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****************%%%%%%%%%%%%%%%%----------------
lame internal error: VBR min %d kbps or VBR max %d kbps not allowed.
[%*i]
%3d%s %.*s%.*s%s
%3d%s %.*s%.*s%*s
%*u:%02u:%02u
%02u:%02u
%.*s%s%.*s%s
%.*s%s%.*s%*s
%5.1f
kbps
mono
IS
LR
MS
%%
long
switch
short
mixed
%5.1f
UUUUUUUUUUUUUUUU
?MP3 file
fskip problem: Mostly the return status of functions is not evaluated, so it is more secure to pollute <stderr>.
Error: internal problem!
Error: number of channels has changed in %s - not supported
Error: sample frequency has changed in %s - not supported
Unsupported number of channels: %d
Unsupported sample rate: %d
WAVEfmt
Unsigned input only supported with bitwidth 8
Only 8, 16, 24 and 32 bit input files supported
Error: unexpected number of samples to read: %d
Error reading input file
Unsupported data format: 0x%04X
Unsupported bits per sample: %d
ERROR: input sound data is not PCM
ERROR: input sound data is not 8, 16, 24 or 32 bits
ERROR: input sound data is not mono or stereo
ERROR: block size of input sound data is not 0 bytes
Can't rewind stream to audio data position
Warning: corrupt or unsupported WAVE format
Warning: corrupt or unsupported AIFF format
Warning: unsupported audio format
Error reading headers in mp3 input file %s.
Could not find "%s".
sorry, vorbis support in LAME is deprecated.
Assuming raw pcm input file
: Forcing byte-swapping
Could not close audio input file
Album ID found. length=%i
Input file is freeformat.
fail to sync...
Input file and Output file are the same. Abort.
Can't init infile '%s'
Can't init outfile '%s'
MPEG-%u%s Layer %s
raw PCM data
Microsoft WAVE
SGI/Apple AIFF
unknown
Internal error. Aborting.
input: %s%s(%g kHz, %i channel%s,
<stdin>
output: %s%s(16 bit, Microsoft WAVE)
<stdout>
skipping initial %i samples (encoder+decoder delay)
skipping final %i samples (encoder padding-decoder delay)
WAVE file contains 0 PCM samples
Very huge WAVE file, can't set filesize accordingly
ReplayGain: %s%.1fdB
WARNING: ReplayGain exceeds the -51dB to +51dB range. Such a result is too
high to be stored in the header.
WARNING: clipping occurs at the current gain. Set your decoder to decrease
the gain by at least %.1fdB or encode again
using --scale %.2f
or less (the value under --scale is approximate).
using --scale <arg>
(For a suggestion on the optimal value of <arg> encode
with --scale 1 first)
The waveform does not clip and is less than 0.1dB away from full scale.
The waveform does not clip and is at least %.1fdB away from full scale.
Writing LAME Tag...
Error writing LAME-tag frame: buffer too small: buffer size=%d frame size=%d
fatal error: can't update LAME-tag frame!
Error writing LAME-tag
Error writing ID3v1 tag: buffer too small: buffer size=%d ID3v1 size=%d
Error writing ID3v1 tag
Error writing ID3v2 tag
mp3 buffer is not big enough...
mp3 internal error: error code=%i
Error writing mp3 output
processing nogap file %d: %s
fatal error during initialization
==> Priority set to Low.
==> Priority set to Normal.
==> Priority set to High.
N881*888#88888
8888888
88888888888
NXEXXX7XXXX.X XXXXX
WARNING: argument missing for '%s'
LAME %s version %s (%s)
LAME version %s (%s)
LAME %s version %s
%*s(%s)
LAME version %s
%*s(%s)
Copyright (c) 1999-2011 by The LAME Project
Copyright (c) 1999,2000,2001 by Mark Taylor
Copyright (c) 1998 by Michael Cheng
Copyright (c) 1995,1996,1997 by Michael Hipp: mpglib
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Library General Public License for more details.
You should have received a copy of the GNU Library General Public
License along with this program. If not, see
<http://www.gnu.org/licenses/>.
usage: %s [options] <infile> [outfile]
<infile> and/or <outfile> can be "-", which means stdin/stdout.
"%s --help" for general usage information
"%s --preset help" for information on suggested predefined settings
"%s --longhelp"
or "%s -?" for a complete options list
usage: %s [options] <infile> [outfile]
<infile> and/or <outfile> can be "-", which means stdin/stdout.
