused variable warnings. */
(void) timeInfo;
(void) statusFlags;
(void) userData;
if( framesLeft < framesPerBuffer )
{
/* final buffer... */
for( i=0; i<framesLeft; i++ )
{
*wptr++ = *rptr++; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */
}
for( ; i<framesPerBuffer; i++ )
{
*wptr++ = 0; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = 0; /* right */
}
data->frameIndex += framesLeft;
finished = paComplete;
}
else
{
for( i=0; i<framesPerBuffer; i++ )
{
*wptr++ = *rptr++; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */
}
data->frameIndex += framesPerBuffer;
finished = paContinue;
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters inputParameters,
outputParameters;
PaStream* stream;
PaError err = paNoError;
paTestData data;
int i;
int totalFrames;
int numSamples;
int numBytes;
SAMPLE max, val;
double average;
printf("patest_record.c\n"); fflush(stdout);
data.maxFrameIndex = totalFrames = NUM_SECONDS * SAMPLE_RATE; /* Record for a few seconds. */
data.frameIndex = 0;
numSamples = totalFrames * NUM_CHANNELS;
numBytes = numSamples * sizeof(SAMPLE);
data.recordedSamples = (SAMPLE *) malloc( numBytes ); /* From now on, recordedSamples is initialised. */
if( data.recordedSamples == NULL )
{
printf("Could not allocate record array.\n");
goto done;
}
for( i=0; i<numSamples; i++ ) data.recordedSamples[i] = 0;
err = Pa_Initialize();
if( err != paNoError ) goto done;
inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
inputParameters.channelCount = 2; /* stereo input */
inputParameters.sampleFormat = PA_SAMPLE_TYPE;
inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
/* Record some audio. -------------------------------------------- */
err = Pa_OpenStream(
&stream,
&inputParameters,
NULL, /* &outputParameters, */
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
recordCallback,
&data );
if( err != paNoError ) goto done;
err = Pa_StartStream( stream );
if( err != paNoError ) goto done;
printf("Now recording!!\n"); fflush(stdout);
while( ( err = Pa_IsStreamActive( stream ) ) == 1 )
{
Pa_Sleep(1000);
printf("index = %d\n", data.frameIndex ); fflush(stdout);
}
if( err < 0 ) goto done;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto done;
/* Measure maximum peak amplitude. */
max = 0;
average = 0.0;
for( i=0; i<numSamples; i++ )
{
val = data.recordedSamples[i];
if( val < 0 ) val = -val; /* ABS */
if( val > max )
{
max = val;
}
average += val;
}
average = average / (double)numSamples;
printf("sample max amplitude = "PRINTF_S_FORMAT"\n", max );
printf("sample average = %lf\n", average );
/* Write recorded data to a file. */
#if 0
{
FILE *fid;
fid = fopen("recorded.raw", "wb");
if( fid == NULL )
{
printf("Could not open file.");
}
else
{
fwrite( data.recordedSamples, NUM_CHANNELS * sizeof(SAMPLE), totalFrames, fid );
fclose( fid );
printf("Wrote data to 'recorded.raw'\n");
}
}
#endif
/* Playback recorded data. -------------------------------------------- */
data.frameIndex = 0;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = PA_SAMPLE_TYPE;
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
printf("Begin playback.\n"); fflush(stdout);
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
playCallback,
&data );
if( err != paNoError ) goto done;
if( stream )
{
err = Pa_StartStream( stream );
if( err != paNoError ) goto done;
printf("Waiting for playback to finish.\n"); fflush(stdout);
while( ( err = Pa_IsStreamActive( stream ) ) == 1 ) Pa_Sleep(100);
if( err < 0 ) goto done;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto done;
printf("Done.\n"); fflush(stdout);
}
done:
Pa_Terminate();
if( data.recordedSamples ) /* Sure it is NULL or valid. */
free( data.recordedSamples );
if( err != paNoError )
{
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
err = 1; /* Always return 0 or 1, but no other return codes. */
}
return err;
}

79
portaudio-v19/pa_tests/patest_ringmix.c
View File

@ -1,79 +0,0 @@
/** @file patest_ringmix.c
@brief Ring modulate inputs to left output, mix inputs to right output.
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include "stdio.h"
#include "portaudio.h"
/* This will be called asynchronously by the PortAudio engine. */
static int myCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
const float *in = (const float *) inputBuffer;
float *out = (float *) outputBuffer;
float leftInput, rightInput;
unsigned int i;
/* Read input buffer, process data, and fill output buffer. */
for( i=0; i<framesPerBuffer; i++ )
{
leftInput = *in++; /* Get interleaved samples from input buffer. */
rightInput = *in++;
*out++ = leftInput * rightInput; /* ring modulation */
*out++ = 0.5f * (leftInput + rightInput); /* mix */
}
return 0;
}
/* Open a PortAudioStream to input and output audio data. */
int main(void)
{
PaStream *stream;
Pa_Initialize();
Pa_OpenDefaultStream(
&stream,
2, 2, /* stereo input and output */
paFloat32, 44100.0,
64, /* 64 frames per buffer */
myCallback, NULL );
Pa_StartStream( stream );
Pa_Sleep( 10000 ); /* Sleep for 10 seconds while processing. */
Pa_StopStream( stream );
Pa_CloseStream( stream );
Pa_Terminate();
return 0;
}

125
portaudio-v19/pa_tests/patest_saw.c
View File

@ -1,125 +0,0 @@
/** @file patest_saw.c
@brief Play a simple (aliasing) sawtooth wave.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (4)
#define SAMPLE_RATE (44100)
typedef struct
{
float left_phase;
float right_phase;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
/* Cast data passed through stream to our structure. */
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned int i;
(void) inputBuffer; /* Prevent unused variable warning. */
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->left_phase; /* left */
*out++ = data->right_phase; /* right */
/* Generate simple sawtooth phaser that ranges between -1.0 and 1.0. */
data->left_phase += 0.01f;
/* When signal reaches top, drop back down. */
if( data->left_phase >= 1.0f ) data->left_phase -= 2.0f;
/* higher pitch so we can distinguish left and right. */
data->right_phase += 0.03f;
if( data->right_phase >= 1.0f ) data->right_phase -= 2.0f;
}
return 0;
}
/*******************************************************************/
static paTestData data;
int main(void);
int main(void)
{
PaStream *stream;
PaError err;
printf("PortAudio Test: output sawtooth wave.\n");
/* Initialize our data for use by callback. */
data.left_phase = data.right_phase = 0.0;
/* Initialize library before making any other calls. */
err = Pa_Initialize();
if( err != paNoError ) goto error;
/* Open an audio I/O stream. */
err = Pa_OpenDefaultStream( &stream,
0, /* no input channels */
2, /* stereo output */
paFloat32, /* 32 bit floating point output */
SAMPLE_RATE,
256, /* frames per buffer */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
/* Sleep for several seconds. */
Pa_Sleep(NUM_SECONDS*1000);
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

151
portaudio-v19/pa_tests/patest_sine.c
View File

@ -1,151 +0,0 @@
/** @file patest_sine.c
@brief Play a sine wave for several seconds.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (5)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (64)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
float sine[TABLE_SIZE];
int left_phase;
int right_phase;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
(void) timeInfo; /* Prevent unused variable warnings. */
(void) statusFlags;
(void) inputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
return paContinue;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
int i;
printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data.left_phase = data.right_phase = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Play for %d seconds.\n", NUM_SECONDS );
Pa_Sleep( NUM_SECONDS * 1000 );
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

212
portaudio-v19/pa_tests/patest_sine8.c
View File

@ -1,212 +0,0 @@
/** @file patest_sine8.c
@brief Test 8 bit data: play a sine wave for several seconds.
@author Ross Bencina <rossb@audiomulch.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (8)
#define SAMPLE_RATE (44100)
#define TABLE_SIZE (200)
#define TEST_UNSIGNED (0)
#if TEST_UNSIGNED
#define TEST_FORMAT paUInt8
#else
#define TEST_FORMAT paInt8
#endif
#ifndef M_PI
#define M_PI (3.14159265)
#endif
typedef struct
{
#if TEST_UNSIGNED
unsigned char sine[TABLE_SIZE];
#else
char sine[TABLE_SIZE];
#endif
int left_phase;
int right_phase;
unsigned int framesToGo;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
char *out = (char*)outputBuffer;
int i;
int framesToCalc;
int finished = 0;
(void) inputBuffer; /* Prevent unused variable warnings. */
if( data->framesToGo < framesPerBuffer )
{
framesToCalc = data->framesToGo;
data->framesToGo = 0;
finished = 1;
}
else
{
framesToCalc = framesPerBuffer;
data->framesToGo -= framesPerBuffer;
}
for( i=0; i<framesToCalc; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
/* zero remainder of final buffer */
for( ; i<(int)framesPerBuffer; i++ )
{
#if TEST_UNSIGNED
*out++ = (unsigned char) 0x80; /* left */
*out++ = (unsigned char) 0x80; /* right */
#else
*out++ = 0; /* left */
*out++ = 0; /* right */
#endif
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream* stream;
PaError err;
paTestData data;
PaTime streamOpened;
int i, totalSamps;
#if TEST_UNSIGNED
printf("PortAudio Test: output UNsigned 8 bit sine wave.\n");
#else
printf("PortAudio Test: output signed 8 bit sine wave.\n");
#endif
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (char) (127.0 * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ));
#if TEST_UNSIGNED
data.sine[i] += (unsigned char) 0x80;
#endif
}
data.left_phase = data.right_phase = 0;
data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */
err = Pa_Initialize();
if( err != paNoError )
goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* Default output device. */
outputParameters.channelCount = 2; /* Stereo output. */
outputParameters.sampleFormat = TEST_FORMAT;
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream( &stream,
NULL, /* No input. */
&outputParameters,
SAMPLE_RATE,
256, /* Frames per buffer. */
paClipOff, /* We won't output out of range samples so don't bother clipping them. */
patestCallback,
&data );
if( err != paNoError )
goto error;
streamOpened = Pa_GetStreamTime( stream ); /* Time in seconds when stream was opened (approx). */
err = Pa_StartStream( stream );
if( err != paNoError )
goto error;
/* Watch until sound is halfway finished. */
/* (Was ( Pa_StreamTime( stream ) < (totalSamps/2) ) in V18. */
while( (Pa_GetStreamTime( stream ) - streamOpened) < (PaTime)NUM_SECONDS / 2.0 )
Pa_Sleep(10);
/* Stop sound until ENTER hit. (Hu? don't see any keyboard-input here.) */
err = Pa_StopStream( stream );
if( err != paNoError )
goto error;
printf("Pause for 2 seconds.\n");
Pa_Sleep( 2000 );
err = Pa_StartStream( stream );
if( err != paNoError )
goto error;
printf("Waiting for sound to finish.\n");
while( ( err = Pa_IsStreamActive( stream ) ) == 1 )
Pa_Sleep(100);
if( err < 0 )
goto error;
err = Pa_CloseStream( stream );
if( err != paNoError )
goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

