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Move mezzo where it going to live

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ra 2010-01-23 19:37:49 +00:00
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30
LICENSE.txt Normal file
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Mezzo: A Cross-Platform Audio Editing Engine
Copyright (c) 2004 Joshua Haberman, Dominic Mazzoni
Permission is hereby granted, free of charge, to any person obtaining
a copy of this software and associated documentation files (the
"Software"), to deal in the Software without restriction, including
without limitation the rights to use, copy, modify, merge, publish,
distribute, sublicense, and/or sell copies of the Software, and to
permit persons to whom the Software is furnished to do so, subject to
the following conditions:
The above copyright notice and this permission notice shall be
included in all copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
In addition, the authors of Mezzo request (but do not require) that if
you modify Mezzo that you share your modifications with the Mezzo
development team so that everyone can benefit. Furthermore, we kindly
suggest that if you use Mezzo in a commercial product, that you make a
donation to the Mezzo development team.

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import string
dbg = Environment(CCFLAGS = '-Wall -g', CPPPATH= ["#/lib", "#/src"])
release = Environment( CPPFLAGS = '-Wall')
env = dbg
# env = release
SConscript('src/SConscript', build_dir='build', duplicate=0, exports='env')
SConscript('lib/SConscript', exports='env')
env.Program('tests/MezzoTest',
source = 'tests/MezzoTest.cpp',
LIBS=['audacity', 'expat', 'posh'], LIBPATH=['build', 'lib'])
env.Program('tests/TestManagedFileContext',
source = 'tests/TestManagedFileContext.cpp',
LIBS=['audacity', 'expat', 'posh'],
LIBPATH=['build', 'lib'])
#env.Program('tests/MezzoRecord',
# source = 'tests/MezzoRecord.cpp',
# LIBS=['audacity', 'expat', 'posh', 'portaudio'],
# LIBPATH=['build', 'lib'],
# LINKFLAGS=['-framework','CoreAudio',
# '-framework','AudioUnit',
# '-framework','AudioToolbox'])
#env.Program('tests/MezzoPlay',
# source = 'tests/MezzoPlay.cpp',
# LIBS=['audacity', 'expat', 'posh', 'portaudio'],
# LIBPATH=['build', 'lib'],
# LINKFLAGS=['-framework','CoreAudio',
# '-framework','AudioUnit',
# '-framework','AudioToolbox'])
#env.Program('tests/MezzoCacheDisplay',
# source = 'tests/MezzoCacheDisplay.cpp',
# LIBS=['audacity', 'expat', 'posh', 'portaudio'],
# LIBPATH=['build', 'lib'],
# LINKFLAGS=['-framework','CoreAudio',
# '-framework','AudioUnit',
# '-framework','AudioToolbox'])
#for test in tests:
# testname = "tests/" + test
# env.Program(testname, source = testname + ".cpp",
# LIBS=['audacity', 'expat'], LIBPATH=['build', 'lib'])

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}q(U libposh.aq(cSCons.Sig
SConsignEntry
qoq}qUbsigqNsbU
libexpat.aq(hoq}q Ubsigq
Nsbu.

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Import('env')
expat_files = [
"expat/xmlparse.c",
"expat/xmltok.c",
"expat/xmlrole.c"
]
posh_files = [
"posh/posh.c"
]
env.StaticLibrary("expat", source = expat_files)
env.StaticLibrary("posh", source = posh_files)

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}q(Uascii.hq(cSCons.Sig
SConsignEntry
qoq}q(U timestampqJÎôÜAUcsigqU 7efed5c4a9dd1c07ba83398064948910qubU
xmlparse.cq (hoq
}q (hJÎôÜAhU 861876a740538dd5618fe47ab0e3249eq ubU
asciitab.hq (hoq}q(hJÎôÜAhU f0619a5f857583f35ef0f7d73d8cac50qubU
xmlparse.hq(hoq}q(hJÎôÜAhU 7bf212dbd89fda0f22a5e4df2ece85cfqubU latin1tab.hq(hoq}q(hJÎôÜAhU 9b4b21bf6e5a7a0a72ca244b3aa3e5eaqubUxmldef.hq(hoq}q(hJÎôÜAhU 9dfdb848e2c85065a37f7c17025f9cffqubU xmlrole.oq(hoq}qUbsigq NsbU nametab.hq!(hoq"}q#(hJÎôÜAhU 0675e6b65472addbf2beafb721063922q$ubU iasciitab.hq%(hoq&}q'(hJÎôÜAhU 07dd75cd360aa4841eb22eeb0e1b1ad5q(ubU
xmlparse.oq)(hoq*}q+h NsbU xmlrole.hq,(hoq-}q.(hJÎôÜAhU 8f8a23073a41ff885fdbb00cfa023527q/ubU utf8tab.hq0(hoq1}q2(hJÎôÜAhU 741d75eb3899f294e9b8baf106fc5fd8q3ubUxmltok.oq4(hoq5}q6h NsbU xmltok_impl.cq7(hoq8}q9(hJÎôÜAhU 4de96a301b55ccebd3a1a69d79c6455cq:ubUxmltok.hq;(hoq<}q=(hJÎôÜAhU 2f64d3aa98e042c4858cc3c7b6be6b06q>ubU xmltok_ns.cq?(hoq@}qA(hJÎôÜAhU a67b4b62a74b99c1c707a3dd24791bf8qBubU xmlrole.cqC(hoqD}qE(hJÎôÜAhU d2dbe6353b46d279ff96392a6e25e276qFubUxmltok.cqG(hoqH}qI(hJÎôÜAhU 3a0f7bfc9e214c1cbfa6a45f63b8ab39qJubU xmltok_impl.hqK(hoqL}qM(hJÎôÜAhU 8474d810ff77d7f7246f8b2a922176ceqNubu.

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/*
Copyright (c) 1998, 1999 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
#define ASCII_A 0x41
#define ASCII_B 0x42
#define ASCII_C 0x43
#define ASCII_D 0x44
#define ASCII_E 0x45
#define ASCII_F 0x46
#define ASCII_G 0x47
#define ASCII_H 0x48
#define ASCII_I 0x49
#define ASCII_J 0x4A
#define ASCII_K 0x4B
#define ASCII_L 0x4C
#define ASCII_M 0x4D
#define ASCII_N 0x4E
#define ASCII_O 0x4F
#define ASCII_P 0x50
#define ASCII_Q 0x51
#define ASCII_R 0x52
#define ASCII_S 0x53
#define ASCII_T 0x54
#define ASCII_U 0x55
#define ASCII_V 0x56
#define ASCII_W 0x57
#define ASCII_X 0x58
#define ASCII_Y 0x59
#define ASCII_Z 0x5A
#define ASCII_a 0x61
#define ASCII_b 0x62
#define ASCII_c 0x63
#define ASCII_d 0x64
#define ASCII_e 0x65
#define ASCII_f 0x66
#define ASCII_g 0x67
#define ASCII_h 0x68
#define ASCII_i 0x69
#define ASCII_j 0x6A
#define ASCII_k 0x6B
#define ASCII_l 0x6C
#define ASCII_m 0x6D
#define ASCII_n 0x6E
#define ASCII_o 0x6F
#define ASCII_p 0x70
#define ASCII_q 0x71
#define ASCII_r 0x72
#define ASCII_s 0x73
#define ASCII_t 0x74
#define ASCII_u 0x75
#define ASCII_v 0x76
#define ASCII_w 0x77
#define ASCII_x 0x78
#define ASCII_y 0x79
#define ASCII_z 0x7A
#define ASCII_0 0x30
#define ASCII_1 0x31
#define ASCII_2 0x32
#define ASCII_3 0x33
#define ASCII_4 0x34
#define ASCII_5 0x35
#define ASCII_6 0x36
#define ASCII_7 0x37
#define ASCII_8 0x38
#define ASCII_9 0x39
#define ASCII_TAB 0x09
#define ASCII_SPACE 0x20
#define ASCII_EXCL 0x21
#define ASCII_QUOT 0x22
#define ASCII_AMP 0x26
#define ASCII_APOS 0x27
#define ASCII_MINUS 0x2D
#define ASCII_PERIOD 0x2E
#define ASCII_COLON 0x3A
#define ASCII_SEMI 0x3B
#define ASCII_LT 0x3C
#define ASCII_EQUALS 0x3D
#define ASCII_GT 0x3E
#define ASCII_LSQB 0x5B
#define ASCII_RSQB 0x5D
#define ASCII_UNDERSCORE 0x5F

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/*
Copyright (c) 1998, 1999 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
/* 0x00 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x04 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x08 */ BT_NONXML, BT_S, BT_LF, BT_NONXML,
/* 0x0C */ BT_NONXML, BT_CR, BT_NONXML, BT_NONXML,
/* 0x10 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x14 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x18 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x1C */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x20 */ BT_S, BT_EXCL, BT_QUOT, BT_NUM,
/* 0x24 */ BT_OTHER, BT_PERCNT, BT_AMP, BT_APOS,
/* 0x28 */ BT_LPAR, BT_RPAR, BT_AST, BT_PLUS,
/* 0x2C */ BT_COMMA, BT_MINUS, BT_NAME, BT_SOL,
/* 0x30 */ BT_DIGIT, BT_DIGIT, BT_DIGIT, BT_DIGIT,
/* 0x34 */ BT_DIGIT, BT_DIGIT, BT_DIGIT, BT_DIGIT,
/* 0x38 */ BT_DIGIT, BT_DIGIT, BT_COLON, BT_SEMI,
/* 0x3C */ BT_LT, BT_EQUALS, BT_GT, BT_QUEST,
/* 0x40 */ BT_OTHER, BT_HEX, BT_HEX, BT_HEX,
/* 0x44 */ BT_HEX, BT_HEX, BT_HEX, BT_NMSTRT,
/* 0x48 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x4C */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x50 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x54 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x58 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_LSQB,
/* 0x5C */ BT_OTHER, BT_RSQB, BT_OTHER, BT_NMSTRT,
/* 0x60 */ BT_OTHER, BT_HEX, BT_HEX, BT_HEX,
/* 0x64 */ BT_HEX, BT_HEX, BT_HEX, BT_NMSTRT,
/* 0x68 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x6C */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x70 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x74 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x78 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_OTHER,
/* 0x7C */ BT_VERBAR, BT_OTHER, BT_OTHER, BT_OTHER,

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/*
Copyright (c) 1998, 1999 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
/* Like asciitab.h, except that 0xD has code BT_S rather than BT_CR */
/* 0x00 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x04 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x08 */ BT_NONXML, BT_S, BT_LF, BT_NONXML,
/* 0x0C */ BT_NONXML, BT_S, BT_NONXML, BT_NONXML,
/* 0x10 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x14 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x18 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x1C */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0x20 */ BT_S, BT_EXCL, BT_QUOT, BT_NUM,
/* 0x24 */ BT_OTHER, BT_PERCNT, BT_AMP, BT_APOS,
/* 0x28 */ BT_LPAR, BT_RPAR, BT_AST, BT_PLUS,
/* 0x2C */ BT_COMMA, BT_MINUS, BT_NAME, BT_SOL,
/* 0x30 */ BT_DIGIT, BT_DIGIT, BT_DIGIT, BT_DIGIT,
/* 0x34 */ BT_DIGIT, BT_DIGIT, BT_DIGIT, BT_DIGIT,
/* 0x38 */ BT_DIGIT, BT_DIGIT, BT_COLON, BT_SEMI,
/* 0x3C */ BT_LT, BT_EQUALS, BT_GT, BT_QUEST,
/* 0x40 */ BT_OTHER, BT_HEX, BT_HEX, BT_HEX,
/* 0x44 */ BT_HEX, BT_HEX, BT_HEX, BT_NMSTRT,
/* 0x48 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x4C */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x50 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x54 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x58 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_LSQB,
/* 0x5C */ BT_OTHER, BT_RSQB, BT_OTHER, BT_NMSTRT,
/* 0x60 */ BT_OTHER, BT_HEX, BT_HEX, BT_HEX,
/* 0x64 */ BT_HEX, BT_HEX, BT_HEX, BT_NMSTRT,
/* 0x68 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x6C */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x70 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x74 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0x78 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_OTHER,
/* 0x7C */ BT_VERBAR, BT_OTHER, BT_OTHER, BT_OTHER,

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/*
Copyright (c) 1998, 1999 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
/* 0x80 */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0x84 */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0x88 */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0x8C */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0x90 */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0x94 */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0x98 */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0x9C */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0xA0 */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0xA4 */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0xA8 */ BT_OTHER, BT_OTHER, BT_NMSTRT, BT_OTHER,
/* 0xAC */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0xB0 */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0xB4 */ BT_OTHER, BT_NMSTRT, BT_OTHER, BT_NAME,
/* 0xB8 */ BT_OTHER, BT_OTHER, BT_NMSTRT, BT_OTHER,
/* 0xBC */ BT_OTHER, BT_OTHER, BT_OTHER, BT_OTHER,
/* 0xC0 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xC4 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xC8 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xCC */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xD0 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xD4 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_OTHER,
/* 0xD8 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xDC */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xE0 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xE4 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xE8 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xEC */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xF0 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xF4 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_OTHER,
/* 0xF8 */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,
/* 0xFC */ BT_NMSTRT, BT_NMSTRT, BT_NMSTRT, BT_NMSTRT,

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static const unsigned namingBitmap[] = {
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF,
0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF,
0x00000000, 0x04000000, 0x87FFFFFE, 0x07FFFFFE,
0x00000000, 0x00000000, 0xFF7FFFFF, 0xFF7FFFFF,
0xFFFFFFFF, 0x7FF3FFFF, 0xFFFFFDFE, 0x7FFFFFFF,
0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFE00F, 0xFC31FFFF,
0x00FFFFFF, 0x00000000, 0xFFFF0000, 0xFFFFFFFF,
0xFFFFFFFF, 0xF80001FF, 0x00000003, 0x00000000,
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0xFFFFD740, 0xFFFFFFFB, 0x547F7FFF, 0x000FFFFD,
0xFFFFDFFE, 0xFFFFFFFF, 0xDFFEFFFF, 0xFFFFFFFF,
0xFFFF0003, 0xFFFFFFFF, 0xFFFF199F, 0x033FCFFF,
0x00000000, 0xFFFE0000, 0x027FFFFF, 0xFFFFFFFE,
0x0000007F, 0x00000000, 0xFFFF0000, 0x000707FF,
0x00000000, 0x07FFFFFE, 0x000007FE, 0xFFFE0000,
0xFFFFFFFF, 0x7CFFFFFF, 0x002F7FFF, 0x00000060,
0xFFFFFFE0, 0x23FFFFFF, 0xFF000000, 0x00000003,
0xFFF99FE0, 0x03C5FDFF, 0xB0000000, 0x00030003,
0xFFF987E0, 0x036DFDFF, 0x5E000000, 0x001C0000,
0xFFFBAFE0, 0x23EDFDFF, 0x00000000, 0x00000001,
0xFFF99FE0, 0x23CDFDFF, 0xB0000000, 0x00000003,
0xD63DC7E0, 0x03BFC718, 0x00000000, 0x00000000,
0xFFFDDFE0, 0x03EFFDFF, 0x00000000, 0x00000003,
0xFFFDDFE0, 0x03EFFDFF, 0x40000000, 0x00000003,
0xFFFDDFE0, 0x03FFFDFF, 0x00000000, 0x00000003,
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0xFFFFFFFE, 0x000D7FFF, 0x0000003F, 0x00000000,
0xFEF02596, 0x200D6CAE, 0x0000001F, 0x00000000,
0x00000000, 0x00000000, 0xFFFFFEFF, 0x000003FF,
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0x00000000, 0xFFFFFFFF, 0xFFFF003F, 0x007FFFFF,
0x0007DAED, 0x50000000, 0x82315001, 0x002C62AB,
0x40000000, 0xF580C900, 0x00000007, 0x02010800,
0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF,
0x0FFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF, 0x03FFFFFF,
0x3F3FFFFF, 0xFFFFFFFF, 0xAAFF3F3F, 0x3FFFFFFF,
0xFFFFFFFF, 0x5FDFFFFF, 0x0FCF1FDC, 0x1FDC1FFF,
0x00000000, 0x00004C40, 0x00000000, 0x00000000,
0x00000007, 0x00000000, 0x00000000, 0x00000000,
0x00000080, 0x000003FE, 0xFFFFFFFE, 0xFFFFFFFF,
0x001FFFFF, 0xFFFFFFFE, 0xFFFFFFFF, 0x07FFFFFF,
0xFFFFFFE0, 0x00001FFF, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF,
0xFFFFFFFF, 0x0000003F, 0x00000000, 0x00000000,
0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF, 0xFFFFFFFF,
0xFFFFFFFF, 0x0000000F, 0x00000000, 0x00000000,
0x00000000, 0x07FF6000, 0x87FFFFFE, 0x07FFFFFE,
0x00000000, 0x00800000, 0xFF7FFFFF, 0xFF7FFFFF,
0x00FFFFFF, 0x00000000, 0xFFFF0000, 0xFFFFFFFF,
0xFFFFFFFF, 0xF80001FF, 0x00030003, 0x00000000,
0xFFFFFFFF, 0xFFFFFFFF, 0x0000003F, 0x00000003,
0xFFFFD7C0, 0xFFFFFFFB, 0x547F7FFF, 0x000FFFFD,
0xFFFFDFFE, 0xFFFFFFFF, 0xDFFEFFFF, 0xFFFFFFFF,
0xFFFF007B, 0xFFFFFFFF, 0xFFFF199F, 0x033FCFFF,
0x00000000, 0xFFFE0000, 0x027FFFFF, 0xFFFFFFFE,
0xFFFE007F, 0xBBFFFFFB, 0xFFFF0016, 0x000707FF,
0x00000000, 0x07FFFFFE, 0x0007FFFF, 0xFFFF03FF,
0xFFFFFFFF, 0x7CFFFFFF, 0xFFEF7FFF, 0x03FF3DFF,
0xFFFFFFEE, 0xF3FFFFFF, 0xFF1E3FFF, 0x0000FFCF,
0xFFF99FEE, 0xD3C5FDFF, 0xB080399F, 0x0003FFCF,
0xFFF987E4, 0xD36DFDFF, 0x5E003987, 0x001FFFC0,
0xFFFBAFEE, 0xF3EDFDFF, 0x00003BBF, 0x0000FFC1,
0xFFF99FEE, 0xF3CDFDFF, 0xB0C0398F, 0x0000FFC3,
0xD63DC7EC, 0xC3BFC718, 0x00803DC7, 0x0000FF80,
0xFFFDDFEE, 0xC3EFFDFF, 0x00603DDF, 0x0000FFC3,
0xFFFDDFEC, 0xC3EFFDFF, 0x40603DDF, 0x0000FFC3,
0xFFFDDFEC, 0xC3FFFDFF, 0x00803DCF, 0x0000FFC3,
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0xFFFFFFFE, 0x07FF7FFF, 0x03FF7FFF, 0x00000000,
0xFEF02596, 0x3BFF6CAE, 0x03FF3F5F, 0x00000000,
0x03000000, 0xC2A003FF, 0xFFFFFEFF, 0xFFFE03FF,
0xFEBF0FDF, 0x02FE3FFF, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x1FFF0000, 0x00000002,
0x000000A0, 0x003EFFFE, 0xFFFFFFFE, 0xFFFFFFFF,
0x661FFFFF, 0xFFFFFFFE, 0xFFFFFFFF, 0x77FFFFFF,
};
static const unsigned char nmstrtPages[] = {
0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, 0x00,
0x00, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, 0x0F,
0x10, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x12, 0x13,
0x00, 0x14, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x15, 0x16, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x17,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x18,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
};
static const unsigned char namePages[] = {
0x19, 0x03, 0x1A, 0x1B, 0x1C, 0x1D, 0x1E, 0x00,
0x00, 0x1F, 0x20, 0x21, 0x22, 0x23, 0x24, 0x25,
0x10, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x12, 0x13,
0x26, 0x14, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x27, 0x16, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x17,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01,
0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x18,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
};

38
lib/expat/utf8tab.h Normal file
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@ -0,0 +1,38 @@
/*
Copyright (c) 1998, 1999 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
/* 0x80 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0x84 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0x88 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0x8C */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0x90 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0x94 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0x98 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0x9C */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0xA0 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0xA4 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0xA8 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0xAC */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0xB0 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0xB4 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0xB8 */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0xBC */ BT_TRAIL, BT_TRAIL, BT_TRAIL, BT_TRAIL,
/* 0xC0 */ BT_LEAD2, BT_LEAD2, BT_LEAD2, BT_LEAD2,
/* 0xC4 */ BT_LEAD2, BT_LEAD2, BT_LEAD2, BT_LEAD2,
/* 0xC8 */ BT_LEAD2, BT_LEAD2, BT_LEAD2, BT_LEAD2,
/* 0xCC */ BT_LEAD2, BT_LEAD2, BT_LEAD2, BT_LEAD2,
/* 0xD0 */ BT_LEAD2, BT_LEAD2, BT_LEAD2, BT_LEAD2,
/* 0xD4 */ BT_LEAD2, BT_LEAD2, BT_LEAD2, BT_LEAD2,
/* 0xD8 */ BT_LEAD2, BT_LEAD2, BT_LEAD2, BT_LEAD2,
/* 0xDC */ BT_LEAD2, BT_LEAD2, BT_LEAD2, BT_LEAD2,
/* 0xE0 */ BT_LEAD3, BT_LEAD3, BT_LEAD3, BT_LEAD3,
/* 0xE4 */ BT_LEAD3, BT_LEAD3, BT_LEAD3, BT_LEAD3,
/* 0xE8 */ BT_LEAD3, BT_LEAD3, BT_LEAD3, BT_LEAD3,
/* 0xEC */ BT_LEAD3, BT_LEAD3, BT_LEAD3, BT_LEAD3,
/* 0xF0 */ BT_LEAD4, BT_LEAD4, BT_LEAD4, BT_LEAD4,
/* 0xF4 */ BT_LEAD4, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0xF8 */ BT_NONXML, BT_NONXML, BT_NONXML, BT_NONXML,
/* 0xFC */ BT_NONXML, BT_NONXML, BT_MALFORM, BT_MALFORM,

52
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/*
Copyright (c) 1998, 1999 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
#include <string.h>
#ifdef XML_WINLIB
#define WIN32_LEAN_AND_MEAN
#define STRICT
#include <windows.h>
#define malloc(x) HeapAlloc(GetProcessHeap(), 0, (x))
#define calloc(x, y) HeapAlloc(GetProcessHeap(), HEAP_ZERO_MEMORY, (x)*(y))
#define free(x) HeapFree(GetProcessHeap(), 0, (x))
#define realloc(x, y) HeapReAlloc(GetProcessHeap(), 0, x, y)
#define abort() /* as nothing */
#else /* not XML_WINLIB */
#include <stdlib.h>
#endif /* not XML_WINLIB */
/* This file can be used for any definitions needed in
particular environments. */
/* Mozilla specific defines */
#ifdef MOZILLA_CLIENT
#include "nspr.h"
#define malloc(x) PR_Malloc((size_t)(x))
#define realloc(x, y) PR_Realloc((x), (size_t)(y))
#define calloc(x, y) PR_Calloc((x),(y))
#define free(x) PR_Free(x)
#if PR_BYTES_PER_INT != 4
#define int int32
#endif
/* Enable Unicode string processing in expat. */
#ifndef XML_UNICODE
#define XML_UNICODE
#endif
/* Enable external parameter entity parsing in expat */
#ifndef XML_DTD
#define XML_DTD 1
#endif
#endif /* MOZILLA_CLIENT */

