Re: [mp3encoder] MP3 CUE cutter <<< VIRUS

2003-07-23 Thread Haas Wernfried
On Fri, Jul 18, 2003 at 01:14:17PM +0200, [EMAIL PROTECTED] wrote:
> Just in case someone's antivirus software doesn't pick it up, don't open
> that message.
where exactly should be a virus here?

regards,
wernfried

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Re: [MP3 ENCODER] Does lame need vorbis?

2002-12-09 Thread Haas Wernfried
hi,
On Thu, Dec 05, 2002 at 01:06:34AM +, Mark Collier wrote:
> Does lame need vorbis? I say not...
afaik it is obsolete, use the vorbis-tools instead.

> I use a program called CDex for CD ripping > Vorbis.
> CDex can work with external encoders as long as they
> accept input via STDIN, which Lame does. I used the
> Vorbis functionality for encoding to Ogg.

> I could easily write myself a Vorbis encoder that
> accepts input from STDIN, so ogg 1.0 encoding can
> still be done from CDex (or use one of the CDex plugis)
doen't oggenc accept stdin too? at least the linux version does.
you might try if it works on windows, if not you could try to ask
on the ogg-vorbis ML for help / request this feature. 

regards,
wernfried

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Re: [MP3 ENCODER] cpu time

2002-12-01 Thread Haas Wernfried

hi,
On Sat, Nov 30, 2002 at 07:44:35PM -0300, Gabriel Bonsoir wrote:
> i use lame for encode mp3 in a AMD 300 and lame 
> use all the processor that he can. That is ok, is not a big processor, but i 
> try with XP 2000 and lame still using me all the cpu time.  Is very fast and 
> really  good encoding, but i want to be able to do something else at the 
> same time, like listen music, but i can't. 
the faster your cpu is, the faster lame will finish encoding, but it still
uses as much cpu time as it can get. 
> There is same way (i'm using linux) for make that for example lame use just 
> 50% of cpu and no more, or that use the cpu time that other running program 
> let.
you should try "nice -n 19 lame -whatever_your_options_are"
now lame is running with a lower scheduling priority, which means that lame
still uses all cpu time it can get, but only if no other application needs
it. your system should behave quite like you were not encoding anything in
background then. 

regards,
wernfried

PS: next time, please start a new thread not by sending a reply and just
changing the topic as this breaks threadsorting.

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Re: [MP3 ENCODER] Clarification regarding Mp3

2002-11-24 Thread Haas Wernfried
hello,
On Sun, Nov 24, 2002 at 01:40:02PM +0530, [EMAIL PROTECTED] wrote:
> I wish to know the following using lame software
> 
> 1.Convert mono wav file to stereo mp3 ? 
> ( I am aware it is possible to convert stereo wav to mono mp3)
as far as i can see in the manpage, this is not possible (anyone corrct me
if i am wrong).
lame -ms (or better -mf) test.wav still creates a mono mp3, you could use
sox (http://sox.sourceforge.net/) to convert the file or also convert it 
in audacity:
duplicate the mono track, select both tracks and export as wav.
I am just not really sure why you want to do this anyway ;)

> 2.If  I use the use variable bit recording is it compatible with all 
> Hardware (cd mp3 players) ?
don't know, sorry...

> 3.Right now I am using Audacity software which uses lame in the 
> background.If I directly use lame.exe with the proper options in dos
> prompt will it affect the quality ?
if you use the same options and the same version of lame, there should not
be a difference. in doubt you could encode the same file with lame/audacity
and lame in dos and check if the files differ.

regards,
wernfried

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Re: [MP3 ENCODER] Bug in lame 3.93..

2002-11-24 Thread Haas Wernfried
On Thu, Nov 21, 2002 at 11:10:37AM -0700, Bowie J. Poag wrote:
> Here's a nice bug..  lame 3.93 (and from the looks of it, every version 
> before it..) identifies files based purely on the friggin suffix of the 
> filename. World-class coding there, guys. (*chuckle*)  Anyway, I've 
> managed to control my laughter long enough to post a note about it.
how generous, this surely helps a lot in increasing the quality of lame.