196
portaudio-v19/pa_tests/patest_sine_formats.c
View File

@ -1,196 +0,0 @@
/** @file patest_sine_formats.c
@brief Play a sine wave for several seconds. Test various data formats.
@author Phil Burk
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (10)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (512)
#define LEFT_FREQ (SAMPLE_RATE/256.0) /* So we hit 1.0 */
#define RIGHT_FREQ (500.0)
#define AMPLITUDE (1.0)
/* Select ONE format for testing. */
#define TEST_UINT8 (0)
#define TEST_INT8 (0)
#define TEST_INT16 (1)
#define TEST_FLOAT32 (0)
#if TEST_UINT8
#define TEST_FORMAT paUInt8
typedef unsigned char SAMPLE_t;
#define SAMPLE_ZERO (0x80)
#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(127.0 * (x)))
#define FORMAT_NAME "Unsigned 8 Bit"
#elif TEST_INT8
#define TEST_FORMAT paInt8
typedef char SAMPLE_t;
#define SAMPLE_ZERO (0)
#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(127.0 * (x)))
#define FORMAT_NAME "Signed 8 Bit"
#elif TEST_INT16
#define TEST_FORMAT paInt16
typedef short SAMPLE_t;
#define SAMPLE_ZERO (0)
#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(32767 * (x)))
#define FORMAT_NAME "Signed 16 Bit"
#elif TEST_FLOAT32
#define TEST_FORMAT paFloat32
typedef float SAMPLE_t;
#define SAMPLE_ZERO (0.0)
#define DOUBLE_TO_SAMPLE(x) ((SAMPLE_t)(x))
#define FORMAT_NAME "Float 32 Bit"
#endif
#ifndef M_PI
#define M_PI (3.14159265)
#endif
typedef struct
{
double left_phase;
double right_phase;
unsigned int framesToGo;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer,
void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
SAMPLE_t *out = (SAMPLE_t *)outputBuffer;
int i;
int framesToCalc;
int finished = 0;
(void) inputBuffer; /* Prevent unused variable warnings. */
if( data->framesToGo < framesPerBuffer )
{
framesToCalc = data->framesToGo;
data->framesToGo = 0;
finished = 1;
}
else
{
framesToCalc = framesPerBuffer;
data->framesToGo -= framesPerBuffer;
}
for( i=0; i<framesToCalc; i++ )
{
data->left_phase += (LEFT_FREQ / SAMPLE_RATE);
if( data->left_phase > 1.0) data->left_phase -= 1.0;
*out++ = DOUBLE_TO_SAMPLE( AMPLITUDE * sin( (data->left_phase * M_PI * 2. )));
data->right_phase += (RIGHT_FREQ / SAMPLE_RATE);
if( data->right_phase > 1.0) data->right_phase -= 1.0;
*out++ = DOUBLE_TO_SAMPLE( AMPLITUDE * sin( (data->right_phase * M_PI * 2. )));
}
/* zero remainder of final buffer */
for( ; i<(int)framesPerBuffer; i++ )
{
*out++ = SAMPLE_ZERO; /* left */
*out++ = SAMPLE_ZERO; /* right */
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStream *stream;
PaStreamParameters outputParameters;
PaError err;
paTestData data;
int totalSamps;
printf("PortAudio Test: output " FORMAT_NAME "\n");
data.left_phase = data.right_phase = 0.0;
data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* Default output device. */
outputParameters.channelCount = 2; /* Stereo output */
outputParameters.sampleFormat = TEST_FORMAT; /* Selected above. */
outputParameters.suggestedLatency = Pa_GetDeviceInfo(outputParameters.device)->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream( &stream,
NULL, /* No input. */
&outputParameters, /* As above. */
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Waiting %d seconds for sound to finish.\n", NUM_SECONDS );
while( ( err = Pa_IsStreamActive( stream ) ) == 1 ) Pa_Sleep(100);
if( err < 0 ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("PortAudio Test Finished: " FORMAT_NAME "\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

208
portaudio-v19/pa_tests/patest_sine_time.c
View File

@ -1,208 +0,0 @@
/** @file patest_sine_time.c
@brief Play a sine wave for several seconds, pausing in the middle.
Uses the Pa_GetStreamTime() call.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#include "pa_util.h"
#define NUM_SECONDS (8)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (64)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TWOPI (M_PI * 2.0)
#define TABLE_SIZE (200)
typedef struct
{
double left_phase;
double right_phase;
volatile PaTime outTime;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned int i;
double left_phaseInc = 0.02;
double right_phaseInc = 0.06;
double left_phase = data->left_phase;
double right_phase = data->right_phase;
(void) statusFlags; /* Prevent unused variable warnings. */
(void) inputBuffer;
data->outTime = timeInfo->outputBufferDacTime;
for( i=0; i<framesPerBuffer; i++ )
{
left_phase += left_phaseInc;
if( left_phase > TWOPI ) left_phase -= TWOPI;
*out++ = (float) sin( left_phase );
right_phase += right_phaseInc;
if( right_phase > TWOPI ) right_phase -= TWOPI;
*out++ = (float) sin( right_phase );
}
data->left_phase = left_phase;
data->right_phase = right_phase;
return paContinue;
}
/*******************************************************************/
static void ReportStreamTime( PaStream *stream, paTestData *data );
static void ReportStreamTime( PaStream *stream, paTestData *data )
{
PaTime streamTime, latency, outTime;
streamTime = Pa_GetStreamTime( stream );
outTime = data->outTime;
if( outTime < 0.0 )
{
printf("Stream time = %8.1f\n", streamTime );
}
else
{
latency = outTime - streamTime;
printf("Stream time = %8.4f, outTime = %8.4f, latency = %8.4f\n",
streamTime, outTime, latency );
}
fflush(stdout);
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
PaTime startTime;
printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
data.left_phase = data.right_phase = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
/* Watch until sound is halfway finished. */
printf("Play for %d seconds.\n", NUM_SECONDS/2 ); fflush(stdout);
data.outTime = -1.0; /* mark time for callback as undefined */
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
startTime = Pa_GetStreamTime( stream );
do
{
ReportStreamTime( stream, &data );
Pa_Sleep(100);
} while( (Pa_GetStreamTime( stream ) - startTime) < (NUM_SECONDS/2) );
/* Stop sound for 2 seconds. */
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
printf("Pause for 2 seconds.\n"); fflush(stdout);
Pa_Sleep( 2000 );
data.outTime = -1.0; /* mark time for callback as undefined */
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
startTime = Pa_GetStreamTime( stream );
printf("Play until sound is finished.\n"); fflush(stdout);
do
{
ReportStreamTime( stream, &data );
Pa_Sleep(100);
} while( (Pa_GetStreamTime( stream ) - startTime) < (NUM_SECONDS/2) );
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

162
portaudio-v19/pa_tests/patest_start_stop.c
View File

@ -1,162 +0,0 @@
/** @file patest_start_stop.c
@brief Play a sine wave for several seconds
- start and stop the stream multiple times.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define OUTPUT_DEVICE Pa_GetDefaultOutputDevice() /* default output device */
#define NUM_SECONDS (3)
#define NUM_LOOPS (4)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (400)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
float sine[TABLE_SIZE];
int left_phase;
int right_phase;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
(void) timeInfo; /* Prevent unused variable warnings. */
(void) statusFlags;
(void) inputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
return paContinue;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
int i;
printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data.left_phase = data.right_phase = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = OUTPUT_DEVICE;
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
for( i=0; i<NUM_LOOPS; i++ )
{
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Play for %d seconds.\n", NUM_SECONDS );
Pa_Sleep( NUM_SECONDS * 1000 );
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
printf("Stopped.\n" );
Pa_Sleep( 1000 );
}
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

313
portaudio-v19/pa_tests/patest_stop.c
View File

@ -1,313 +0,0 @@
/** @file patest_stop.c
@brief Test different ways of stopping audio.
Test the three ways of stopping audio:
- calling Pa_StopStream(),
- calling Pa_AbortStream(),
- and returning a 1 from the callback function.
A long latency is set up so that you can hear the difference.
Then a simple 8 note sequence is repeated twice.
The program will print what you should hear.
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define OUTPUT_DEVICE (Pa_GetDefaultOutputDevice())
#define SLEEP_DUR (200)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (256)
#define LATENCY_SECONDS (3.f)
#define FRAMES_PER_NOTE (SAMPLE_RATE/2)
#define MAX_REPEATS (2)
#define FUNDAMENTAL (400.0f / SAMPLE_RATE)
#define NOTE_0 (FUNDAMENTAL * 1.0f / 1.0f)
#define NOTE_1 (FUNDAMENTAL * 5.0f / 4.0f)
#define NOTE_2 (FUNDAMENTAL * 4.0f / 3.0f)
#define NOTE_3 (FUNDAMENTAL * 3.0f / 2.0f)
#define NOTE_4 (FUNDAMENTAL * 2.0f / 1.0f)
#define MODE_FINISH (0)
#define MODE_STOP (1)
#define MODE_ABORT (2)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (400)
typedef struct
{
float waveform[TABLE_SIZE + 1]; /* Add one for guard point for interpolation. */
float phase_increment;
float phase;
float *tune;
int notesPerTune;
int frameCounter;
int noteCounter;
int repeatCounter;
PaTime outTime;
int stopMode;
int done;
}
paTestData;
/************* Prototypes *****************************/
int TestStopMode( paTestData *data );
float LookupWaveform( paTestData *data, float phase );
/******************************************************
* Convert phase between 0.0 and 1.0 to waveform value
* using linear interpolation.
*/
float LookupWaveform( paTestData *data, float phase )
{
float fIndex = phase*TABLE_SIZE;
int index = (int) fIndex;
float fract = fIndex - index;
float lo = data->waveform[index];
float hi = data->waveform[index+1];
float val = lo + fract*(hi-lo);
return val;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
float value;
unsigned int i = 0;
int finished = paContinue;
(void) inputBuffer; /* Prevent unused variable warnings. */
(void) timeInfo;
(void) statusFlags;
/* data->outTime = outTime; */
if( !data->done )
{
for( i=0; i<framesPerBuffer; i++ )
{
/* Are we done with this note? */
if( data->frameCounter >= FRAMES_PER_NOTE )
{
data->noteCounter += 1;
data->frameCounter = 0;
/* Are we done with this tune? */
if( data->noteCounter >= data->notesPerTune )
{
data->noteCounter = 0;
data->repeatCounter += 1;
/* Are we totally done? */
if( data->repeatCounter >= MAX_REPEATS )
{
data->done = 1;
if( data->stopMode == MODE_FINISH )
{
finished = paComplete;
break;
}
}
}
data->phase_increment = data->tune[data->noteCounter];
}
value = LookupWaveform(data, data->phase);
*out++ = value; /* left */
*out++ = value; /* right */
data->phase += data->phase_increment;
if( data->phase >= 1.0f ) data->phase -= 1.0f;
data->frameCounter += 1;
}
}
/* zero remainder of final buffer */
for( ; i<framesPerBuffer; i++ )
{
*out++ = 0; /* left */
*out++ = 0; /* right */
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
paTestData data;
int i;
float simpleTune[] = { NOTE_0, NOTE_1, NOTE_2, NOTE_3, NOTE_4, NOTE_3, NOTE_2, NOTE_1 };
printf("PortAudio Test: play song and test stopping. ask for %f seconds latency\n", LATENCY_SECONDS );
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.waveform[i] = (float) (
(0.2 * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. )) +
(0.2 * sin( ((double)(3*i)/(double)TABLE_SIZE) * M_PI * 2. )) +
(0.1 * sin( ((double)(5*i)/(double)TABLE_SIZE) * M_PI * 2. ))
);
}
data.waveform[TABLE_SIZE] = data.waveform[0]; /* Set guard point. */
data.tune = &simpleTune[0];
data.notesPerTune = sizeof(simpleTune) / sizeof(float);
printf("Test MODE_FINISH - callback returns 1.\n");
printf("Should hear entire %d note tune repeated twice.\n", data.notesPerTune);
data.stopMode = MODE_FINISH;
if( TestStopMode( &data ) != paNoError )
{
printf("Test of MODE_FINISH failed!\n");
goto error;
}
printf("Test MODE_STOP - stop when song is done.\n");
printf("Should hear entire %d note tune repeated twice.\n", data.notesPerTune);
data.stopMode = MODE_STOP;
if( TestStopMode( &data ) != paNoError )
{
printf("Test of MODE_STOP failed!\n");
goto error;
}
printf("Test MODE_ABORT - abort immediately.\n");
printf("Should hear last repetition cut short by %f seconds.\n", LATENCY_SECONDS);
data.stopMode = MODE_ABORT;
if( TestStopMode( &data ) != paNoError )
{
printf("Test of MODE_ABORT failed!\n");
goto error;
}
return 0;
error:
return 1;
}
int TestStopMode( paTestData *data )
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
data->done = 0;
data->phase = 0.0;
data->frameCounter = 0;
data->noteCounter = 0;
data->repeatCounter = 0;
data->phase_increment = data->tune[data->noteCounter];
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = OUTPUT_DEVICE;
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = LATENCY_SECONDS;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
if( data->stopMode == MODE_FINISH )
{
while( ( err = Pa_IsStreamActive( stream ) ) == 1 )
{
/*printf("outTime = %g, note# = %d, repeat# = %d\n", data->outTime,
data->noteCounter, data->repeatCounter );
fflush(stdout); */
Pa_Sleep( SLEEP_DUR );
}
if( err < 0 ) goto error;
}
else
{
while( data->repeatCounter < MAX_REPEATS )
{
/*printf("outTime = %g, note# = %d, repeat# = %d\n", data->outTime,
data->noteCounter, data->repeatCounter );
fflush(stdout); */
Pa_Sleep( SLEEP_DUR );
}
}
if( data->stopMode == MODE_ABORT )
{
printf("Call Pa_AbortStream()\n");
err = Pa_AbortStream( stream );
}
else
{
printf("Call Pa_StopStream()\n");
err = Pa_StopStream( stream );
}
if( err != paNoError ) goto error;
printf("Call Pa_CloseStream()\n"); fflush(stdout);
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