3925
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527
lib/expat/xmlparse.h Normal file
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/*
Copyright (c) 1998, 1999, 2000 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
#ifndef XmlParse_INCLUDED
#define XmlParse_INCLUDED 1
#ifdef __cplusplus
extern "C" {
#endif
#ifndef XMLPARSEAPI
#define XMLPARSEAPI /* as nothing */
#endif
typedef void *XML_Parser;
#ifdef XML_UNICODE_WCHAR_T
/* XML_UNICODE_WCHAR_T will work only if sizeof(wchar_t) == 2 and wchar_t
uses Unicode. */
/* Information is UTF-16 encoded as wchar_ts */
#ifndef XML_UNICODE
#define XML_UNICODE
#endif
#include <stddef.h>
typedef wchar_t XML_Char;
typedef wchar_t XML_LChar;
#else /* not XML_UNICODE_WCHAR_T */
#ifdef XML_UNICODE
/* Information is UTF-16 encoded as unsigned shorts */
typedef unsigned short XML_Char;
typedef char XML_LChar;
#else /* not XML_UNICODE */
/* Information is UTF-8 encoded. */
typedef char XML_Char;
typedef char XML_LChar;
#endif /* not XML_UNICODE */
#endif /* not XML_UNICODE_WCHAR_T */
/* Constructs a new parser; encoding is the encoding specified by the external
protocol or null if there is none specified. */
XML_Parser XMLPARSEAPI
XML_ParserCreate(const XML_Char *encoding);
/* Constructs a new parser and namespace processor. Element type names
and attribute names that belong to a namespace will be expanded;
unprefixed attribute names are never expanded; unprefixed element type
names are expanded only if there is a default namespace. The expanded
name is the concatenation of the namespace URI, the namespace separator character,
and the local part of the name. If the namespace separator is '\0' then
the namespace URI and the local part will be concatenated without any
separator. When a namespace is not declared, the name and prefix will be
passed through without expansion. */
XML_Parser XMLPARSEAPI
XML_ParserCreateNS(const XML_Char *encoding, XML_Char namespaceSeparator);
/* atts is array of name/value pairs, terminated by 0;
names and values are 0 terminated. */
typedef void (*XML_StartElementHandler)(void *userData,
const XML_Char *name,
const XML_Char **atts);
typedef void (*XML_EndElementHandler)(void *userData,
const XML_Char *name);
/* s is not 0 terminated. */
typedef void (*XML_CharacterDataHandler)(void *userData,
const XML_Char *s,
int len);
/* target and data are 0 terminated */
typedef void (*XML_ProcessingInstructionHandler)(void *userData,
const XML_Char *target,
const XML_Char *data);
/* data is 0 terminated */
typedef void (*XML_CommentHandler)(void *userData, const XML_Char *data);
typedef void (*XML_StartCdataSectionHandler)(void *userData);
typedef void (*XML_EndCdataSectionHandler)(void *userData);
/* This is called for any characters in the XML document for
which there is no applicable handler. This includes both
characters that are part of markup which is of a kind that is
not reported (comments, markup declarations), or characters
that are part of a construct which could be reported but
for which no handler has been supplied. The characters are passed
exactly as they were in the XML document except that
they will be encoded in UTF-8. Line boundaries are not normalized.
Note that a byte order mark character is not passed to the default handler.
There are no guarantees about how characters are divided between calls
to the default handler: for example, a comment might be split between
multiple calls. */
typedef void (*XML_DefaultHandler)(void *userData,
const XML_Char *s,
int len);
/* This is called for the start of the DOCTYPE declaration when the
name of the DOCTYPE is encountered. */
typedef void (*XML_StartDoctypeDeclHandler)(void *userData,
const XML_Char *doctypeName);
/* This is called for the start of the DOCTYPE declaration when the
closing > is encountered, but after processing any external subset. */
typedef void (*XML_EndDoctypeDeclHandler)(void *userData);
/* This is called for a declaration of an unparsed (NDATA)
entity. The base argument is whatever was set by XML_SetBase.
The entityName, systemId and notationName arguments will never be null.
The other arguments may be. */
typedef void (*XML_UnparsedEntityDeclHandler)(void *userData,
const XML_Char *entityName,
const XML_Char *base,
const XML_Char *systemId,
const XML_Char *publicId,
const XML_Char *notationName);
/* This is called for a declaration of notation.
The base argument is whatever was set by XML_SetBase.
The notationName will never be null. The other arguments can be. */
typedef void (*XML_NotationDeclHandler)(void *userData,
const XML_Char *notationName,
const XML_Char *base,
const XML_Char *systemId,
const XML_Char *publicId);
typedef void (*XML_ExternalParsedEntityDeclHandler)(void *userData,
const XML_Char *entityName,
const XML_Char *base,
const XML_Char *systemId,
const XML_Char *publicId);
typedef void (*XML_InternalParsedEntityDeclHandler)(void *userData,
const XML_Char *entityName,
const XML_Char *replacementText,
int replacementTextLength);
/* When namespace processing is enabled, these are called once for
each namespace declaration. The call to the start and end element
handlers occur between the calls to the start and end namespace
declaration handlers. For an xmlns attribute, prefix will be null.
For an xmlns="" attribute, uri will be null. */
typedef void (*XML_StartNamespaceDeclHandler)(void *userData,
const XML_Char *prefix,
const XML_Char *uri);
typedef void (*XML_EndNamespaceDeclHandler)(void *userData,
const XML_Char *prefix);
/* This is called if the document is not standalone (it has an
external subset or a reference to a parameter entity, but does not
have standalone="yes"). If this handler returns 0, then processing
will not continue, and the parser will return a
XML_ERROR_NOT_STANDALONE error. */
typedef int (*XML_NotStandaloneHandler)(void *userData);
/* This is called for a reference to an external parsed general entity.
The referenced entity is not automatically parsed.
The application can parse it immediately or later using
XML_ExternalEntityParserCreate.
The parser argument is the parser parsing the entity containing the reference;
it can be passed as the parser argument to XML_ExternalEntityParserCreate.
The systemId argument is the system identifier as specified in the entity declaration;
it will not be null.
The base argument is the system identifier that should be used as the base for
resolving systemId if systemId was relative; this is set by XML_SetBase;
it may be null.
The publicId argument is the public identifier as specified in the entity declaration,
or null if none was specified; the whitespace in the public identifier
will have been normalized as required by the XML spec.
The context argument specifies the parsing context in the format
expected by the context argument to
XML_ExternalEntityParserCreate; context is valid only until the handler
returns, so if the referenced entity is to be parsed later, it must be copied.
The handler should return 0 if processing should not continue because of
a fatal error in the handling of the external entity.
In this case the calling parser will return an XML_ERROR_EXTERNAL_ENTITY_HANDLING
error.
Note that unlike other handlers the first argument is the parser, not userData. */
typedef int (*XML_ExternalEntityRefHandler)(XML_Parser parser,
const XML_Char *context,
const XML_Char *base,
const XML_Char *systemId,
const XML_Char *publicId);
/* This structure is filled in by the XML_UnknownEncodingHandler
to provide information to the parser about encodings that are unknown
to the parser.
The map[b] member gives information about byte sequences
whose first byte is b.
If map[b] is c where c is >= 0, then b by itself encodes the Unicode scalar value c.
If map[b] is -1, then the byte sequence is malformed.
If map[b] is -n, where n >= 2, then b is the first byte of an n-byte
sequence that encodes a single Unicode scalar value.
The data member will be passed as the first argument to the convert function.
The convert function is used to convert multibyte sequences;
s will point to a n-byte sequence where map[(unsigned char)*s] == -n.
The convert function must return the Unicode scalar value
represented by this byte sequence or -1 if the byte sequence is malformed.
The convert function may be null if the encoding is a single-byte encoding,
that is if map[b] >= -1 for all bytes b.
When the parser is finished with the encoding, then if release is not null,
it will call release passing it the data member;
once release has been called, the convert function will not be called again.
Expat places certain restrictions on the encodings that are supported
using this mechanism.
1. Every ASCII character that can appear in a well-formed XML document,
other than the characters
$@\^`{}~
must be represented by a single byte, and that byte must be the
same byte that represents that character in ASCII.
2. No character may require more than 4 bytes to encode.
3. All characters encoded must have Unicode scalar values <= 0xFFFF,
(ie characters that would be encoded by surrogates in UTF-16
are not allowed). Note that this restriction doesn't apply to
the built-in support for UTF-8 and UTF-16.
4. No Unicode character may be encoded by more than one distinct sequence
of bytes. */
typedef struct {
int map[256];
void *data;
int (*convert)(void *data, const char *s);
void (*release)(void *data);
} XML_Encoding;
/* This is called for an encoding that is unknown to the parser.
The encodingHandlerData argument is that which was passed as the
second argument to XML_SetUnknownEncodingHandler.
The name argument gives the name of the encoding as specified in
the encoding declaration.
If the callback can provide information about the encoding,
it must fill in the XML_Encoding structure, and return 1.
Otherwise it must return 0.
If info does not describe a suitable encoding,
then the parser will return an XML_UNKNOWN_ENCODING error. */
typedef int (*XML_UnknownEncodingHandler)(void *encodingHandlerData,
const XML_Char *name,
XML_Encoding *info);
void XMLPARSEAPI
XML_SetElementHandler(XML_Parser parser,
XML_StartElementHandler start,
XML_EndElementHandler end);
void XMLPARSEAPI
XML_SetCharacterDataHandler(XML_Parser parser,
XML_CharacterDataHandler handler);
void XMLPARSEAPI
XML_SetProcessingInstructionHandler(XML_Parser parser,
XML_ProcessingInstructionHandler handler);
void XMLPARSEAPI
XML_SetCommentHandler(XML_Parser parser,
XML_CommentHandler handler);
void XMLPARSEAPI
XML_SetCdataSectionHandler(XML_Parser parser,
XML_StartCdataSectionHandler start,
XML_EndCdataSectionHandler end);
/* This sets the default handler and also inhibits expansion of internal entities.
The entity reference will be passed to the default handler. */
void XMLPARSEAPI
XML_SetDefaultHandler(XML_Parser parser,
XML_DefaultHandler handler);
/* This sets the default handler but does not inhibit expansion of internal entities.
The entity reference will not be passed to the default handler. */
void XMLPARSEAPI
XML_SetDefaultHandlerExpand(XML_Parser parser,
XML_DefaultHandler handler);
void XMLPARSEAPI
XML_SetDoctypeDeclHandler(XML_Parser parser,
XML_StartDoctypeDeclHandler start,
XML_EndDoctypeDeclHandler end);
void XMLPARSEAPI
XML_SetUnparsedEntityDeclHandler(XML_Parser parser,
XML_UnparsedEntityDeclHandler handler);
void XMLPARSEAPI
XML_SetNotationDeclHandler(XML_Parser parser,
XML_NotationDeclHandler handler);
void XMLPARSEAPI
XML_SetExternalParsedEntityDeclHandler(XML_Parser parser,
XML_ExternalParsedEntityDeclHandler handler);
void XMLPARSEAPI
XML_SetInternalParsedEntityDeclHandler(XML_Parser parser,
XML_InternalParsedEntityDeclHandler handler);
void XMLPARSEAPI
XML_SetNamespaceDeclHandler(XML_Parser parser,
XML_StartNamespaceDeclHandler start,
XML_EndNamespaceDeclHandler end);
void XMLPARSEAPI
XML_SetNotStandaloneHandler(XML_Parser parser,
XML_NotStandaloneHandler handler);
void XMLPARSEAPI
XML_SetExternalEntityRefHandler(XML_Parser parser,
XML_ExternalEntityRefHandler handler);
/* If a non-null value for arg is specified here, then it will be passed
as the first argument to the external entity ref handler instead
of the parser object. */
void XMLPARSEAPI
XML_SetExternalEntityRefHandlerArg(XML_Parser, void *arg);
void XMLPARSEAPI
XML_SetUnknownEncodingHandler(XML_Parser parser,
XML_UnknownEncodingHandler handler,
void *encodingHandlerData);
/* This can be called within a handler for a start element, end element,
processing instruction or character data. It causes the corresponding
markup to be passed to the default handler. */
void XMLPARSEAPI XML_DefaultCurrent(XML_Parser parser);
/* This value is passed as the userData argument to callbacks. */
void XMLPARSEAPI
XML_SetUserData(XML_Parser parser, void *userData);
/* Returns the last value set by XML_SetUserData or null. */
#define XML_GetUserData(parser) (*(void **)(parser))
/* This is equivalent to supplying an encoding argument
to XML_ParserCreate. It must not be called after XML_Parse
or XML_ParseBuffer. */
int XMLPARSEAPI
XML_SetEncoding(XML_Parser parser, const XML_Char *encoding);
/* If this function is called, then the parser will be passed
as the first argument to callbacks instead of userData.
The userData will still be accessible using XML_GetUserData. */
void XMLPARSEAPI
XML_UseParserAsHandlerArg(XML_Parser parser);
/* Sets the base to be used for resolving relative URIs in system identifiers in
declarations. Resolving relative identifiers is left to the application:
this value will be passed through as the base argument to the
XML_ExternalEntityRefHandler, XML_NotationDeclHandler
and XML_UnparsedEntityDeclHandler. The base argument will be copied.
Returns zero if out of memory, non-zero otherwise. */
int XMLPARSEAPI
XML_SetBase(XML_Parser parser, const XML_Char *base);
const XML_Char XMLPARSEAPI *
XML_GetBase(XML_Parser parser);
/* Returns the number of the attribute/value pairs passed in last call
to the XML_StartElementHandler that were specified in the start-tag
rather than defaulted. Each attribute/value pair counts as 2; thus
this correspondds to an index into the atts array passed to the
XML_StartElementHandler. */
int XMLPARSEAPI XML_GetSpecifiedAttributeCount(XML_Parser parser);
/* Returns the index of the ID attribute passed in the last call to
XML_StartElementHandler, or -1 if there is no ID attribute. Each
attribute/value pair counts as 2; thus this correspondds to an index
into the atts array passed to the XML_StartElementHandler. */
int XMLPARSEAPI XML_GetIdAttributeIndex(XML_Parser parser);
/* Parses some input. Returns 0 if a fatal error is detected.
The last call to XML_Parse must have isFinal true;
len may be zero for this call (or any other). */
int XMLPARSEAPI
XML_Parse(XML_Parser parser, const char *s, int len, int isFinal);
void XMLPARSEAPI *
XML_GetBuffer(XML_Parser parser, int len);
int XMLPARSEAPI
XML_ParseBuffer(XML_Parser parser, int len, int isFinal);
/* Creates an XML_Parser object that can parse an external general entity;
context is a '\0'-terminated string specifying the parse context;
encoding is a '\0'-terminated string giving the name of the externally specified encoding,
or null if there is no externally specified encoding.
The context string consists of a sequence of tokens separated by formfeeds (\f);
a token consisting of a name specifies that the general entity of the name
is open; a token of the form prefix=uri specifies the namespace for a particular
prefix; a token of the form =uri specifies the default namespace.
This can be called at any point after the first call to an ExternalEntityRefHandler
so longer as the parser has not yet been freed.
The new parser is completely independent and may safely be used in a separate thread.
The handlers and userData are initialized from the parser argument.
Returns 0 if out of memory. Otherwise returns a new XML_Parser object. */
XML_Parser XMLPARSEAPI
XML_ExternalEntityParserCreate(XML_Parser parser,
const XML_Char *context,
const XML_Char *encoding);
enum XML_ParamEntityParsing {
XML_PARAM_ENTITY_PARSING_NEVER,
XML_PARAM_ENTITY_PARSING_UNLESS_STANDALONE,
XML_PARAM_ENTITY_PARSING_ALWAYS
};
/* Controls parsing of parameter entities (including the external DTD
subset). If parsing of parameter entities is enabled, then references
to external parameter entities (including the external DTD subset)
will be passed to the handler set with
XML_SetExternalEntityRefHandler. The context passed will be 0.
Unlike external general entities, external parameter entities can only
be parsed synchronously. If the external parameter entity is to be
parsed, it must be parsed during the call to the external entity ref
handler: the complete sequence of XML_ExternalEntityParserCreate,
XML_Parse/XML_ParseBuffer and XML_ParserFree calls must be made during
this call. After XML_ExternalEntityParserCreate has been called to
create the parser for the external parameter entity (context must be 0
for this call), it is illegal to make any calls on the old parser
until XML_ParserFree has been called on the newly created parser. If
the library has been compiled without support for parameter entity
parsing (ie without XML_DTD being defined), then
XML_SetParamEntityParsing will return 0 if parsing of parameter
entities is requested; otherwise it will return non-zero. */
int XMLPARSEAPI
XML_SetParamEntityParsing(XML_Parser parser,
enum XML_ParamEntityParsing parsing);
enum XML_Error {
XML_ERROR_NONE,
XML_ERROR_NO_MEMORY,
XML_ERROR_SYNTAX,
XML_ERROR_NO_ELEMENTS,
XML_ERROR_INVALID_TOKEN,
XML_ERROR_UNCLOSED_TOKEN,
XML_ERROR_PARTIAL_CHAR,
XML_ERROR_TAG_MISMATCH,
XML_ERROR_DUPLICATE_ATTRIBUTE,
XML_ERROR_JUNK_AFTER_DOC_ELEMENT,
XML_ERROR_PARAM_ENTITY_REF,
XML_ERROR_UNDEFINED_ENTITY,
XML_ERROR_RECURSIVE_ENTITY_REF,
XML_ERROR_ASYNC_ENTITY,
XML_ERROR_BAD_CHAR_REF,
XML_ERROR_BINARY_ENTITY_REF,
XML_ERROR_ATTRIBUTE_EXTERNAL_ENTITY_REF,
XML_ERROR_MISPLACED_XML_PI,
XML_ERROR_UNKNOWN_ENCODING,
XML_ERROR_INCORRECT_ENCODING,
XML_ERROR_UNCLOSED_CDATA_SECTION,
XML_ERROR_EXTERNAL_ENTITY_HANDLING,
XML_ERROR_NOT_STANDALONE
};
/* If XML_Parse or XML_ParseBuffer have returned 0, then XML_GetErrorCode
returns information about the error. */
enum XML_Error XMLPARSEAPI XML_GetErrorCode(XML_Parser parser);
/* These functions return information about the current parse location.
They may be called when XML_Parse or XML_ParseBuffer return 0;
in this case the location is the location of the character at which
the error was detected.
They may also be called from any other callback called to report
some parse event; in this the location is the location of the first
of the sequence of characters that generated the event. */
int XMLPARSEAPI XML_GetCurrentLineNumber(XML_Parser parser);
int XMLPARSEAPI XML_GetCurrentColumnNumber(XML_Parser parser);
long XMLPARSEAPI XML_GetCurrentByteIndex(XML_Parser parser);
/* Return the number of bytes in the current event.
Returns 0 if the event is in an internal entity. */
int XMLPARSEAPI XML_GetCurrentByteCount(XML_Parser parser);
/* For backwards compatibility with previous versions. */
#define XML_GetErrorLineNumber XML_GetCurrentLineNumber
#define XML_GetErrorColumnNumber XML_GetCurrentColumnNumber
#define XML_GetErrorByteIndex XML_GetCurrentByteIndex
/* Frees memory used by the parser. */
void XMLPARSEAPI
XML_ParserFree(XML_Parser parser);
/* Returns a string describing the error. */
const XML_LChar XMLPARSEAPI *XML_ErrorString(int code);
#ifdef __cplusplus
}
#endif
#endif /* not XmlParse_INCLUDED */

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/*
Copyright (c) 1998, 1999 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
#ifndef XmlRole_INCLUDED
#define XmlRole_INCLUDED 1
#include "xmltok.h"
#ifdef __cplusplus
extern "C" {
#endif
enum {
XML_ROLE_ERROR = -1,
XML_ROLE_NONE = 0,
XML_ROLE_XML_DECL,
XML_ROLE_INSTANCE_START,
XML_ROLE_DOCTYPE_NAME,
XML_ROLE_DOCTYPE_SYSTEM_ID,
XML_ROLE_DOCTYPE_PUBLIC_ID,
XML_ROLE_DOCTYPE_CLOSE,
XML_ROLE_GENERAL_ENTITY_NAME,
XML_ROLE_PARAM_ENTITY_NAME,
XML_ROLE_ENTITY_VALUE,
XML_ROLE_ENTITY_SYSTEM_ID,
XML_ROLE_ENTITY_PUBLIC_ID,
XML_ROLE_ENTITY_NOTATION_NAME,
XML_ROLE_NOTATION_NAME,
XML_ROLE_NOTATION_SYSTEM_ID,
XML_ROLE_NOTATION_NO_SYSTEM_ID,
XML_ROLE_NOTATION_PUBLIC_ID,
XML_ROLE_ATTRIBUTE_NAME,
XML_ROLE_ATTRIBUTE_TYPE_CDATA,
XML_ROLE_ATTRIBUTE_TYPE_ID,
XML_ROLE_ATTRIBUTE_TYPE_IDREF,
XML_ROLE_ATTRIBUTE_TYPE_IDREFS,
XML_ROLE_ATTRIBUTE_TYPE_ENTITY,
XML_ROLE_ATTRIBUTE_TYPE_ENTITIES,
XML_ROLE_ATTRIBUTE_TYPE_NMTOKEN,
XML_ROLE_ATTRIBUTE_TYPE_NMTOKENS,
XML_ROLE_ATTRIBUTE_ENUM_VALUE,
XML_ROLE_ATTRIBUTE_NOTATION_VALUE,
XML_ROLE_ATTLIST_ELEMENT_NAME,
XML_ROLE_IMPLIED_ATTRIBUTE_VALUE,
XML_ROLE_REQUIRED_ATTRIBUTE_VALUE,
XML_ROLE_DEFAULT_ATTRIBUTE_VALUE,
XML_ROLE_FIXED_ATTRIBUTE_VALUE,
XML_ROLE_ELEMENT_NAME,
XML_ROLE_CONTENT_ANY,
XML_ROLE_CONTENT_EMPTY,
XML_ROLE_CONTENT_PCDATA,
XML_ROLE_GROUP_OPEN,
XML_ROLE_GROUP_CLOSE,
XML_ROLE_GROUP_CLOSE_REP,
XML_ROLE_GROUP_CLOSE_OPT,
XML_ROLE_GROUP_CLOSE_PLUS,
XML_ROLE_GROUP_CHOICE,
XML_ROLE_GROUP_SEQUENCE,
XML_ROLE_CONTENT_ELEMENT,
XML_ROLE_CONTENT_ELEMENT_REP,
XML_ROLE_CONTENT_ELEMENT_OPT,
XML_ROLE_CONTENT_ELEMENT_PLUS,
#ifdef XML_DTD
XML_ROLE_TEXT_DECL,
XML_ROLE_IGNORE_SECT,
XML_ROLE_INNER_PARAM_ENTITY_REF,
#endif /* XML_DTD */
XML_ROLE_PARAM_ENTITY_REF,
XML_ROLE_EXTERNAL_GENERAL_ENTITY_NO_NOTATION
};
typedef struct prolog_state {
int (*handler)(struct prolog_state *state,
int tok,
const char *ptr,
const char *end,
const ENCODING *enc);
unsigned level;
#ifdef XML_DTD
unsigned includeLevel;
int documentEntity;
#endif /* XML_DTD */
} PROLOG_STATE;
void XMLTOKAPI XmlPrologStateInit(PROLOG_STATE *);
#ifdef XML_DTD
void XMLTOKAPI XmlPrologStateInitExternalEntity(PROLOG_STATE *);
#endif /* XML_DTD */
#define XmlTokenRole(state, tok, ptr, end, enc) \
(((state)->handler)(state, tok, ptr, end, enc))
#ifdef __cplusplus
}
#endif
#endif /* not XmlRole_INCLUDED */