> How to reproduce the bug:
> [..]
of course lame could check better, but this is like using a microwave
ofen to dry your dog...

how to avoid the bug:
don't try stupid things and/or go buy some professional commercial software.

> Whoever is responsible for that piece of crap-code aught to be shot, 
> hung, drawn, quartered, buried, dug up, their bones crushed into powder, 
> the phosphorous extracted from them and used in fireworks that spell out 
> painful insults directed at members of his immediate family.
sending a patch is usually more useful.

wernfried

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Re: [MP3 ENCODER] --alt-preset ultimate description

2002-11-16 Thread Haas Wernfried

hi,
On Sat, Nov 16, 2002 at 12:35:46PM +, Dominique wrote:
> Could someone please refer me to explanations of "--alt-preset" 
> switches? I have searched for a long while and all I could find was 
> broken links or flames for asking ;/
lame --alt-preset  help
is this what you are looking for?

> Another thing: is it possible to pass --alt-preset's to libmp3lame? I 
> mean e.g. for MEncoding?
don't know, anyone else?
(my guess is no, at least i wouldn't know how to do this with 
mplayer/mencoder, so i think it is not supported.)

regards,
wernfried

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Re: [MP3 ENCODER] multiple files

2002-11-13 Thread Haas Wernfried
hi,
On Wed, Nov 13, 2002 at 11:58:40AM -0300, Pablo Morales wrote:
> you need a script to do that. yf you are using DS download cd3mp2 from amy
> home page
> http://netwarrior.esmartweb.com it works for freesd , but you can tune it or
> remove the code you need .
there's also a script in the misc directory of the lame package called
mlame, adopt the options (options_high) at your needs.
and probly there are thousands of other frontends :)

regards,
wernfried

> Bytes
> 
> - Original Message -
> From: "mike" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, November 12, 2002 10:31 AM
> Subject: [MP3 ENCODER] multiple files
> 
> 
> > Just wondering...
> >
> > how can I encode multiple wav-files in one rush?
> > Stupid as I am, I tried "lame [options] *.wav,
> > (that works with ogg, all wav- files in the directory
> > are encoded one after the other), but it doesn't work
> > with
> > Lame..
> > I tried to find some options in the help- pages but
> > didn't find any.
> > Any help appreciated,
> >
> > cheers,
> >
> > Michael
> >
> > __
> > Do you Yahoo!?
> > U2 on LAUNCH - Exclusive greatest hits videos
> > http://launch.yahoo.com/u2
> > ___
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Re: [MP3 ENCODER] Quality problem reencoding

2002-11-13 Thread Haas Wernfried
hi,
there's a recent thread about reencoding mp3, you can find it in the
archives at:
http://minnie.tuhs.org/pipermail/mp3encoder/2002-August/005028.html
to sum it up in short terms: reencoding is generally not a very 
good idea.
maybe you could switch to ogg vorbis, it is designed to be able to peel
a stream, which means reducing bitrate without reencoding, but afaik
the tools for peeling are quite experimental / not available for the public
yet. if you already have mp3s and not the original wav files, ogg would
not help you anyway, as reencoding mp3 to ogg is a very bad idea, too.
hope no one's annoyed because im advertising ogg here, i personally like
both lame/mp3 and ogg vorbis.

regards,
wernfried

On Wed, Nov 13, 2002 at 09:22:45AM -0500, User BS wrote:
> I have a number of files created with lame 3.92 using the --r3mix
> preset.  They all sound great.  The problem is when I reencode them down
> for streaming.  It doesn't matter if I reencode on the fly or reencode
> first.  The effect reminds me of a phase shifter commonly used on a
> guitar.  You can actually hear it here with winamp, xmms or whatever
> you like here:  http://caroline.pop4.net:8004   The stream is 56k @
> 44.1KHz - which should sound good.
...