RECOMMENDED:
lame -V2 input.wav output.mp3
OPTIONS:
-b bitrate set the bitrate, default 128 kbps
-h higher quality, but a little slower.
-f fast mode (lower quality)
-V n quality setting for VBR. default n=%i
0=high quality,bigger files. 9.999=smaller files
--preset type type must be "medium", "standard", "extreme", "insane",
or a value for an average desired bitrate and depending
on the value specified, appropriate quality settings will
be used.
"--preset help" gives more info on these
--priority type sets the process priority
0,1 = Low priority
2 = normal priority
3,4 = High priority
--help id3 ID3 tagging related options
--longhelp full list of options
--license print License information
ID3 tag options:
--tt <title> audio/song title (max 30 chars for version 1 tag)
--ta <artist> audio/song artist (max 30 chars for version 1 tag)
--tl <album> audio/song album (max 30 chars for version 1 tag)
--ty <year> audio/song year of issue (1 to 9999)
--tc <comment> user-defined text (max 30 chars for v1 tag, 28 for v1.1)
--tn <track[/total]> audio/song track number and (optionally) the total
number of tracks on the original recording. (track
and total each 1 to 255. just the track number
creates v1.1 tag, providing a total forces v2.0).
--tg <genre> audio/song genre (name or number in list)
--ti <file> audio/song albumArt (jpeg/png/gif file, v2.3 tag)
--tv <id=value> user-defined frame specified by id and value (v2.3 tag)
syntax: --tv "TXXX=description=content"
--add-id3v2 force addition of version 2 tag
--id3v1-only add only a version 1 tag
--id3v2-only add only a version 2 tag
--id3v2-utf16 add following options in unicode text encoding
--id3v2-latin1 add following options in latin-1 text encoding
--space-id3v1 pad version 1 tag with spaces instead of nulls
--pad-id3v2 same as '--pad-id3v2-size 128'
--pad-id3v2-size <value> adds version 2 tag, pad with extra <value> bytes
--genre-list print alphabetically sorted ID3 genre list and exit
--ignore-tag-errors ignore errors in values passed for tags
Note: A version 2 tag will NOT be added unless one of the input fields
won't fit in a version 1 tag (e.g. the title string is longer than 30
characters), or the '--add-id3v2' or '--id3v2-only' options are used,
or output is redirected to stdout.
Note: Almost all of the following switches aren't available in this build!
ATH related:
--noath turns ATH down to a flat noise floor
--athshort ignore GPSYCHO for short blocks, use ATH only
--athonly ignore GPSYCHO completely, use ATH only
--athtype n selects between different ATH types [0-4]
--athlower x lowers ATH by x dB
--athaa-type n ATH auto adjust: 0 'no' else 'loudness based'
--athaa-sensitivity x activation offset in -/+ dB for ATH auto-adjustment
PSY related:
--short use short blocks when appropriate
--noshort do not use short blocks
--allshort use only short blocks
(1) --temporal-masking x x=0 disables, x=1 enables temporal masking effect
--nssafejoint M/S switching criterion
--nsmsfix <arg> M/S switching tuning [effective 0-3.5]
(2) --interch x adjust inter-channel masking ratio
--ns-bass x adjust masking for sfbs 0 - 6 (long) 0 - 5 (short)
--ns-alto x adjust masking for sfbs 7 - 13 (long) 6 - 10 (short)
--ns-treble x adjust masking for sfbs 14 - 21 (long) 11 - 12 (short)
--ns-sfb21 x change ns-treble by x dB for sfb21
--shortthreshold x,y short block switching threshold,
x for L/R/M channel, y for S channel
-Z [n] always do calculate short block maskings
Noise Shaping related:
(1) --substep n use pseudo substep noise shaping method types 0-2
(1) -X n[,m] selects between different noise measurements
n for long block, m for short. if m is omitted, m = n
1: CBR, ABR and VBR-old encoding modes only
2: ignored
OPTIONS:
Input options:
--scale <arg> scale input (multiply PCM data) by <arg>
--scale-l <arg> scale channel 0 (left) input (multiply PCM data) by <arg>
--scale-r <arg> scale channel 1 (right) input (multiply PCM data) by <arg>
--swap-channel swap L/R channels
--ignorelength ignore file length in WAV header
--gain <arg> apply Gain adjustment in decibels, range -20.0 to +12.0
--mp1input input file is a MPEG Layer I file
--mp2input input file is a MPEG Layer II file
--mp3input input file is a MPEG Layer III file
--nogap <file1> <file2> <...>
gapless encoding for a set of contiguous files
--nogapout <dir>
output dir for gapless encoding (must precede --nogap)
--nogaptags allow the use of VBR tags in gapless encoding
--out-dir <dir> output dir, must exist
Input options for RAW PCM:
-r input is raw pcm
-s sfreq sampling frequency of input file (kHz) - default 44.1 kHz
--signed input is signed (default)
--unsigned input is unsigned
--bitwidth w input bit width is w (default 16)
-x force byte-swapping of input
--little-endian input is little-endian (default)
--big-endian input is big-endian
-a downmix from stereo to mono file for mono encoding
Operational options:
-m <mode> (j)oint, (s)imple, (f)orce, (d)ual-mono, (m)ono (l)eft (r)ight
default is (j)
joint = Uses the best possible of MS and LR stereo
simple = force LR stereo on all frames
force = force MS stereo on all frames.