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/*
Copyright (c) 1998, 1999 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
#ifndef XmlTok_INCLUDED
#define XmlTok_INCLUDED 1
#ifdef __cplusplus
extern "C" {
#endif
#ifndef XMLTOKAPI
#define XMLTOKAPI /* as nothing */
#endif
/* The following token may be returned by XmlContentTok */
#define XML_TOK_TRAILING_RSQB -5 /* ] or ]] at the end of the scan; might be start of
illegal ]]> sequence */
/* The following tokens may be returned by both XmlPrologTok and XmlContentTok */
#define XML_TOK_NONE -4 /* The string to be scanned is empty */
#define XML_TOK_TRAILING_CR -3 /* A CR at the end of the scan;
might be part of CRLF sequence */
#define XML_TOK_PARTIAL_CHAR -2 /* only part of a multibyte sequence */
#define XML_TOK_PARTIAL -1 /* only part of a token */
#define XML_TOK_INVALID 0
/* The following tokens are returned by XmlContentTok; some are also
returned by XmlAttributeValueTok, XmlEntityTok, XmlCdataSectionTok */
#define XML_TOK_START_TAG_WITH_ATTS 1
#define XML_TOK_START_TAG_NO_ATTS 2
#define XML_TOK_EMPTY_ELEMENT_WITH_ATTS 3 /* empty element tag <e/> */
#define XML_TOK_EMPTY_ELEMENT_NO_ATTS 4
#define XML_TOK_END_TAG 5
#define XML_TOK_DATA_CHARS 6
#define XML_TOK_DATA_NEWLINE 7
#define XML_TOK_CDATA_SECT_OPEN 8
#define XML_TOK_ENTITY_REF 9
#define XML_TOK_CHAR_REF 10 /* numeric character reference */
/* The following tokens may be returned by both XmlPrologTok and XmlContentTok */
#define XML_TOK_PI 11 /* processing instruction */
#define XML_TOK_XML_DECL 12 /* XML decl or text decl */
#define XML_TOK_COMMENT 13
#define XML_TOK_BOM 14 /* Byte order mark */
/* The following tokens are returned only by XmlPrologTok */
#define XML_TOK_PROLOG_S 15
#define XML_TOK_DECL_OPEN 16 /* <!foo */
#define XML_TOK_DECL_CLOSE 17 /* > */
#define XML_TOK_NAME 18
#define XML_TOK_NMTOKEN 19
#define XML_TOK_POUND_NAME 20 /* #name */
#define XML_TOK_OR 21 /* | */
#define XML_TOK_PERCENT 22
#define XML_TOK_OPEN_PAREN 23
#define XML_TOK_CLOSE_PAREN 24
#define XML_TOK_OPEN_BRACKET 25
#define XML_TOK_CLOSE_BRACKET 26
#define XML_TOK_LITERAL 27
#define XML_TOK_PARAM_ENTITY_REF 28
#define XML_TOK_INSTANCE_START 29
/* The following occur only in element type declarations */
#define XML_TOK_NAME_QUESTION 30 /* name? */
#define XML_TOK_NAME_ASTERISK 31 /* name* */
#define XML_TOK_NAME_PLUS 32 /* name+ */
#define XML_TOK_COND_SECT_OPEN 33 /* <![ */
#define XML_TOK_COND_SECT_CLOSE 34 /* ]]> */
#define XML_TOK_CLOSE_PAREN_QUESTION 35 /* )? */
#define XML_TOK_CLOSE_PAREN_ASTERISK 36 /* )* */
#define XML_TOK_CLOSE_PAREN_PLUS 37 /* )+ */
#define XML_TOK_COMMA 38
/* The following token is returned only by XmlAttributeValueTok */
#define XML_TOK_ATTRIBUTE_VALUE_S 39
/* The following token is returned only by XmlCdataSectionTok */
#define XML_TOK_CDATA_SECT_CLOSE 40
/* With namespace processing this is returned by XmlPrologTok
for a name with a colon. */
#define XML_TOK_PREFIXED_NAME 41
#ifdef XML_DTD
#define XML_TOK_IGNORE_SECT 42
#endif /* XML_DTD */
#ifdef XML_DTD
#define XML_N_STATES 4
#else /* not XML_DTD */
#define XML_N_STATES 3
#endif /* not XML_DTD */
#define XML_PROLOG_STATE 0
#define XML_CONTENT_STATE 1
#define XML_CDATA_SECTION_STATE 2
#ifdef XML_DTD
#define XML_IGNORE_SECTION_STATE 3
#endif /* XML_DTD */
#define XML_N_LITERAL_TYPES 2
#define XML_ATTRIBUTE_VALUE_LITERAL 0
#define XML_ENTITY_VALUE_LITERAL 1
/* The size of the buffer passed to XmlUtf8Encode must be at least this. */
#define XML_UTF8_ENCODE_MAX 4
/* The size of the buffer passed to XmlUtf16Encode must be at least this. */
#define XML_UTF16_ENCODE_MAX 2
typedef struct position {
/* first line and first column are 0 not 1 */
unsigned long lineNumber;
unsigned long columnNumber;
} POSITION;
typedef struct {
const char *name;
const char *valuePtr;
const char *valueEnd;
char normalized;
} ATTRIBUTE;
struct encoding;
typedef struct encoding ENCODING;
struct encoding {
int (*scanners[XML_N_STATES])(const ENCODING *,
const char *,
const char *,
const char **);
int (*literalScanners[XML_N_LITERAL_TYPES])(const ENCODING *,
const char *,
const char *,
const char **);
int (*sameName)(const ENCODING *,
const char *, const char *);
int (*nameMatchesAscii)(const ENCODING *,
const char *, const char *, const char *);
int (*nameLength)(const ENCODING *, const char *);
const char *(*skipS)(const ENCODING *, const char *);
int (*getAtts)(const ENCODING *enc, const char *ptr,
int attsMax, ATTRIBUTE *atts);
int (*charRefNumber)(const ENCODING *enc, const char *ptr);
int (*predefinedEntityName)(const ENCODING *, const char *, const char *);
void (*updatePosition)(const ENCODING *,
const char *ptr,
const char *end,
POSITION *);
int (*isPublicId)(const ENCODING *enc, const char *ptr, const char *end,
const char **badPtr);
void (*utf8Convert)(const ENCODING *enc,
const char **fromP,
const char *fromLim,
char **toP,
const char *toLim);
void (*utf16Convert)(const ENCODING *enc,
const char **fromP,
const char *fromLim,
unsigned short **toP,
const unsigned short *toLim);
int minBytesPerChar;
char isUtf8;
char isUtf16;
};
/*
Scan the string starting at ptr until the end of the next complete token,
but do not scan past eptr. Return an integer giving the type of token.
Return XML_TOK_NONE when ptr == eptr; nextTokPtr will not be set.
Return XML_TOK_PARTIAL when the string does not contain a complete token;
nextTokPtr will not be set.
Return XML_TOK_INVALID when the string does not start a valid token; nextTokPtr
will be set to point to the character which made the token invalid.
Otherwise the string starts with a valid token; nextTokPtr will be set to point
to the character following the end of that token.
Each data character counts as a single token, but adjacent data characters
may be returned together. Similarly for characters in the prolog outside
literals, comments and processing instructions.
*/
#define XmlTok(enc, state, ptr, end, nextTokPtr) \
(((enc)->scanners[state])(enc, ptr, end, nextTokPtr))
#define XmlPrologTok(enc, ptr, end, nextTokPtr) \
XmlTok(enc, XML_PROLOG_STATE, ptr, end, nextTokPtr)
#define XmlContentTok(enc, ptr, end, nextTokPtr) \
XmlTok(enc, XML_CONTENT_STATE, ptr, end, nextTokPtr)
#define XmlCdataSectionTok(enc, ptr, end, nextTokPtr) \
XmlTok(enc, XML_CDATA_SECTION_STATE, ptr, end, nextTokPtr)
#ifdef XML_DTD
#define XmlIgnoreSectionTok(enc, ptr, end, nextTokPtr) \
XmlTok(enc, XML_IGNORE_SECTION_STATE, ptr, end, nextTokPtr)
#endif /* XML_DTD */
/* This is used for performing a 2nd-level tokenization on
the content of a literal that has already been returned by XmlTok. */
#define XmlLiteralTok(enc, literalType, ptr, end, nextTokPtr) \
(((enc)->literalScanners[literalType])(enc, ptr, end, nextTokPtr))
#define XmlAttributeValueTok(enc, ptr, end, nextTokPtr) \
XmlLiteralTok(enc, XML_ATTRIBUTE_VALUE_LITERAL, ptr, end, nextTokPtr)
#define XmlEntityValueTok(enc, ptr, end, nextTokPtr) \
XmlLiteralTok(enc, XML_ENTITY_VALUE_LITERAL, ptr, end, nextTokPtr)
#define XmlSameName(enc, ptr1, ptr2) (((enc)->sameName)(enc, ptr1, ptr2))
#define XmlNameMatchesAscii(enc, ptr1, end1, ptr2) \
(((enc)->nameMatchesAscii)(enc, ptr1, end1, ptr2))
#define XmlNameLength(enc, ptr) \
(((enc)->nameLength)(enc, ptr))
#define XmlSkipS(enc, ptr) \
(((enc)->skipS)(enc, ptr))
#define XmlGetAttributes(enc, ptr, attsMax, atts) \
(((enc)->getAtts)(enc, ptr, attsMax, atts))
#define XmlCharRefNumber(enc, ptr) \
(((enc)->charRefNumber)(enc, ptr))
#define XmlPredefinedEntityName(enc, ptr, end) \
(((enc)->predefinedEntityName)(enc, ptr, end))
#define XmlUpdatePosition(enc, ptr, end, pos) \
(((enc)->updatePosition)(enc, ptr, end, pos))
#define XmlIsPublicId(enc, ptr, end, badPtr) \
(((enc)->isPublicId)(enc, ptr, end, badPtr))
#define XmlUtf8Convert(enc, fromP, fromLim, toP, toLim) \
(((enc)->utf8Convert)(enc, fromP, fromLim, toP, toLim))
#define XmlUtf16Convert(enc, fromP, fromLim, toP, toLim) \
(((enc)->utf16Convert)(enc, fromP, fromLim, toP, toLim))
typedef struct {
ENCODING initEnc;
const ENCODING **encPtr;
} INIT_ENCODING;
int XMLTOKAPI XmlParseXmlDecl(int isGeneralTextEntity,
const ENCODING *enc,
const char *ptr,
const char *end,
const char **badPtr,
const char **versionPtr,
const char **encodingNamePtr,
const ENCODING **namedEncodingPtr,
int *standalonePtr);
int XMLTOKAPI XmlInitEncoding(INIT_ENCODING *, const ENCODING **, const char *name);
const ENCODING XMLTOKAPI *XmlGetUtf8InternalEncoding(void);
const ENCODING XMLTOKAPI *XmlGetUtf16InternalEncoding(void);
int XMLTOKAPI XmlUtf8Encode(int charNumber, char *buf);
int XMLTOKAPI XmlUtf16Encode(int charNumber, unsigned short *buf);
int XMLTOKAPI XmlSizeOfUnknownEncoding(void);
ENCODING XMLTOKAPI *
XmlInitUnknownEncoding(void *mem,
int *table,
int (*conv)(void *userData, const char *p),
void *userData);
int XMLTOKAPI XmlParseXmlDeclNS(int isGeneralTextEntity,
const ENCODING *enc,
const char *ptr,
const char *end,
const char **badPtr,
const char **versionPtr,
const char **encodingNamePtr,
const ENCODING **namedEncodingPtr,
int *standalonePtr);
int XMLTOKAPI XmlInitEncodingNS(INIT_ENCODING *, const ENCODING **, const char *name);
const ENCODING XMLTOKAPI *XmlGetUtf8InternalEncodingNS(void);
const ENCODING XMLTOKAPI *XmlGetUtf16InternalEncodingNS(void);
ENCODING XMLTOKAPI *
XmlInitUnknownEncodingNS(void *mem,
int *table,
int (*conv)(void *userData, const char *p),
void *userData);
#ifdef __cplusplus
}
#endif
#endif /* not XmlTok_INCLUDED */

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/*
Copyright (c) 1998, 1999 Thai Open Source Software Center Ltd
See the file copying.txt for copying permission.
*/
enum {
BT_NONXML,
BT_MALFORM,
BT_LT,
BT_AMP,
BT_RSQB,
BT_LEAD2,
BT_LEAD3,
BT_LEAD4,
BT_TRAIL,
BT_CR,
BT_LF,
BT_GT,
BT_QUOT,
BT_APOS,
BT_EQUALS,
BT_QUEST,
BT_EXCL,
BT_SOL,
BT_SEMI,
BT_NUM,
BT_LSQB,
BT_S,
BT_NMSTRT,
BT_COLON,
BT_HEX,
BT_DIGIT,
BT_NAME,
BT_MINUS,
BT_OTHER, /* known not to be a name or name start character */
BT_NONASCII, /* might be a name or name start character */
BT_PERCNT,
BT_LPAR,
BT_RPAR,
BT_AST,
BT_PLUS,
BT_COMMA,
BT_VERBAR
};
#include <stddef.h>

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const ENCODING *NS(XmlGetUtf8InternalEncoding)(void)
{
return &ns(internal_utf8_encoding).enc;
}
const ENCODING *NS(XmlGetUtf16InternalEncoding)(void)
{
#if XML_BYTE_ORDER == 12
return &ns(internal_little2_encoding).enc;
#elif XML_BYTE_ORDER == 21
return &ns(internal_big2_encoding).enc;
#else
const short n = 1;
return *(const char *)&n ? &ns(internal_little2_encoding).enc : &ns(internal_big2_encoding).enc;
#endif
}
static
const ENCODING *NS(encodings)[] = {
&ns(latin1_encoding).enc,
&ns(ascii_encoding).enc,
&ns(utf8_encoding).enc,
&ns(big2_encoding).enc,
&ns(big2_encoding).enc,
&ns(little2_encoding).enc,
&ns(utf8_encoding).enc /* NO_ENC */
};
static
int NS(initScanProlog)(const ENCODING *enc, const char *ptr, const char *end,
const char **nextTokPtr)
{
return initScan(NS(encodings), (const INIT_ENCODING *)enc, XML_PROLOG_STATE, ptr, end, nextTokPtr);
}
static
int NS(initScanContent)(const ENCODING *enc, const char *ptr, const char *end,
const char **nextTokPtr)
{
return initScan(NS(encodings), (const INIT_ENCODING *)enc, XML_CONTENT_STATE, ptr, end, nextTokPtr);
}
int NS(XmlInitEncoding)(INIT_ENCODING *p, const ENCODING **encPtr, const char *name)
{
int i = getEncodingIndex(name);
if (i == UNKNOWN_ENC)
return 0;
SET_INIT_ENC_INDEX(p, i);
p->initEnc.scanners[XML_PROLOG_STATE] = NS(initScanProlog);
p->initEnc.scanners[XML_CONTENT_STATE] = NS(initScanContent);
p->initEnc.updatePosition = initUpdatePosition;
p->encPtr = encPtr;
*encPtr = &(p->initEnc);
return 1;
}
static
const ENCODING *NS(findEncoding)(const ENCODING *enc, const char *ptr, const char *end)
{
#define ENCODING_MAX 128
char buf[ENCODING_MAX];
char *p = buf;
int i;
XmlUtf8Convert(enc, &ptr, end, &p, p + ENCODING_MAX - 1);
if (ptr != end)
return 0;
*p = 0;
if (streqci(buf, KW_UTF_16) && enc->minBytesPerChar == 2)
return enc;
i = getEncodingIndex(buf);
if (i == UNKNOWN_ENC)
return 0;
return NS(encodings)[i];
}
int NS(XmlParseXmlDecl)(int isGeneralTextEntity,
const ENCODING *enc,
const char *ptr,
const char *end,
const char **badPtr,
const char **versionPtr,
const char **encodingName,
const ENCODING **encoding,
int *standalone)
{
return doParseXmlDecl(NS(findEncoding),
isGeneralTextEntity,
enc,
ptr,
end,
badPtr,
versionPtr,
encodingName,
encoding,
standalone);
}

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/LICENSE.txt/1.3/Sun Mar 2 08:01:30 2003//
/README.txt/1.3/Sun Mar 2 08:01:31 2003//
/index.html/1.3/Sun Mar 2 08:01:32 2003//
D

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A D/docs////
A D/macproj////
A D/pa_common////
A D/pa_linux_oss////
A D/pa_mac////
A D/pa_mac_core////
A D/pa_tests////
A D/pa_unix_oss////
A D/pa_win_ds////
A D/pa_win_wmme////
A D/pablio////
A D/winproj////

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/cvsroot/audacity/lib-src/portaudio

1
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:ext:habes@cvs.sourceforge.net:/cvsroot/audacity

65
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Portable header file to contain:
/*
* PortAudio Portable Real-Time Audio Library
* PortAudio API Header File
* Latest version available at: 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.
*
*/
Implementation files to contain:
/*
* PortAudio Portable Real-Time Audio Library
* Latest version at: http://www.audiomulch.com/portaudio/
* <platform> Implementation
* Copyright (c) 1999-2000 <author(s)>
*
* 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.
*
*/

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README for PortAudio
Implementations for PC DirectSound and Mac SoundManager
/*
* PortAudio Portable Real-Time Audio Library
* Latest Version at: http://www.portaudio.com//
*
* Copyright (c) 1999-2000 Phil Burk and Ross Bencina
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*/
PortAudio is a portable audio I/O library designed for cross-platform
support of audio. It uses a callback mechanism to request audio processing.
Audio can be generated in various formats, including 32 bit floating point,
and will be converted to the native format internally.
Documentation:
See "pa_common/portaudio.h" for API spec.
See docs folder for a tutorial.
Also see http://www.portaudio.com/docs/
And see "pa_tests/patest_saw.c" for an example.
For information on compiling programs with PortAudio, please see the
tutorial at:
http://www.portaudio.com/docs/pa_tutorial.html
Important Files and Folders:
pa_common/ = platform independant code
pa_common/portaudio.h = header file for PortAudio API. Specifies API.
pa_common/pa_lib.c = host independant code for all implementations.
pablio = simple blocking read/write interface
Platform Implementations
pa_asio = ASIO for Windows and Macintosh
pa_beos = BeOS
pa_mac = Macintosh Sound Manager for OS 8,9 and Carbon
pa_mac_core = Macintosh Core Audio for OS X
pa_sgi = Silicon Graphics AL
pa_unix_oss = OSS implementation for various Unixes
pa_win_ds = Windows Direct Sound
pa_win_wmme = Windows MME (most widely supported)
Test Programs
pa_tests/pa_fuzz.c = guitar fuzz box
pa_tests/pa_devs.c = print a list of available devices
pa_tests/pa_minlat.c = determine minimum latency for your machine
pa_tests/paqa_devs.c = self test that opens all devices
pa_tests/paqa_errs.c = test error detection and reporting
pa_tests/patest_clip.c = hear a sine wave clipped and unclipped
pa_tests/patest_dither.c = hear effects of dithering (extremely subtle)
pa_tests/patest_pink.c = fun with pink noise
pa_tests/patest_record.c = record and playback some audio
pa_tests/patest_maxsines.c = how many sine waves can we play? Tests Pa_GetCPULoad().
pa_tests/patest_sine.c = output a sine wave in a simple PA app
pa_tests/patest_sync.c = test syncronization of audio and video
pa_tests/patest_wire.c = pass input to output, wire simulator

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/pa_tut_asio.html/1.1.1.1/Mon Nov 26 05:47:25 2001//
/pa_tut_callback.html/1.2/Mon Nov 26 05:56:05 2001//
/pa_tut_devs.html/1.2/Mon Nov 26 05:56:05 2001//
/pa_tut_explore.html/1.2/Mon Nov 26 05:56:05 2001//
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/pa_tut_rw.html/1.2/Mon Nov 26 05:56:05 2001//
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/pa_tut_util.html/1.2/Mon Nov 26 05:56:05 2001//
/pa_tutorial.html/1.2/Mon Nov 26 05:56:05 2001//
/portaudio_h.txt/1.2/Mon Nov 26 05:56:05 2001//
/portaudio_icmc2001.pdf/1.1.1.1/Mon Nov 26 05:47:28 2001//
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="PortAudio Docs, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Docs</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Documentation</h1></center>
</td>
</tr>
</table></center>
<p>Copyright 2000 Phil Burk and Ross Bencina
<h2>
<a href="pa_tutorial.html">Tutorial</a></h2>
<blockquote>Describes how to write audio programs using the PortAudio API.</blockquote>
<h2>
<a href="pa_impl_guide.html">Implementation Guide</a></h2>
<blockquote>Describes how to write an implementation of PortAudio for a
new computer platform.</blockquote>
<h2>
<a href="portaudio_icmc2001.pdf">Paper Presented at ICMC2001</a> (PDF)</h2>
<blockquote>Describes the PortAudio API and discusses implementation issues.
Written July 2001.</blockquote>
<p><br><a href="http://www.portaudio.com/">Return to PortAudio Home Page</a>
</body>
</html>

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.75 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Internal docs. How a stream is started or stopped.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Implementation - Start/Stop</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Implementation Guide</h1></center>
</td>
</tr>
</table></center>
<p>This document describes how to implement the PortAudio API on a new
computer platform. Implementing PortAudio on a new platform, makes it possible
to port many existing audio applications to that platform.
<p>By Phil Burk
<br>Copyright 2000 Phil Burk and Ross Bencina
<p>Note that the license says: <b>"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."</b>.
So when you have finished a new implementation, please send it back to
us at&nbsp; "<a href="http://www.portaudio.com">http://www.portaudio.com</a>"
so that we can make it available for other users. Thank you!
<h2>
Download the Latest PortAudio Implementation</h2>
Always start with the latest implementation available at "<a href="http://www.portaudio.com">http://www.portaudio.com</a>".
<h2>
Select an Existing Implementation as a Basis</h2>
The fastest way to get started is to take an existing implementation and
translate it for your new platform. Choose an implementation whose architecture
is as close as possible to your target.
<ul>
<li>
DirectSound Implementation - pa_win_ds - Uses a timer callback for the
background "thread". Polls a circular buffer and writes blocks of data
to keep it full.</li>
<li>
Windows MME - pa_win_wmme - Spawns an actual Win32 thread. Writes blocks
of data to the HW device and waits for events that signal buffer completion.</li>
<li>
Linux OSS - pa_linux - Spawns a real thread that writes to the "/dev/dsp"
stream using blocking I/O calls.</li>
</ul>
When you write a new implementation, you will be using some code that is
in common with all implementations. This code is in the folder "pa_common".
It provides various functions such as parameter checking, error code to
text conversion, sample format conversion, clipping and dithering, etc.
<p>The code that you write will go into a separate folder called "pa_{os}_{api}".
For example, code specific to the DirectSound interface for Windows goes
in "pa_win_ds".
<h2>
Read Docs and Code</h2>
Famialiarize yourself with the system by reading the documentation provided.
here is a suggested order:
<ol>
<li>
User Programming <a href="pa_tutorial.html">Tutorial</a></li>
<li>
Header file "pa_common/portaudio.h" which defines API.</li>
<li>
Header file "pa_common/pa_host.h" for host dependant code. This definces
the routine you will need to provide.</li>
<li>
Shared code in "pa_common/pa_lib.c".</li>
<li>
Docs on Implementation of <a href="pa_impl_startstop.html">Start/Stop</a>
code.</li>
</ol>
<h2>
Implement&nbsp; Output to Default Device</h2>
Now we are ready to crank some code. For instant gratification, let's try
to play a sine wave.
<ol>
<li>
Link the test program "pa_tests/patest_sine.c" with the file "pa_lib.c"
and the implementation specific file you are creating.</li>
<li>
For now, just stub out the device query code and the audio input code.</li>
<li>
Modify PaHost_OpenStream() to open your default target device and get everything
setup.</li>
<li>
Modify PaHost_StartOutput() to start playing audio.</li>
<li>
Modify PaHost_StopOutput() to stop audio.</li>
<li>
Modify PaHost_CloseStream() to clean up. Free all memory that you allocated
in PaHost_OpenStream().</li>
<li>
Keep cranking until you can play a sine wave using "patest_sine.c".</li>
<li>
Once that works, try "patest_pink.c", "patest_clip.c", "patest_sine8.c".</li>
<li>
To test your Open and Close code, try "patest_many.c".</li>
<li>
Now test to make sure that the three modes of stopping are properly supported
by running "patest_stop.c".</li>
<li>
Test your implementation of time stamping with "patest_sync.c".</li>
</ol>
<h2>
Implement Device Queries</h2>
Now that output is working, lets implement the code for querying what devices
are available to the user. Run "pa_tests/pa_devs.c". It should print all
of the devices available and their characteristics.
<h2>
Implement Input</h2>
Implement audio input and test it with:
<ol>
<li>
patest_record.c - record in half duplex, play back as recorded.</li>
<li>
patest_wire.c - full duplex, copies input to output. Note that some HW
may not support full duplex.</li>
<li>
patest_fuzz.c - plug in your guitar and get a feel for why latency is an
important issue in computer music.</li>
<li>
paqa_devs.c - try to open every device and use it with every possible format</li>
</ol>
<h2>
Debugging Tools</h2>
You generally cannot use printf() calls to debug real-time processes because
they disturb the timing. Also calling printf() from your background thread
or interrupt could crash the machine. So PA includes a tool for capturing
events and storing the information while it is running. It then prints
the events when Pa_Terminate() is called.
<ol>
<li>
To enable trace mode, change TRACE_REALTIME_EVENTS in "pa_common/pa_trace.h"
from a (0) to a (1).</li>
<li>
Link with "pa_common/pa_trace.c".</li>
<li>
Add trace messages to your code by calling:<br>
<tt>&nbsp;&nbsp; void AddTraceMessage( char *msg, int data );<br>
for example<br>
&nbsp;&nbsp; AddTraceMessage("Pa_TimeSlice: past_NumCallbacks ", past->past_NumCallbacks
);</tt></li>
<li>
Run your program. You will get a dump of events at the end.</li>
<li>
You can leave the trace messages in your code. They will turn to NOOPs
when you change TRACE_REALTIME_EVENTS back to (0).</li>
</ol>
<h2>
Delivery</h2>
Please send your new code along with notes on the implementation back to
us at "<a href="http://www.portaudio.com">http://www.portaudio.com</a>".
We will review the implementation and post it with your name. If you had
to make any modifications to the code in "pa_common" or "pa_tests" <b>please</b>
send us those modifications and your notes. We will try to merge your changes
so that the "pa_common" code works with <b>all</b> implementations.
<p>If you have suggestions for how to make future implementations easier,
please let us know.
<br>THANKS!
<br>&nbsp;
</body>
</html>