432
portaudio-v19/pa_tests/patest_stop_playout.c
View File

@ -1,432 +0,0 @@
/** @file patest_stop_playout.c
@brief Test whether all queued samples are played when Pa_StopStream()
is used with a callback or read/write stream, or when the callback
returns paComplete.
@author Ross Bencina <rossb@audiomulch.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com/
* Copyright (c) 1999-2004 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (1024)
#define TONE_SECONDS (1) /* long tone */
#define TONE_FADE_SECONDS (.04) /* fades at start and end of long tone */
#define GAP_SECONDS (.25) /* gap between long tone and blip */
#define BLIP_SECONDS (.035) /* short blip */
#define NUM_REPEATS (3)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (2048)
typedef struct
{
float sine[TABLE_SIZE+1];
int repeatCount;
double phase;
double lowIncrement, highIncrement;
int gap1Length, toneLength, toneFadesLength, gap2Length, blipLength;
int gap1Countdown, toneCountdown, gap2Countdown, blipCountdown;
}
TestData;
static void RetriggerTestSignalGenerator( TestData *data )
{
data->phase = 0.;
data->gap1Countdown = data->gap1Length;
data->toneCountdown = data->toneLength;
data->gap2Countdown = data->gap2Length;
data->blipCountdown = data->blipLength;
}
static void ResetTestSignalGenerator( TestData *data )
{
data->repeatCount = 0;
RetriggerTestSignalGenerator( data );
}
static void InitTestSignalGenerator( TestData *data )
{
int signalLengthModBufferLength, i;
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data->sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data->sine[TABLE_SIZE] = data->sine[0]; /* guard point for linear interpolation */
data->lowIncrement = (330. / SAMPLE_RATE) * TABLE_SIZE;
data->highIncrement = (1760. / SAMPLE_RATE) * TABLE_SIZE;
data->gap1Length = GAP_SECONDS * SAMPLE_RATE;
data->toneLength = TONE_SECONDS * SAMPLE_RATE;
data->toneFadesLength = TONE_FADE_SECONDS * SAMPLE_RATE;
data->gap2Length = GAP_SECONDS * SAMPLE_RATE;
data->blipLength = BLIP_SECONDS * SAMPLE_RATE;
/* adjust signal length to be a multiple of the buffer length */
signalLengthModBufferLength = (data->gap1Length + data->toneLength + data->gap2Length + data->blipLength) % FRAMES_PER_BUFFER;
if( signalLengthModBufferLength > 0 )
data->toneLength += signalLengthModBufferLength;
ResetTestSignalGenerator( data );
}
#define MIN( a, b ) (((a)<(b))?(a):(b))
static void GenerateTestSignal( TestData *data, float *stereo, int frameCount )
{
int framesGenerated = 0;
float output;
long index;
float fraction;
int count, i;
while( framesGenerated < frameCount && data->repeatCount < NUM_REPEATS )
{
if( framesGenerated < frameCount && data->gap1Countdown > 0 ){
count = MIN( frameCount - framesGenerated, data->gap1Countdown );
for( i=0; i < count; ++i )
{
*stereo++ = 0.f;
*stereo++ = 0.f;
}
data->gap1Countdown -= count;
framesGenerated += count;
}
if( framesGenerated < frameCount && data->toneCountdown > 0 ){
count = MIN( frameCount - framesGenerated, data->toneCountdown );
for( i=0; i < count; ++i )
{
/* tone with data->lowIncrement phase increment */
index = (long)data->phase;
fraction = data->phase - index;
output = data->sine[ index ] + (data->sine[ index + 1 ] - data->sine[ index ]) * fraction;
data->phase += data->lowIncrement;
while( data->phase >= TABLE_SIZE )
data->phase -= TABLE_SIZE;
/* apply fade to ends */
if( data->toneCountdown < data->toneFadesLength )
{
/* cosine-bell fade out at end */
output *= (-cos(((float)data->toneCountdown / (float)data->toneFadesLength) * M_PI) + 1.) * .5;
}
else if( data->toneCountdown > data->toneLength - data->toneFadesLength )
{
/* cosine-bell fade in at start */
output *= (cos(((float)(data->toneCountdown - (data->toneLength - data->toneFadesLength)) / (float)data->toneFadesLength) * M_PI) + 1.) * .5;
}
output *= .5; /* play tone half as loud as blip */
*stereo++ = output;
*stereo++ = output;
data->toneCountdown--;
}
framesGenerated += count;
}
if( framesGenerated < frameCount && data->gap2Countdown > 0 ){
count = MIN( frameCount - framesGenerated, data->gap2Countdown );
for( i=0; i < count; ++i )
{
*stereo++ = 0.f;
*stereo++ = 0.f;
}
data->gap2Countdown -= count;
framesGenerated += count;
}
if( framesGenerated < frameCount && data->blipCountdown > 0 ){
count = MIN( frameCount - framesGenerated, data->blipCountdown );
for( i=0; i < count; ++i )
{
/* tone with data->highIncrement phase increment */
index = (long)data->phase;
fraction = data->phase - index;
output = data->sine[ index ] + (data->sine[ index + 1 ] - data->sine[ index ]) * fraction;
data->phase += data->highIncrement;
while( data->phase >= TABLE_SIZE )
data->phase -= TABLE_SIZE;
/* cosine-bell envelope over whole blip */
output *= (-cos( ((float)data->blipCountdown / (float)data->blipLength) * 2. * M_PI) + 1.) * .5;
*stereo++ = output;
*stereo++ = output;
data->blipCountdown--;
}
framesGenerated += count;
}
if( data->blipCountdown == 0 )
{
RetriggerTestSignalGenerator( data );
data->repeatCount++;
}
}
if( framesGenerated < frameCount )
{
count = frameCount - framesGenerated;
for( i=0; i < count; ++i )
{
*stereo++ = 0.f;
*stereo++ = 0.f;
}
}
}
static int IsTestSignalFinished( TestData *data )
{
if( data->repeatCount >= NUM_REPEATS )
return 1;
else
return 0;
}
static int TestCallback1( const void *inputBuffer, void *outputBuffer,
unsigned long frameCount,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
(void) inputBuffer; /* Prevent unused variable warnings. */
(void) timeInfo;
(void) statusFlags;
GenerateTestSignal( (TestData*)userData, (float*)outputBuffer, frameCount );
if( IsTestSignalFinished( (TestData*)userData ) )
return paComplete;
else
return paContinue;
}
volatile int testCallback2Finished = 0;
static int TestCallback2( const void *inputBuffer, void *outputBuffer,
unsigned long frameCount,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
(void) inputBuffer; /* Prevent unused variable warnings. */
(void) timeInfo;
(void) statusFlags;
GenerateTestSignal( (TestData*)userData, (float*)outputBuffer, frameCount );
if( IsTestSignalFinished( (TestData*)userData ) )
testCallback2Finished = 1;
return paContinue;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
TestData data;
float writeBuffer[ FRAMES_PER_BUFFER * 2 ];
printf("PortAudio Test: check that stopping stream plays out all queued samples. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
InitTestSignalGenerator( &data );
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultHighOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
/* test paComplete ---------------------------------------------------------- */
ResetTestSignalGenerator( &data );
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
TestCallback1,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("\nPlaying 'tone-blip' %d times using callback, stops by returning paComplete from callback.\n", NUM_REPEATS );
printf("If final blip is not intact, callback+paComplete implementation may be faulty.\n\n" );
while( (err = Pa_IsStreamActive( stream )) == 1 )
Pa_Sleep( 5 );
if( err != 0 ) goto error;
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Sleep( 500 );
/* test Pa_StopStream() with callback --------------------------------------- */
ResetTestSignalGenerator( &data );
testCallback2Finished = 0;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
TestCallback2,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("\nPlaying 'tone-blip' %d times using callback, stops by calling Pa_StopStream.\n", NUM_REPEATS );
printf("If final blip is not intact, callback+Pa_StopStream implementation may be faulty.\n\n" );
/* note that polling a volatile flag is not a good way to synchronise with
the callback, but it's the best we can do portably. */
while( !testCallback2Finished )
Pa_Sleep( 2 );
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Sleep( 500 );
/* test Pa_StopStream() with Pa_WriteStream --------------------------------- */
ResetTestSignalGenerator( &data );
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
NULL, /* no callback, use blocking API */
NULL ); /* no callback, so no callback userData */
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("\nPlaying 'tone-blip' %d times using Pa_WriteStream, stops by calling Pa_StopStream.\n", NUM_REPEATS );
printf("If final blip is not intact, Pa_WriteStream+Pa_StopStream implementation may be faulty.\n\n" );
do{
GenerateTestSignal( &data, writeBuffer, FRAMES_PER_BUFFER );
err = Pa_WriteStream( stream, writeBuffer, FRAMES_PER_BUFFER );
if( err != paNoError ) goto error;
}while( !IsTestSignalFinished( &data ) );
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
/* -------------------------------------------------------------------------- */
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