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Re: [MP3 ENCODER] Lame truncates long files...

2002-11-11 Thread Haas Wernfried
hello,
On Fri, Nov 08, 2002 at 10:13:07PM -0800, Richard Brockie wrote:
> I am using Lame (--r3mix) to mp3 encode large (~2hr) wav files. I edit 
> the intro and outro of the wav files to something pleasing, but find 
> that the mp3 file is truncated by an amount which seems to be 
> proportional to the length of the file. Is this known behaviour and is 
> there something which can be done to get round this?
hmm, i don't exactly understand what you mean by "truncated by an amount 
which seems to be proportional to the length of the file".
i just checked my two longest mp3s (approx 2.0 and 2.5 hours long), both
of them encoded with lame 3.92 --alt-preset standard. they play fine,
the only thing i could see was, that when i went to the end of the 
files in xmms they still played for about 10-20 seconds though the
time-counter was at end of file. maybe this is the same thing you are
experiencing?

i also remember that i once encoded something quite long with an older
version of lame, i had trouble seeking in that file after about 1.5 hours
with xmms, but as i had already switched to lame 3.92 at that time and
the problem already had disappeared in 3.92, i did not really care
about seeking a bug in an old version ;)

which version of lame are you using and with which software are you
decoding your mp3s?

regards,
wernfried

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Re: [MP3 ENCODER] split album.mp3 + album.cue into tracks

2002-11-04 Thread Haas Wernfried
hi,
On Sat, Nov 02, 2002 at 10:01:13AM -0500, Chris Holt wrote:
> Get MP3 Direct Cut.  It's freeware and does not decode the MP3 to cut it up.
> Ooops, I'm busted...  You probably needed a Linux based program didn't you?

thanks for pointing me at this. though it's windows based, it can
be run more or less through wine. (i've got some windows 3.11 installation
left somewhere, i never boot it, but it's handy to for trying to run
things with wine).

> I'm pretty sure it will work with the cue sheet, but if it does not there is
> a handy silence detection feature that will set the cues automatically.
unluckily the cue sheet doesn't seem to work right (it's an vbr encoded mp3),
cutting by ear fails (i wasn't able to use sound though i downloaded
mpglib.dll) and silence detection crashes.
i think it is very likely that the wine/win 3.11 system isn't the best 
environment for runnning mp3 direct cut and i think the described problems
are not a problem of mp3 direct cut.

> Download the manual and READ IT.  
:)))

mp3 direct cut reminded me of the fact that there is also a editor
for linux (http://minnie.tuhs.org/Programs/Mpcut/), it does not seem
to support .cue sheets, but cutting by ear seems to be a nice backup
plan if no one else comes up with an idea for an automated solution.

regards,
wernfried

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[MP3 ENCODER] split album.mp3 + album.cue into tracks

2002-11-02 Thread Haas Wernfried
hello,
i apologize if this is too much off topic on this list.
i downloaded some music on the net, when i unpacked it, i got 2 files:
album.mp3 and album.cue. album.mp3 is the whole album as one big mp3 file,
album.cue is a textfile, that looks like this:
PERFORMER "foo"
TITLE "bar"
FILE "album.mp3" MP3
  TRACK 01 AUDIO
TITLE "track1"
PERFORMER "foo"
INDEX 01 00:00:00
  TRACK 02 AUDIO
TITLE "track2"
PERFORMER "foo"
INDEX 01 01:02:23
etc...

i searched the web for information about this, the .cue file holds 
the information, where each track is starting and there are some plugins
for winamp, but i did not really find out if it is possible to use the 
queue file with popular players for linux.
does anyone have hint, how i could either teachs xmms to use the .cue file
or even better how to split the big .mp3 into tracks (preferred solution
if possible without decoding to wav, splitting and reencoding, which
would be possible with bchunk).

thanks,
wernfried

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Re: [MP3 ENCODER] How to get 20x performance on MP3 encoding..