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.75 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Internal docs. How a stream is started or stopped.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Implementation - Start/Stop</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Implementation</h1></center>
</td>
</tr>
</table></center>
<h2>
Starting and Stopping Streams</h2>
PortAudio is generally executed in two "threads". The foreground thread
is the application thread. The background "thread" may be implemented as
an actual thread, an interrupt handler, or a callback from a timer thread.
<p>There are three ways that PortAudio can stop a stream. In each case
we look at the sequence of events and the messages sent between the two
threads. The following variables are contained in the internalPortAudioStream.
<blockquote><tt>int&nbsp;&nbsp; past_IsActive;&nbsp;&nbsp;&nbsp;&nbsp;
/* Background is still playing. */</tt>
<br><tt>int&nbsp;&nbsp; past_StopSoon;&nbsp;&nbsp;&nbsp;&nbsp; /* Stop
when last buffer done. */</tt>
<br><tt>int&nbsp;&nbsp; past_StopNow;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /*
Stop IMMEDIATELY. */</tt></blockquote>
<h3>
Pa_AbortStream()</h3>
This function causes the background thread to terminate as soon as possible
and audio I/O to stop abruptly.
<br>&nbsp;
<table BORDER COLS=2 WIDTH="60%" >
<tr>
<td><b>Foreground Thread</b></td>
<td><b>Background Thread</b></td>
</tr>
<tr>
<td>sets <tt>StopNow</tt></td>
<td></td>
</tr>
<tr>
<td></td>
<td>sees <tt>StopNow</tt>,&nbsp;</td>
</tr>
<tr>
<td></td>
<td>clears IsActive, stops thread</td>
</tr>
<tr>
<td>waits for thread to exit</td>
<td></td>
</tr>
<tr>
<td>turns off audio I/O</td>
<td></td>
</tr>
</table>
<h3>
Pa_StopStream()</h3>
This function stops the user callback function from being called and then
waits for all audio data written to the output buffer to be played. In
a system with very low latency, you may not hear any difference between
<br>&nbsp;
<table BORDER COLS=2 WIDTH="60%" >
<tr>
<td><b>Foreground Thread</b></td>
<td><b>Background Thread</b></td>
</tr>
<tr>
<td>sets StopSoon</td>
<td></td>
</tr>
<tr>
<td></td>
<td>stops calling user callback</td>
</tr>
<tr>
<td></td>
<td>continues until output buffer empty</td>
</tr>
<tr>
<td></td>
<td>clears IsActive, stops thread</td>
</tr>
<tr>
<td>waits for thread to exit</td>
<td></td>
</tr>
<tr>
<td>turns off audio I/O</td>
<td></td>
</tr>
</table>
<h3>
User callback returns one.</h3>
If the user callback returns one then the user callback function will no
longer be called. Audio output will continue until all audio data written
to the output buffer has been played. Then the audio I/O is stopped, the
background thread terminates, and the stream becomes inactive.
<br>&nbsp;
<table BORDER COLS=2 WIDTH="60%" >
<tr>
<td><b>Foreground Thread</b></td>
<td><b>Background Thread</b></td>
</tr>
<tr>
<td></td>
<td>callback returns 1</td>
</tr>
<tr>
<td></td>
<td>sets StopSoon</td>
</tr>
<tr>
<td></td>
<td>stops calling user callback</td>
</tr>
<tr>
<td></td>
<td>continues until output buffer empty</td>
</tr>
<tr>
<td></td>
<td>clears IsActive, stops thread</td>
</tr>
<tr>
<td>waits for thread to exit</td>
<td></td>
</tr>
<tr>
<td>turns off audio I/O</td>
<td></td>
</tr>
</table>
<br>&nbsp;
</body>
</html>

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Compiling for ASIO (Windows or Macintosh)</h2>
<blockquote>ASIO is a low latency audio API from Steinberg. To compile
an ASIO application, you must first <a href="http://www.steinberg.net/developers/ASIO2SDKAbout.phtml">download
the ASIO SDK</a> from Steinberg. You also need to obtain ASIO drivers for
your audio device.
<p>Note: I am using '/' as a file separator below. On Macintosh replace
'/' with ':'. On Windows, replace '/' with '\'.
<p>&nbsp;To use ASIO with the PortAudio library add the following source
files to your project:
<blockquote>
<pre>pa_asio/pa_asio.cpp</pre>
</blockquote>
and also these files from the ASIO SDK:
<blockquote>
<pre>common/asio.cpp
host/asiodrivers.cpp
host/asiolist.cpp</pre>
</blockquote>
and add these directories to the path for include files
<blockquote>
<pre>asiosdk2/host/pc&nbsp;&nbsp; (for Windows)
asiosdk2/common
asiosdk2/host</pre>
</blockquote>
You may try compiling the "pa_tests/patest_saw.c" file first because it
is the simplest.</blockquote>
<font size=+2><a href="http://www.portaudio.com/">home</a> |
<a href="pa_tutorial.html">contents</a>
| <a href="pa_tut_over.html">previous</a> |&nbsp; <a href="pa_tut_callback.html">next</a></font>
</body>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Writing a Callback Function</h2>
<blockquote>To write a program using PortAudio, you must include the "portaudio.h"
include file. You may wish to read "<a href="portaudio_h.txt">portaudio.h</a>"
because it contains a complete description of the PortAudio functions and
constants.
<blockquote>
<pre>#include "portaudio.h"</pre>
</blockquote>
The next task is to write your custom callback function. It is a function
that is called by the PortAudio engine whenever it has captured audio data,
or when it needs more audio data for output.
<p>Your callback function is often called by an interrupt, or low level
process so you should not do any complex system activities like allocating
memory, or reading or writing files, or printf(). Just crunch numbers and
generate audio signals. What is safe or not safe will vary from platform
to platform. On the Macintosh, for example, you can only call "interrupt
safe" routines. Also do not call any PortAudio functions in the callback
except for Pa_StreamTime() and Pa_GetCPULoad().
<p>Your callback function must return an int and accept the exact parameters
specified in this typedef:
<blockquote>
<pre>typedef int (PortAudioCallback)(
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; void *inputBuffer, void *outputBuffer,
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; unsigned long framesPerBuffer,
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PaTimestamp outTime, void *userData );</pre>
</blockquote>
Here is an example callback function from the test file "patests/patest_saw.c".
It calculates a simple left and right sawtooth signal and writes it to
the output buffer. Notice that in this example, the signals are of <tt>float</tt>
data type. The signals must be between -1.0 and +1.0. You can also use
16 bit integers or other formats which are specified during setup. You
can pass a pointer to your data structure through PortAudio which will
appear as <tt>userData</tt>.
<blockquote>
<pre>int patestCallback(&nbsp; void *inputBuffer, void *outputBuffer,
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; unsigned long framesPerBuffer,
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PaTimestamp outTime, void *userData )
{
&nbsp;&nbsp;&nbsp; unsigned int i;
/* Cast data passed through stream to our structure type. */
&nbsp;&nbsp;&nbsp; paTestData *data = (paTestData*)userData;
&nbsp;&nbsp;&nbsp; float *out = (float*)outputBuffer;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp; for( i=0; i&lt;framesPerBuffer; i++ )
&nbsp;&nbsp;&nbsp; {
&nbsp;&nbsp;&nbsp; /* Stereo channels are interleaved. */
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; *out++ = data->left_phase;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* left */
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; *out++ = data->right_phase;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* right */
&nbsp;&nbsp;&nbsp; /* Generate simple sawtooth phaser that ranges between -1.0 and 1.0. */
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; data->left_phase += 0.01f;
&nbsp;&nbsp;&nbsp; /* When signal reaches top, drop back down. */
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; if( data->left_phase >= 1.0f ) data->left_phase -= 2.0f;
&nbsp;&nbsp;&nbsp; /* higher pitch so we can distinguish left and right. */
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; data->right_phase += 0.03f;&nbsp;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; if( data->right_phase >= 1.0f ) data->right_phase -= 2.0f;
&nbsp;&nbsp;&nbsp; }
&nbsp;&nbsp;&nbsp; return 0;
}</pre>
</blockquote>
</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.75 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Querying for Available Devices</h2>
<blockquote>There are often several different audio devices available in
a computer with different capabilities. They can differ in the sample rates
supported, bit widths, etc. PortAudio provides a simple way to query for
the available devices, and then pass the selected device to Pa_OpenStream().
For an example, see the file "pa_tests/pa_devs.c".
<p>To determine the number of devices:
<blockquote>
<pre>numDevices = Pa_CountDevices();</pre>
</blockquote>
You can then query each device in turn by calling Pa_GetDeviceInfo() with
an index.
<blockquote>
<pre>for( i=0; i&lt;numDevices; i++ ) {
&nbsp;&nbsp;&nbsp;&nbsp; pdi = Pa_GetDeviceInfo( i );</pre>
</blockquote>
It will return a pointer to a <tt>PaDeviceInfo</tt> structure which is
defined as:
<blockquote>
<pre>typedef struct{
&nbsp;&nbsp;&nbsp; int structVersion;&nbsp;
&nbsp;&nbsp;&nbsp; const char *name;
&nbsp;&nbsp;&nbsp; int maxInputChannels;
&nbsp;&nbsp;&nbsp; int maxOutputChannels;
/* Number of discrete rates, or -1 if range supported. */
&nbsp;&nbsp;&nbsp; int numSampleRates;
/* Array of supported sample rates, or {min,max} if range supported. */
&nbsp;&nbsp;&nbsp; const double *sampleRates;
&nbsp;&nbsp;&nbsp; PaSampleFormat nativeSampleFormat;
}PaDeviceInfo;</pre>
</blockquote>
If the device supports a continuous range of sample rates, then numSampleRates
will equal -1, and the sampleRates array will have two values, the minimum&nbsp;
and maximum rate.
<p>The device information is allocated by Pa_Initialize() and freed by
Pa_Terminate() so you do not have to free() the structure returned by Pa_GetDeviceInfo().</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.73 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Exploring PortAudio</h2>
<blockquote>Now that you have a good idea of how PortAudio works, you can
try out the test programs.
<ul>
<li>
For an example of playing a sine wave, see "pa_tests/patest_sine.c".</li>
<li>
For an example of recording and playing back a sound, see&nbsp; "pa_tests/patest_record.c".</li>
</ul>
I also encourage you to examine the source for the PortAudio libraries.
If you have suggestions on ways to improve them, please let us know. if
you want to implement PortAudio on a new platform, please let us know as
well so we can coordinate people's efforts.</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.73 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Initializing PortAudio</h2>
<blockquote>Before making any other calls to PortAudio, you must call <tt>Pa_Initialize</tt>().
This will trigger a scan of available devices which can be queried later.
Like most PA functions, it will return a result of type <tt>paError</tt>.
If the result is not <tt>paNoError</tt>, then an error has occurred.
<blockquote>
<pre>err = Pa_Initialize();
if( err != paNoError ) goto error;</pre>
</blockquote>
You can get a text message that explains the error message by passing it
to
<blockquote>
<pre>printf(&nbsp; "PortAudio error: %s\n", Pa_GetErrorText( err ) );</pre>
</blockquote>
</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Compiling for Macintosh</h2>
<blockquote>To compile a Macintosh application with the PortAudio library,
add the following source files to your project:
<blockquote>
<pre>pa_mac:pa_mac.c
pa_common:pa_lib.c
pa_common:portaudio.h
pa_common:pa_host.h</pre>
</blockquote>
Also add the Apple <b>SoundLib</b> to your project.
<p>You may try compiling the "pa_tests:patest_saw.c" file first because
it is the simplest.</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.73 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Opening a Stream using Defaults</h2>
<blockquote>The next step is to open a stream which is similar to opening
a file. You can specify whether you want audio input and/or output, how
many channels, the data format, sample rate, etc. There are two calls for
opening streams, <tt>Pa_OpenStream</tt>() and <tt>Pa_OpenDefaultStream</tt>().
<p><tt>Pa_OpenStream()</tt> takes extra&nbsp; parameters which give you
more control. You can normally just use <tt>Pa_OpenDefaultStream</tt>()
which just calls <tt>Pa_OpenStream()</tt> <tt>with</tt> some reasonable
default values.&nbsp; Let's open a stream for stereo output, using floating
point data, at 44100 Hz.
<blockquote>
<pre>err = Pa_OpenDefaultStream(
&nbsp;&nbsp;&nbsp; &amp;stream,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* passes back stream pointer */
&nbsp;&nbsp;&nbsp; 0,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* no input channels */
&nbsp;&nbsp;&nbsp; 2,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* stereo output */
&nbsp;&nbsp;&nbsp; paFloat32,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* 32 bit floating point output */
&nbsp;&nbsp;&nbsp; 44100,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* sample rate */
&nbsp;&nbsp;&nbsp; 256,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* frames per buffer */
&nbsp;&nbsp;&nbsp; 0,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* number of buffers, if zero then use default minimum */
&nbsp;&nbsp;&nbsp; patestCallback, /* specify our custom callback */
&nbsp;&nbsp;&nbsp; &amp;data );&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /* pass our data through to callback */</pre>
</blockquote>
If you want to use 16 bit integer data, pass <tt>paInt16</tt> instead of
<tt>paFloat32</tt>.</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.75 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Compiling for Linux</h2>
<blockquote>[Skip this page if you are not using Linux]
<p>We currently only support the OSS audio drivers for Linux. We hope to
someday support the newer ALSA drivers.
<ol>
<li>
cd to pa_linux_oss directory</li>
<li>
Edit the Makefile and uncomment one of the tests. You may try compiling
the "patest_sine.c" file first because it is very simple.</li>
<li>
gmake run</li>
</ol>
</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Overview of PortAudio</h2>
<blockquote>PortAudio is a library that provides streaming audio input
and output. It is a cross-platform API (Application Programming Interface)
that works on Windows and Macintosh, and perhaps other platforms by the
time you read this. This means that you can write a simple 'C' program
to process or generate an audio signal, and that program can run on several
different computers just by recompiling the source code.
<p>Here are the steps to writing a PortAudio application:
<ol>
<li>
Write a callback function that will be called by PortAudio when audio processing
is needed.</li>
<li>
Initialize the PA library and open a stream for audio I/O.</li>
<li>
Start the stream. Your callback function will be now be called repeatedly
by PA in the background.</li>
<li>
In your callback you can read audio data from the inputBuffer and/or write
data to the outputBuffer.</li>
<li>
Stop the stream by returning 1 from your callback, or by calling a stop
function.</li>
<li>
Close the stream and terminate the library.</li>
</ol>
</blockquote>
<blockquote>There is also <a href="pa_tut_rw.html">another interface</a>
provided that allows you to generate audio in the foreground. You then
simply write data to the stream and the tool will not return until it is
ready to accept more data. This interface is simpler to use but is usually
not preferred for large applications because it requires that you launch
a thread to perform the synthesis. Launching a thread may be difficult
on non-multi-tasking systems such as the Macintosh prior to MacOS X.
<p>Let's continue by building a simple application that will play a sawtooth
wave.
<p>Please select the page for the implementation you would like to use:
<ul>
<li>
<a href="pa_tut_pc.html">Windows (WMME or DirectSound)</a></li>
<li>
<a href="pa_tut_mac.html">Macintosh SoundManager</a></li>
<li>
<a href="pa_tut_asio.html">ASIO on Windows or Macintosh</a></li>
<li>
<a href="pa_tut_oss.html">Unix OSS</a></li>
</ul>
</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Compiling for Windows (WMME or DirectSound)</h2>
<blockquote>To compile PortAudio for Windows, you can choose between two
options. One implementation uses the DirectSound API. The other uses the
Windows MultiMedia Extensions API (aka WMME or WAVE).
<p>Some advantages of using DirectSound are that DirectSound may have lower
latency than WMME, and supports effects processing plugins. But one disadvantage
is that DirectSound is not installed on all PCs, and is not well supported
under Windows NT.
<p>For either implementation add the following source files to your project:
<blockquote>
<pre>pa_common\pa_lib.c
pa_common\portaudio.h
pa_common\pa_host.h</pre>
<pre>Link with the system library: winmm.lib</pre>
</blockquote>
<b>WMME</b> - To use the WMME implementation, add the following source
files to your project:
<blockquote><tt>pa_win_wmme/pa_win_wmme.c</tt></blockquote>
<b>DirectSound</b> - If you want to use the DirectSound implementation
of PortAudio then you must have a recent copy of the free
<a href="http://www.microsoft.com/directx/download.asp">DirectX</a>
SDK for Developers from Microsoft installed on your computer. To compile
an application add the following source files to your project:
<blockquote>
<pre>pa_win_ds\dsound_wrapper.c
pa_win_ds\pa_dsound.c</pre>
</blockquote>
<blockquote>
<pre>Link with the system library: dsound.lib</pre>
</blockquote>
You may try compiling the "pa_tests\patest_saw.c" file first because it
is the simplest.</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.73 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Starting and Stopping a Stream</h2>
<blockquote>The stream will not start running until you call Pa_StartStream().
Then it will start calling your callback function to perform the audio
processing.
<blockquote>
<pre>err = Pa_StartStream( stream );
if( err != paNoError ) goto error;</pre>
</blockquote>
At this point, audio is being generated. You can communicate to your callback
routine through the data structure you passed in on the open call, or through
global variables, or using other interprocess communication techniques.
Please be aware that your callback function may be called at interrupt
time when your foreground process is least expecting it. So avoid sharing
complex data structures that are easily corrupted like double linked lists.
<p>In many of the tests we simply sleep for a few seconds so we can hear
the sound. This is easy to do with Pa_Sleep() which will sleep for some
number of milliseconds. Do not rely on this function for accurate scheduling.
it is mostly for writing examples.
<blockquote>
<pre>/* Sleep for several seconds. */
Pa_Sleep(NUM_SECONDS*1000);</pre>
</blockquote>
When you are through, you can stop the stream from the foreground.
<blockquote>
<pre>err = Pa_StopStream( stream );
if( err != paNoError ) goto error;</pre>
</blockquote>
You can also stop the stream by returning 1 from your custom callback function.</blockquote>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.75 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Blocking Read/Write Functions</h2>
<blockquote>[Note: These functions are not part of the official PortAudio
API. They are simply built on top of PortAudio as an extra utility. Also
note that they are under evaluation and their definition may change.]
<p>There are two fundamentally different ways to design an audio API. One
is to use callback functions the way we have already shown. The callback
function operates under an interrupt or background thread This leaves the
foreground application free to do other things while the audio just runs
in the background. But this can sometimes be awkward.
<p>So we have provided an alternative technique that lets a program generate
audio in the foreground and then just write it to the audio stream as if
it was a file. If there is not enough room in the audio buffer for more
data, then the write function will just block until more room is available.
This can make it very easy to write an audio example. To use this tool,
you must add the files "pablio/pablio.c" and "pablio/ringbuffer.c" to your
project. You must also:
<blockquote>
<pre>#include "pablio.h"</pre>
</blockquote>
Here is a short excerpt of a program that opens a stream for input and
output. It then read a pair of samples at a time from input, and writes
them to output, in a loop. The complete example can be found in "pablio/test_rw.c".
<blockquote>
<pre>&nbsp;&nbsp;&nbsp; SAMPLE&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; samples[2];
&nbsp;&nbsp;&nbsp; PaError&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; err;
&nbsp;&nbsp;&nbsp; PABLIO_Stream&nbsp; *aStream;
/* Open simplified blocking I/O layer on top of PortAudio. */
&nbsp;&nbsp;&nbsp; err = OpenAudioStream( &amp;rwbl, SAMPLE_RATE, paFloat32,
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (PABLIO_READ_WRITE | PABLIO_STEREO) );
&nbsp;&nbsp;&nbsp; if( err != paNoError ) goto error;
/* Process samples in the foreground. */
&nbsp;&nbsp;&nbsp; for( i=0; i&lt;(NUM_SECONDS * SAMPLE_RATE); i++ )
&nbsp;&nbsp;&nbsp; {
&nbsp;&nbsp;&nbsp; /* Read one frame of data into sample array from audio input. */
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ReadAudioStream( aStream, samples, 1 );
&nbsp;&nbsp;&nbsp; /*
&nbsp;&nbsp;&nbsp; ** At this point you could process the data in samples[0] and samples[1],
&nbsp;&nbsp;&nbsp; ** and write the result back to the same samples array.
&nbsp;&nbsp;&nbsp; */
&nbsp;&nbsp;&nbsp; /* Write that same frame of data to output. */
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; WriteAudioStream( aStream, samples, 1 );
&nbsp;&nbsp;&nbsp; }
&nbsp;&nbsp;&nbsp; CloseAudioStream( aStream );</pre>
</blockquote>
</blockquote>
<font size=+2><a href="http://www.portaudio.com/">home</a> |
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.73 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Terminating PortAudio</h2>
<blockquote>You can start and stop a stream as many times as you like.
But when you are done using it, you should close it by calling:</blockquote>
<blockquote>
<blockquote>
<pre>err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;</pre>
</blockquote>
Then when you are done using PortAudio, you should terminate the whole
system by calling:
<blockquote>
<pre>Pa_Terminate();</pre>
</blockquote>
That's basically it. You can now write an audio program in 'C' that will
run on multiple platforms, for example PCs and Macintosh.
<p>In the rest of the tutorial we will look at some additional utility
functions, and a different way of using PortAudio that does not require
the use of a callback function.</blockquote>
<font size=+2><a href="http://www.portaudio.com/">home</a> | <a href="pa_tutorial.html">contents</a>
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.75 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<h2>
Utility Functions</h2>
<blockquote>Here are several more functions that are not critical, but
may be handy when using PortAudio.
<p>Pa_StreamActive() returns one when the stream in playing audio, zero
when not playing, or a negative error number if the stream is invalid.
The stream is active between calls to Pa_StartStream() and Pa_StopStream(),
but may also become inactive if the callback returns a non-zero value.
In the latter case, the stream is considered inactive after the last buffer
has finished playing.
<blockquote>
<pre>PaError Pa_StreamActive( PortAudioStream *stream );</pre>
</blockquote>
Pa_StreamTime() returns the number of samples that have been generated.
PaTimeStamp is a double precision number which is a convenient way to pass
big numbers around even though we only need integers.
<blockquote>
<pre>PaTimestamp Pa_StreamTime( PortAudioStream *stream );</pre>
</blockquote>
The "CPU Load" is a fraction of total CPU time consumed by the stream's
audio processing. A value of 0.5 would imply that PortAudio and the sound
generating callback was consuming roughly 50% of the available CPU time.
This function may be called from the callback function or the application.
<blockquote>
<pre>double Pa_GetCPULoad( PortAudioStream* stream );</pre>
</blockquote>
</blockquote>
<font size=+2><a href="http://www.portaudio.com/">home</a> |
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Tutorial</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio Tutorial</h1></center>
</td>
</tr>
</table></center>
<p>Copyright 2000 Phil Burk and Ross Bencina
<h2>
Table of Contents</h2>
<blockquote><a href="pa_tut_over.html">Overview of PortAudio</a>
<br><a href="pa_tut_mac.html">Compiling for Macintosh</a>
<br><a href="pa_tut_pc.html">Compiling for Windows (DirectSound and WMME)</a>
<br><a href="pa_tut_asio.html">Compiling for ASIO on Windows or Macintosh</a>
<br><a href="pa_tut_linux.html">Compiling for Unix OSS</a>
<br><a href="pa_tut_callback.html">Writing a Callback Function</a>
<br><a href="pa_tut_init.html">Initializing PortAudio</a>
<br><a href="pa_tut_open.html">Opening a Stream using Defaults</a>
<br><a href="pa_tut_run.html">Starting and Stopping a Stream</a>
<br><a href="pa_tut_term.html">Cleaning Up</a>
<br><a href="pa_tut_util.html">Utilities</a>
<br><a href="pa_tut_devs.html">Querying for Devices</a>
<br><a href="pa_tut_rw.html">Blocking Read/Write Functions</a>
<br><a href="pa_tut_explore.html">Exploring the PortAudio Package</a></blockquote>
<font size=+2><a href="http://www.portaudio.com/">home</a> | contents |
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#ifndef PORT_AUDIO_H
#define PORT_AUDIO_H
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
/*
* PortAudio Portable Real-Time Audio Library
* PortAudio API Header File
* Latest version available at: 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.
*
*/
typedef int PaError;
typedef enum {
paNoError = 0,
paHostError = -10000,
paInvalidChannelCount,
paInvalidSampleRate,
paInvalidDeviceId,
paInvalidFlag,
paSampleFormatNotSupported,
paBadIODeviceCombination,
paInsufficientMemory,
paBufferTooBig,
paBufferTooSmall,
paNullCallback,
paBadStreamPtr,
paTimedOut,
paInternalError
} PaErrorNum;
/*
Pa_Initialize() is the library initialisation function - call this before
using the library.
*/
PaError Pa_Initialize( void );
/*
Pa_Terminate() is the library termination function - call this after
using the library.
*/
PaError Pa_Terminate( void );
/*
Return host specific error.
This can be called after receiving a paHostError.
*/
long Pa_GetHostError( void );
/*
Translate the error number into a human readable message.
*/
const char *Pa_GetErrorText( PaError errnum );
/*
Sample formats
These are formats used to pass sound data between the callback and the
stream. Each device has a "native" format which may be used when optimum
efficiency or control over conversion is required.
Formats marked "always available" are supported (emulated) by all devices.
The floating point representation uses +1.0 and -1.0 as the respective
maximum and minimum.
*/
typedef unsigned long PaSampleFormat;
#define paFloat32 ((PaSampleFormat) (1<<0)) /*always available*/
#define paInt16 ((PaSampleFormat) (1<<1)) /*always available*/
#define paInt32 ((PaSampleFormat) (1<<2)) /*always available*/
#define paInt24 ((PaSampleFormat) (1<<3))
#define paPackedInt24 ((PaSampleFormat) (1<<4))
#define paInt8 ((PaSampleFormat) (1<<5))
#define paUInt8 ((PaSampleFormat) (1<<6)) /* unsigned 8 bit, 128 is "ground" */
#define paCustomFormat ((PaSampleFormat) (1<<16))
/*
Device enumeration mechanism.
Device ids range from 0 to Pa_CountDevices()-1.
Devices may support input, output or both. Device 0 is always the "default"
device and should support at least stereo in and out if that is available
on the taget platform _even_ if this involves kludging an input/output
device on platforms that usually separate input from output. Other platform
specific devices are specified by positive device ids.
*/
typedef int PaDeviceID;
#define paNoDevice -1
typedef struct{
int structVersion;
const char *name;
int maxInputChannels;
int maxOutputChannels;
/* Number of discrete rates, or -1 if range supported. */
int numSampleRates;
/* Array of supported sample rates, or {min,max} if range supported. */
const double *sampleRates;
PaSampleFormat nativeSampleFormats;
} PaDeviceInfo;
int Pa_CountDevices();
/*
Pa_GetDefaultInputDeviceID(), Pa_GetDefaultOutputDeviceID()
Return the default device ID or paNoDevice if there is no devices.
The result can be passed to Pa_OpenStream().
On the PC, the user can specify a default device by
setting an environment variable. For example, to use device #1.
set PA_RECOMMENDED_OUTPUT_DEVICE=1
The user should first determine the available device ID by using
the supplied application "pa_devs".
*/
PaDeviceID Pa_GetDefaultInputDeviceID( void );
PaDeviceID Pa_GetDefaultOutputDeviceID( void );
/*
PaTimestamp is used to represent a continuous sample clock with arbitrary
start time useful for syncronisation. The type is used in the outTime
argument to the callback function and the result of Pa_StreamTime()
*/
typedef double PaTimestamp;
/*
Pa_GetDeviceInfo() returns a pointer to an immutable PaDeviceInfo structure
referring to the device specified by id.
If id is out of range the function returns NULL.
The returned structure is owned by the PortAudio implementation and must
not be manipulated or freed. The pointer is guaranteed to be valid until
between calls to Pa_Initialize() and Pa_Terminate().
*/
const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID devID );
/*
PortAudioCallback is implemented by clients of the portable audio api.
inputBuffer and outputBuffer are arrays of interleaved samples,
the format, packing and number of channels used by the buffers are
determined by parameters to Pa_OpenStream() (see below).
framesPerBuffer is the number of sample frames to be processed by the callback.
outTime is the time in samples when the buffer(s) processed by
this callback will begin being played at the audio output.
See also Pa_StreamTime()
userData is the value of a user supplied pointer passed to Pa_OpenStream()
intended for storing synthesis data etc.
return value:
The callback can return a nonzero value to stop the stream. This may be
useful in applications such as soundfile players where a specific duration
of output is required. However, it is not necessary to utilise this mechanism
as StopStream() will also terminate the stream. A callback returning a
nonzero value must fill the entire outputBuffer.
NOTE: None of the other stream functions may be called from within the
callback function except for Pa_GetCPULoad().
*/
typedef int (PortAudioCallback)(
void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData );
/*
Stream flags
These flags may be supplied (ored together) in the streamFlags argument to
the Pa_OpenStream() function.
[ suggestions? ]
*/
#define paNoFlag (0)
#define paClipOff (1<<0) /* disable defult clipping of out of range samples */
#define paDitherOff (1<<1) /* disable default dithering */
#define paPlatformSpecificFlags (0x00010000)
typedef unsigned long PaStreamFlags;
/*
A single PortAudioStream provides multiple channels of real-time
input and output audio streaming to a client application.
Pointers to PortAudioStream objects are passed between PortAudio functions.
*/
typedef void PortAudioStream;
#define PaStream PortAudioStream
/*
Pa_OpenStream() opens a stream for either input, output or both.
stream is the address of a PortAudioStream pointer which will receive
a pointer to the newly opened stream.
inputDevice is the id of the device used for input (see PaDeviceID above.)
inputDevice may be paNoDevice to indicate that an input device is not required.
numInputChannels is the number of channels of sound to be delivered to the
callback. It can range from 1 to the value of maxInputChannels in the
device input record for the device specified in the inputDevice parameter.
If inputDevice is paNoDevice numInputChannels is ignored.
inputSampleFormat is the format of inputBuffer provided to the callback
function. inputSampleFormat may be any of the formats described by the
PaSampleFormat enumeration (see above). PortAudio guarantees support for
the sound devices native formats (nativeSampleFormats in the device info
record) and additionally 16 and 32 bit integer and 32 bit floating point
formats. Support for other formats is implementation defined.
inputDriverInfo is a pointer to an optional driver specific data structure
containing additional information for device setup or stream processing.
inputDriverInfo is never required for correct operation. If not used
inputDriverInfo should be NULL.
outputDevice is the id of the device used for output (see PaDeviceID above.)
outputDevice may be paNoDevice to indicate that an output device is not required.
numOutputChannels is the number of channels of sound to be supplied by the
callback. See the definition of numInputChannels above for more details.
outputSampleFormat is the sample format of the outputBuffer filled by the
callback function. See the definition of inputSampleFormat above for more
details.
outputDriverInfo is a pointer to an optional driver specific data structure
containing additional information for device setup or stream processing.
outputDriverInfo is never required for correct operation. If not used
outputDriverInfo should be NULL.
sampleRate is the desired sampleRate for input and output
framesPerBuffer is the length in sample frames of all internal sample buffers
used for communication with platform specific audio routines. Wherever
possible this corresponds to the framesPerBuffer parameter passed to the
callback function.
numberOfBuffers is the number of buffers used for
multibuffered communication with the platform specific audio
routines. If you pass zero, then an optimum value will be
chosen for you internally. This parameter is provided only
as a guide - and does not imply that an implementation must
use multibuffered i/o when reliable double buffering is
available (such as SndPlayDoubleBuffer() on the Macintosh.)
streamFlags may contain a combination of flags ORed together.
These flags modify the behavior of the
streaming process. Some flags may only be relevant to certain buffer formats.
callback is a pointer to a client supplied function that is responsible
for processing and filling input and output buffers (see above for details.)
userData is a client supplied pointer which is passed to the callback
function. It could for example, contain a pointer to instance data necessary
for processing the audio buffers.
return value:
Apon success Pa_OpenStream() returns PaNoError and places a pointer to a
valid PortAudioStream in the stream argument. The stream is inactive (stopped).
If a call to Pa_OpenStream() fails a nonzero error code is returned (see
PAError above) and the value of stream is invalid.
*/
PaError Pa_OpenStream( PortAudioStream** stream,
PaDeviceID inputDevice,
int numInputChannels,
PaSampleFormat inputSampleFormat,
void *inputDriverInfo,
PaDeviceID outputDevice,
int numOutputChannels,
PaSampleFormat outputSampleFormat,
void *outputDriverInfo,
double sampleRate,
unsigned long framesPerBuffer,
unsigned long numberOfBuffers,
PaStreamFlags streamFlags,
PortAudioCallback *callback,
void *userData );
/*
Pa_OpenDefaultStream() is a simplified version of Pa_OpenStream() that
opens the default input and/or ouput devices. Most parameters have
identical meaning to their Pa_OpenStream() counterparts, with the following
exceptions:
If either numInputChannels or numOutputChannels is 0 the respective device
is not opened (same as passing paNoDevice in the device arguments to Pa_OpenStream() )
sampleFormat applies to both the input and output buffers.
*/
PaError Pa_OpenDefaultStream( PortAudioStream** stream,
int numInputChannels,
int numOutputChannels,
PaSampleFormat sampleFormat,
double sampleRate,
unsigned long framesPerBuffer,
unsigned long numberOfBuffers,
PortAudioCallback *callback,
void *userData );
/*
Pa_CloseStream() closes an audio stream, flushing any pending buffers.
*/
PaError Pa_CloseStream( PortAudioStream* );
/*
Pa_StartStream() and Pa_StopStream() begin and terminate audio processing.
Pa_StopStream() waits until all pending audio buffers have been played.
Pa_AbortStream() stops playing immediately without waiting for pending
buffers to complete.
*/
PaError Pa_StartStream( PortAudioStream *stream );
PaError Pa_StopStream( PortAudioStream *stream );
PaError Pa_AbortStream( PortAudioStream *stream );
/*
Pa_StreamActive() returns one when the stream is playing audio,
zero when not playing, or a negative error number if the
stream is invalid.
The stream is active between calls to Pa_StartStream() and Pa_StopStream(),
but may also become inactive if the callback returns a non-zero value.
In the latter case, the stream is considered inactive after the last
buffer has finished playing.
*/
PaError Pa_StreamActive( PortAudioStream *stream );
/*
Pa_StreamTime() returns the current output time for the stream in samples.
This time may be used as a time reference (for example syncronising audio to
MIDI).
*/
PaTimestamp Pa_StreamTime( PortAudioStream *stream );
/*
The "CPU Load" is a fraction of total CPU time consumed by the
stream's audio processing.
A value of 0.5 would imply that PortAudio and the sound generating
callback was consuming roughly 50% of the available CPU time.
This function may be called from the callback function or the application.
*/
double Pa_GetCPULoad( PortAudioStream* stream );
/*
Use Pa_GetMinNumBuffers() to determine minimum number of buffers required for
the current host based on minimum latency.
On the PC, for the DirectSound implementation, latency can be optionally set
by user by setting an environment variable.
For example, to set latency to 200 msec, put:
set PA_MIN_LATENCY_MSEC=200
in the AUTOEXEC.BAT file and reboot.
If the environment variable is not set, then the latency will be determined
based on the OS. Windows NT has higher latency than Win95.
*/
int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate );
/*
Sleep for at least 'msec' milliseconds.
You may sleep longer than the requested time so don't rely
on this for accurate musical timing.
*/
void Pa_Sleep( long msec );
/*
Return size in bytes of a single sample in a given PaSampleFormat
or paSampleFormatNotSupported.
*/
PaError Pa_GetSampleSize( PaSampleFormat format );
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* PORT_AUDIO_H */