--preset type type must be "medium", "standard", "extreme", "insane",
or a value for an average desired bitrate and depending
on the value specified, appropriate quality settings will
be used.
"--preset help" gives more info on these
--comp <arg> choose bitrate to achieve a compression ratio of <arg>
--replaygain-fast compute RG fast but slightly inaccurately (default)
--replaygain-accurate compute RG more accurately and find the peak sample
--noreplaygain disable ReplayGain analysis
--clipdetect enable --replaygain-accurate and print a message whether
clipping occurs and how far the waveform is from full scale
--flush flush output stream as soon as possible
--freeformat produce a free format bitstream
--decode input=mp3 file, output=wav
-t disable writing wav header when using --decode
Verbosity:
--disptime <arg>print progress report every arg seconds
-S don't print progress report, VBR histograms
--nohist disable VBR histogram display
--quiet don't print anything on screen
--silent don't print anything on screen, but fatal errors
--brief print more useful information
--verbose print a lot of useful information
Noise shaping & psycho acoustic algorithms:
-q <arg> <arg> = 0...9. Default -q 3
-q 0: Highest quality, very slow
-q 9: Poor quality, but fast
-h Same as -q 2.
-f Same as -q 7. Fast, ok quality
CBR (constant bitrate, the default) options:
-b <bitrate> set the bitrate in kbps, default 128 kbps
--cbr enforce use of constant bitrate
ABR options:
--abr <bitrate> specify average bitrate desired (instead of quality)
VBR options:
-V n quality setting for VBR. default n=%i
0=high quality,bigger files. 9=smaller files
-v the same as -V 4
--vbr-old use old variable bitrate (VBR) routine
--vbr-new use new variable bitrate (VBR) routine (default)
-Y lets LAME ignore noise in sfb21, like in CBR
(Default for V3 to V9.999)
-b <bitrate> specify minimum allowed bitrate, default 32 kbps
-B <bitrate> specify maximum allowed bitrate, default 320 kbps
-F strictly enforce the -b option, for use with players that
do not support low bitrate mp3
-t disable writing LAME Tag
-T enable and force writing LAME Tag
MP3 header/stream options:
-e <emp> de-emphasis n/5/c (obsolete)
-c mark as copyright
-o mark as non-original
-p error protection. adds 16 bit checksum to every frame
(the checksum is computed correctly)
--nores disable the bit reservoir
--strictly-enforce-ISO comply as much as possible to ISO MPEG spec
--buffer-constraint <constraint> available values for constraint:
default, strict, maximum
Filter options:
--lowpass <freq> frequency(kHz), lowpass filter cutoff above freq
--lowpass-width <freq> frequency(kHz) - default 15%% of lowpass freq
--highpass <freq> frequency(kHz), highpass filter cutoff below freq
--highpass-width <freq> frequency(kHz) - default 15%% of highpass freq
--resample <sfreq> sampling frequency of output file(kHz)- default=automatic
MS-Windows-specific options:
--priority <type> sets the process priority:
0,1 = Low priority (IDLE_PRIORITY_CLASS)
2 = normal priority (NORMAL_PRIORITY_CLASS, default)
3,4 = High priority (HIGH_PRIORITY_CLASS))
Note: Calling '--priority' without a parameter will select priority 0.
--license print License information
Platform specific:
--noasm <instructions> disable assembly optimizations for mmx/3dnow/sse
MPEG-%-3s layer III sample frequencies (kHz): %2d %2d %g
bitrates (kbps):
The --preset switches are aliases over LAME settings.