259
portaudio-v19/pa_tests/patest_sync.c
View File

@ -1,259 +0,0 @@
/** @file patest_sync.c
@brief Test time stamping and synchronization of audio and video.
A high latency is used so we can hear the difference in time.
Random durations are used so we know we are hearing the right beep
and not the one before or after.
Sequence of events:
-# Foreground requests a beep.
-# Background randomly schedules a beep.
-# Foreground waits for the beep to be heard based on PaUtil_GetTime().
-# Foreground outputs video (printf) in sync with audio.
-# Repeat.
@author Phil Burk http://www.softsynth.com
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#include "pa_util.h"
#define NUM_BEEPS (6)
#define SAMPLE_RATE (44100)
#define SAMPLE_PERIOD (1.0/44100.0)
#define FRAMES_PER_BUFFER (256)
#define BEEP_DURATION (400)
#define LATENCY_MSEC (2000)
#define SLEEP_MSEC (10)
#define TIMEOUT_MSEC (15000)
#define STATE_BKG_IDLE (0)
#define STATE_BKG_PENDING (1)
#define STATE_BKG_BEEPING (2)
typedef struct
{
float left_phase;
float right_phase;
int state;
int requestBeep; /* Set by foreground, cleared by background. */
PaTime beepTime;
int beepCount;
double latency; /* For debugging. */
}
paTestData;
static unsigned long GenerateRandomNumber( void );
/************************************************************/
/* Calculate pseudo-random 32 bit number based on linear congruential method. */
static unsigned long GenerateRandomNumber( void )
{
static unsigned long randSeed = 99887766; /* Change this for different random sequences. */
randSeed = (randSeed * 196314165) + 907633515;
return randSeed;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo *timeInfo,
PaStreamCallbackFlags statusFlags, void *userData )
{
/* Cast data passed through stream to our structure. */
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned int i;
(void) inputBuffer;
data->latency = timeInfo->outputBufferDacTime - timeInfo->currentTime;
for( i=0; i<framesPerBuffer; i++ )
{
switch( data->state )
{
case STATE_BKG_IDLE:
/* Schedule beep at some random time in the future. */
if( data->requestBeep )
{
int random = GenerateRandomNumber() >> 14;
data->beepTime = timeInfo->outputBufferDacTime + (( (double)(random + SAMPLE_RATE)) * SAMPLE_PERIOD );
data->state = STATE_BKG_PENDING;
}
*out++ = 0.0; /* left */
*out++ = 0.0; /* right */
break;
case STATE_BKG_PENDING:
if( (timeInfo->outputBufferDacTime + (i*SAMPLE_PERIOD)) >= data->beepTime )
{
data->state = STATE_BKG_BEEPING;
data->beepCount = BEEP_DURATION;
data->left_phase = data->right_phase = 0.0;
}
*out++ = 0.0; /* left */
*out++ = 0.0; /* right */
break;
case STATE_BKG_BEEPING:
if( data->beepCount <= 0 )
{
data->state = STATE_BKG_IDLE;
data->requestBeep = 0;
*out++ = 0.0; /* left */
*out++ = 0.0; /* right */
}
else
{
/* Play sawtooth wave. */
*out++ = data->left_phase; /* left */
*out++ = data->right_phase; /* right */
/* Generate simple sawtooth phaser that ranges between -1.0 and 1.0. */
data->left_phase += 0.01f;
/* When signal reaches top, drop back down. */
if( data->left_phase >= 1.0f ) data->left_phase -= 2.0f;
/* higher pitch so we can distinguish left and right. */
data->right_phase += 0.03f;
if( data->right_phase >= 1.0f ) data->right_phase -= 2.0f;
}
data->beepCount -= 1;
break;
default:
data->state = STATE_BKG_IDLE;
break;
}
}
return 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStream *stream;
PaError err;
paTestData DATA;
int i, timeout;
PaTime previousTime;
PaStreamParameters outputParameters;
printf("PortAudio Test: you should see BEEP at the same time you hear it.\n");
printf("Wait for a few seconds random delay between BEEPs.\n");
printf("BEEP %d times.\n", NUM_BEEPS );
/* Initialize our DATA for use by callback. */
DATA.left_phase = DATA.right_phase = 0.0;
DATA.state = STATE_BKG_IDLE;
DATA.requestBeep = 0;
/* Initialize library before making any other calls. */
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice();
outputParameters.channelCount = 2;
outputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.sampleFormat = paFloat32;
outputParameters.suggestedLatency = (double)LATENCY_MSEC / 1000;
/* Open an audio I/O stream. */
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&DATA );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("started\n");
fflush(stdout);
previousTime = Pa_GetStreamTime( stream );
for( i=0; i<NUM_BEEPS; i++ )
{
/* Request a beep from background. */
DATA.requestBeep = 1;
/* Wait for background to acknowledge request. */
timeout = TIMEOUT_MSEC;
while( (DATA.requestBeep == 1) && (timeout-- > 0 ) ) Pa_Sleep(SLEEP_MSEC);
if( timeout <= 0 )
{
fprintf( stderr, "Timed out waiting for background to acknowledge request.\n" );
goto error;
}
printf("calc beep for %9.3f, latency = %6.3f\n", DATA.beepTime, DATA.latency );
fflush(stdout);
/* Wait for scheduled beep time. */
timeout = TIMEOUT_MSEC + (10000/SLEEP_MSEC);
while( (Pa_GetStreamTime( stream ) < DATA.beepTime) && (timeout-- > 0 ) )
{
Pa_Sleep(SLEEP_MSEC);
}
if( timeout <= 0 )
{
fprintf( stderr, "Timed out waiting for time. Now = %9.3f, Beep for %9.3f.\n",
PaUtil_GetTime(), DATA.beepTime );
goto error;
}
/* Beep should be sounding now so print synchronized BEEP. */
printf("hear \"BEEP\" at %9.3f, delta = %9.3f\n",
Pa_GetStreamTime( stream ), (DATA.beepTime - previousTime) );
fflush(stdout);
previousTime = DATA.beepTime;
}
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

173
portaudio-v19/pa_tests/patest_toomanysines.c
View File

@ -1,173 +0,0 @@
/** @file patest_toomanysines.c
@brief Play more sine waves than we can handle in real time as a stress test.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define MAX_SINES (500)
#define MAX_LOAD (1.2)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (512)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TWOPI (M_PI * 2.0)
typedef struct paTestData
{
int numSines;
double phases[MAX_SINES];
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
int j;
int finished = 0;
(void) inputBuffer; /* Prevent unused variable warning. */
for( i=0; i<framesPerBuffer; i++ )
{
float output = 0.0;
double phaseInc = 0.02;
double phase;
for( j=0; j<data->numSines; j++ )
{
/* Advance phase of next oscillator. */
phase = data->phases[j];
phase += phaseInc;
if( phase > TWOPI ) phase -= TWOPI;
phaseInc *= 1.02;
if( phaseInc > 0.5 ) phaseInc *= 0.5;
/* This is not a very efficient way to calc sines. */
output += (float) sin( phase );
data->phases[j] = phase;
}
*out++ = (float) (output / data->numSines);
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
int numStress;
paTestData data = {0};
double load;
printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d. MAX_LOAD = %f\n",
SAMPLE_RATE, FRAMES_PER_BUFFER, MAX_LOAD );
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 1; /* mono output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
/* Determine number of sines required to get to 50% */
do
{
data.numSines++;
Pa_Sleep( 100 );
load = Pa_GetStreamCpuLoad( stream );
printf("numSines = %d, CPU load = %f\n", data.numSines, load );
}
while( load < 0.5 );
/* Calculate target stress value then ramp up to that level*/
numStress = (int) (2.0 * data.numSines * MAX_LOAD );
for( ; data.numSines < numStress; data.numSines++ )
{
Pa_Sleep( 200 );
load = Pa_GetStreamCpuLoad( stream );
printf("STRESSING: numSines = %d, CPU load = %f\n", data.numSines, load );
}
printf("Suffer for 5 seconds.\n");
Pa_Sleep( 5000 );
printf("Stop stream.\n");
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

168
portaudio-v19/pa_tests/patest_two_rates.c
View File

@ -1,168 +0,0 @@
/*
* $Id$
* patest_two_rates.c
* Play two streams at different rates to make sure they don't interfere.
*
* Author: Phil Burk http://www.softsynth.com
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID())
#define SAMPLE_RATE_1 (44100)
#define SAMPLE_RATE_2 (44100)
#define FRAMES_PER_BUFFER (256)
#define FREQ_INCR (0.1)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
typedef struct
{
double phase;
double numFrames;
} paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
int frameIndex, channelIndex;
int finished = 0;
(void) outTime; /* Prevent unused variable warnings. */
(void) inputBuffer;
for( frameIndex=0; frameIndex<(int)framesPerBuffer; frameIndex++ )
{
/* Generate sine wave. */
float value = (float) 0.3 * sin(data->phase);
/* Stereo - two channels. */
*out++ = value;
*out++ = value;
data->phase += FREQ_INCR;
if( data->phase >= (2.0 * M_PI) ) data->phase -= (2.0 * M_PI);
}
return 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError err;
PortAudioStream *stream1;
PortAudioStream *stream2;
paTestData data1 = {0};
paTestData data2 = {0};
printf("PortAudio Test: two rates.\n" );
err = Pa_Initialize();
if( err != paNoError ) goto error;
/* Start first stream. **********************/
err = Pa_OpenStream(
&stream1,
paNoDevice, /* default input device */
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
OUTPUT_DEVICE,
2, /* Stereo */
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE_1,
FRAMES_PER_BUFFER, /* frames per buffer */
0, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data1 );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream1 );
if( err != paNoError ) goto error;
Pa_Sleep( 3 * 1000 );
/* Start second stream. **********************/
err = Pa_OpenStream(
&stream2,
paNoDevice, /* default input device */
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
OUTPUT_DEVICE,
2, /* Stereo */
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE_2,
FRAMES_PER_BUFFER, /* frames per buffer */
0, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data2 );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream2 );
if( err != paNoError ) goto error;
Pa_Sleep( 3 * 1000 );
err = Pa_StopStream( stream2 );
if( err != paNoError ) goto error;
Pa_Sleep( 3 * 1000 );
err = Pa_StopStream( stream1 );
if( err != paNoError ) goto error;
Pa_CloseStream( stream2 );
Pa_CloseStream( stream1 );
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

151
portaudio-v19/pa_tests/patest_underflow.c
View File

@ -1,151 +0,0 @@
/** @file patest_underflow.c
@brief Simulate an output buffer underflow condition.
Tests whether the stream can be stopped when underflowing buffers.
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.audiomulch.com/portaudio/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (20)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (2048)
#define MSEC_PER_BUFFER ( (FRAMES_PER_BUFFER * 1000) / SAMPLE_RATE )
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
float sine[TABLE_SIZE];
int left_phase;
int right_phase;
int sleepTime;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
int finished = 0;
(void) inputBuffer; /* Prevent unused variable warnings. */
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
}
/* Cause underflow to occur. */
if( data->sleepTime > 0 ) Pa_Sleep( data->sleepTime );
data->sleepTime += 1;
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
int i;
printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data.left_phase = data.right_phase = data.sleepTime = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
while( data.sleepTime < (2 * MSEC_PER_BUFFER) )
{
printf("SleepTime = %d\n", data.sleepTime );
Pa_Sleep( data.sleepTime );
}
printf("Try to stop stream.\n");
err = Pa_StopStream( stream ); /* */
err = Pa_AbortStream( stream ); /* */
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