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="PortAudio is a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio Implementations for DirectSound</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio - Release Notes</h1></center>
</td>
</tr>
</table></center>
<p>Link to <a href="http://www.portaudio.com">PortAudio Home Page</a>
<h2>
<b>V17 - 10/15/01</b></h2>
<blockquote><b>Unix OSS</b>
<ul>
<li>
Set num channels back to two after device query for ALSA. This fixed a
bug in V16 that sometimes caused a failure when querying for the sample
rates. Thanks Stweart Greenhill.</li>
</ul>
</blockquote>
<blockquote>
<h4>
<b>Macintosh Sound Manager</b></h4>
<ul>
<li>
Use NewSndCallBackUPP() for CARBON compatibility.</li>
</ul>
</blockquote>
<h2>
<b>V16 - 9/27/01</b></h2>
<blockquote><b>Added Alpha implementations for ASIO, SGI, and BeOS!</b>
<br>&nbsp;
<li>
CPULoad is now calculated based on the time spent to generate a known number
of frames. This is more accurate than a simple percentage of real-time.
Implemented in pa_unix_oss, pa_win_wmme and pa_win_ds.</li>
<li>
Fix dither and shift for recording PaUInt8 format data.</li>
<li>
Added "patest_maxsines.c" which tests <tt>Pa_GetCPULoad().</tt></li>
</blockquote>
<blockquote>
<h4>
Windows WMME</h4>
<ul>
<li>
sDevicePtrs now allocated using <tt>GlobalAlloc()</tt>. This prevents a
crash in Pa_Terminate() on Win2000. Thanks Mike Berry for finding this.
Thanks Mike Berry.</li>
<li>
Pass process instead of thread to <tt>SetPriorityClass</tt>(). This fixes
a bug that caused the priority to not be increased. Thanks to Alberto di
Bene for spotting this.</li>
</ul>
<h4>
Windows DirectSound</h4>
<ul>
<li>
Casts for compiling with __MWERKS__ CodeWarrior.</li>
</ul>
<h4>
UNIX OSS</h4>
<ul>
<li>
Derived from Linux OSS implementation.</li>
<li>
Numerous patches from Heiko Purnhagen, Stephen Brandon, etc.</li>
<li>
Improved query mechanism which often bailed out unnecessarily.</li>
<li>
Removed sNumDevices and potential related bugs,</li>
<li>
Use <tt>getenv("PA_MIN_LATENCY_MSEC")</tt> in code to set desired latency.
User can set by entering:</li>
<br>&nbsp;&nbsp;&nbsp; <tt>export PA_MIN_LATENCY_MSEC=40</tt></ul>
<h4>
Macintosh Sound Manager</h4>
<ul>
<li>
Pass unused event to WaitNextEvent instead of NULL to prevent Mac OSX crash.
Thanks Dominic Mazzoni.</li>
<br>&nbsp;</ul>
</blockquote>
<h2>
<b>V15 - 5/29/01</b></h2>
<blockquote>
<ul>
<li>
<b>New Linux OSS Beta</b></li>
</ul>
<h4>
Windows WMME</h4>
<ul>
<li>
&nbsp;sDevicePtrs now allocated based on sizeof(pointer). Was allocating
too much space.</li>
<li>
&nbsp;Check for excessive numbers of channels. Some drivers reported bogus
numbers.</li>
<li>
Apply Mike Berry's changes for CodeWarrior on PC including condition including
of memory.h, and explicit typecasting on memory allocation.</li>
</ul>
<h4>
Macintosh Sound Manager</h4>
<ul>
<li>
ScanInputDevices was setting sDefaultOutputDeviceID instead of sDefaultInputDeviceID.</li>
<li>
Device Scan was crashing for anything other than siBadSoundInDevice, but
some Macs may return other errors! Caused failure to init on some G4s under
OS9.</li>
<li>
Fix TIMEOUT in record mode.</li>
<li>
Change CARBON_COMPATIBLE to TARGET_API_MAC_CARBON</li>
</ul>
</blockquote>
<h2>
<b>V14 - 2/6/01</b></h2>
<blockquote>
<ul>
<li>
Added implementation for Windows MultiMedia Extensions (WMME) by Ross and
Phil</li>
<li>
Changed Pa_StopStream() so that it waits for the buffers to drain.</li>
<li>
Added Pa_AbortStream() that stops immediately without waiting.</li>
<li>
Added new test: patest_stop.c to test above two mods.</li>
<li>
Fixed Pa_StreamTime() so that it returns current play position instead
of the write position. Added "patest_sync.c" to demo audio/video sync.</li>
<li>
Improved stability of Macintosh implementation. Added timeouts to prevent
hangs.</li>
<li>
Added Pa_GetSampleSize( PaSampleFormat format );</li>
<li>
Changes some "int"s to "long"s so that PA works properly on Macintosh which
often compiles using 16 bit ints.</li>
<li>
Added Implementation Guide</li>
</ul>
</blockquote>
<h2>
<b>V12 - 1/9/01</b></h2>
<blockquote>
<ul>
<li>
Mac now scans for and queries all devices. But it does not yet support
selecting any other than the default device.</li>
<li>
Blocking I/O calls renamed to separate them from the PortAudio API.</li>
<li>
Cleaned up indentation problems with tabs versus spaces.</li>
<li>
Now attempts to correct bogus sample rate info returned from DirectSound
device queries.</li>
</ul>
</blockquote>
</body>
</html>

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]">
<meta name="Author" content="Phil Burk">
<meta name="Description" content="PortAudio is a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function.">
<meta name="KeyWords" content="audio, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,">
<title>PortAudio - Cross-Platform Audio API</title>
</head>
<body>
&nbsp;
<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" >
<tr>
<td>
<center>
<h1>
PortAudio - Portable Audio Library</h1></center>
</td>
</tr>
</table></center>
<p>Last updated 5/6/02.
<p>PortAudio is a cross platform, <a href="#License">open-source</a>, audio
I/O library proposed by <b>Ross Bencina</b> to the <a href="http://shoko.calarts.edu/~glmrboy/musicdsp/music-dsp.html">music-dsp</a>
mailing list. It lets you write simple audio programs in 'C' that will
compile and run on <b>Windows, Macintosh, Unix, BeOS</b>. PortAudio is
intended to promote the exchange of audio synthesis software between developers
on different platforms.
<p>For complete information on PortAudio and to download the latest releases,
please visit "<b><font size=+2><a href="http://www.portaudio.com">http://www.portaudio.com</a></font></b>".
<br>&nbsp;
<br>&nbsp;
<center>
<h2>
<b><a href="docs/index.html">Click here for Documentation</a></b></h2></center>
<h2>
<b><font size=+2></font></b></h2>
<h2>
<b><font size=+2>Contacts and E-Mail List</font></b></h2>
<ul>
<li>
If you are using or implementing PortAudio then please join the <b><font size=+1><a href="http://techweb.rfa.org/mailman/listinfo/portaudio">PortAudio
mail list</a></font><font size=+2> </font></b>generously administered by
<b>Bill
Eldridge</b>.</li>
<li>
If you find bugs in one of these implementations, or have suggestions,
please e-mail them to <a href="mailto:philburk@softsynth.com">Phil Burk</a>.</li>
<li>
If you make improvements to the library, please send them to us so we can
incorporate the improvements.</li>
</ul>
<h2>
<a NAME="License"></a>License</h2>
PortAudio Portable Real-Time Audio Library
<br>Copyright (c) 1999-2000 Ross Bencina and Phil Burk
<p>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:
<ul>
<li>
The above copyright notice and this permission notice shall be included
in all copies or substantial portions of the Software.</li>
<li>
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.</li>
</ul>
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 ON INFRINGEMENT.
<br>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.
<br>&nbsp;
</body>
</html>

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/pa_convert.c/1.2/Sun Mar 2 08:01:33 2003//
/pa_host.h/1.5/Sun Mar 2 08:01:34 2003//
/pa_lib.c/1.6/Sun Mar 2 08:01:34 2003//
/pa_trace.c/1.5/Sun Mar 2 08:01:34 2003//
/pa_trace.h/1.5/Sun Mar 2 08:01:35 2003//
/portaudio.h/1.6/Sun Mar 2 08:01:35 2003//
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/*
* pa_conversions.c
* portaudio
*
* Created by Phil Burk on Mon Mar 18 2002.
*
*/
#include <stdio.h>
#include "portaudio.h"
#include "pa_host.h"
#define CLIP( val, min, max ) { val = ((val) < (min)) ? min : (((val) < (max)) ? (max) : (val)); }
/*************************************************************************/
static void PaConvert_Float32_Int16(
float *sourceBuffer, int sourceStride,
short *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
short samp = (short) (*sourceBuffer * (32767.0f));
*targetBuffer = samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_Int16_Clip(
float *sourceBuffer, int sourceStride,
short *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
long samp = (long) (*sourceBuffer * (32767.0f));
CLIP( samp, -0x8000, 0x7FFF );
*targetBuffer = (short) samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_Int16_ClipDither(
float *sourceBuffer, int sourceStride,
short *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
// use smaller scaler to prevent overflow when we add the dither
float dither = PaConvert_TriangularDither() * PA_DITHER_SCALE;
float dithered = (*sourceBuffer * (32766.0f)) + dither;
long samp = (long) dithered;
CLIP( samp, -0x8000, 0x7FFF );
*targetBuffer = (short) samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_Int16_Dither(
float *sourceBuffer, int sourceStride,
short *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
// use smaller scaler to prevent overflow when we add the dither
float dither = PaConvert_TriangularDither() * PA_DITHER_SCALE;
float dithered = (*sourceBuffer * (32766.0f)) + dither;
*targetBuffer = (short) dithered;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Int16_Float32(
short *sourceBuffer, int sourceStride,
float *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
float samp = *sourceBuffer * (1.0f / 32768.0f);
*targetBuffer = samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_Int8(
float *sourceBuffer, int sourceStride,
char *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
char samp = (char) (*sourceBuffer * (127.0));
*targetBuffer = samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_Int8_Clip(
float *sourceBuffer, int sourceStride,
char *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
long samp = (long) (*sourceBuffer * 127.0f);
CLIP( samp, -0x80, 0x7F );
*targetBuffer = (char) samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_Int8_ClipDither(
float *sourceBuffer, int sourceStride,
char *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
// use smaller scaler to prevent overflow when we add the dither
float dither = PaConvert_TriangularDither() * PA_DITHER_SCALE;
float dithered = (*sourceBuffer * (126.0f)) + dither;
long samp = (long) dithered;
CLIP( samp, -0x80, 0x7F );
*targetBuffer = (char) samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_Int8_Dither(
float *sourceBuffer, int sourceStride,
char *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
// use smaller scaler to prevent overflow when we add the dither
float dither = PaConvert_TriangularDither() * PA_DITHER_SCALE; //FIXME
float dithered = (*sourceBuffer * (126.0f)) + dither;
long samp = (long) dithered;
*targetBuffer = (char) samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Int8_Float32(
char *sourceBuffer, int sourceStride,
float *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
float samp = *sourceBuffer * (1.0f / 128.0f);
*targetBuffer = samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_UInt8(
float *sourceBuffer, int sourceStride,
unsigned char *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
unsigned char samp = (unsigned char)(128 + (*sourceBuffer * (127.0)));
*targetBuffer = samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_UInt8_Float32(
unsigned char *sourceBuffer, int sourceStride,
float *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
float samp = (*sourceBuffer - 128) * (1.0f / 128.0f);
*targetBuffer = samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_Int32(
float *sourceBuffer, int sourceStride,
long *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
int samp = (int) (*sourceBuffer * 0x7FFFFFFF);
*targetBuffer = samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Float32_Int32_Clip(
float *sourceBuffer, int sourceStride,
long *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
int samp;
float fs = *sourceBuffer;
CLIP( fs, -1.0, 0.999999 );
samp = (int) (*sourceBuffer * 0x7FFFFFFF);
*targetBuffer = samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static void PaConvert_Int32_Float32(
long *sourceBuffer, int sourceStride,
float *targetBuffer, int targetStride,
int numSamples )
{
int i;
for( i=0; i<numSamples; i++ )
{
float samp = *sourceBuffer * (1.0f / 0x7FFFFFFF);
*targetBuffer = samp;
sourceBuffer += sourceStride;
targetBuffer += targetStride;
}
}
/*************************************************************************/
static PortAudioConverter *PaConvert_SelectProc( PaSampleFormat sourceFormat,
PaSampleFormat targetFormat, int ifClip, int ifDither )
{
PortAudioConverter *proc = NULL;
switch( sourceFormat )
{
case paUInt8:
switch( targetFormat )
{
case paFloat32:
proc = (PortAudioConverter *) PaConvert_UInt8_Float32;
break;
default:
break;
}
break;
case paInt8:
switch( targetFormat )
{
case paFloat32:
proc = (PortAudioConverter *) PaConvert_Int8_Float32;
break;
default:
break;
}
break;
case paInt16:
switch( targetFormat )
{
case paFloat32:
proc = (PortAudioConverter *) PaConvert_Int16_Float32;
break;
default:
break;
}
break;
case paInt32:
switch( targetFormat )
{
case paFloat32:
proc = (PortAudioConverter *) PaConvert_Int32_Float32;
break;
default:
break;
}
break;
case paFloat32:
switch( targetFormat )
{
case paUInt8:
proc = (PortAudioConverter *) PaConvert_Float32_UInt8;
break;
case paInt8:
if( ifClip && ifDither ) proc = (PortAudioConverter *) PaConvert_Float32_Int8_ClipDither;
else if( ifClip ) proc = (PortAudioConverter *) PaConvert_Float32_Int8_Clip;
else if( ifDither ) proc = (PortAudioConverter *) PaConvert_Float32_Int8_Dither;
else proc = (PortAudioConverter *) PaConvert_Float32_Int8;
break;
case paInt16:
if( ifClip && ifDither ) proc = (PortAudioConverter *) PaConvert_Float32_Int16_ClipDither;
else if( ifClip ) proc = (PortAudioConverter *) PaConvert_Float32_Int16_Clip;
else if( ifDither ) proc = (PortAudioConverter *) PaConvert_Float32_Int16_Dither;
else proc = (PortAudioConverter *) PaConvert_Float32_Int16;
break;
case paInt32:
/* Don't bother dithering a 32 bit integer! */
if( ifClip ) proc = (PortAudioConverter *) PaConvert_Float32_Int32_Clip;
else proc = (PortAudioConverter *) PaConvert_Float32_Int32;
break;
default:
break;
}
break;
default:
break;
}
return proc;
}
/*************************************************************************/
PaError PaConvert_SetupInput( internalPortAudioStream *past,
PaSampleFormat nativeInputSampleFormat )
{
past->past_NativeInputSampleFormat = nativeInputSampleFormat;
past->past_InputConversionSourceStride = 1;
past->past_InputConversionTargetStride = 1;
if( nativeInputSampleFormat != past->past_InputSampleFormat )
{
int ifDither = (past->past_Flags & paDitherOff) == 0;
past->past_InputConversionProc = PaConvert_SelectProc( nativeInputSampleFormat,
past->past_InputSampleFormat, 0, ifDither );
if( past->past_InputConversionProc == NULL ) return paSampleFormatNotSupported;
}
else
{
past->past_InputConversionProc = NULL; /* no conversion necessary */
}
return paNoError;
}
/*************************************************************************/
PaError PaConvert_SetupOutput( internalPortAudioStream *past,
PaSampleFormat nativeOutputSampleFormat )
{
past->past_NativeOutputSampleFormat = nativeOutputSampleFormat;
past->past_OutputConversionSourceStride = 1;
past->past_OutputConversionTargetStride = 1;
if( nativeOutputSampleFormat != past->past_OutputSampleFormat )
{
int ifDither = (past->past_Flags & paDitherOff) == 0;
int ifClip = (past->past_Flags & paClipOff) == 0;
past->past_OutputConversionProc = PaConvert_SelectProc( past->past_OutputSampleFormat,
nativeOutputSampleFormat, ifClip, ifDither );
if( past->past_OutputConversionProc == NULL ) return paSampleFormatNotSupported;
}
else
{
past->past_OutputConversionProc = NULL; /* no conversion necessary */
}
return paNoError;
}
/*************************************************************************
** Called by host code.
** Convert input from native format to user format,
** call user code,
** then convert output to native format.
** Returns result from user callback.
*/
long PaConvert_Process( internalPortAudioStream *past,
void *nativeInputBuffer,
void *nativeOutputBuffer )
{
int userResult;
void *inputBuffer = NULL;
void *outputBuffer = NULL;
/* Get native input data. */
if( (past->past_NumInputChannels > 0) && (nativeInputBuffer != NULL) )
{
if( past->past_InputSampleFormat == past->past_NativeInputSampleFormat )
{
/* Already in native format so just read directly from native buffer. */
inputBuffer = nativeInputBuffer;
}
else
{
inputBuffer = past->past_InputBuffer;
/* Convert input data to user format. */
(*past->past_InputConversionProc)(nativeInputBuffer, past->past_InputConversionSourceStride,
inputBuffer, past->past_InputConversionTargetStride,
past->past_FramesPerUserBuffer * past->past_NumInputChannels );
}
}
/* Are we doing output? */
if( (past->past_NumOutputChannels > 0) && (nativeOutputBuffer != NULL) )
{
outputBuffer = (past->past_OutputConversionProc == NULL) ?
nativeOutputBuffer : past->past_OutputBuffer;
}
/*
AddTraceMessage("Pa_CallConvertInt16: inputBuffer = ", (int) inputBuffer );
AddTraceMessage("Pa_CallConvertInt16: outputBuffer = ", (int) outputBuffer );
*/
/* Call user callback routine. */
userResult = past->past_Callback(
inputBuffer,
outputBuffer,
past->past_FramesPerUserBuffer,
past->past_FrameCount,
past->past_UserData );
/* Advance frame counter for timestamp. */
past->past_FrameCount += past->past_FramesPerUserBuffer; // FIXME - should this be in here?
/* Convert to native format if necessary. */
if( (past->past_OutputConversionProc != NULL ) && (outputBuffer != NULL) )
{
(*past->past_OutputConversionProc)( outputBuffer, past->past_OutputConversionSourceStride,
nativeOutputBuffer, past->past_OutputConversionTargetStride,
past->past_FramesPerUserBuffer * past->past_NumOutputChannels );
}
return userResult;
}