To activate these presets:
For VBR modes (generally highest quality):
--preset medium This preset should provide near transparency to most
people on most music.
--preset standard This preset should generally be transparent to most
people on most music and is already quite high
in quality.
--preset extreme If you have extremely good hearing and similar
equipment, this preset will generally provide
slightly higher quality than the "standard" mode.
For CBR 320kbps (highest quality possible from the --preset switches):
--preset insane This preset will usually be overkill for most people
and most situations, but if you must have the
absolute highest quality with no regard to filesize,
this is the way to go.
For ABR modes (high quality per given bitrate but not as high as VBR):
--preset <kbps> Using this preset will usually give you good quality
at a specified bitrate. Depending on the bitrate
entered, this preset will determine the optimal
settings for that particular situation. For example:
"--preset 185" activates this preset and uses 185
as an average kbps.
"cbr" - If you use the ABR mode (read above) with a significant
bitrate such as 80, 96, 112, 128, 160, 192, 224, 256, 320,
you can use the "cbr" option to force CBR mode encoding
instead of the standard abr mode. ABR does provide higher
quality but CBR may be useful in situations such as when
streaming an mp3 over the internet may be important.
For example:
--preset standard <input file> <output file>
or --preset cbr 192 <input file> <output file>
or --preset 172 <input file> <output file>
or --preset extreme <input file> <output file>
A few aliases are also available for ABR mode:
phone => 16kbps/mono phon+/lw/mw-eu/sw => 24kbps/mono
mw-us => 40kbps/mono voice => 56kbps/mono
fm/radio/tape => 112kbps hifi => 160kbps
cd => 192kbps studio => 256kbps
studio
medium
insane
extreme
standard
Error: The bitrate specified is out of the valid range for this preset
When using this mode you must enter a value between "32" and "320"
For further information try: "%s --preset help"
Error: You did not enter a valid profile and/or options with --preset
Available profiles are:
medium
standard
extreme
insane
<cbr> (ABR Mode) - The ABR Mode is implied. To use it,
simply specify a bitrate. For example:
"--preset 185" activates this
preset and uses 185 as an average kbps.
Some examples:
or "%s --preset standard <input file> <output file>"
or "%s --preset cbr 192 <input file> <output file>"
or "%s --preset 172 <input file> <output file>"
or "%s --preset extreme <input file> <output file>"
For further information try: "%s --preset help"
%3d %s
Illegal resample frequency: %.3f kHz
error: output file name too long
Could not find: '%s'.
Insufficient memory for reading the albumart.
Read error: '%s'.
Unsupported image: '%s'.
Specify JPEG/PNG/GIF image
WARNING: ignoring developer-only switch --%s %s
WARNING: ignoring developer-only switch --%s
unknown buffer constraint '%s'
maximum
strict
default
The track number has to be between 1 and 255 for ID3v1.
The track number has to be between 1 and 255 for ID3v1, ignored for ID3v1.
Unknown ID3v1 genre number: '%s'.
Unknown ID3v1 genre: '%s'.
Unknown ID3v1 genre: '%s'. Setting ID3v1 genre to 'Other'
Internal error.
Invalid field value: '%s'. Ignored
Must specify lowpass with --lowpass freq, freq >= 0.001 kHz
Must specify lowpass width with --lowpass-width freq, freq >= 0.001 kHz
Must specify highpass with --highpass freq, freq >= 0.001 kHz
Must specify highpass width with --highpass-width freq, freq >= 0.001 kHz
Must specify compression ratio >= 1.0
%s: %s argument length (%d) exceeds limit (%d)
%s: unrecognized option --%s
debug-file
athaa-type
athtype
athlower
athshort
athonly
sfscale
sbgain
substep
no-preset-tune
athcurve
maskingadjustshort
maskingadjust
shortthreshold
ns-sfb21
ns-treble
ns-alto
ns-bass
nsmsfix
nssafejoint
nspsytune
temporal-masking
interch
notemp
allshort
noshort
athaa-sensitivity
ignorelength
swap-channel
out-dir
nogapout
nogaptags
disptime
alt-preset
preset
longhelp
license
version
verbose
silent
highpass-width
highpass
lowpass-width
lowpass
genre-list
pad-id3v2-size
pad-id3v2
space-id3v1
id3v2-only
id3v1-only
add-id3v2
ignore-tag-errors
id3v2-latin1
id3v2-ucs2
id3v2-utf16
priority
nohist
clipdetect
noreplaygain
replaygain-accurate
replaygain-fast
freeformat
scale-r
scale-l
buffer-constraint
strictly-enforce-ISO
decode-mp3delay
decode
ogginput
mp3input
mp2input
mp1input
big-endian
little-endian
unsigned
signed
bitwidth
vbr-mtrh
vbr-new
vbr-old
resample
%s: -m mode must be s/d/f/j/m/l/r not %s
WARNING: -%c is obsolete.