284
portaudio-v19/pa_tests/patest_wire.c
View File

@ -1,284 +0,0 @@
/** @file patest_wire.c
@brief Pass input directly to output.
Note that some HW devices, for example many ISA audio cards
on PCs, do NOT support full duplex! For a PC, you normally need
a PCI based audio card such as the SBLive.
@author Phil Burk http://www.softsynth.com
While adapting to V19-API, I excluded configs with framesPerCallback=0
because of an assert in file pa_common/pa_process.c. Pieter, Oct 9, 2003.
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define SAMPLE_RATE (44100)
typedef struct WireConfig_s
{
int isInputInterleaved;
int isOutputInterleaved;
int numInputChannels;
int numOutputChannels;
int framesPerCallback;
} WireConfig_t;
#define USE_FLOAT_INPUT (1)
#define USE_FLOAT_OUTPUT (1)
/* Latencies set to defaults. */
#if USE_FLOAT_INPUT
#define INPUT_FORMAT paFloat32
typedef float INPUT_SAMPLE;
#else
#define INPUT_FORMAT paInt16
typedef short INPUT_SAMPLE;
#endif
#if USE_FLOAT_OUTPUT
#define OUTPUT_FORMAT paFloat32
typedef float OUTPUT_SAMPLE;
#else
#define OUTPUT_FORMAT paInt16
typedef short OUTPUT_SAMPLE;
#endif
double gInOutScaler = 1.0;
#define CONVERT_IN_TO_OUT(in) ((OUTPUT_SAMPLE) ((in) * gInOutScaler))
#define INPUT_DEVICE (Pa_GetDefaultInputDevice())
#define OUTPUT_DEVICE (Pa_GetDefaultOutputDevice())
static PaError TestConfiguration( WireConfig_t *config );
static int wireCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData );
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int wireCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
INPUT_SAMPLE *in;
OUTPUT_SAMPLE *out;
int inStride;
int outStride;
int inDone = 0;
int outDone = 0;
WireConfig_t *config = (WireConfig_t *) userData;
unsigned int i;
int inChannel, outChannel;
/* This may get called with NULL inputBuffer during initial setup. */
if( inputBuffer == NULL) return 0;
inChannel=0, outChannel=0;
while( !(inDone && outDone) )
{
if( config->isInputInterleaved )
{
in = ((INPUT_SAMPLE*)inputBuffer) + inChannel;
inStride = config->numInputChannels;
}
else
{
in = ((INPUT_SAMPLE**)inputBuffer)[inChannel];
inStride = 1;
}
if( config->isOutputInterleaved )
{
out = ((OUTPUT_SAMPLE*)outputBuffer) + outChannel;
outStride = config->numOutputChannels;
}
else
{
out = ((OUTPUT_SAMPLE**)outputBuffer)[outChannel];
outStride = 1;
}
for( i=0; i<framesPerBuffer; i++ )
{
*out = CONVERT_IN_TO_OUT( *in );
out += outStride;
in += inStride;
}
if(inChannel < (config->numInputChannels - 1)) inChannel++;
else inDone = 1;
if(outChannel < (config->numOutputChannels - 1)) outChannel++;
else outDone = 1;
}
return 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError err;
WireConfig_t CONFIG;
WireConfig_t *config = &CONFIG;
int configIndex = 0;;
err = Pa_Initialize();
if( err != paNoError ) goto error;
printf("Please connect audio signal to input and listen for it on output!\n");
printf("input format = %lu\n", INPUT_FORMAT );
printf("output format = %lu\n", OUTPUT_FORMAT );
printf("input device ID = %d\n", INPUT_DEVICE );
printf("output device ID = %d\n", OUTPUT_DEVICE );
if( INPUT_FORMAT == OUTPUT_FORMAT )
{
gInOutScaler = 1.0;
}
else if( (INPUT_FORMAT == paInt16) && (OUTPUT_FORMAT == paFloat32) )
{
gInOutScaler = 1.0/32768.0;
}
else if( (INPUT_FORMAT == paFloat32) && (OUTPUT_FORMAT == paInt16) )
{
gInOutScaler = 32768.0;
}
for( config->isInputInterleaved = 0; config->isInputInterleaved < 2; config->isInputInterleaved++ )
{
for( config->isOutputInterleaved = 0; config->isOutputInterleaved < 2; config->isOutputInterleaved++ )
{
for( config->numInputChannels = 1; config->numInputChannels < 3; config->numInputChannels++ )
{
for( config->numOutputChannels = 1; config->numOutputChannels < 3; config->numOutputChannels++ )
{
/* If framesPerCallback = 0, assertion fails in file pa_common/pa_process.c, line 1413: EX. */
for( config->framesPerCallback = 64; config->framesPerCallback < 129; config->framesPerCallback += 64 )
{
printf("-----------------------------------------------\n" );
printf("Configuration #%d\n", configIndex++ );
err = TestConfiguration( config );
/* Give user a chance to bail out. */
if( err == 1 )
{
err = paNoError;
goto done;
}
else if( err != paNoError ) goto error;
}
}
}
}
}
done:
Pa_Terminate();
printf("Full duplex sound test complete.\n"); fflush(stdout);
printf("Hit ENTER to quit.\n"); fflush(stdout);
getchar();
return 0;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
printf("Hit ENTER to quit.\n"); fflush(stdout);
getchar();
return -1;
}
static PaError TestConfiguration( WireConfig_t *config )
{
int c;
PaError err;
PaStream *stream;
PaStreamParameters inputParameters, outputParameters;
printf("input %sinterleaved!\n", (config->isInputInterleaved ? " " : "NOT ") );
printf("output %sinterleaved!\n", (config->isOutputInterleaved ? " " : "NOT ") );
printf("input channels = %d\n", config->numInputChannels );
printf("output channels = %d\n", config->numOutputChannels );
printf("framesPerCallback = %d\n", config->framesPerCallback );
inputParameters.device = INPUT_DEVICE; /* default input device */
inputParameters.channelCount = config->numInputChannels;
inputParameters.sampleFormat = INPUT_FORMAT | (config->isInputInterleaved ? 0 : paNonInterleaved);
inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
outputParameters.device = OUTPUT_DEVICE; /* default output device */
outputParameters.channelCount = config->numOutputChannels;
outputParameters.sampleFormat = OUTPUT_FORMAT | (config->isOutputInterleaved ? 0 : paNonInterleaved);
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
&inputParameters,
&outputParameters,
SAMPLE_RATE,
config->framesPerCallback, /* frames per buffer */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
wireCallback,
config );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Hit ENTER for next configuration, or 'q' to quit.\n"); fflush(stdout);
c = getchar();
printf("Closing stream.\n");
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
if( c == 'q' ) return 1;
error:
return err;
}

147
portaudio-v19/pa_tests/patest_write_sine.c
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@ -1,147 +0,0 @@
/** @file patest_write_sine.c
@brief Play a sine wave for several seconds using the blocking API (Pa_WriteStream())
@author Ross Bencina <rossb@audiomulch.com>
@author Phil Burk <philburk@softsynth.com>
*/
/*
* $Id$
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (5)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (1024)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
float buffer[FRAMES_PER_BUFFER][2]; /* stereo output buffer */
float sine[TABLE_SIZE]; /* sine wavetable */
int left_phase = 0;
int right_phase = 0;
int left_inc = 1;
int right_inc = 3; /* higher pitch so we can distinguish left and right. */
int i, j, k;
int bufferCount;
printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
outputParameters.channelCount = 2; /* stereo output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
NULL, /* no callback, use blocking API */
NULL ); /* no callback, so no callback userData */
if( err != paNoError ) goto error;
printf( "Play 3 times, higher each time.\n" );
for( k=0; k < 3; ++k )
{
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Play for %d seconds.\n", NUM_SECONDS );
bufferCount = ((NUM_SECONDS * SAMPLE_RATE) / FRAMES_PER_BUFFER);
for( i=0; i < bufferCount; i++ )
{
for( j=0; j < FRAMES_PER_BUFFER; j++ )
{
buffer[j][0] = sine[left_phase]; /* left */
buffer[j][1] = sine[right_phase]; /* right */
left_phase += left_inc;
if( left_phase >= TABLE_SIZE ) left_phase -= TABLE_SIZE;
right_phase += right_inc;
if( right_phase >= TABLE_SIZE ) right_phase -= TABLE_SIZE;
}
err = Pa_WriteStream( stream, buffer, FRAMES_PER_BUFFER );
if( err != paNoError ) goto error;
}
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
++left_inc;
++right_inc;
Pa_Sleep( 1000 );
}
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}

BIN
portaudio-v19/pa_unix_oss/low_latency_tip.txt
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portaudio-v19/pa_unix_oss/pa_unix_oss.c
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1922
portaudio-v19/pa_unix_oss/pa_unix_oss.c.orig
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114
portaudio-v19/pa_unix_oss/recplay.c
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@ -1,114 +0,0 @@
/*
* 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 );
}