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#ifndef PA_HOST_H
#define PA_HOST_H
/*
* $Id: pa_host.h,v 1.5 2003/03/02 08:01:34 dmazzoni Exp $
* Host dependant internal API for PortAudio
*
* Author: Phil Burk <philburk@softsynth.com>
*
* PortAudio Portable Real-Time Audio Library
* Latest Version at: http://www.softsynth.com/portaudio/
* DirectSound and Macintosh Implementation
* Copyright (c) 1999-2000 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 "portaudio.h"
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
#ifndef SUPPORT_AUDIO_CAPTURE
#define SUPPORT_AUDIO_CAPTURE (1)
#endif
#ifndef int32
typedef long int32;
#endif
#ifndef uint32
typedef unsigned long uint32;
#endif
#ifndef int16
typedef short int16;
#endif
#ifndef uint16
typedef unsigned short uint16;
#endif
/* Used to convert between various sample formats. */
typedef void (PortAudioConverter)(
void *inputBuffer, int inputStride,
void *outputBuffer, int outputStride,
int numSamples );
#define PA_MAGIC (0x18273645)
/************************************************************************************/
/****************** Structures ******************************************************/
/************************************************************************************/
typedef struct internalPortAudioStream
{
uint32 past_Magic; /* ID for struct to catch bugs. */
/* Begin user specified information. */
uint32 past_FramesPerUserBuffer;
uint32 past_NumUserBuffers;
double past_SampleRate; /* Closest supported sample rate. */
int past_NumInputChannels;
int past_NumOutputChannels;
PaDeviceID past_InputDeviceID;
PaDeviceID past_OutputDeviceID;
PaSampleFormat past_InputSampleFormat;
PaSampleFormat past_OutputSampleFormat;
PortAudioCallback *past_Callback;
void *past_UserData;
uint32 past_Flags;
/* End user specified information. */
void *past_DeviceData;
PaSampleFormat past_NativeOutputSampleFormat;
PaSampleFormat past_NativeInputSampleFormat;
/* Flags for communicating between foreground and background. */
volatile int past_IsActive; /* Background is still playing. */
volatile int past_StopSoon; /* Background should keep playing when buffers empty. */
volatile int past_StopNow; /* Background should stop playing now. */
/* These buffers are used when the native format does not match the user format. */
void *past_InputBuffer;
uint32 past_InputBufferSize; /* Size in bytes of the input buffer. */
void *past_OutputBuffer;
uint32 past_OutputBufferSize;
/* Measurements */
uint32 past_NumCallbacks;
PaTimestamp past_FrameCount; /* Frames output to buffer. */
/* For measuring CPU utilization. */
double past_AverageInsideCount;
double past_AverageTotalCount;
double past_Usage;
int past_IfLastExitValid;
/* Format Conversion */
/* These are setup by PaConversion_Setup() */
PortAudioConverter *past_InputConversionProc;
int past_InputConversionSourceStride;
int past_InputConversionTargetStride;
PortAudioConverter *past_OutputConversionProc;
int past_OutputConversionSourceStride;
int past_OutputConversionTargetStride;
}
internalPortAudioStream;
/************************************************************************************/
/******** These functions must be provided by a platform implementation. ************/
/************************************************************************************/
PaError PaHost_Init( void );
PaError PaHost_Term( void );
PaError PaHost_OpenStream( internalPortAudioStream *past );
PaError PaHost_CloseStream( internalPortAudioStream *past );
PaError PaHost_StartOutput( internalPortAudioStream *past );
PaError PaHost_StopOutput( internalPortAudioStream *past, int abort );
PaError PaHost_StartInput( internalPortAudioStream *past );
PaError PaHost_StopInput( internalPortAudioStream *past, int abort );
PaError PaHost_StartEngine( internalPortAudioStream *past );
PaError PaHost_StopEngine( internalPortAudioStream *past, int abort );
PaError PaHost_StreamActive( internalPortAudioStream *past );
void *PaHost_AllocateFastMemory( long numBytes );
void PaHost_FreeFastMemory( void *addr, long numBytes );
/* This only called if PA_VALIDATE_RATE IS CALLED. */
PaError PaHost_ValidateSampleRate( PaDeviceID id, double requestedFrameRate,
double *closestFrameRatePtr );
/**********************************************************************/
/************ Common Utility Routines provided by PA ******************/
/**********************************************************************/
/* PaHost_IsInitialized() returns non-zero if PA is initialized, 0 otherwise */
int PaHost_IsInitialized( void );
internalPortAudioStream* PaHost_GetStreamRepresentation( PortAudioStream *stream );
int PaHost_FindClosestTableEntry( double allowableError, const double *rateTable,
int numRates, double frameRate );
long Pa_CallConvertInt16( internalPortAudioStream *past,
short *nativeInputBuffer,
short *nativeOutputBuffer );
/* Calculate 2 LSB dither signal with a triangular distribution.
** Ranged properly for adding to a 32 bit 1.31 fixed point value prior to >>15.
** Range of output is +/- 65535
** Multiply by PA_DITHER_SCALE to get a float between -2.0 and 2.0. */
#define PA_DITHER_BITS (15)
#define PA_DITHER_SCALE (1.0f / ((1<<PA_DITHER_BITS)-1))
long PaConvert_TriangularDither( void );
PaError PaConvert_SetupInput( internalPortAudioStream *past,
PaSampleFormat nativeInputSampleFormat );
PaError PaConvert_SetupOutput( internalPortAudioStream *past,
PaSampleFormat nativeOutputSampleFormat );
long PaConvert_Process( internalPortAudioStream *past,
void *nativeInputBuffer,
void *nativeOutputBuffer );
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* PA_HOST_H */

806
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/*
* $Id: pa_lib.c,v 1.6 2003/03/02 08:01:34 dmazzoni Exp $
* Portable Audio I/O Library
* Host Independant Layer
*
* Based on the Open Source API proposed by Ross Bencina
* Copyright (c) 1999-2000 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.
*
*/
/* Modification History:
PLB20010422 - apply Mike Berry's changes for CodeWarrior on PC
PLB20010820 - fix dither and shift for recording PaUInt8 format
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
/* PLB20010422 - "memory.h" doesn't work on CodeWarrior for PC. Thanks Mike Berry for the mod. */
#ifdef _WIN32
#ifndef __MWERKS__
#include <memory.h>
#endif /* __MWERKS__ */
#else /* !_WIN32 */
#include <memory.h>
#endif /* _WIN32 */
#include "portaudio.h"
#include "pa_host.h"
#include "pa_trace.h"
/* The reason we might NOT want to validate the rate before opening the stream
* is because many DirectSound drivers lie about the rates they actually support.
*/
#define PA_VALIDATE_RATE (0) /* If true validate sample rate against driver info. */
/*
O- maybe not allocate past_InputBuffer and past_OutputBuffer if not needed for conversion
*/
#ifndef FALSE
#define FALSE (0)
#define TRUE (!FALSE)
#endif
#define PRINT(x) { printf x; fflush(stdout); }
#define ERR_RPT(x) PRINT(x)
#define DBUG(x) /* PRINT(x) */
#define DBUGX(x) /* PRINT(x) */
static int gInitCount = 0; /* Count number of times Pa_Initialize() called to allow nesting and overlapping. */
static PaError Pa_KillStream( PortAudioStream *stream, int abort );
/***********************************************************************/
int PaHost_FindClosestTableEntry( double allowableError, const double *rateTable, int numRates, double frameRate )
{
double err, minErr = allowableError;
int i, bestFit = -1;
for( i=0; i<numRates; i++ )
{
err = fabs( frameRate - rateTable[i] );
if( err < minErr )
{
minErr = err;
bestFit = i;
}
}
return bestFit;
}
/**************************************************************************
** Make sure sample rate is legal and also convert to enumeration for driver.
*/
PaError PaHost_ValidateSampleRate( PaDeviceID id, double requestedFrameRate,
double *closestFrameRatePtr )
{
long bestRateIndex;
const PaDeviceInfo *pdi;
pdi = Pa_GetDeviceInfo( id );
if( pdi == NULL )
{
return paInvalidDeviceId;
}
if( pdi->numSampleRates == -1 )
{
/* Is it out of range? */
if( (requestedFrameRate < pdi->sampleRates[0]) ||
(requestedFrameRate > pdi->sampleRates[1]) )
{
return paInvalidSampleRate;
}
*closestFrameRatePtr = requestedFrameRate;
}
else
{
bestRateIndex = PaHost_FindClosestTableEntry( 1.0, pdi->sampleRates, pdi->numSampleRates, requestedFrameRate );
if( bestRateIndex < 0 ) return paInvalidSampleRate;
*closestFrameRatePtr = pdi->sampleRates[bestRateIndex];
}
return paNoError;
}
/*************************************************************************/
PaError Pa_OpenStream(
PortAudioStream** streamPtrPtr,
PaDeviceID inputDeviceID,
int numInputChannels,
PaSampleFormat inputSampleFormat,
void *inputDriverInfo,
PaDeviceID outputDeviceID,
int numOutputChannels,
PaSampleFormat outputSampleFormat,
void *outputDriverInfo,
double sampleRate,
unsigned long framesPerBuffer,
unsigned long numberOfBuffers,
unsigned long streamFlags,
PortAudioCallback *callback,
void *userData )
{
internalPortAudioStream *past = NULL;
PaError result = paNoError;
int bitsPerInputSample;
int bitsPerOutputSample;
/* Print passed parameters. */
DBUG(("Pa_OpenStream( %p, %d, %d, %d, %p, /* input */ \n",
streamPtrPtr, inputDeviceID, numInputChannels,
inputSampleFormat, inputDriverInfo ));
DBUG((" %d, %d, %d, %p, /* output */\n",
outputDeviceID, numOutputChannels,
outputSampleFormat, outputDriverInfo ));
DBUG((" %g, %d, %d, 0x%x, , %p )\n",
sampleRate, framesPerBuffer, numberOfBuffers,
streamFlags, userData ));
/* Check for parameter errors. */
if( (streamFlags & ~(paClipOff | paDitherOff)) != 0 ) return paInvalidFlag;
if( streamPtrPtr == NULL ) return paBadStreamPtr;
if( inputDriverInfo != NULL ) return paHostError; /* REVIEW */
if( outputDriverInfo != NULL ) return paHostError; /* REVIEW */
if( (inputDeviceID < 0) && ( outputDeviceID < 0) ) return paInvalidDeviceId;
if( (outputDeviceID >= Pa_CountDevices()) || (inputDeviceID >= Pa_CountDevices()) )
{
return paInvalidDeviceId;
}
if( (numInputChannels <= 0) && ( numOutputChannels <= 0) ) return paInvalidChannelCount;
#if SUPPORT_AUDIO_CAPTURE
if( inputDeviceID >= 0 )
{
PaError size = Pa_GetSampleSize( inputSampleFormat );
if( size < 0 ) return size;
bitsPerInputSample = 8 * size;
if( (numInputChannels <= 0) ) return paInvalidChannelCount;
}
#else
if( inputDeviceID >= 0 )
{
return paInvalidChannelCount;
}
#endif /* SUPPORT_AUDIO_CAPTURE */
else
{
if( numInputChannels > 0 ) return paInvalidChannelCount;
bitsPerInputSample = 0;
}
if( outputDeviceID >= 0 )
{
PaError size = Pa_GetSampleSize( outputSampleFormat );
if( size < 0 ) return size;
bitsPerOutputSample = 8 * size;
if( (numOutputChannels <= 0) ) return paInvalidChannelCount;
}
else
{
if( numOutputChannels > 0 ) return paInvalidChannelCount;
bitsPerOutputSample = 0;
}
if( callback == NULL ) return paNullCallback;
/* Allocate and clear stream structure. */
past = (internalPortAudioStream *) PaHost_AllocateFastMemory( sizeof(internalPortAudioStream) );
if( past == NULL ) return paInsufficientMemory;
memset( past, 0, sizeof(internalPortAudioStream) );
AddTraceMessage("Pa_OpenStream: past", (long) past );
past->past_Magic = PA_MAGIC; /* Set ID to catch bugs. */
past->past_FramesPerUserBuffer = framesPerBuffer;
past->past_NumUserBuffers = numberOfBuffers; /* NOTE - PaHost_OpenStream() MUST CHECK FOR ZERO! */
past->past_Callback = callback;
past->past_UserData = userData;
past->past_OutputSampleFormat = outputSampleFormat;
past->past_InputSampleFormat = inputSampleFormat;
past->past_OutputDeviceID = outputDeviceID;
past->past_InputDeviceID = inputDeviceID;
past->past_NumInputChannels = numInputChannels;
past->past_NumOutputChannels = numOutputChannels;
past->past_Flags = streamFlags;
/* Check for absurd sample rates. */
if( (sampleRate < 1000.0) || (sampleRate > 200000.0) )
{
result = paInvalidSampleRate;
goto cleanup;
}
/* Allocate buffers that may be used for format conversion from user to native buffers. */
if( numInputChannels > 0 )
{
#if PA_VALIDATE_RATE
result = PaHost_ValidateSampleRate( inputDeviceID, sampleRate, &past->past_SampleRate );
if( result < 0 )
{
goto cleanup;
}
#else
past->past_SampleRate = sampleRate;
#endif
/* Allocate single Input buffer for passing formatted samples to user callback. */
past->past_InputBufferSize = framesPerBuffer * numInputChannels * ((bitsPerInputSample+7) / 8);
past->past_InputBuffer = PaHost_AllocateFastMemory(past->past_InputBufferSize);
if( past->past_InputBuffer == NULL )
{
result = paInsufficientMemory;
goto cleanup;
}
}
else
{
past->past_InputBuffer = NULL;
}
/* Allocate single Output buffer. */
if( numOutputChannels > 0 )
{
#if PA_VALIDATE_RATE
result = PaHost_ValidateSampleRate( outputDeviceID, sampleRate, &past->past_SampleRate );
if( result < 0 )
{
goto cleanup;
}
#else
past->past_SampleRate = sampleRate;
#endif
past->past_OutputBufferSize = framesPerBuffer * numOutputChannels * ((bitsPerOutputSample+7) / 8);
past->past_OutputBuffer = PaHost_AllocateFastMemory(past->past_OutputBufferSize);
if( past->past_OutputBuffer == NULL )
{
result = paInsufficientMemory;
goto cleanup;
}
}
else
{
past->past_OutputBuffer = NULL;
}
result = PaHost_OpenStream( past );
if( result < 0 ) goto cleanup;
*streamPtrPtr = (void *) past;
return result;
cleanup:
if( past != NULL ) Pa_CloseStream( past );
*streamPtrPtr = NULL;
return result;
}
/*************************************************************************/
PaError Pa_OpenDefaultStream( PortAudioStream** stream,
int numInputChannels,
int numOutputChannels,
PaSampleFormat sampleFormat,
double sampleRate,
unsigned long framesPerBuffer,
unsigned long numberOfBuffers,
PortAudioCallback *callback,
void *userData )
{
return Pa_OpenStream(
stream,
((numInputChannels > 0) ? Pa_GetDefaultInputDeviceID() : paNoDevice),
numInputChannels, sampleFormat, NULL,
((numOutputChannels > 0) ? Pa_GetDefaultOutputDeviceID() : paNoDevice),
numOutputChannels, sampleFormat, NULL,
sampleRate, framesPerBuffer, numberOfBuffers, paNoFlag, callback, userData );
}
/*************************************************************************/
PaError Pa_CloseStream( PortAudioStream* stream)
{
PaError result;
internalPortAudioStream *past;
DBUG(("Pa_CloseStream()\n"));
if( stream == NULL ) return paBadStreamPtr;
past = (internalPortAudioStream *) stream;
Pa_AbortStream( past );
result = PaHost_CloseStream( past );
if( past->past_InputBuffer ) PaHost_FreeFastMemory( past->past_InputBuffer, past->past_InputBufferSize );
if( past->past_OutputBuffer ) PaHost_FreeFastMemory( past->past_OutputBuffer, past->past_OutputBufferSize );
PaHost_FreeFastMemory( past, sizeof(internalPortAudioStream) );
return result;
}
/*************************************************************************/
PaError Pa_StartStream( PortAudioStream *stream )
{
PaError result = paHostError;
internalPortAudioStream *past;
if( stream == NULL ) return paBadStreamPtr;
past = (internalPortAudioStream *) stream;
past->past_FrameCount = 0.0;
if( past->past_NumInputChannels > 0 )
{
result = PaHost_StartInput( past );
DBUG(("Pa_StartStream: PaHost_StartInput returned = 0x%X.\n", result));
if( result < 0 ) goto error;
}
if( past->past_NumOutputChannels > 0 )
{
result = PaHost_StartOutput( past );
DBUG(("Pa_StartStream: PaHost_StartOutput returned = 0x%X.\n", result));
if( result < 0 ) goto error;
}
result = PaHost_StartEngine( past );
DBUG(("Pa_StartStream: PaHost_StartEngine returned = 0x%X.\n", result));
if( result < 0 ) goto error;
return paNoError;
error:
return result;
}
/*************************************************************************/
PaError Pa_StopStream( PortAudioStream *stream )
{
return Pa_KillStream( stream, 0 );
}
/*************************************************************************/
PaError Pa_AbortStream( PortAudioStream *stream )
{
return Pa_KillStream( stream, 1 );
}
/*************************************************************************/
static PaError Pa_KillStream( PortAudioStream *stream, int abort )
{
PaError result = paNoError;
internalPortAudioStream *past;
DBUG(("Pa_StopStream().\n"));
if( stream == NULL ) return paBadStreamPtr;
past = (internalPortAudioStream *) stream;
if( (past->past_NumInputChannels > 0) || (past->past_NumOutputChannels > 0) )
{
result = PaHost_StopEngine( past, abort );
DBUG(("Pa_StopStream: PaHost_StopEngine returned = 0x%X.\n", result));
if( result < 0 ) goto error;
}
if( past->past_NumInputChannels > 0 )
{
result = PaHost_StopInput( past, abort );
DBUG(("Pa_StopStream: PaHost_StopInput returned = 0x%X.\n", result));
if( result != paNoError ) goto error;
}
if( past->past_NumOutputChannels > 0 )
{
result = PaHost_StopOutput( past, abort );
DBUG(("Pa_StopStream: PaHost_StopOutput returned = 0x%X.\n", result));
if( result != paNoError ) goto error;
}
error:
past->past_Usage = 0;
past->past_IfLastExitValid = 0;
return result;
}
/*************************************************************************/
PaError Pa_StreamActive( PortAudioStream *stream )
{
internalPortAudioStream *past;
if( stream == NULL ) return paBadStreamPtr;
past = (internalPortAudioStream *) stream;
return PaHost_StreamActive( past );
}
/*************************************************************************/
const char *Pa_GetErrorText( PaError errnum )
{
const char *msg;
switch(errnum)
{
case paNoError: msg = "Success"; break;
case paHostError: msg = "Host error."; break;
case paInvalidChannelCount: msg = "Invalid number of channels."; break;
case paInvalidSampleRate: msg = "Invalid sample rate."; break;
case paInvalidDeviceId: msg = "Invalid device ID."; break;
case paInvalidFlag: msg = "Invalid flag."; break;
case paSampleFormatNotSupported: msg = "Sample format not supported"; break;
case paBadIODeviceCombination: msg = "Illegal combination of I/O devices."; break;
case paInsufficientMemory: msg = "Insufficient memory."; break;
case paBufferTooBig: msg = "Buffer too big."; break;
case paBufferTooSmall: msg = "Buffer too small."; break;
case paNullCallback: msg = "No callback routine specified."; break;
case paBadStreamPtr: msg = "Invalid stream pointer."; break;
case paTimedOut : msg = "Wait Timed Out."; break;
case paInternalError: msg = "Internal PortAudio Error."; break;
case paDeviceUnavailable: msg = "Device Unavailable."; break;
default: msg = "Illegal error number."; break;
}
return msg;
}
/*
Get CPU Load as a fraction of total CPU time.
A value of 0.5 would imply that PortAudio and the sound generating
callback was consuming roughly 50% of the available CPU time.
The amount may vary depending on CPU load.
This function may be called from the callback function.
*/
double Pa_GetCPULoad( PortAudioStream* stream)
{
internalPortAudioStream *past;
if( stream == NULL ) return (double) paBadStreamPtr;
past = (internalPortAudioStream *) stream;
return past->past_Usage;
}
/*************************************************************************/
internalPortAudioStream* PaHost_GetStreamRepresentation( PortAudioStream *stream )
{
internalPortAudioStream* result = (internalPortAudioStream*) stream;
if( result == NULL || result->past_Magic != PA_MAGIC )
return NULL;
else
return result;
}
/*************************************************************
** Calculate 2 LSB dither signal with a triangular distribution.
** Ranged properly for adding to a 32 bit integer prior to >>15.
** Range of output is +/- 32767
*/
#define PA_DITHER_BITS (15)
#define PA_DITHER_SCALE (1.0f / ((1<<PA_DITHER_BITS)-1))
long PaConvert_TriangularDither( void )
{
static unsigned long previous = 0;
static unsigned long randSeed1 = 22222;
static unsigned long randSeed2 = 5555555;
long current, highPass;
/* Generate two random numbers. */
randSeed1 = (randSeed1 * 196314165) + 907633515;
randSeed2 = (randSeed2 * 196314165) + 907633515;
/* Generate triangular distribution about 0.
* Shift before adding to prevent overflow which would skew the distribution.
* Also shift an extra bit for the high pass filter.
*/
#define DITHER_SHIFT ((32 - PA_DITHER_BITS) + 1)
current = (((long)randSeed1)>>DITHER_SHIFT) + (((long)randSeed2)>>DITHER_SHIFT);
/* High pass filter to reduce audibility. */
highPass = current - previous;
previous = current;
return highPass;
}
/*************************************************************************
** Called by host code.
** Convert input from Int16, call user code, then convert output
** to Int16 format for native use.
** Assumes host native format is paInt16.
** Returns result from user callback.
*/
long Pa_CallConvertInt16( internalPortAudioStream *past,
short *nativeInputBuffer,
short *nativeOutputBuffer )
{
long temp;
int userResult;
unsigned int i;
void *inputBuffer = NULL;
void *outputBuffer = NULL;
#if SUPPORT_AUDIO_CAPTURE
/* Get native data from DirectSound. */
if( (past->past_NumInputChannels > 0) && (nativeInputBuffer != NULL) )
{
/* Convert from native format to PA format. */
unsigned int samplesPerBuffer = past->past_FramesPerUserBuffer * past->past_NumInputChannels;
switch(past->past_InputSampleFormat)
{
case paFloat32:
{
float *inBufPtr = (float *) past->past_InputBuffer;
inputBuffer = past->past_InputBuffer;
for( i=0; i<samplesPerBuffer; i++ )
{
inBufPtr[i] = nativeInputBuffer[i] * (1.0f / 32767.0f);
}
break;
}
case paInt32:
{
/* Convert 16 bit data to 32 bit integers */
int *inBufPtr = (int *) past->past_InputBuffer;
inputBuffer = past->past_InputBuffer;
for( i=0; i<samplesPerBuffer; i++ )
{
inBufPtr[i] = nativeInputBuffer[i] << 16;
}
break;
}
case paInt16:
{
/* Already in correct format so don't copy. */
inputBuffer = nativeInputBuffer;
break;
}
case paInt8:
{
/* Convert 16 bit data to 8 bit chars */
char *inBufPtr = (char *) past->past_InputBuffer;
inputBuffer = past->past_InputBuffer;
if( past->past_Flags & paDitherOff )
{
for( i=0; i<samplesPerBuffer; i++ )
{
inBufPtr[i] = (char)(nativeInputBuffer[i] >> 8);
}
}
else
{
for( i=0; i<samplesPerBuffer; i++ )
{
temp = nativeInputBuffer[i];
temp += PaConvert_TriangularDither() >> 8; /* PLB20010820 */
temp = ((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp));
inBufPtr[i] = (char)(temp >> 8);
}
}
break;
}
case paUInt8:
{
/* Convert 16 bit data to 8 bit unsigned chars */
unsigned char *inBufPtr = (unsigned char *) past->past_InputBuffer;
inputBuffer = past->past_InputBuffer;
if( past->past_Flags & paDitherOff )
{
for( i=0; i<samplesPerBuffer; i++ )
{
inBufPtr[i] = (unsigned char)((nativeInputBuffer[i] >> 8) + 0x80);
}
}
else
{
/* If you dither then you have to clip because dithering could push the signal out of range! */
for( i=0; i<samplesPerBuffer; i++ )
{
temp = nativeInputBuffer[i];
temp += PaConvert_TriangularDither() >> 8; /* PLB20010820 */
temp = ((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp));
inBufPtr[i] = (unsigned char)((temp>>8) + 0x80); /* PLB20010820 */
}
}
break;
}
default:
break;
}
}
#endif /* SUPPORT_AUDIO_CAPTURE */
/* Are we doing output time? */
if( (past->past_NumOutputChannels > 0) && (nativeOutputBuffer != NULL) )
{
/* May already be in native format so just write directly to native buffer. */
outputBuffer = (past->past_OutputSampleFormat == paInt16) ?
nativeOutputBuffer : past->past_OutputBuffer;
}
/*
AddTraceMessage("Pa_CallConvertInt16: inputBuffer = ", (int) inputBuffer );
AddTraceMessage("Pa_CallConvertInt16: outputBuffer = ", (int) outputBuffer );
*/
/* Call user callback routine. */
userResult = past->past_Callback(
inputBuffer,
outputBuffer,
past->past_FramesPerUserBuffer,
past->past_FrameCount,
past->past_UserData );
past->past_FrameCount += (PaTimestamp) past->past_FramesPerUserBuffer;
/* Convert to native format if necessary. */
if( outputBuffer != NULL )
{
unsigned int samplesPerBuffer = past->past_FramesPerUserBuffer * past->past_NumOutputChannels;
switch(past->past_OutputSampleFormat)
{
case paFloat32:
{
float *outBufPtr = (float *) past->past_OutputBuffer;
if( past->past_Flags & paDitherOff )
{
if( past->past_Flags & paClipOff ) /* NOTHING */
{
for( i=0; i<samplesPerBuffer; i++ )
{
*nativeOutputBuffer++ = (short) (outBufPtr[i] * (32767.0f));
}
}
else /* CLIP */
{
for( i=0; i<samplesPerBuffer; i++ )
{
temp = (long)(outBufPtr[i] * 32767.0f);
*nativeOutputBuffer++ = (short)((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp));
}
}
}
else
{
/* If you dither then you have to clip because dithering could push the signal out of range! */
for( i=0; i<samplesPerBuffer; i++ )
{
float dither = PaConvert_TriangularDither()*PA_DITHER_SCALE;
float dithered = (outBufPtr[i] * (32767.0f)) + dither;
temp = (long) (dithered);
*nativeOutputBuffer++ = (short)((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp));
}
}
break;
}
case paInt32:
{
int *outBufPtr = (int *) past->past_OutputBuffer;
if( past->past_Flags & paDitherOff )
{
for( i=0; i<samplesPerBuffer; i++ )
{
*nativeOutputBuffer++ = (short) (outBufPtr[i] >> 16 );
}
}
else
{
for( i=0; i<samplesPerBuffer; i++ )
{
/* Shift one bit down before dithering so that we have room for overflow from add. */
temp = (outBufPtr[i] >> 1) + PaConvert_TriangularDither();
temp = temp >> 15;
*nativeOutputBuffer++ = (short)((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp));
}
}
break;
}
case paInt8:
{
char *outBufPtr = (char *) past->past_OutputBuffer;
for( i=0; i<samplesPerBuffer; i++ )
{
*nativeOutputBuffer++ = (short) (((int)outBufPtr[i]) << 8);
}
break;
}
case paUInt8:
{
unsigned char *outBufPtr = (unsigned char *) past->past_OutputBuffer;
for( i=0; i<samplesPerBuffer; i++ )
{
*nativeOutputBuffer++ = (short) (((int)(outBufPtr[i] - 0x80)) << 8);
}
break;
}
default:
break;
}
}
return userResult;
}
/*************************************************************************/
PaError Pa_Initialize( void )
{
if( gInitCount++ > 0 ) return paNoError;
ResetTraceMessages();
return PaHost_Init();
}
PaError Pa_Terminate( void )
{
PaError result = paNoError;
if( gInitCount == 0 ) return paNoError;
else if( --gInitCount == 0 )
{
result = PaHost_Term();
DumpTraceMessages();
}
return result;
}
int PaHost_IsInitialized()
{
return gInitCount;
}
/*************************************************************************/
PaError Pa_GetSampleSize( PaSampleFormat format )
{
int size;
switch(format )
{
case paUInt8:
case paInt8:
size = 1;
break;
case paInt16:
size = 2;
break;
case paPackedInt24:
size = 3;
break;
case paFloat32:
case paInt32:
case paInt24:
size = 4;
break;
default:
size = paSampleFormatNotSupported;
break;
}
return (PaError) size;
}