2002-11-02 Thread Haas Wernfried
hi,
On Fri, Nov 01, 2002 at 08:42:38PM -0500, Ishaan Dalal wrote:
> Encoding Speedproportional toQualityandCPU Speed
> 
> MusicMatch gets that speed by forsaking qualitysimple as that. I believe
> it uses the Xing encoder...though newer versions also have the Fraunhofer
> (FhG) VBR encoder.
you _can_ also do that with lame:
lame -q 9 foo.wav foo.mp3
Using polyphase lowpass  filter, transition band: 15115 Hz - 15648 Hz
Encoding foo.wav to foo.mp3
Encoding as 44.1 kHz 128 kbps j-stereo MPEG-1 Layer III (11x) qval=9
Frame  |  CPU time/estim | REAL time/estim | play/CPU |ETA
  9173/147000 ( 6%)|0:13/3:36|0:13/3:41|   17.697x|3:27
average: 128.0 kbpsMS: 9176 (100.0%)

cpu is a duron 800, so this is approximately as fast as musicmatch.
btw i can listen to the mp3 without having blood running out of my
ears, i remember having downloaded mp3s that sounded worse.

however i usually encode with -q 2 and vbr, which brings my speed
down to realtime again.

regards,
wernfried

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Re: [MP3 ENCODER] Abig question: mp3 to wave format

2002-09-13 Thread Haas Wernfried

hello,
On Wed, Sep 11, 2002 at 09:16:53AM +0300, [EMAIL PROTECTED] wrote:
> I would like to knew if there is some open soure program (or exe
> program like lame) that convert mp3 to wav file?
> please let me knew if Yes or No.
Yes.

regards,
wernfried

PS: one solution: "lame --decode file.mp3"

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Re: [MP3 ENCODER] alt-preset "low"?

2002-09-09 Thread Haas Wernfried

hello,
On Mon, Sep 09, 2002 at 01:37:11PM +0300, Vincent Kargatis wrote:
[..]
> At the moment I'm using 'alt-preset 144[or 160] -b96' as a compromise to
> retain a smaller filesize.  With that goal in mind, are there any better
> encoding option choices?

if i understand the documentation (lame --alt-preset help) right, in 
this case lame is working in ABR mode. it also says that abr is not nearly as
flexible as vbr. (see last point)


> [..] Does
> the 'standard' methodology allow resolution scaling like that, or does it
> use heuristics that don't scale like that?  Just curious - I'm ignorant
> about the innards.

afaik "standard" invokes some high-quality settings (vbr with -q=2 
(good algorithm, takes more time), vbr-quality high (-V=2), mininum bitrate
128 and some other settings that are more special and i don't really know 
about them).
however you can override settings, i use 
--alt-preset standard -V 4 -b 112
for quite some stuff, bitrate is of course lower than just "standard", 
quality good enough for my issues.
question to all experts: is this useful or would it make more sense
to invoke lame just with --alt-preset standard -V 4 -b 112 -q=2 ?

> One aspect I'm wondering about with my current choice is that it doesn't
> seem to take a 'global' look at the file: say the middle section is very
> quiet and sparse, the 'alt-preset NNN' doesn't seem to decide it could use
> the unneeded space from there on the busier parts elsewhere - it just varies
> the resolution on a very whort time scale.  (Heck, I don't even know if
> 'standard' does that, but I was assuming it did.)
i think this is the difference between vbr and abr (which i think you
are using in this case, see first point). afaik abr tries to maintain
the specified bitrate all over the file, vbr tries maintain the specified 
quality. vbr possibly spends high bitrates for some parts and low on other
(works fine for silence, which is encoded at 32kbps). however filesize of
vbr isn't (that well) predictible.

i'm no expert on encoding at all, so forgive me if i am wrong about the 
abr vs. vbr points, expert's comments welcome.

regards
wernfried

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Re: [MP3 ENCODER] Lame Help?