%s: -e emp must be n/5/c not %s
%s: unrecognized option -%c
Error: 'nogap option'. Calling program does not allow nogap option, or
you have exceeded maximum number of input files for the nogap option
%s: excess arg %s
combination of nogap and decode not supported!
Note: Disabling VBR Xing/Info tag since it interferes with --nogap
For free format, specify a bitrate between 8 and 640 kbps
with the -b <bitrate> option
LAMEOPT
? %2u:%02u%c
%2lu:%02u:%02u%c
%6lu h%c
Frame | CPU time/estim | REAL time/estim | play/CPU | ETA
0/ ( 0%%)| 0:00/ : | 0:00/ : | x| :
%6i/%-6i
(%2d%%)|
(%3.3d%%)|
%9.4fx|
%#9.5gx|
Encoding %s%s to %s
Encoding as %g kHz
stereo
j-stereo
dual-ch
single-ch
force-ms
%s MPEG-%u%s Layer III VBR(q=%g) qval=%i
%s MPEG-%u%s Layer III VBR(q=%g)
%s MPEG-%u%s Layer III (%gx) average %d kbps qval=%i
%s MPEG-%u%s Layer III (%gx) %3d kbps qval=%i
Frame#%6i/%-6i %3i kbps
%s %c
Aff>A
Warning: highpass filter disabled. highpass frequency too small
LAME %s %s (%s)
MMX (ASM used)
3DNow! (ASM used)
3DNow!
SSE (ASM used)
CPU features: %s
Autoconverting from stereo to mono. Setting encoding to mono mode.
Resampling: input %g kHz output %g kHz
Using polyphase highpass filter, transition band: %5.0f Hz - %5.0f Hz
Using polyphase lowpass filter, transition band: %5.0f Hz - %5.0f Hz
polyphase lowpass filter disabled
Warning: many decoders cannot handle free format bitstreams
Warning: many decoders cannot handle free format bitrates >320 kbps (see documentation)
scaling: %g
ch0 (left) scaling: %g
ch1 (right) scaling: %g
normal
best (outside loop)
best (inside loop, slow)
huffman search: %s
experimental Y=%d
stream format:
MPEG-%s Layer 3
joint stereo
dual channel
not set (error)
unknown (error)
%d channel - %s
padding: %s
(default)
(free format)
constant bitrate - CBR %s
variable bitrate - ABR %s
variable bitrate - VBR rh %s
variable bitrate - VBR mt %s
variable bitrate - VBR mtrh %s
?? oops, some new one ??
using LAME Tag
psychoacoustic:
allowed
channel coupled
dispensed
forced
using short blocks: %s
subblock gain: %d
adjust masking: %g dB
adjust masking short: %g dB
quantization comparison: %d
^ comparison short blocks: %d
noise shaping: %d
^ amplification: %d
^ stopping: %d
the only masking for short blocks
the only masking
not used
ATH: %s
^ type: %d
^ shape: %g%s
(only for type 4)
^ level adjustement: %g dB
^ adjust type: %d
^ adjust sensitivity power: %f
experimental psy tunings by Naoki Shibata
adjust masking bass=%g dB, alto=%g dB, treble=%g dB, sfb21=%g dB
using temporal masking effect: %s
interchannel masking ratio: %g
Error: can't allocate in_buffer buffer
Classic Rock
Country
Grunge
Hip-Hop
New Age
Oldies
Reggae
Techno
Industrial
Alternative
Death Metal
Pranks
Soundtrack
Euro-Techno
Ambient
Trip-Hop
Jazz+Funk
Fusion
Trance
Classical
Instrumental
Sound Clip
Gospel
Alternative Rock
Meditative
Instrumental Pop
Instrumental Rock
Ethnic
Gothic
Darkwave
Techno-Industrial
Electronic
Pop-Folk
Eurodance
Southern Rock
Comedy
Gangsta
Top 40
Christian Rap
Pop/Funk
Jungle
Native US
Cabaret
New Wave
Psychedelic
Showtunes
Trailer
Tribal
Acid Punk
Acid Jazz
Musical
Rock & Roll
Hard Rock
Folk-Rock
National Folk
Fast Fusion
Revival
Celtic
Bluegrass
Avantgarde
Gothic Rock
Progressive Rock
Psychedelic Rock
Symphonic Rock
Slow Rock
Big Band
Chorus
Easy Listening
Acoustic
Humour
Speech
Chanson
Chamber Music
Sonata
Symphony
Booty Bass
Primus
Porn Groove
Satire
Slow Jam
Folklore
Ballad
Power Ballad
Rhythmic Soul
Freestyle
Punk Rock
Drum Solo