616
portaudio-v19/pa_win_ds/dsound_wrapper.c
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@ -1,616 +0,0 @@
/*
* $Id$
* Simplified DirectSound interface.
*
* Author: Phil Burk & Robert Marsanyi
*
* PortAudio Portable Real-Time Audio Library
* For more information see: http://www.softsynth.com/portaudio/
* DirectSound Implementation
* Copyright (c) 1999-2000 Phil Burk & Robert Marsanyi
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include "dsound_wrapper.h"
#include "pa_trace.h"
/*
Rather than linking with dxguid.a or using "#define INITGUID" to force a
header file to instantiate the required GUID(s), we define them directly
below.
*/
#include <initguid.h> // needed for the DEFINE_GUID macro
DEFINE_GUID(IID_IDirectSoundNotify, 0xb0210783, 0x89cd, 0x11d0, 0xaf, 0x8, 0x0, 0xa0, 0xc9, 0x25, 0xcd, 0x16);
/************************************************************************************/
DSoundEntryPoints dswDSoundEntryPoints = { 0, 0, 0, 0, 0, 0, 0 };
/************************************************************************************/
static HRESULT WINAPI DummyDirectSoundCreate(LPGUID lpcGuidDevice, LPDIRECTSOUND *ppDS, LPUNKNOWN pUnkOuter)
{
(void)lpcGuidDevice; /* unused parameter */
(void)ppDS; /* unused parameter */
(void)pUnkOuter; /* unused parameter */
return E_NOTIMPL;
}
static HRESULT WINAPI DummyDirectSoundEnumerateW(LPDSENUMCALLBACKW lpDSEnumCallback, LPVOID lpContext)
{
(void)lpDSEnumCallback; /* unused parameter */
(void)lpContext; /* unused parameter */
return E_NOTIMPL;
}
static HRESULT WINAPI DummyDirectSoundEnumerateA(LPDSENUMCALLBACKA lpDSEnumCallback, LPVOID lpContext)
{
(void)lpDSEnumCallback; /* unused parameter */
(void)lpContext; /* unused parameter */
return E_NOTIMPL;
}
static HRESULT WINAPI DummyDirectSoundCaptureCreate(LPGUID lpcGUID, LPDIRECTSOUNDCAPTURE *lplpDSC, LPUNKNOWN pUnkOuter)
{
(void)lpcGUID; /* unused parameter */
(void)lplpDSC; /* unused parameter */
(void)pUnkOuter; /* unused parameter */
return E_NOTIMPL;
}
static HRESULT WINAPI DummyDirectSoundCaptureEnumerateW(LPDSENUMCALLBACKW lpDSCEnumCallback, LPVOID lpContext)
{
(void)lpDSCEnumCallback; /* unused parameter */
(void)lpContext; /* unused parameter */
return E_NOTIMPL;
}
static HRESULT WINAPI DummyDirectSoundCaptureEnumerateA(LPDSENUMCALLBACKA lpDSCEnumCallback, LPVOID lpContext)
{
(void)lpDSCEnumCallback; /* unused parameter */
(void)lpContext; /* unused parameter */
return E_NOTIMPL;
}
/************************************************************************************/
void DSW_InitializeDSoundEntryPoints(void)
{
dswDSoundEntryPoints.hInstance_ = LoadLibrary("dsound.dll");
if( dswDSoundEntryPoints.hInstance_ != NULL )
{
dswDSoundEntryPoints.DirectSoundCreate =
(HRESULT (WINAPI *)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN))
GetProcAddress( dswDSoundEntryPoints.hInstance_, "DirectSoundCreate" );
if( dswDSoundEntryPoints.DirectSoundCreate == NULL )
dswDSoundEntryPoints.DirectSoundCreate = DummyDirectSoundCreate;
dswDSoundEntryPoints.DirectSoundEnumerateW =
(HRESULT (WINAPI *)(LPDSENUMCALLBACKW, LPVOID))
GetProcAddress( dswDSoundEntryPoints.hInstance_, "DirectSoundEnumerateW" );
if( dswDSoundEntryPoints.DirectSoundEnumerateW == NULL )
dswDSoundEntryPoints.DirectSoundEnumerateW = DummyDirectSoundEnumerateW;
dswDSoundEntryPoints.DirectSoundEnumerateA =
(HRESULT (WINAPI *)(LPDSENUMCALLBACKA, LPVOID))
GetProcAddress( dswDSoundEntryPoints.hInstance_, "DirectSoundEnumerateA" );
if( dswDSoundEntryPoints.DirectSoundEnumerateA == NULL )
dswDSoundEntryPoints.DirectSoundEnumerateA = DummyDirectSoundEnumerateA;
dswDSoundEntryPoints.DirectSoundCaptureCreate =
(HRESULT (WINAPI *)(LPGUID, LPDIRECTSOUNDCAPTURE *, LPUNKNOWN))
GetProcAddress( dswDSoundEntryPoints.hInstance_, "DirectSoundCaptureCreate" );
if( dswDSoundEntryPoints.DirectSoundCaptureCreate == NULL )
dswDSoundEntryPoints.DirectSoundCaptureCreate = DummyDirectSoundCaptureCreate;
dswDSoundEntryPoints.DirectSoundCaptureEnumerateW =
(HRESULT (WINAPI *)(LPDSENUMCALLBACKW, LPVOID))
GetProcAddress( dswDSoundEntryPoints.hInstance_, "DirectSoundCaptureEnumerateW" );
if( dswDSoundEntryPoints.DirectSoundCaptureEnumerateW == NULL )
dswDSoundEntryPoints.DirectSoundCaptureEnumerateW = DummyDirectSoundCaptureEnumerateW;
dswDSoundEntryPoints.DirectSoundCaptureEnumerateA =
(HRESULT (WINAPI *)(LPDSENUMCALLBACKA, LPVOID))
GetProcAddress( dswDSoundEntryPoints.hInstance_, "DirectSoundCaptureEnumerateA" );
if( dswDSoundEntryPoints.DirectSoundCaptureEnumerateA == NULL )
dswDSoundEntryPoints.DirectSoundCaptureEnumerateA = DummyDirectSoundCaptureEnumerateA;
}
else
{
/* initialize with dummy entry points to make live easy when ds isn't present */
dswDSoundEntryPoints.DirectSoundCreate = DummyDirectSoundCreate;
dswDSoundEntryPoints.DirectSoundEnumerateW = DummyDirectSoundEnumerateW;
dswDSoundEntryPoints.DirectSoundEnumerateA = DummyDirectSoundEnumerateA;
dswDSoundEntryPoints.DirectSoundCaptureCreate = DummyDirectSoundCaptureCreate;
dswDSoundEntryPoints.DirectSoundCaptureEnumerateW = DummyDirectSoundCaptureEnumerateW;
dswDSoundEntryPoints.DirectSoundCaptureEnumerateA = DummyDirectSoundCaptureEnumerateA;
}
}
/************************************************************************************/
void DSW_TerminateDSoundEntryPoints(void)
{
if( dswDSoundEntryPoints.hInstance_ != NULL )
{
FreeLibrary( dswDSoundEntryPoints.hInstance_ );
dswDSoundEntryPoints.hInstance_ = NULL;
/* ensure that we crash reliably if the entry points arent initialised */
dswDSoundEntryPoints.DirectSoundCreate = 0;
dswDSoundEntryPoints.DirectSoundEnumerateW = 0;
dswDSoundEntryPoints.DirectSoundEnumerateA = 0;
dswDSoundEntryPoints.DirectSoundCaptureCreate = 0;
dswDSoundEntryPoints.DirectSoundCaptureEnumerateW = 0;
dswDSoundEntryPoints.DirectSoundCaptureEnumerateA = 0;
}
}
/************************************************************************************/
void DSW_Term( DSoundWrapper *dsw )
{
// Cleanup the sound buffers
if (dsw->dsw_OutputBuffer)
{
IDirectSoundBuffer_Stop( dsw->dsw_OutputBuffer );
IDirectSoundBuffer_Release( dsw->dsw_OutputBuffer );
dsw->dsw_OutputBuffer = NULL;
}
if (dsw->dsw_InputBuffer)
{
IDirectSoundCaptureBuffer_Stop( dsw->dsw_InputBuffer );
IDirectSoundCaptureBuffer_Release( dsw->dsw_InputBuffer );
dsw->dsw_InputBuffer = NULL;
}
if (dsw->dsw_pDirectSoundCapture)
{
IDirectSoundCapture_Release( dsw->dsw_pDirectSoundCapture );
dsw->dsw_pDirectSoundCapture = NULL;
}
if (dsw->dsw_pDirectSound)
{
IDirectSound_Release( dsw->dsw_pDirectSound );
dsw->dsw_pDirectSound = NULL;
}
}
/************************************************************************************/
HRESULT DSW_Init( DSoundWrapper *dsw )
{
memset( dsw, 0, sizeof(DSoundWrapper) );
return 0;
}
/************************************************************************************/
HRESULT DSW_InitOutputDevice( DSoundWrapper *dsw, LPGUID lpGUID )
{
// Create the DS object
HRESULT hr = dswDSoundEntryPoints.DirectSoundCreate( lpGUID, &dsw->dsw_pDirectSound, NULL );
if( hr != DS_OK ) return hr;
return hr;
}
/************************************************************************************/
HRESULT DSW_InitOutputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, WORD nChannels, int bytesPerBuffer )
{
DWORD dwDataLen;
DWORD playCursor;
HRESULT result;
LPDIRECTSOUNDBUFFER pPrimaryBuffer;
HWND hWnd;
HRESULT hr;
WAVEFORMATEX wfFormat;
DSBUFFERDESC primaryDesc;
DSBUFFERDESC secondaryDesc;
unsigned char* pDSBuffData;
LARGE_INTEGER counterFrequency;
dsw->dsw_OutputSize = bytesPerBuffer;
dsw->dsw_OutputRunning = FALSE;
dsw->dsw_OutputUnderflows = 0;
dsw->dsw_FramesWritten = 0;
dsw->dsw_BytesPerOutputFrame = nChannels * sizeof(short);
// We were using getForegroundWindow() but sometimes the ForegroundWindow may not be the
// applications's window. Also if that window is closed before the Buffer is closed
// then DirectSound can crash. (Thanks for Scott Patterson for reporting this.)
// So we will use GetDesktopWindow() which was suggested by Miller Puckette.
// hWnd = GetForegroundWindow();
//
// FIXME: The example code I have on the net creates a hidden window that
// is managed by our code - I think we should do that - one hidden
// window for the whole of Pa_DS
//
hWnd = GetDesktopWindow();
// Set cooperative level to DSSCL_EXCLUSIVE so that we can get 16 bit output, 44.1 KHz.
// Exclusize also prevents unexpected sounds from other apps during a performance.
if ((hr = IDirectSound_SetCooperativeLevel( dsw->dsw_pDirectSound,
hWnd, DSSCL_EXCLUSIVE)) != DS_OK)
{
return hr;
}
// -----------------------------------------------------------------------
// Create primary buffer and set format just so we can specify our custom format.
// Otherwise we would be stuck with the default which might be 8 bit or 22050 Hz.
// Setup the primary buffer description
ZeroMemory(&primaryDesc, sizeof(DSBUFFERDESC));
primaryDesc.dwSize = sizeof(DSBUFFERDESC);
primaryDesc.dwFlags = DSBCAPS_PRIMARYBUFFER; // all panning, mixing, etc done by synth
primaryDesc.dwBufferBytes = 0;
primaryDesc.lpwfxFormat = NULL;
// Create the buffer
if ((result = IDirectSound_CreateSoundBuffer( dsw->dsw_pDirectSound,
&primaryDesc, &pPrimaryBuffer, NULL)) != DS_OK) return result;
// Define the buffer format
wfFormat.wFormatTag = WAVE_FORMAT_PCM;
wfFormat.nChannels = nChannels;
wfFormat.nSamplesPerSec = nFrameRate;
wfFormat.wBitsPerSample = 8 * sizeof(short);
wfFormat.nBlockAlign = (WORD)(wfFormat.nChannels * (wfFormat.wBitsPerSample / 8));
wfFormat.nAvgBytesPerSec = wfFormat.nSamplesPerSec * wfFormat.nBlockAlign;
wfFormat.cbSize = 0; /* No extended format info. */
// Set the primary buffer's format
if((result = IDirectSoundBuffer_SetFormat( pPrimaryBuffer, &wfFormat)) != DS_OK) return result;
// ----------------------------------------------------------------------
// Setup the secondary buffer description
ZeroMemory(&secondaryDesc, sizeof(DSBUFFERDESC));
secondaryDesc.dwSize = sizeof(DSBUFFERDESC);
secondaryDesc.dwFlags = DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2;
secondaryDesc.dwBufferBytes = bytesPerBuffer;
secondaryDesc.lpwfxFormat = &wfFormat;
// Create the secondary buffer
if ((result = IDirectSound_CreateSoundBuffer( dsw->dsw_pDirectSound,
&secondaryDesc, &dsw->dsw_OutputBuffer, NULL)) != DS_OK) return result;
// Lock the DS buffer
if ((result = IDirectSoundBuffer_Lock( dsw->dsw_OutputBuffer, 0, dsw->dsw_OutputSize, (LPVOID*)&pDSBuffData,
&dwDataLen, NULL, 0, 0)) != DS_OK) return result;
// Zero the DS buffer
ZeroMemory(pDSBuffData, dwDataLen);
// Unlock the DS buffer
if ((result = IDirectSoundBuffer_Unlock( dsw->dsw_OutputBuffer, pDSBuffData, dwDataLen, NULL, 0)) != DS_OK) return result;
if( QueryPerformanceFrequency( &counterFrequency ) )
{
int framesInBuffer = bytesPerBuffer / (nChannels * sizeof(short));
dsw->dsw_CounterTicksPerBuffer.QuadPart = (counterFrequency.QuadPart * framesInBuffer) / nFrameRate;
}
else
{
dsw->dsw_CounterTicksPerBuffer.QuadPart = 0;
}
// Let DSound set the starting write position because if we set it to zero, it looks like the
// buffer is full to begin with. This causes a long pause before sound starts when using large buffers.
hr = IDirectSoundBuffer_GetCurrentPosition( dsw->dsw_OutputBuffer, &playCursor, &dsw->dsw_WriteOffset );
if( hr != DS_OK )
{
return hr;
}
dsw->dsw_FramesWritten = dsw->dsw_WriteOffset / dsw->dsw_BytesPerOutputFrame;
/* printf("DSW_InitOutputBuffer: playCursor = %d, writeCursor = %d\n", playCursor, dsw->dsw_WriteOffset ); */
return DS_OK;
}
/************************************************************************************/
HRESULT DSW_StartOutput( DSoundWrapper *dsw )
{
HRESULT hr;
QueryPerformanceCounter( &dsw->dsw_LastPlayTime );
dsw->dsw_LastPlayCursor = 0;
dsw->dsw_FramesPlayed = 0;
hr = IDirectSoundBuffer_SetCurrentPosition( dsw->dsw_OutputBuffer, 0 );
if( hr != DS_OK )
{
return hr;
}
// Start the buffer playback in a loop.
if( dsw->dsw_OutputBuffer != NULL )
{
hr = IDirectSoundBuffer_Play( dsw->dsw_OutputBuffer, 0, 0, DSBPLAY_LOOPING );
if( hr != DS_OK )
{
return hr;
}
dsw->dsw_OutputRunning = TRUE;
}
return 0;
}
/************************************************************************************/
HRESULT DSW_StopOutput( DSoundWrapper *dsw )
{
// Stop the buffer playback
if( dsw->dsw_OutputBuffer != NULL )
{
dsw->dsw_OutputRunning = FALSE;
return IDirectSoundBuffer_Stop( dsw->dsw_OutputBuffer );
}
else return 0;
}
/************************************************************************************/
HRESULT DSW_QueryOutputFilled( DSoundWrapper *dsw, long *bytesFilledPtr )
{
HRESULT hr;
DWORD playCursor;
DWORD writeCursor;
long bytesFilled;
// Query to see where play position is.
// We don't need the writeCursor but sometimes DirectSound doesn't handle NULLS correctly
// so let's pass a pointer just to be safe.
hr = IDirectSoundBuffer_GetCurrentPosition( dsw->dsw_OutputBuffer, &playCursor, &writeCursor );
if( hr != DS_OK )
{
return hr;
}
bytesFilled = dsw->dsw_WriteOffset - playCursor;
if( bytesFilled < 0 ) bytesFilled += dsw->dsw_OutputSize; // unwrap offset
*bytesFilledPtr = bytesFilled;
return hr;
}
/************************************************************************************
* Determine how much space can be safely written to in DS buffer.
* Detect underflows and overflows.
* Does not allow writing into safety gap maintained by DirectSound.
*/
HRESULT DSW_QueryOutputSpace( DSoundWrapper *dsw, long *bytesEmpty )
{
HRESULT hr;
DWORD playCursor;
DWORD writeCursor;
long numBytesEmpty;
long playWriteGap;
// Query to see how much room is in buffer.
hr = IDirectSoundBuffer_GetCurrentPosition( dsw->dsw_OutputBuffer, &playCursor, &writeCursor );
if( hr != DS_OK )
{
return hr;
}
// Determine size of gap between playIndex and WriteIndex that we cannot write into.
playWriteGap = writeCursor - playCursor;
if( playWriteGap < 0 ) playWriteGap += dsw->dsw_OutputSize; // unwrap
/* DirectSound doesn't have a large enough playCursor so we cannot detect wrap-around. */
/* Attempt to detect playCursor wrap-around and correct it. */
if( dsw->dsw_OutputRunning && (dsw->dsw_CounterTicksPerBuffer.QuadPart != 0) )
{
/* How much time has elapsed since last check. */
LARGE_INTEGER currentTime;
LARGE_INTEGER elapsedTime;
long bytesPlayed;
long bytesExpected;
long buffersWrapped;
QueryPerformanceCounter( &currentTime );
elapsedTime.QuadPart = currentTime.QuadPart - dsw->dsw_LastPlayTime.QuadPart;
dsw->dsw_LastPlayTime = currentTime;
/* How many bytes does DirectSound say have been played. */
bytesPlayed = playCursor - dsw->dsw_LastPlayCursor;
if( bytesPlayed < 0 ) bytesPlayed += dsw->dsw_OutputSize; // unwrap
dsw->dsw_LastPlayCursor = playCursor;
/* Calculate how many bytes we would have expected to been played by now. */
bytesExpected = (long) ((elapsedTime.QuadPart * dsw->dsw_OutputSize) / dsw->dsw_CounterTicksPerBuffer.QuadPart);
buffersWrapped = (bytesExpected - bytesPlayed) / dsw->dsw_OutputSize;
if( buffersWrapped > 0 )
{
playCursor += (buffersWrapped * dsw->dsw_OutputSize);
bytesPlayed += (buffersWrapped * dsw->dsw_OutputSize);
}
/* Maintain frame output cursor. */
dsw->dsw_FramesPlayed += (bytesPlayed / dsw->dsw_BytesPerOutputFrame);
}
numBytesEmpty = playCursor - dsw->dsw_WriteOffset;
if( numBytesEmpty < 0 ) numBytesEmpty += dsw->dsw_OutputSize; // unwrap offset
/* Have we underflowed? */
if( numBytesEmpty > (dsw->dsw_OutputSize - playWriteGap) )
{
if( dsw->dsw_OutputRunning )
{
dsw->dsw_OutputUnderflows += 1;
}
dsw->dsw_WriteOffset = writeCursor;
numBytesEmpty = dsw->dsw_OutputSize - playWriteGap;
}
*bytesEmpty = numBytesEmpty;
return hr;
}
/************************************************************************************/
HRESULT DSW_ZeroEmptySpace( DSoundWrapper *dsw )
{
HRESULT hr;
LPBYTE lpbuf1 = NULL;
LPBYTE lpbuf2 = NULL;
DWORD dwsize1 = 0;
DWORD dwsize2 = 0;
long bytesEmpty;
hr = DSW_QueryOutputSpace( dsw, &bytesEmpty ); // updates dsw_FramesPlayed
if (hr != DS_OK) return hr;
if( bytesEmpty == 0 ) return DS_OK;
// Lock free space in the DS
hr = IDirectSoundBuffer_Lock( dsw->dsw_OutputBuffer, dsw->dsw_WriteOffset, bytesEmpty, (void **) &lpbuf1, &dwsize1,
(void **) &lpbuf2, &dwsize2, 0);
if (hr == DS_OK)
{
// Copy the buffer into the DS
ZeroMemory(lpbuf1, dwsize1);
if(lpbuf2 != NULL)
{
ZeroMemory(lpbuf2, dwsize2);
}
// Update our buffer offset and unlock sound buffer
dsw->dsw_WriteOffset = (dsw->dsw_WriteOffset + dwsize1 + dwsize2) % dsw->dsw_OutputSize;
IDirectSoundBuffer_Unlock( dsw->dsw_OutputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2);
dsw->dsw_FramesWritten += bytesEmpty / dsw->dsw_BytesPerOutputFrame;
}
return hr;
}
/************************************************************************************/
HRESULT DSW_WriteBlock( DSoundWrapper *dsw, char *buf, long numBytes )
{
HRESULT hr;
LPBYTE lpbuf1 = NULL;
LPBYTE lpbuf2 = NULL;
DWORD dwsize1 = 0;
DWORD dwsize2 = 0;
// Lock free space in the DS
hr = IDirectSoundBuffer_Lock( dsw->dsw_OutputBuffer, dsw->dsw_WriteOffset, numBytes, (void **) &lpbuf1, &dwsize1,
(void **) &lpbuf2, &dwsize2, 0);
if (hr == DS_OK)
{
// Copy the buffer into the DS
CopyMemory(lpbuf1, buf, dwsize1);
if(lpbuf2 != NULL)
{
CopyMemory(lpbuf2, buf+dwsize1, dwsize2);
}
// Update our buffer offset and unlock sound buffer
dsw->dsw_WriteOffset = (dsw->dsw_WriteOffset + dwsize1 + dwsize2) % dsw->dsw_OutputSize;
IDirectSoundBuffer_Unlock( dsw->dsw_OutputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2);
dsw->dsw_FramesWritten += numBytes / dsw->dsw_BytesPerOutputFrame;
}
return hr;
}
/************************************************************************************/
DWORD DSW_GetOutputStatus( DSoundWrapper *dsw )
{
DWORD status;
if (IDirectSoundBuffer_GetStatus( dsw->dsw_OutputBuffer, &status ) != DS_OK)
return( DSERR_INVALIDPARAM );
else
return( status );
}
/* These routines are used to support audio input.
* Do NOT compile these calls when using NT4 because it does
* not support the entry points.
*/
/************************************************************************************/
HRESULT DSW_InitInputDevice( DSoundWrapper *dsw, LPGUID lpGUID )
{
HRESULT hr = dswDSoundEntryPoints.DirectSoundCaptureCreate( lpGUID, &dsw->dsw_pDirectSoundCapture, NULL );
if( hr != DS_OK ) return hr;
return hr;
}
/************************************************************************************/
HRESULT DSW_InitInputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, WORD nChannels, int bytesPerBuffer )
{
DSCBUFFERDESC captureDesc;
WAVEFORMATEX wfFormat;
HRESULT result;
dsw->dsw_BytesPerInputFrame = nChannels * sizeof(short);
// Define the buffer format
wfFormat.wFormatTag = WAVE_FORMAT_PCM;
wfFormat.nChannels = nChannels;
wfFormat.nSamplesPerSec = nFrameRate;
wfFormat.wBitsPerSample = 8 * sizeof(short);
wfFormat.nBlockAlign = (WORD)(wfFormat.nChannels * (wfFormat.wBitsPerSample / 8));
wfFormat.nAvgBytesPerSec = wfFormat.nSamplesPerSec * wfFormat.nBlockAlign;
wfFormat.cbSize = 0; /* No extended format info. */
dsw->dsw_InputSize = bytesPerBuffer;
// ----------------------------------------------------------------------
// Setup the secondary buffer description
ZeroMemory(&captureDesc, sizeof(DSCBUFFERDESC));
captureDesc.dwSize = sizeof(DSCBUFFERDESC);
captureDesc.dwFlags = 0;
captureDesc.dwBufferBytes = bytesPerBuffer;
captureDesc.lpwfxFormat = &wfFormat;
// Create the capture buffer
if ((result = IDirectSoundCapture_CreateCaptureBuffer( dsw->dsw_pDirectSoundCapture,
&captureDesc, &dsw->dsw_InputBuffer, NULL)) != DS_OK) return result;
dsw->dsw_ReadOffset = 0; // reset last read position to start of buffer
return DS_OK;
}
/************************************************************************************/
HRESULT DSW_StartInput( DSoundWrapper *dsw )
{
// Start the buffer playback
if( dsw->dsw_InputBuffer != NULL )
{
return IDirectSoundCaptureBuffer_Start( dsw->dsw_InputBuffer, DSCBSTART_LOOPING );
}
else return 0;
}
/************************************************************************************/
HRESULT DSW_StopInput( DSoundWrapper *dsw )
{
// Stop the buffer playback
if( dsw->dsw_InputBuffer != NULL )
{
return IDirectSoundCaptureBuffer_Stop( dsw->dsw_InputBuffer );
}
else return 0;
}
/************************************************************************************/
HRESULT DSW_QueryInputFilled( DSoundWrapper *dsw, long *bytesFilled )
{
HRESULT hr;
DWORD capturePos;
DWORD readPos;
long filled;
// Query to see how much data is in buffer.
// We don't need the capture position but sometimes DirectSound doesn't handle NULLS correctly
// so let's pass a pointer just to be safe.
hr = IDirectSoundCaptureBuffer_GetCurrentPosition( dsw->dsw_InputBuffer, &capturePos, &readPos );
if( hr != DS_OK )
{
return hr;
}
filled = readPos - dsw->dsw_ReadOffset;
if( filled < 0 ) filled += dsw->dsw_InputSize; // unwrap offset
*bytesFilled = filled;
return hr;
}
/************************************************************************************/
HRESULT DSW_ReadBlock( DSoundWrapper *dsw, char *buf, long numBytes )
{
HRESULT hr;
LPBYTE lpbuf1 = NULL;
LPBYTE lpbuf2 = NULL;
DWORD dwsize1 = 0;
DWORD dwsize2 = 0;
// Lock free space in the DS
hr = IDirectSoundCaptureBuffer_Lock ( dsw->dsw_InputBuffer, dsw->dsw_ReadOffset, numBytes, (void **) &lpbuf1, &dwsize1,
(void **) &lpbuf2, &dwsize2, 0);
if (hr == DS_OK)
{
// Copy from DS to the buffer
CopyMemory( buf, lpbuf1, dwsize1);
if(lpbuf2 != NULL)
{
CopyMemory( buf+dwsize1, lpbuf2, dwsize2);
}
// Update our buffer offset and unlock sound buffer
dsw->dsw_ReadOffset = (dsw->dsw_ReadOffset + dwsize1 + dwsize2) % dsw->dsw_InputSize;
IDirectSoundCaptureBuffer_Unlock ( dsw->dsw_InputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2);
}
return hr;
}