2002-08-15 Thread Haas Wernfried

On Wed, Aug 14, 2002 at 09:48:54PM -0600, Preston Lord wrote:
> When installing lame-3.91 as described below I get following errors:
btw 3.92 is already released (some months)
> Using SuSE 8 Pro, KDE 3, GCC 2.95.3
> 
> ./configure --with-fileio=lame --without-vorbis --disable-gtktest
> --enable-expopt=full --prefix=/usr
  
just tried to compile with this option on my system - no way.
doesn't seem to be a default setting, which seems to be --disable-expopt,
you could also try --enable-expopt=norm, which even compiles with 
gcc 2.95.2. 
> 
> all goes well on configure BUT then I get this on make
> (if installed without switches, will not work with BlackIce)
if this blackice thing (whatever it is, some intrusion detections system?)
really wants you to use a switch, use the ones above.

> cc1: Invalid option `-fsched-interblock'
i guess this is some compiler optimization that your (and my) too old 
compiler don't understand...
> make[2]: *** [common.lo] Error 1
> make[2]: Leaving directory
> `/home/preston/Documents/dev/lame-3.91/mpglib'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/home/preston/Documents/dev/lame-3.91'
> make: *** [all] Error 2

regards,
wernfried

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Re: [MP3 ENCODER] Trouple with 8khz, 16bit PCM file

2002-08-11 Thread Haas Wernfried

On Sat, Aug 10, 2002 at 10:02:39AM -0600, [EMAIL PROTECTED] wrote:
> 66.37.140.181/orig.wav
> If I try to encode it with LAME I just get noise.
> If I use soundforge to resample it to 32khz, then lame can encode it
> just fine.

looks like some "unusual" (non-)pcm data file to me. whatever - i'm not
an expert on this.
> 
> I need to convert files like these in a scripted, command-line 
> environment... anyone know of a windows 2000 compatible command line 
> utility to resample wav files and/or is there a magic parameter that 
> will allow me to encode that file?

no problem using linux:
"sox orig.wav -w ori2.wav" and "lame ori2.wav orig.mp3" worked fine, but
produced an MPEG-2.5 layer III file (MPEG-1 layer III needs at least a 
samplerate of 32khz according to lame-docs). if you want mpeg-1 layer III
use "sox orig.wav -r 32000 -w ori2.wav" to resample (seems to work better
than mpeg-2.5 to me, but files get bigger). of course this was linux now.
for windows 2000 (are you sure you don't want linux :) check out
http://sourceforge.net/projects/sox/ for the sox-win package. i guess
there should not be great differences and it should be no problem to do
the same with windows-commandline.

regards
wernfried



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Re: [MP3 ENCODER] 768 sample delay for psychoacoustic model

2002-07-22 Thread Haas Wernfried

sorry, really don't know, but your clock seems to be quite wrong:
Date: Sun, 22 Jul 2001 14:46:22 +0530
wrong year ;)

regards
wernfried


On Sun, Jul 22, 2001 at 02:46:22PM +0530, sandeep wrote:
> Hi,
> I am trying to play with psychoacoustics.I am messed with the delay concept
> Can anyone tell me how to give the delay for psychoacoustic model ?
> Should it be given for every 576 samples  or every 1152 samples or a one
> time delay of 768 samples is sufficient for whole process?
> 
> I looked for some material in net and landed with Ramapriya Rangachars
> material.I didn't the way samples should be systematically delayed .
> Can anyone help me in this regard??
> 
> Thanx in advance.
> Regards
> Sandeep
> 
> 
> 
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Re: [MP3 ENCODER] Performance of Lame on Pentium III 500