A Cappella
Euro-House
Dance Hall
Drum & Bass
Club-House
Hardcore
Terror
BritPop
Negerpunk
Polsk Punk
Christian Gangsta
Heavy Metal
Black Metal
Crossover
Contemporary Christian
Christian Rock
Merengue
Thrash Metal
SynthPop
LAME %s version %s (%s)
LAME version %s (%s)
image/jpeg
image/png
image/gif
Error: can't allocate VbrFrames buffer
3.100.1
LAME3.100r1
LAME3.100r
http://lame.sf.net
32bits
@333333
333333
@ffffff!@
333333
333333
@333333
@a2U0*
D?333333
Aq=Z@o
Error: MAX_HEADER_BUF too small in bitstream.c
strange error flushing buffer ...
Internal buffer inconsistency. flushbits <> ResvSize
bit reservoir error:
l3_side->main_data_begin: %i
Resvoir size: %i
resv drain (post) %i
resv drain (pre) %i
header and sideinfo: %i
data bits: %i
total bits: %i (remainder: %i)
bitsperframe: %i
This is a fatal error. It has several possible causes:
90%% LAME compiled with buggy version of gcc using advanced optimizations
9%% Your system is overclocked
1%% bug in LAME encoding library
Q&>c0A
v?E+n?n
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intensity stereo not implemented yet
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INTERNAL ERROR IN VBR NEW CODE, please send bug report
INTERNAL ERROR IN VBR NEW CODE (986), please send bug report
INTERNAL ERROR IN VBR NEW CODE (1313), please send bug report
maxbits=%d usedbits=%d
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%s%s: %s error
%s%s(0,0): %s error
%s%s(0,negative): %s error
%s%s(negative,non-integer): %s error
acosdq
acoshq
annuity
annuityq
asindq
atan2d
atan2dq
atan2q
atanhq
cdfnorm
cdfnorminv
compound
compoundq
erfcinv
erfcxq
erfinv
exp10q
expm1q
gammaq
hypotq
ilogbq
ldexpq
lgamma
lgammaq
llrint
llrintq
llround
llroundq
log10q
log1pq
lrintq
lround
lroundq
nextafter
nextafterq
nexttoward
nexttowardq
remainder
remainderq
remquo
remquoq
scalbln
scalblnq
scalbn
scalbnq
scalbq
sinhcosh
sinhcoshq
tgamma
pow3o2
acospi
asinpi
atan2pi
%lu/%s
libmUI.dll
Cannot open message catalog %s
__dummy
DOMAIN
OVERFLOW
UNDERFLOW
#dummy
#printf
#printf_s
#__iob_func
#fclose
#fflush
#fopen
#fopen_s
#fprintf
#fprintf_s
#fwprintf
#fwprintf_s
#_snprintf
#_snprintf_s
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#sprintf_s
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#sscanf_s
#swprintf_s
#vfprintf
#vfprintf_s
#vprintf
#vprintf_s
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#vsprintf
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#_vsnprintf
#_vsnprintf_s
#vswprintf
#wprintf
#wprintf_s
#_wfopen
#_wfopen_s
#fread
#fwrite
#fgets
#fgetws
#fgetc
#fgetwc
#fputs
#fputws
#fputc
#fputwc
#ftell
#ferror
#freopen
#freopen_s
#_wfreopen
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#__stdio_common_vsprintf
#__stdio_common_vswprintf
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ucrtbase.dll
msvcrt.dll
SYSTEMROOT
\system32\
Constant propagation error (%s substitution):
FORMAL
RETURN
GLOBAL
I32 %lx != %lx
SI32 %ld != %ld
F32 %f != %f
I64 %lx:%lx != %lx:%lx
SI64 %ld:%ld != %ld:%ld
F64 %f != %f
Bad second argument
Fatal Error: Can not initiate the Heap
Usage: %s input_file output_file
Usage: %s segment_size input_file [-trace]
Conversion from text file %s to binary %s completed
segment_size = 0x%x = %d
Cannot allocate memory to hold segment (size = 0x%x)
Input file: %s corrupted
routine_name = '%s'
file_name = '%s'
prof_dir = '%s'
Dynamic profile created from file %s completed
Fatal Error: This program was not built to run on the processor in your system.