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portaudio-v19/pa_win_ds/dsound_wrapper.h
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#ifndef __DSOUND_WRAPPER_H
#define __DSOUND_WRAPPER_H
/*
* $Id$
* Simplified DirectSound interface.
*
* Author: Phil Burk & Robert Marsanyi
*
* For PortAudio Portable Real-Time Audio Library
* For more information see: http://www.softsynth.com/portaudio/
* DirectSound Implementation
* Copyright (c) 1999-2000 Phil Burk & Robert Marsanyi
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
/* on Borland compilers, WIN32 doesn't seem to be defined by default, which
breaks DSound.h. Adding the define here fixes the problem. - rossb. */
#ifdef __BORLANDC__
#if !defined(WIN32)
#define WIN32
#endif
#endif
/*
We are only using DX3 in here, no need to polute the namespace - davidv
*/
#define DIRECTSOUND_VERSION 0x0300
#include <DSound.h>
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
typedef struct
{
HINSTANCE hInstance_;
HRESULT (WINAPI *DirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
HRESULT (WINAPI *DirectSoundEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
HRESULT (WINAPI *DirectSoundEnumerateA)(LPDSENUMCALLBACKA, LPVOID);
HRESULT (WINAPI *DirectSoundCaptureCreate)(LPGUID, LPDIRECTSOUNDCAPTURE *, LPUNKNOWN);
HRESULT (WINAPI *DirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
HRESULT (WINAPI *DirectSoundCaptureEnumerateA)(LPDSENUMCALLBACKA, LPVOID);
}DSoundEntryPoints;
extern DSoundEntryPoints dswDSoundEntryPoints;
void DSW_InitializeDSoundEntryPoints(void);
void DSW_TerminateDSoundEntryPoints(void);
#define DSW_NUM_POSITIONS (4)
#define DSW_NUM_EVENTS (5)
#define DSW_TERMINATION_EVENT (DSW_NUM_POSITIONS)
typedef struct
{
/* Output */
LPDIRECTSOUND dsw_pDirectSound;
LPDIRECTSOUNDBUFFER dsw_OutputBuffer;
DWORD dsw_WriteOffset; /* last write position */
INT dsw_OutputSize;
INT dsw_BytesPerOutputFrame;
/* Try to detect play buffer underflows. */
LARGE_INTEGER dsw_CounterTicksPerBuffer; /* counter ticks it should take to play a full buffer */
LARGE_INTEGER dsw_LastPlayTime;
UINT dsw_LastPlayCursor;
UINT dsw_OutputUnderflows;
BOOL dsw_OutputRunning;
/* use double which lets us can play for several thousand years with enough precision */
double dsw_FramesWritten;
double dsw_FramesPlayed;
/* Input */
INT dsw_BytesPerInputFrame;
LPDIRECTSOUNDCAPTURE dsw_pDirectSoundCapture;
LPDIRECTSOUNDCAPTUREBUFFER dsw_InputBuffer;
UINT dsw_ReadOffset; /* last read position */
UINT dsw_InputSize;
} DSoundWrapper;
HRESULT DSW_Init( DSoundWrapper *dsw );
void DSW_Term( DSoundWrapper *dsw );
HRESULT DSW_InitOutputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate,
WORD nChannels, int bufSize );
HRESULT DSW_StartOutput( DSoundWrapper *dsw );
HRESULT DSW_StopOutput( DSoundWrapper *dsw );
DWORD DSW_GetOutputStatus( DSoundWrapper *dsw );
HRESULT DSW_WriteBlock( DSoundWrapper *dsw, char *buf, long numBytes );
HRESULT DSW_ZeroEmptySpace( DSoundWrapper *dsw );
HRESULT DSW_QueryOutputSpace( DSoundWrapper *dsw, long *bytesEmpty );
HRESULT DSW_Enumerate( DSoundWrapper *dsw );
HRESULT DSW_InitInputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate,
WORD nChannels, int bufSize );
HRESULT DSW_StartInput( DSoundWrapper *dsw );
HRESULT DSW_StopInput( DSoundWrapper *dsw );
HRESULT DSW_ReadBlock( DSoundWrapper *dsw, char *buf, long numBytes );
HRESULT DSW_QueryInputFilled( DSoundWrapper *dsw, long *bytesFilled );
HRESULT DSW_QueryOutputFilled( DSoundWrapper *dsw, long *bytesFilled );
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* __DSOUND_WRAPPER_H */