2002-07-18 Thread Haas Wernfried

On Thu, Jul 18, 2002 at 09:48:53AM -0700, RSRamesh wrote:
> Hi all, 
> Is there a performance measurement of Lame on PIII
> 500?
> I am looking if Lame on PIII 500 can achieve 3x to 5x
> encoding speeds or not. 
> Thanks for your inputs in advance
> Ramesh
hello,
using an amd duron 800 with alt-preset standard (vbr, high quality, q=2 
(rtfm about this)) is a bit higher 1x on my system. standard cbr (cbr is 
usually faster than vbr) is a lot faster. especially the choice between 
cbr and vbr and most of all the setting for qval (-q 0 to -q 9 (means 
very best and slow algorithms or fast and lousy algorithms) determine
encoding speed more than your processor.
if you want to achieve 3x to 5x encoding speed and this is the most
important thing for you in encoding, choose between cbr and vbr and then
check which setting for qval is fast enough for you...
i guess there are some other things that determine your encoding speed,
too. check the manual/docs and maybe there are some special compiler
flags to optimise speed too, don't really know about this...

regards,
wernfried

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Re: [MP3 ENCODER] Lame vbr and vbr-new difference?

2002-07-16 Thread Haas Wernfried

hello,
On Tue, Jul 16, 2002 at 07:24:22PM +0200, Vladimir Mincev wrote:
> R> You should read more careful the included documentation.  -vbr or
> R> --vbr-old is the "old" way of doing Variable Bitrate Encoding.  Slower,
> R> much tested.  As of 2.91 if I recall well the --vbr-new is as good as
> R> --vbr-old when it comes to quality.  But keep in mind that is a
> R> DIFFERENT METHOD.  It is a lot faster also.  That's about it.
yeah, that's in the manual. 
> 
> R> As the other list memebers have told you - RTFM.
rtfm helps for many problems, but there's not much more in either manpage 
nor in the docs directory, so i think rtfm is a bit of a short answer.
or did i miss something in the docs?
 
> I  have  read  the manual, but there is no tech info about differences
> between  those  two  VBR  settings.  Both  work  fine  when decoded so
> question  was,  should I stick to old (slower an well tested) or go to
> new (faster) metod!?  Have somebody had a problems using new VBR?
would be interesting for me, too. especially which of them will be used
in upcoming versions. both? will vbr-new replace vbr-old? will vbr-new
become standard for vbr-encoding? will mulder and scully finally kiss?

regards,
wernfried

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Re: [MP3 ENCODER] Lame: options that most economic to convert voice wav to mp3

2002-07-03 Thread Haas Wernfried

On Tue, Jul 02, 2002 at 02:01:13PM -0400, Kang Sun wrote:
>   We get about 1.5GB wav files daily that contain purely telephone
> conversions between our customer service representatives and customers.
> I like to convert these wav files to mp3 to save some storage space
> using LAME. The question is what is the best option to convert mono,
> telephone quality wav file to mp3?
well, depends on the quality you actually need, check the manpage of lame
for settings. most interesting for could be --preset help -> --preset phone
regards
wernfried

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Re: [MP3 ENCODER] 8-bit sample-size is not supported

2002-07-03 Thread Haas Wernfried

hello,
i guess it would be best to check the sox project site on
http://sourceforge.net/projects/sox
for help about this...
regards 
wernfried

On Tue, Jul 02, 2002 at 10:12:27AM -0400, Kang Sun wrote:
> Thanks for the suggestion.
> I am trying to install sox on a SUN box running Solaris 8 with gcc. I
> got the following error. Help!
> 
> gcc -g -O2 -Wall -DSUNAUDIO_PLAYER  -DWORDS_BIGENDIAN=1 -DHAVE_LIBM=1
> -DHAVE_UNISTD_H=1 -DHAVE_MALLOC_H=1 -DHAVE_GETOPT=1 -DHAVE_STRERROR=1
> -DHAVE_MEMMOVE=1 -DHAVE_RAND=1 -DHAVE_SYS_AUDIOIO_H=1  -I.   -c -o au.o
> au.c
> au.c: In function `auwriteheader':
> au.c:401: warning: implicit declaration of function `strlen'
> /usr/ccs/bin/as: "/var/tmp/cc6rWfHD.s", line 353: error: unknown opcode
> ".subsection"
> /usr/ccs/bin/as: "/var/tmp/cc6rWfHD.s", line 353: error: statement
> syntax
> /usr/ccs/bin/as: "/var/tmp/cc6rWfHD.s", line 383: error: unknown opcode
> ".previous"
> /usr/ccs/bin/as: "/var/tmp/cc6rWfHD.s", line 383: error: statement
> syntax
> make: *** [au.o] Error 1
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]] On Behalf Of Haas Wernfried
> Sent: Friday, June 28, 2002 4:56 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [MP3 ENCODER] 8-bit sample-size is not supported
> 
> On Fri, Jun 28, 2002 at 03:05:21PM +1000, Erik de Castro Lopo wrote:
> > > I am trying to convert wav files into mp3 with LAME. I
> got
> > > this error: 8-bit sample-size is not supported!
> 
> > When someone does the mods to LAME so that LAME can use the new
> version of 
> > libsndfile your problem will be addressed. 
> > 
> until then you could convert the file to 16 bit. there's quite a lot of
> software that can do that and i guess that sox is quite surely installed
> on
> your system.
> try "sox file.wav -w file2.wav" 
> regards
>   wernfried
> 
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Re: [MP3 ENCODER] 8-bit sample-size is not supported