83
lib/portaudio/pa_common/pa_trace.c Normal file
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@ -0,0 +1,83 @@
/*
* $Id: pa_trace.c,v 1.5 2003/03/02 08:01:34 dmazzoni Exp $
* Portable Audio I/O Library Trace Facility
* Store trace information in real-time for later printing.
*
* Based on the Open Source API proposed by Ross Bencina
* Copyright (c) 1999-2000 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 <string.h>
#include "pa_trace.h"
#if TRACE_REALTIME_EVENTS
static char *traceTextArray[MAX_TRACE_RECORDS];
static int traceIntArray[MAX_TRACE_RECORDS];
static int traceIndex = 0;
static int traceBlock = 0;
/*********************************************************************/
void ResetTraceMessages()
{
traceIndex = 0;
}
/*********************************************************************/
void DumpTraceMessages()
{
int i;
int numDump = (traceIndex < MAX_TRACE_RECORDS) ? traceIndex : MAX_TRACE_RECORDS;
printf("DumpTraceMessages: traceIndex = %d\n", traceIndex );
for( i=0; i<numDump; i++ )
{
printf("%3d: %s = 0x%08X\n",
i, traceTextArray[i], traceIntArray[i] );
}
ResetTraceMessages();
fflush(stdout);
}
/*********************************************************************/
void AddTraceMessage( char *msg, int data )
{
if( (traceIndex == MAX_TRACE_RECORDS) && (traceBlock == 0) )
{
traceBlock = 1;
/* DumpTraceMessages(); */
}
else if( traceIndex < MAX_TRACE_RECORDS )
{
traceTextArray[traceIndex] = msg;
traceIntArray[traceIndex] = data;
traceIndex++;
}
}
#endif

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#ifndef PA_TRACE_H
#define PA_TRACE_H
/*
* $Id: pa_trace.h,v 1.5 2003/03/02 08:01:35 dmazzoni Exp $
* Portable Audio I/O Library Trace Facility
* Store trace information in real-time for later printing.
*
* Based on the Open Source API proposed by Ross Bencina
* Copyright (c) 1999-2000 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.
*/
#define TRACE_REALTIME_EVENTS (0) /* Keep log of various real-time events. */
#define MAX_TRACE_RECORDS (2048)
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
/************************************************************************************/
/****************** Prototypes ******************************************************/
/************************************************************************************/
#if TRACE_REALTIME_EVENTS
void DumpTraceMessages();
void ResetTraceMessages();
void AddTraceMessage( char *msg, int data );
#else
#define AddTraceMessage(msg,data) /* noop */
#define ResetTraceMessages() /* noop */
#define DumpTraceMessages() /* noop */
#endif
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* PA_TRACE_H */

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#ifndef PORT_AUDIO_H
#define PORT_AUDIO_H
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
/*
* $Id: portaudio.h,v 1.6 2003/03/02 08:01:35 dmazzoni Exp $
* PortAudio Portable Real-Time Audio Library
* PortAudio API Header File
* Latest version available at: 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.
*
*/
typedef int PaError;
typedef enum {
paNoError = 0,
paHostError = -10000,
paInvalidChannelCount,
paInvalidSampleRate,
paInvalidDeviceId,
paInvalidFlag,
paSampleFormatNotSupported,
paBadIODeviceCombination,
paInsufficientMemory,
paBufferTooBig,
paBufferTooSmall,
paNullCallback,
paBadStreamPtr,
paTimedOut,
paInternalError,
paDeviceUnavailable
} PaErrorNum;
/*
Pa_Initialize() is the library initialisation function - call this before
using the library.
*/
PaError Pa_Initialize( void );
/*
Pa_Terminate() is the library termination function - call this after
using the library.
*/
PaError Pa_Terminate( void );
/*
Pa_GetHostError() returns a host specific error code.
This can be called after receiving a PortAudio error code of paHostError.
*/
long Pa_GetHostError( void );
/*
Pa_GetErrorText() translates the supplied PortAudio error number
into a human readable message.
*/
const char *Pa_GetErrorText( PaError errnum );
/*
Sample formats
These are formats used to pass sound data between the callback and the
stream. Each device has a "native" format which may be used when optimum
efficiency or control over conversion is required.
Formats marked "always available" are supported (emulated) by all
PortAudio implementations.
The floating point representation (paFloat32) uses +1.0 and -1.0 as the
maximum and minimum respectively.
paUInt8 is an unsigned 8 bit format where 128 is considered "ground"
*/
typedef unsigned long PaSampleFormat;
#define paFloat32 ((PaSampleFormat) (1<<0)) /*always available*/
#define paInt16 ((PaSampleFormat) (1<<1)) /*always available*/
#define paInt32 ((PaSampleFormat) (1<<2)) /*always available*/
#define paInt24 ((PaSampleFormat) (1<<3))
#define paPackedInt24 ((PaSampleFormat) (1<<4))
#define paInt8 ((PaSampleFormat) (1<<5))
#define paUInt8 ((PaSampleFormat) (1<<6))
#define paCustomFormat ((PaSampleFormat) (1<<16))
/*
Device enumeration mechanism.
Device ids range from 0 to Pa_CountDevices()-1.
Devices may support input, output or both.
*/
typedef int PaDeviceID;
#define paNoDevice -1
int Pa_CountDevices( void );
typedef struct
{
int structVersion;
const char *name;
int maxInputChannels;
int maxOutputChannels;
/* Number of discrete rates, or -1 if range supported. */
int numSampleRates;
/* Array of supported sample rates, or {min,max} if range supported. */
const double *sampleRates;
PaSampleFormat nativeSampleFormats;
}
PaDeviceInfo;
/*
Pa_GetDefaultInputDeviceID(), Pa_GetDefaultOutputDeviceID() return the
default device ids for input and output respectively, or paNoDevice if
no device is available.
The result can be passed to Pa_OpenStream().
On the PC, the user can specify a default device by
setting an environment variable. For example, to use device #1.
set PA_RECOMMENDED_OUTPUT_DEVICE=1
The user should first determine the available device ids by using
the supplied application "pa_devs".
*/
PaDeviceID Pa_GetDefaultInputDeviceID( void );
PaDeviceID Pa_GetDefaultOutputDeviceID( void );
/*
Pa_GetDeviceInfo() returns a pointer to an immutable PaDeviceInfo structure
for the device specified.
If the device parameter is out of range the function returns NULL.
PortAudio manages the memory referenced by the returned pointer, the client
must not manipulate or free the memory. The pointer is only guaranteed to be
valid between calls to Pa_Initialize() and Pa_Terminate().
*/
const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID device );
/*
PaTimestamp is used to represent a continuous sample clock with arbitrary
start time that can be used for syncronization. The type is used for the
outTime argument to the PortAudioCallback and as the result of Pa_StreamTime()
*/
typedef double PaTimestamp;
/*
PortAudioCallback is implemented by PortAudio clients.
inputBuffer and outputBuffer are arrays of interleaved samples,
the format, packing and number of channels used by the buffers are
determined by parameters to Pa_OpenStream() (see below).
framesPerBuffer is the number of sample frames to be processed by the callback.
outTime is the time in samples when the buffer(s) processed by
this callback will begin being played at the audio output.
See also Pa_StreamTime()
userData is the value of a user supplied pointer passed to Pa_OpenStream()
intended for storing synthesis data etc.
return value:
The callback can return a non-zero value to stop the stream. This may be
useful in applications such as soundfile players where a specific duration
of output is required. However, it is not necessary to utilise this mechanism
as StopStream() will also terminate the stream. A callback returning a
non-zero value must fill the entire outputBuffer.
NOTE: None of the other stream functions may be called from within the
callback function except for Pa_GetCPULoad().
*/
typedef int (PortAudioCallback)(
void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData );
/*
Stream flags
These flags may be supplied (ored together) in the streamFlags argument to
the Pa_OpenStream() function.
*/
#define paNoFlag (0)
#define paClipOff (1<<0) /* disable default clipping of out of range samples */
#define paDitherOff (1<<1) /* disable default dithering */
#define paPlatformSpecificFlags (0x00010000)
typedef unsigned long PaStreamFlags;
/*
A single PortAudioStream provides multiple channels of real-time
input and output audio streaming to a client application.
Pointers to PortAudioStream objects are passed between PortAudio functions.
*/
typedef void PortAudioStream;
#define PaStream PortAudioStream
/*
Pa_OpenStream() opens a stream for either input, output or both.
stream is the address of a PortAudioStream pointer which will receive
a pointer to the newly opened stream.
inputDevice is the id of the device used for input (see PaDeviceID above.)
inputDevice may be paNoDevice to indicate that an input device is not required.
numInputChannels is the number of channels of sound to be delivered to the
callback. It can range from 1 to the value of maxInputChannels in the
PaDeviceInfo record for the device specified by the inputDevice parameter.
If inputDevice is paNoDevice numInputChannels is ignored.
inputSampleFormat is the sample format of inputBuffer provided to the callback
function. inputSampleFormat may be any of the formats described by the
PaSampleFormat enumeration (see above). PortAudio guarantees support for
the device's native formats (nativeSampleFormats in the device info record)
and additionally 16 and 32 bit integer and 32 bit floating point formats.
Support for other formats is implementation defined.
inputDriverInfo is a pointer to an optional driver specific data structure
containing additional information for device setup or stream processing.
inputDriverInfo is never required for correct operation. If not used
inputDriverInfo should be NULL.
outputDevice is the id of the device used for output (see PaDeviceID above.)
outputDevice may be paNoDevice to indicate that an output device is not required.
numOutputChannels is the number of channels of sound to be supplied by the
callback. See the definition of numInputChannels above for more details.
outputSampleFormat is the sample format of the outputBuffer filled by the
callback function. See the definition of inputSampleFormat above for more
details.
outputDriverInfo is a pointer to an optional driver specific data structure
containing additional information for device setup or stream processing.
outputDriverInfo is never required for correct operation. If not used
outputDriverInfo should be NULL.
sampleRate is the desired sampleRate. For full-duplex streams it is the
sample rate for both input and output
framesPerBuffer is the length in sample frames of all internal sample buffers
used for communication with platform specific audio routines. Wherever
possible this corresponds to the framesPerBuffer parameter passed to the
callback function.
numberOfBuffers is the number of buffers used for multibuffered communication
with the platform specific audio routines. If you pass zero, then an optimum
value will be chosen for you internally. This parameter is provided only
as a guide - and does not imply that an implementation must use multibuffered
i/o when reliable double buffering is available (such as SndPlayDoubleBuffer()
on the Macintosh.)
streamFlags may contain a combination of flags ORed together.
These flags modify the behaviour of the streaming process. Some flags may only
be relevant to certain buffer formats.
callback is a pointer to a client supplied function that is responsible
for processing and filling input and output buffers (see above for details.)
userData is a client supplied pointer which is passed to the callback
function. It could for example, contain a pointer to instance data necessary
for processing the audio buffers.
return value:
Upon success Pa_OpenStream() returns PaNoError and places a pointer to a
valid PortAudioStream in the stream argument. The stream is inactive (stopped).
If a call to Pa_OpenStream() fails a non-zero error code is returned (see
PaError above) and the value of stream is invalid.
*/
PaError Pa_OpenStream( PortAudioStream** stream,
PaDeviceID inputDevice,
int numInputChannels,
PaSampleFormat inputSampleFormat,
void *inputDriverInfo,
PaDeviceID outputDevice,
int numOutputChannels,
PaSampleFormat outputSampleFormat,
void *outputDriverInfo,
double sampleRate,
unsigned long framesPerBuffer,
unsigned long numberOfBuffers,
PaStreamFlags streamFlags,
PortAudioCallback *callback,
void *userData );
/*
Pa_OpenDefaultStream() is a simplified version of Pa_OpenStream() that opens
the default input and/or output devices. Most parameters have identical meaning
to their Pa_OpenStream() counterparts, with the following exceptions:
If either numInputChannels or numOutputChannels is 0 the respective device
is not opened. This has the same effect as passing paNoDevice in the device
arguments to Pa_OpenStream().
sampleFormat applies to both the input and output buffers.
*/
PaError Pa_OpenDefaultStream( PortAudioStream** stream,
int numInputChannels,
int numOutputChannels,
PaSampleFormat sampleFormat,
double sampleRate,
unsigned long framesPerBuffer,
unsigned long numberOfBuffers,
PortAudioCallback *callback,
void *userData );
/*
Pa_CloseStream() closes an audio stream, flushing any pending buffers.
*/
PaError Pa_CloseStream( PortAudioStream* );
/*
Pa_StartStream() and Pa_StopStream() begin and terminate audio processing.
Pa_StopStream() waits until all pending audio buffers have been played.
Pa_AbortStream() stops playing immediately without waiting for pending
buffers to complete.
*/
PaError Pa_StartStream( PortAudioStream *stream );
PaError Pa_StopStream( PortAudioStream *stream );
PaError Pa_AbortStream( PortAudioStream *stream );
/*
Pa_StreamActive() returns one (1) when the stream is active (ie playing
or recording audio), zero (0) when not playing, or a negative error number
if the stream is invalid.
The stream is active between calls to Pa_StartStream() and Pa_StopStream(),
but may also become inactive if the callback returns a non-zero value.
In the latter case, the stream is considered inactive after the last
buffer has finished playing.
*/
PaError Pa_StreamActive( PortAudioStream *stream );
/*
Pa_StreamTime() returns the current output time in samples for the stream.
This time may be used as a time reference (for example synchronizing audio to
MIDI).
*/
PaTimestamp Pa_StreamTime( PortAudioStream *stream );
/*
Pa_GetCPULoad() returns the CPU Load for the stream.
The "CPU Load" is a fraction of total CPU time consumed by the stream's
audio processing routines including, but not limited to the client supplied
callback.
A value of 0.5 would imply that PortAudio and the sound generating
callback was consuming roughly 50% of the available CPU time.
This function may be called from the callback function or the application.
*/
double Pa_GetCPULoad( PortAudioStream* stream );
/*
Pa_GetMinNumBuffers() returns the minimum number of buffers required by
the current host based on minimum latency.
On the PC, for the DirectSound implementation, latency can be optionally set
by user by setting an environment variable.
For example, to set latency to 200 msec, put:
set PA_MIN_LATENCY_MSEC=200
in the AUTOEXEC.BAT file and reboot.
If the environment variable is not set, then the latency will be determined
based on the OS. Windows NT has higher latency than Win95.
*/
int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate );
/*
Pa_Sleep() puts the caller to sleep for at least 'msec' milliseconds.
You may sleep longer than the requested time so don't rely on this for
accurate musical timing.
Pa_Sleep() is provided as a convenience for authors of portable code (such as
the tests and examples in the PortAudio distribution.)
*/
void Pa_Sleep( long msec );
/*
Pa_GetSampleSize() returns the size in bytes of a single sample in the
supplied PaSampleFormat, or paSampleFormatNotSupported if the format is
no supported.
*/
PaError Pa_GetSampleSize( PaSampleFormat format );
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* PORT_AUDIO_H */

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/deprecated/1.1/Mon Nov 26 06:23:31 2001//
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Use pa_unix_oss instead

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/pa_mac.c/1.7/Sun Mar 2 08:01:36 2003//
/patest_devinfo.c/1.2/Mon Nov 26 05:56:05 2001//
D

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/Makefile/1.4/Fri Oct 18 05:38:13 2002//
/pa_mac_core.c/1.9/Mon Sep 22 05:15:18 2003//
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# Make PortAudio Library for Mac OS X / CoreAudio
CC = cc
CFLAGS = -g -I../pa_common -I../pablio
PASRC = ../pa_common/pa_lib.c ../pa_common/pa_convert.c \
../pablio/ringbuffer.c pa_mac_core.c
PAINC = ../pa_common/portaudio.h
PAOBJ = ../pa_common/pa_lib.o ../pa_common/pa_convert.o \
../pablio/ringbuffer.o pa_mac_core.o
portaudio.a: $(PAOBJ)
ar ruv portaudio.a $(PAOBJ)
ranlib portaudio.a
clean:
rm -f portaudio.a *.o
%.o: %.c $(PAINC) Makefile
$(CC) -c $(CFLAGS) $< -o $@

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/debug_dual.c/1.3/Sun Mar 2 08:01:39 2003//
/debug_multi_in.c/1.2/Sun Mar 2 08:01:39 2003//
/debug_multi_out.c/1.2/Sun Mar 2 08:01:39 2003//
/debug_record.c/1.2/Sun Mar 2 08:01:40 2003//
/debug_sine.c/1.3/Sun Mar 2 08:01:40 2003//
/debug_test1.c/1.2/Sun Mar 2 08:01:40 2003//
/pa_devs.c/1.2/Sun Mar 2 08:01:40 2003//
/pa_fuzz.c/1.3/Sun Mar 2 08:01:40 2003//
/pa_minlat.c/1.2/Sun Mar 2 08:01:40 2003//
/paqa_devs.c/1.2/Sun Mar 2 08:01:40 2003//
/paqa_errs.c/1.2/Sun Mar 2 08:01:41 2003//
/patest1.c/1.3/Sun Mar 2 08:01:41 2003//
/patest_clip.c/1.2/Sun Mar 2 08:01:41 2003//
/patest_dither.c/1.2/Sun Mar 2 08:01:42 2003//
/patest_latency.c/1.2/Sun Mar 2 08:01:42 2003//
/patest_leftright.c/1.3/Sun Mar 2 08:01:42 2003//
/patest_longsine.c/1.2/Sun Mar 2 08:01:42 2003//
/patest_many.c/1.2/Sun Mar 2 08:01:42 2003//
/patest_maxsines.c/1.2/Sun Mar 2 08:01:42 2003//
/patest_pink.c/1.2/Sun Mar 2 08:01:42 2003//
/patest_record.c/1.3/Sun Mar 2 08:01:42 2003//
/patest_ringmix.c/1.2/Sun Mar 2 08:01:42 2003//
/patest_saw.c/1.2/Sun Mar 2 08:01:42 2003//
/patest_sine.c/1.3/Sun Mar 2 08:01:43 2003//
/patest_sine8.c/1.2/Sun Mar 2 08:01:43 2003//
/patest_sine_time.c/1.3/Sun Mar 2 08:01:43 2003//
/patest_stop.c/1.2/Sun Mar 2 08:01:43 2003//
/patest_sync.c/1.2/Sun Mar 2 08:01:43 2003//
/patest_wire.c/1.3/Sun Mar 2 08:01:43 2003//
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/*
* $Id: debug_dual.c,v 1.3 2003/03/02 08:01:39 dmazzoni Exp $
* 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
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/*
* $Id: debug_multi_in.c,v 1.2 2003/03/02 08:01:39 dmazzoni Exp $
* 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
lib/portaudio/pa_tests/debug_multi_out.c Normal file
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/*
* $Id: debug_multi_out.c,v 1.2 2003/03/02 08:01:39 dmazzoni Exp $
* 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;
}