The allowed processors are: %s.
Run-Time Check Failure: The variable '%s' is being used without being initialized
Error: Buffer overrun occurred, forced exit
Initialization of symbol handler failed. Error code %d
NTDLL module not found
RtlCaptureContext function not found in ntdll.dll
Windows XP 64-bit Edition Version 2003 or newer should be used.
StackWalk is terminated abnormally. Error code %d
Exception is raised during stack walking
Signal %s is raised
Signal %s is raised at 0x%p
SIGSEGV
SIGILL
SIGBUS
SIGFPE
No error
You must link with libunwind to use traceback functionality
Intel(R) Core(TM) Duo processors and compatible Intel processors with supplemental Streaming SIMD Extensions 3 (SSSE3) instruction support
Intel(R) Pentium(R) 4 and compatible Intel processors with Intel(R) Streaming SIMD Extensions 3 (Intel(R) SSE3) instruction support
Intel(R) Pentium(R) M and compatible Intel processors
Intel(R) Pentium(R) 4 and compatible Intel processors. Enables new optimizations in addition to Intel processor-specific optimizations
Intel(R) processors with Swing New Instructions support
Intel(R) processors with SSE4.2 and POPCNT instructions support
Intel(R) processors with MOVBE instructions support
Fatal Error: This program was not built to run in your system.
Please verify that both the operating system and the processor support Intel(R) AVX.
Please verify that both the operating system and the processor support Intel(R) AVX, F16C and RDRAND instructions.
Please verify that both the operating system and the processor support Intel(R) AVX2, BMI, LZCNT, HLE, RTM and FMA instructions.
Boundary Run-Time Check Failure for variable '%s'
Boundary Run-Time Check Failure
Conversion Run-Time Check Failure
Please verify that both the operating system and the processor support Intel(R) %s instructions.
Please verify that your application was built with compatible Intel(R) libirc library
Use of incompatible or internally inconsistent Intel(R) libirc library
Run-Time Check Failure: The variable '%s' is being used in %s without being initialized
Cache size is incorrectly reported as 0. Please engage your hardware and/or OS support to address the problem.
irc_msg.dll
GENERIC_IA32
FXSAVE
SSE4_1
SSE4_2
POPCNT
PCLMULQDQ
AVX512DQ
PTWRITE
AVX512F
RDSEED
AVX512IFMA52
AVX512ER
AVX512PF
AVX512CD
AVX512BW
AVX512VL
AVX512VBMI
AVX512_4FMAPS
AVX512_4VNNIW
AVX512_VPOPCNTDQ
AVX512_BITALG
AVX512_VBMI2
VPCLMULQDQ
AVX512_VNNI
WBNOINVD
PCONFIG
__INTEL_LIBIRC_DEBUG
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`scalar deleting destructor'
`vector constructor iterator'
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`vector vbase constructor iterator'
`virtual displacement map'
`eh vector constructor iterator'
`eh vector destructor iterator'
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`copy constructor closure'
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new[]
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`omni callsig'
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Sunday
Monday
Tuesday
Wednesday
Thursday
Friday
Saturday
January
February
August
September
October
November
December
MM/dd/yy
dddd, MMMM dd, yyyy
HH:mm:ss
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CompareStringEx
GetCurrentPackageId
GetFileInformationByHandleEx
GetSystemTimePreciseAsFileTime
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hip: addbuf() Out of memory!
hip: Fatal error! tried to read past mp buffer
hip: bitstream problem, resyncing skipping %d bytes...
hip: wordpointer trashed. size=%i (%i) bytes=%i
hip: error audio data exceeds framesize by %d bytes
hip: invalid layer %d
hip: fatal error. MAXFRAMESIZE not large enough.
hip: Insufficient memory for decoding buffer %d
hip: out space too small for unclipped mode
MPEG-2.5 is supported by Layer3 only
Stream error
Frame size too big.