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portaudio-v19/pa_win_ds/pa_win_ds.c
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portaudio-v19/pa_win_wdmks/pa_win_wdmks.c
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portaudio-v19/pa_win_wdmks/readme.txt
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@ -1,82 +0,0 @@
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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#ifndef PA_WIN_WMME_H
#define PA_WIN_WMME_H
/*
* $Id$
* PortAudio Portable Real-Time Audio Library
* MME specific extensions
*
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
/** @file
@brief WMME-specific PortAudio API extension header file.
*/
#include "portaudio.h"
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
#define paWinMmeUseLowLevelLatencyParameters (0x01)
#define paWinMmeUseMultipleDevices (0x02) /* use mme specific multiple device feature */
/* By default, the mme implementation drops the processing thread's priority
to THREAD_PRIORITY_NORMAL and sleeps the thread if the CPU load exceeds 100%
This flag disables any priority throttling. The processing thread will always
run at THREAD_PRIORITY_TIME_CRITICAL.
*/
#define paWinMmeDontThrottleOverloadedProcessingThread (0x08)
typedef struct PaWinMmeDeviceAndChannelCount{
PaDeviceIndex device;
int channelCount;
}PaWinMmeDeviceAndChannelCount;
typedef struct PaWinMmeStreamInfo{
unsigned long size; /**< sizeof(PaWinMmeStreamInfo) */
PaHostApiTypeId hostApiType; /**< paMME */
unsigned long version; /**< 1 */
unsigned long flags;
/* low-level latency setting support
These settings control the number and size of host buffers in order
to set latency. They will be used instead of the generic parameters
to Pa_OpenStream() if flags contains the PaWinMmeUseLowLevelLatencyParameters
flag.
If PaWinMmeStreamInfo structures with PaWinMmeUseLowLevelLatencyParameters
are supplied for both input and output in a full duplex stream, then the
input and output framesPerBuffer must be the same, or the larger of the
two must be a multiple of the smaller, otherwise a
paIncompatibleHostApiSpecificStreamInfo error will be returned from
Pa_OpenStream().
*/
unsigned long framesPerBuffer;
unsigned long bufferCount; /* formerly numBuffers */
/* multiple devices per direction support
If flags contains the PaWinMmeUseMultipleDevices flag,
this functionality will be used, otherwise the device parameter to
Pa_OpenStream() will be used instead.
If devices are specified here, the corresponding device parameter
to Pa_OpenStream() should be set to paUseHostApiSpecificDeviceSpecification,
otherwise an paInvalidDevice error will result.
The total number of channels accross all specified devices
must agree with the corresponding channelCount parameter to
Pa_OpenStream() otherwise a paInvalidChannelCount error will result.
*/
PaWinMmeDeviceAndChannelCount *devices;
unsigned long deviceCount;
}PaWinMmeStreamInfo;
/** Retrieve the number of wave in handles used by a PortAudio WinMME stream.
Returns zero if the stream is output only.
@return A non-negative value indicating the number of wave in handles
or, a PaErrorCode (which are always negative) if PortAudio is not initialized
or an error is encountered.
@see PaWinMME_GetStreamInputHandle
*/
int PaWinMME_GetStreamInputHandleCount( PaStream* stream );
/** Retrieve a wave in handle used by a PortAudio WinMME stream.
@param stream The stream to query.
@param handleIndex The zero based index of the wave in handle to retrieve. This
should be in the range [0, PaWinMME_GetStreamInputHandle(stream)-1].
@return A valid wave in handle, or NULL if an error occurred.
@see PaWinMME_GetStreamInputHandle
*/
HWAVEIN PaWinMME_GetStreamInputHandle( PaStream* stream, int handleIndex );
/** Retrieve the number of wave out handles used by a PortAudio WinMME stream.
Returns zero if the stream is input only.
@return A non-negative value indicating the number of wave out handles
or, a PaErrorCode (which are always negative) if PortAudio is not initialized
or an error is encountered.
@see PaWinMME_GetStreamOutputHandle
*/
int PaWinMME_GetStreamOutputHandleCount( PaStream* stream );
/** Retrieve a wave out handle used by a PortAudio WinMME stream.
@param stream The stream to query.
@param handleIndex The zero based index of the wave out handle to retrieve.
This should be in the range [0, PaWinMME_GetStreamOutputHandleCount(stream)-1].
@return A valid wave out handle, or NULL if an error occurred.
@see PaWinMME_GetStreamOutputHandleCount
*/
HWAVEOUT PaWinMME_GetStreamOutputHandle( PaStream* stream, int handleIndex );
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* PA_WIN_WMME_H */