338
lib/portaudio/pa_tests/debug_record.c Normal file
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/*
* $Id: debug_record.c,v 1.2 2003/03/02 08:01:40 dmazzoni Exp $
* patest_record.c
* Record input into an array.
* Save array to a file.
* Playback recorded data.
*
* 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 "portaudio.h"
#define SAMPLE_RATE (22050)
#define NUM_SECONDS (6)
#define NUM_CHANNELS (2)
#define FRAMES_PER_BUFFER (64)
/* #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)
#elif 0
#define PA_SAMPLE_TYPE paInt32
typedef long SAMPLE;
#define SAMPLE_SILENCE (0)
#elif 0
#define PA_SAMPLE_TYPE paInt16
typedef short SAMPLE;
#define SAMPLE_SILENCE (0)
#elif 0
#define PA_SAMPLE_TYPE paInt8
typedef char SAMPLE;
#define SAMPLE_SILENCE (0)
#else
#define PA_SAMPLE_TYPE paUInt8
typedef unsigned char SAMPLE;
#define SAMPLE_SILENCE (128)
#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( 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 * NUM_CHANNELS];
long framesToCalc;
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++ = 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( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, 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) outTime;
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 = 1;
}
else
{
for( i=0; i<framesPerBuffer; i++ )
{
*wptr++ = *rptr++; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */
}
data->frameIndex += framesPerBuffer;
finished = 0;
}
return finished;
}
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *stream;
PaError err;
paTestData data;
int i;
int totalFrames;
int numSamples;
int numBytes;
SAMPLE max, average, val;
printf("debug_record.c, sampleRate = %d, numChannels = %d\n",
SAMPLE_RATE, NUM_CHANNELS );
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 );
if( data.recordedSamples == NULL )
{
printf("Could not allocate record array.\n");
exit(1);
}
for( i=0; i<numSamples; i++ ) data.recordedSamples[i] = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
/* Record some audio. -------------------------------------------- */
err = Pa_OpenStream(
&stream,
Pa_GetDefaultInputDeviceID(),
NUM_CHANNELS, /* stereo input */
PA_SAMPLE_TYPE,
NULL,
paNoDevice,
0,
PA_SAMPLE_TYPE,
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER, /* frames per buffer */
0, /* number of buffers, if zero then use default minimum */
0, //paDitherOff, /* flags */
recordCallback,
&data );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Start recording!!\n"); fflush(stdout);
while( Pa_StreamActive( stream ) )
{
Pa_Sleep(1000);
printf("index = %d\n", data.frameIndex ); fflush(stdout);
}
printf("Stop recording!!\n"); fflush(stdout);
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
/* Measure maximum peak amplitude. */
max = 0;
average = 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 / numSamples;
if( PA_SAMPLE_TYPE == paFloat32 )
{
printf("sample max amplitude = %f\n", (double) max );
printf("sample average = %f\n", (double) average );
}
else
{
printf("sample max amplitude = %d\n", (int) max );
printf("sample average = %d\n", (int) 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;
printf("Begin playback.\n"); fflush(stdout);
err = Pa_OpenStream(
&stream,
paNoDevice,
0, /* NO input */
PA_SAMPLE_TYPE,
NULL,
Pa_GetDefaultOutputDeviceID(),
NUM_CHANNELS, /* 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,
&data );
if( err != paNoError ) goto error;
if( stream )
{
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Start playback!!\n"); fflush(stdout);
while( Pa_StreamActive( stream ) )
{
Pa_Sleep(1000);
printf("index = %d\n", data.frameIndex ); fflush(stdout);
}
printf("Stop playback!!\n"); fflush(stdout);
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
printf("Done.\n"); fflush(stdout);
}
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 ) );
return -1;
}

201
lib/portaudio/pa_tests/debug_sine.c Normal file
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/*
* $Id: debug_sine.c,v 1.3 2003/03/02 08:01:40 dmazzoni Exp $
* 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 (4096)
#define MSEC_PER_BUFFER (1000 * FRAMES_PER_BUFFER / SAMPLE_RATE)
#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 */
}
// Pa_Sleep( 3 * MSEC_PER_BUFFER / 4 );
// Pa_Sleep( MSEC_PER_BUFFER / 3 );
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 );
printf("MSEC_PER_BUFFER = %d\n", MSEC_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
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;
}

114
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/*
* $Id: debug_test1.c,v 1.2 2003/03/02 08:01:40 dmazzoni Exp $
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;
}

99
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/*
* $Id: pa_devs.c,v 1.2 2003/03/02 08:01:40 dmazzoni Exp $
* pa_devs.c
* List available devices.
*
* 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"
/*******************************************************************/
int main(void);
int main(void)
{
int i,j;
int numDevices;
const PaDeviceInfo *pdi;
PaError err;
Pa_Initialize();
numDevices = Pa_CountDevices();
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++ )
{
pdi = Pa_GetDeviceInfo( i );
printf("---------------------------------------------- #%d", i );
if( i == Pa_GetDefaultInputDeviceID() ) printf(" DefaultInput");
if( i == Pa_GetDefaultOutputDeviceID() ) printf(" DefaultOutput");
printf("\nName = %s\n", pdi->name );
printf("Max Inputs = %d", pdi->maxInputChannels );
printf(", Max Outputs = %d\n", pdi->maxOutputChannels );
if( pdi->numSampleRates == -1 )
{
printf("Sample Rate Range = %f to %f\n", pdi->sampleRates[0], pdi->sampleRates[1] );
}
else
{
printf("Sample Rates =");
for( j=0; j<pdi->numSampleRates; j++ )
{
printf(" %8.2f,", pdi->sampleRates[j] );
}
printf("\n");
}
printf("Native Sample Formats = ");
if( pdi->nativeSampleFormats & paInt8 ) printf("paInt8, ");
if( pdi->nativeSampleFormats & paUInt8 ) printf("paUInt8, ");
if( pdi->nativeSampleFormats & paInt16 ) printf("paInt16, ");
if( pdi->nativeSampleFormats & paInt32 ) printf("paInt32, ");
if( pdi->nativeSampleFormats & paFloat32 ) printf("paFloat32, ");
if( pdi->nativeSampleFormats & paInt24 ) printf("paInt24, ");
if( pdi->nativeSampleFormats & paPackedInt24 ) printf("paPackedInt24, ");
printf("\n");
}
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;
}

156
lib/portaudio/pa_tests/pa_fuzz.c Normal file
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/*
* $Id: pa_fuzz.c,v 1.3 2003/03/02 08:01:40 dmazzoni Exp $
* pa_fuzz.c
* Distort input like a fuzz boz.
*
* 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"
/*
** 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( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, 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( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
SAMPLE *out = (SAMPLE*)outputBuffer;
SAMPLE *in = (SAMPLE*)inputBuffer;
unsigned int i;
(void) outTime; /* Prevent unused variable warnings. */
(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 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *stream;
PaError err;
err = Pa_Initialize();
if( err != paNoError ) goto error;
err = Pa_OpenStream(
&stream,
Pa_GetDefaultInputDeviceID(), /* default output device */
2, /* stereo input */
PA_SAMPLE_TYPE,
NULL,
Pa_GetDefaultOutputDeviceID(), /* default output device */
2, /* stereo output */
PA_SAMPLE_TYPE,
NULL,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
0, /* number of buffers, if zero then use default minimum */
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");
fflush(stdout);
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;
}

172
lib/portaudio/pa_tests/pa_minlat.c Normal file
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/*
* $Id: pa_minlat.c,v 1.2 2003/03/02 08:01:40 dmazzoni Exp $
* paminlat.c
* Experiment with different numbers of buffers to determine the
* minimum latency for a computer.
*
* 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 <math.h>
#include "portaudio.h"
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TWOPI (M_PI * 2.0)
#define DEFAULT_BUFFER_SIZE (64)
typedef struct
{
double left_phase;
double right_phase;
}
paTestData;
/* Very simple synthesis routine to generate two sine waves. */
static int paminlatCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, 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;
}
void main( int argc, char **argv );
void main( int argc, char **argv )
{
PortAudioStream *stream;
PaError err;
paTestData data;
int go;
int numBuffers = 0;
int minBuffers = 0;
int framesPerBuffer;
double sampleRate = 44100.0;
char str[256];
printf("paminlat - Determine minimum latency for your computer.\n");
printf(" usage: paminlat {framesPerBuffer}\n");
printf(" for example: paminlat 256\n");
printf("Adjust your stereo until you hear a smooth tone in each speaker.\n");
printf("Then try to find the smallest number of buffers 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;
/* Ask PortAudio for the recommended minimum number of buffers. */
numBuffers = minBuffers = Pa_GetMinNumBuffers( framesPerBuffer, sampleRate );
printf("NumBuffers set to %d based on a call to Pa_GetMinNumBuffers()\n", numBuffers );
/* Try different numBuffers in a loop. */
go = 1;
while( go )
{
printf("Latency = framesPerBuffer * numBuffers = %d * %d = %d frames = %d msecs.\n",
framesPerBuffer, numBuffers, framesPerBuffer*numBuffers,
(int)((1000 * framesPerBuffer * numBuffers) / sampleRate) );
err = Pa_OpenStream(
&stream,
paNoDevice,
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
Pa_GetDefaultOutputDeviceID(), /* default output device */
2, /* stereo output */
paFloat32, /* 32 bit floating point output */
NULL,
sampleRate,
framesPerBuffer,
numBuffers, /* number of buffers */
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 number of buffers. */
printf("\nMove windows around to see if the sound glitches.\n");
printf("NumBuffers currently %d, enter new number, or 'q' to quit: ", numBuffers );
gets( str );
if( str[0] == 'q' ) go = 0;
else
{
numBuffers = atol( str );
if( numBuffers < minBuffers )
{
printf( "numBuffers below minimum of %d! Set to minimum!!!\n", minBuffers );
numBuffers = minBuffers;
}
}
/* 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;
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 ) );
}

322
lib/portaudio/pa_tests/paqa_devs.c Normal file
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/*
* $Id: paqa_devs.c,v 1.2 2003/03/02 08:01:40 dmazzoni Exp $
* paqa_devs.c
* 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
*
* 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, PaDeviceID deviceID, double sampleRate,
int numChannels );
static int TestAdvance( int mode, PaDeviceID deviceID, double sampleRate,
int numChannels, PaSampleFormat format );
static int QaCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, 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( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
unsigned long i;
short phase;
PaQaData *data = (PaQaData *) userData;
(void) inputBuffer;
(void) outTime;
/* Play simple 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 )
{
AddTraceMessage("QaCallback: running. framesLeft", data->framesLeft );
data->framesLeft -= framesPerBuffer;
return 0;
}
else
{
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 );
}
/*******************************************************************
* Try each output device, through its full range of capabilities. */
static void TestDevices( int mode )
{
int id,jc,kr;
int maxChannels;
const PaDeviceInfo *pdi;
int numDevices = Pa_CountDevices();
/* Iterate through all devices. */
for( id=0; id<numDevices; id++ )
{
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 legal sample rate. */
if( pdi->numSampleRates == -1 )
{
double low, high;
low = pdi->sampleRates[0];
high = pdi->sampleRates[1];
if( low < 8000.0 ) low = 8000.0;
TestFormats( mode, id, low, jc );
#define TESTSR(sr) {if(((sr)>=low) && ((sr)<=high)) TestFormats( mode, id, (sr), jc ); }
TESTSR(11025.0);
TESTSR(22050.0);
TESTSR(34567.0);
TESTSR(44100.0);
TestFormats( mode, id, high, jc );
}
else
{
for( kr=0; kr<pdi->numSampleRates; kr++ )
{
TestFormats( mode, id, pdi->sampleRates[kr], jc );
}
}
}
}
}
/*******************************************************************/
static void TestFormats( int mode, PaDeviceID deviceID, double sampleRate,
int numChannels )
{
TestAdvance( mode, deviceID, sampleRate, numChannels, paFloat32 ); /* */
TestAdvance( mode, deviceID, sampleRate, numChannels, paInt16 ); /* */
TestAdvance( mode, deviceID, sampleRate, numChannels, paInt32 ); /* */
}
/*******************************************************************/
static int TestAdvance( int mode, PaDeviceID deviceID, double sampleRate,
int numChannels, PaSampleFormat format )
{
PortAudioStream *stream = NULL;
PaError result;
PaQaData myData;
#define FRAMES_PER_BUFFER (64)
printf("------ TestAdvance: %s, device = %d, rate = %g, numChannels = %d, format = %d -------\n",
( mode == MODE_INPUT ) ? "INPUT" : "OUTPUT",
deviceID, sampleRate, numChannels, format);
fflush(stdout);
/* 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;
}
EXPECT( ((result = Pa_OpenStream(
&stream,
( mode == MODE_INPUT ) ? deviceID : paNoDevice,
( mode == MODE_INPUT ) ? numChannels : 0,
format,
NULL,
( mode == MODE_OUTPUT ) ? deviceID : paNoDevice,
( mode == MODE_OUTPUT ) ? numChannels : 0,
format,
NULL,
sampleRate,
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 */
QaCallback,
&myData )
) == 0) );
if( stream )
{
PaTimestamp oldStamp, newStamp;
unsigned long oldFrames;
int minDelay = ( mode == MODE_INPUT ) ? 1000 : 400;
int minNumBuffers = Pa_GetMinNumBuffers( FRAMES_PER_BUFFER, sampleRate );
int msec = (int) ((minNumBuffers * 3 * 1000.0 * FRAMES_PER_BUFFER) / sampleRate);
if( msec < minDelay ) msec = minDelay;
printf("msec = %d\n", msec); /**/
EXPECT( ((result=Pa_StartStream( stream )) == 0) );
/* Check to make sure PortAudio is advancing timeStamp. */
result = paNoError;
oldStamp = Pa_StreamTime(stream);
fflush(stdout);
Pa_Sleep(msec);
newStamp = Pa_StreamTime(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 = %d, myData.framesLeft = %d\n", oldFrames, myData.framesLeft ); /**/
EXPECT( (oldFrames > myData.framesLeft) );
EXPECT( ((result=Pa_CloseStream( stream )) == 0) );
stream = NULL;
}
error:
if( stream != NULL ) Pa_CloseStream( stream );
fflush(stdout);
return result;
}

330
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/*
* $Id: paqa_errs.c,v 1.2 2003/03/02 08:01:41 dmazzoni Exp $
* paqa_devs.c
* Self Testing Quality Assurance app for PortAudio
* Do lots of bad things to test error reporting.
*
* 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"
/****************************************** Definitions ***********/
#define MODE_INPUT (0)
#define MODE_OUTPUT (1)
#define FRAMES_PER_BUFFER (64)
#define SAMPLE_RATE (44100.0)
#define NUM_BUFFERS (0)
typedef struct PaQaData
{
unsigned long framesLeft;
int numChannels;
int bytesPerSample;
int mode;
}
PaQaData;
/****************************************** Prototypes ***********/
static void TestDevices( int mode );
static void TestFormats( int mode, PaDeviceID deviceID, double sampleRate,
int numChannels );
static int TestBadOpens( void );
static int TestBadActions( void );
static int QaCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, 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( msg, _exp) \
do \
{ \
if ((_exp)) {\
gNumPassed++; \
} \
else { \
printf("\nERROR %s\n - 0x%x - %s for %s\n", (msg), result, Pa_GetErrorText(result), #_exp ); \
gNumFailed++; \
goto error; \
} \
} while(0)
#define HOPEFOR( msg, _exp) \
do \
{ \
if ((_exp)) {\
gNumPassed++; \
} \
else { \
printf("\nERROR %s\n - 0x%x - %s for %s\n", (msg), 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( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
unsigned long i;
unsigned char *out = (unsigned char *) outputBuffer;
PaQaData *data = (PaQaData *) userData;
(void) inputBuffer; /* Prevent "unused variable" warnings. */
(void) outTime;
/* 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;
}
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError result;
EXPECT( "init", ((result=Pa_Initialize()) == 0) );
TestBadActions();
TestBadOpens();
error:
Pa_Terminate();
printf("QA Report: %d passed, %d failed.\n", gNumPassed, gNumFailed );
return 0;
}
/*******************************************************************/
static int TestBadOpens( void )
{
PortAudioStream *stream = NULL;
PaError result;
PaQaData myData;
/* Setup data for synthesis thread. */
myData.framesLeft = (unsigned long) (SAMPLE_RATE * 100); /* 100 seconds */
myData.numChannels = 1;
myData.mode = MODE_OUTPUT;
HOPEFOR( "No devices specified.",(
(result = Pa_OpenStream(
&stream,
paNoDevice, 0, paFloat32, NULL,
paNoDevice, 0, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidDeviceId) );
HOPEFOR( "Out of range input device specified.",(
(result = Pa_OpenStream(
&stream,
Pa_CountDevices(), 0, paFloat32, NULL,
paNoDevice, 0, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidDeviceId) );
HOPEFOR( "Out of range output device specified.",(
(result = Pa_OpenStream(
&stream,
paNoDevice, 0, paFloat32, NULL,
Pa_CountDevices(), 0, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidDeviceId) );
HOPEFOR( "Zero input channels.",(
(result = Pa_OpenStream(
&stream,
Pa_GetDefaultInputDeviceID(), 0, paFloat32, NULL,
paNoDevice, 0, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidChannelCount) );
HOPEFOR( "Zero output channels.",(
(result = Pa_OpenStream(
&stream,
paNoDevice, 0, paFloat32, NULL,
Pa_GetDefaultOutputDeviceID(), 0, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidChannelCount) );
HOPEFOR( "Nonzero input channels but no device.",(
(result = Pa_OpenStream(
&stream,
Pa_GetDefaultInputDeviceID(), 2, paFloat32, NULL,
paNoDevice, 2, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidChannelCount) );
HOPEFOR( "Nonzero output channels but no device.",(
(result = Pa_OpenStream(
&stream,
paNoDevice, 2, paFloat32, NULL,
Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidChannelCount) );
HOPEFOR( "NULL stream pointer.",(
(result = Pa_OpenStream(
NULL,
paNoDevice, 0, paFloat32, NULL,
Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paBadStreamPtr) );
HOPEFOR( "Low sample rate.",(
(result = Pa_OpenStream(
&stream,
paNoDevice, 0, paFloat32, NULL,
Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL,
1.0, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidSampleRate) );
HOPEFOR( "High sample rate.",(
(result = Pa_OpenStream(
&stream,
paNoDevice, 0, paFloat32, NULL,
Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL,
10000000.0, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidSampleRate) );
HOPEFOR( "NULL callback.",(
(result = Pa_OpenStream(
&stream,
paNoDevice, 0, paFloat32, NULL,
Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
NULL,
&myData )
) == paNullCallback) );
HOPEFOR( "Bad flag.",(
(result = Pa_OpenStream(
&stream,
paNoDevice, 0, paFloat32, NULL,
Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
(1<<3),
QaCallback,
&myData )
) == paInvalidFlag) );
#if 1 /* FIXME - this is legal for some implementations. */
HOPEFOR( "Use input device as output device.",(
(result = Pa_OpenStream(
&stream,
paNoDevice, 0, paFloat32, NULL,
Pa_GetDefaultInputDeviceID(), 2, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidDeviceId) );
HOPEFOR( "Use output device as input device.",(
(result = Pa_OpenStream(
&stream,
Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL,
paNoDevice, 0, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == paInvalidDeviceId) );
#endif
if( stream != NULL ) Pa_CloseStream( stream );
return result;
}
/*******************************************************************/
static int TestBadActions( void )
{
PortAudioStream *stream = NULL;
PaError result;
PaQaData myData;
/* Setup data for synthesis thread. */
myData.framesLeft = (unsigned long) (SAMPLE_RATE * 100); /* 100 seconds */
myData.numChannels = 1;
myData.mode = MODE_OUTPUT;
/* Default output. */
EXPECT( "TestBadActions", ((result = Pa_OpenStream(
&stream,
paNoDevice, 0, paFloat32, NULL,
Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL,
SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS,
paClipOff,
QaCallback,
&myData )
) == 0) );
HOPEFOR( "start", ((result = Pa_StartStream( NULL )) == paBadStreamPtr) );
HOPEFOR( "stop", ((result = Pa_StopStream( NULL )) == paBadStreamPtr) );
HOPEFOR( "active?", ((result = Pa_StreamActive( NULL )) == paBadStreamPtr) );
HOPEFOR( "close", ((result = Pa_CloseStream( NULL )) == paBadStreamPtr) );
HOPEFOR( "time?", ((result = (PaError)Pa_StreamTime( NULL )) != 0) );
HOPEFOR( "CPULoad?", ((result = (PaError)Pa_GetCPULoad( NULL )) != 0) );
error:
if( stream != NULL ) Pa_CloseStream( stream );
return result;
}

114
lib/portaudio/pa_tests/patest1.c Normal file
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/*
$Id: patest1.c,v 1.3 2003/03/02 08:01:41 dmazzoni Exp $
patest1.c
Ring modulate the audio input with a 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;
/* Check to see if any input data is available. */
if(inputBuffer == NULL) return 0;
if( data->sampsToGo < bufferFrames )
{
framesToCalc = data->sampsToGo;
finished = 1;
}
for( i=0; 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<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 * 20; // 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.,
512, /* small buffers */
0, /* let PA determine number of 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;
}

156
lib/portaudio/pa_tests/patest_clip.c Normal file
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/*
* $Id: patest_clip.c,v 1.2 2003/03/02 08:01:41 dmazzoni Exp $
* patest_clip.c
* Play a sine wave using the Portable Audio api for several seconds
* at an amplitude that would require clipping.
*
* 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 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( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
float amplitude = data->amplitude;
unsigned int i;
(void) inputBuffer; /* Prevent "unused variable" warnings. */
(void) outTime;
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 )
{
PortAudioStream *stream;
PaError err;
data->left_phase = data->right_phase = 0;
data->amplitude = amplitude;
err = Pa_Initialize();
if( err != paNoError ) goto error;
err = Pa_OpenStream(
&stream,
paNoDevice,/* default input device */
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
Pa_GetDefaultOutputDeviceID(), /* default output device */
2, /* stereo output */
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE,
1024,
0, /* number of buffers, if zero then use default minimum */
flags, /* we won't output out of range samples so don't bother clipping them */
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_GetCPULoad( stream ) );
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
return paNoError;
error:
return err;
}

152
lib/portaudio/pa_tests/patest_dither.c Normal file
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@ -0,0 +1,152 @@
/*
* $Id: patest_dither.c,v 1.2 2003/03/02 08:01:42 dmazzoni Exp $
* patest_dither.c
* 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
*
* 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, PaStreamFlags flags, float amplitude );
static int sineCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, 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 sineCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
float amplitude = data->amplitude;
unsigned int i;
(void) outTime;
(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;
}
/*******************************************************************/
int main(void);
int main(void)
{
PaError err;
paTestData DATA;
int i;
float amplitude = 32.0 / (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 error;
printf("\nClip.\n");
fflush(stdout);
err = PlaySine( &DATA, paDitherOff, amplitude );
if( err < 0 ) goto error;
printf("\nClip and Dither.\n");
fflush(stdout);
err = PlaySine( &DATA, paNoFlag, amplitude );
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, PaStreamFlags flags, float amplitude )
{
PortAudioStream *stream;
PaError err;
data->left_phase = data->right_phase = 0;
data->amplitude = amplitude;
err = Pa_Initialize();
if( err != paNoError ) goto error;
err = Pa_OpenStream(
&stream,
paNoDevice,/* default input device */
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
Pa_GetDefaultOutputDeviceID(), /* default output device */
2, /* stereo output */
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE,
1024,
0, /* number of buffers, if zero then use default minimum */
flags, /* we won't output out of range samples so don't bother clipping them */
sineCallback,
(void *)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_GetCPULoad( stream ) );
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
return paNoError;
error:
return err;
}

176
lib/portaudio/pa_tests/patest_latency.c Normal file
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@ -0,0 +1,176 @@
/*
* $Id: patest_latency.c,v 1.2 2003/03/02 08:01:42 dmazzoni Exp $
* 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
*
* 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 (64)
#if 0
#define MIN_LATENCY_MSEC (2000)
#define NUM_BUFFERS ((MIN_LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000))
#else
#define NUM_BUFFERS (0)
#endif
#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( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
int 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;
}
return 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *stream;
PaError err;
paTestData data;
int i;
int done = 0;
printf("PortAudio Test: enter letter then hit ENTER. numBuffers = %d\n", NUM_BUFFERS );
/* 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 );
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("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;
}

168
lib/portaudio/pa_tests/patest_leftright.c Normal file
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@ -0,0 +1,168 @@
/*
* $Id: patest_leftright.c,v 1.3 2003/03/02 08:01:42 dmazzoni Exp $
* patest_leftright.c
* Play different tone sine waves that alternate between left and right channel.
* The low tone should be on the left channel.
*
* 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 (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( 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++ )
{
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)
{
PortAudioStream *stream;
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;
err = Pa_OpenStream(
&stream,
paNoDevice,/* default input device */
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
Pa_GetDefaultOutputDeviceID(), /* default output device */
2, /* stereo output */
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("Play for several seconds.\n");
timeout = NUM_SECONDS * 4;
while( timeout > 0 )
{
Pa_Sleep( 300 );
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;
}

137
lib/portaudio/pa_tests/patest_longsine.c Normal file
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@ -0,0 +1,137 @@
/*
* $Id: patest_longsine.c,v 1.2 2003/03/02 08:01:42 dmazzoni Exp $
* patest_longsine.c
* Play a sine wave using the Portable Audio api until ENTER hit.
*
* 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 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( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned int i;
(void) outTime; /* Prevent unused variable warnings. */
(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 0;
}
/*******************************************************************/
int main(void);
int main(void)
{
PortAudioStream *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;
err = Pa_OpenStream(
&stream,
paNoDevice,/* default input device */
0, /* no input */
paFloat32, /* 32 bit floating point input */
NULL,
Pa_GetDefaultOutputDeviceID(), /* default output device */
2, /* stereo output */
paFloat32, /* 32 bit floating point output */
NULL,
SAMPLE_RATE,
256, /* 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 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;
}

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