Sorry, layer %d not supported
hip: Can't step back %ld bytes!
hip: Aborting layer 1 decode, illegal bit allocation value
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big_values too large! %i
Blocktype == 0 and window-switching == 1 not allowed.
region0index=%d > 22
region1index=%d > 22
hip: Bogus region length (%d)
hip: OOPS, part2remain=%d l3=%d cb=%d bv=%d region1=%d region2=%d b-type=%d mixed=%d
hip: OOPS, 'gainpow2' buffer overflow lsf=%d i-stereo=%d b-type=%d mixed=%d
hip: Can't rewind stream by %d bits!
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.CRT$XIAC
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.CRT$XPA
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.idata$4
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GetStdHandle
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GetConsoleScreenBufferInfo
WideCharToMultiByte
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SetPriorityClass
GetCurrentProcess
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GetConsoleOutputCP
KERNEL32.dll
LoadLibraryA
GetThreadLocale
FormatMessageA
UnhandledExceptionFilter
SetUnhandledExceptionFilter
TerminateProcess
IsProcessorFeaturePresent
QueryPerformanceCounter
GetCurrentProcessId
GetCurrentThreadId
GetSystemTimeAsFileTime
InitializeSListHead
IsDebuggerPresent
GetStartupInfoW
GetModuleHandleW
InitializeCriticalSection
LoadLibraryExA
LeaveCriticalSection
GetProcAddress
GetModuleHandleExA
GetEnvironmentVariableA
EnterCriticalSection
RtlUnwind
GetLastError
SetLastError
DeleteCriticalSection
InitializeCriticalSectionAndSpinCount
TlsAlloc
TlsGetValue
TlsSetValue
TlsFree
FreeLibrary
LoadLibraryExW
ReadFile
CreateFileW
CloseHandle
PeekNamedPipe
SystemTimeToTzSpecificLocalTime
FileTimeToSystemTime
QueryPerformanceFrequency
GetTimeZoneInformation
ExitProcess
GetModuleHandleExW
WriteFile
GetModuleFileNameW
GetCommandLineA
GetCommandLineW
GetACP
HeapFree
HeapAlloc
CompareStringW
LCMapStringW
FlushFileBuffers
GetConsoleCP
GetConsoleMode
HeapReAlloc
ReadConsoleW
SetFilePointerEx
GetCurrentDirectoryW
SetStdHandle
GetStringTypeW
FindFirstFileExW
FindNextFileW
IsValidCodePage
GetOEMCP
GetCPInfo
GetEnvironmentStringsW
FreeEnvironmentStringsW
SetEnvironmentVariableA
SetEnvironmentVariableW
GetProcessHeap
WriteConsoleW
HeapSize
SetEndOfFile
RaiseException
abcdefghijklmnopqrstuvwxyz
ABCDEFGHIJKLMNOPQRSTUVWXYZ
abcdefghijklmnopqrstuvwxyz
ABCDEFGHIJKLMNOPQRSTUVWXYZ
Unknown
_ldexp
Unknown
___libm_sse2_pow
Unknown
___libm_sse2_log10
Unknown
___libm_sse2_cos
Unknown
___libm_sse2_powf
Unknown
___libm_sse2_exp
Unknown
___libm_sse2_atan
Unknown
___libm_sse2_log10f
Unknown
___libm_sse2_log
Unknown
___libm_ldexp_dispatch_table_init
Unknown
___libm_sincos_huge
Unknown
___libm_feature_flag_init
Unknown
___libm_ldexp_px
Unknown
___libm_ldexp_w7
Unknown
___libm_reduce_pi04l
Unknown
___libm_error_support
Unknown
___libm_matherrl
Unknown
___libm_matherr
Unknown
___libm_matherrf
Unknown
___libm_matherrl_px
Unknown
___libm_matherr_px
Unknown
___libm_matherrf_px
Unknown
___libm_sse2_exp2
.itt_notify_tab
optimization_report_version
optimization_report
.itt_notify_tab
optimization_report_version
optimization_report
.itt_notify_tab
optimization_report_version
optimization_report
.itt_notify_tab
optimization_report_version
optimization_report
.itt_notify_tab
optimization_report_version
optimization_report
@.itt_notify_tab
optimization_report_version
optimization_report
.itt_notify_tab
optimization_report_version
optimization_report
.itt_notify_tab
optimization_report_version
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