2002-06-28 Thread Haas Wernfried

On Fri, Jun 28, 2002 at 03:05:21PM +1000, Erik de Castro Lopo wrote:
> > I am trying to convert wav files into mp3 with LAME. I got
> > this error: 8-bit sample-size is not supported!

> When someone does the mods to LAME so that LAME can use the new version of 
> libsndfile your problem will be addressed. 
> 
until then you could convert the file to 16 bit. there's quite a lot of
software that can do that and i guess that sox is quite surely installed on
your system.
try "sox file.wav -w file2.wav" 
regards
wernfried

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Re: [MP3 ENCODER] Adding id3 tags to an existing file.

2002-06-15 Thread Haas Wernfried

try this:
amnenion@xover:~/testr > mp3info test.mp3
test.mp3 does not have an ID3 1.x tag.

amnenion@xover:~/testr > mp3info -a testartist test.mp3
amnenion@xover:~/testr > mp3info test.mp3
File: test.mp3
Title:  Track:
Artist:  testartist
Album:  Year:
Comment:Genre:  [255]

seems to work :)

it also has an interactive mode (console+gtk) and can be found at
http://metalab.unc.edu/mp3info/

regards
wernfried


On Sat, Jun 15, 2002 at 03:12:56PM +0100, Michel SUCH wrote:
> 
> Is there a way, with lame or other command line program, to add id3 tags
> to an already existing mp3 file?
> 
> Michel SUCH TEAM OS/2 FRANCE  
> ICQ # 51654489
> 
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Re: [MP3 ENCODER] Re-encoding an mp3 file

2002-05-24 Thread Haas Wernfried


maybe the --mp3input option is of interest for you, but i do not know if lame
does something else than just decoding and re-encoding. at least it should 
save the disk space of the .wav file...

regards
wernfried haas


On Thu, May 23, 2002 at 10:21:52PM +, Yosi wrote:
> Hi,
> 
> I have a certain mp3 file that I would like to re-encode with Lame. By 
> "re-encode" I mean, to let lame encode the mp3 file again (and yes, I know 
> that it is lossy). The obvious thing will be to let lame decode the mp3 into 
> wave, and then run lame again and encode the produced wave file. Is there a 
> better way to do it, in terms of preserving as much quality as possible?
> 
> Thanks,
> Yosi
> 
> 
> 
> _
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Re: [MP3 ENCODER] bug in lame 3.91 (estim time)?

2002-02-03 Thread Haas Wernfried

On Thu, Jan 31, 2002 at 08:10:12AM +1000, Mark Taylor wrote:
> > the cpu time / estim dissapears:
> > Frame  |  CPU time/estim | REAL time/estim | play/CPU |ETA
> > 134280/134282(100%)|1193046 h/0:00| 1:07:06/ 1:07:06|   0.x|0:00
> The frame counter display is just an estimate
> based on the input filesize - it has no effect on the encoding.
> the fact that it was within 2 out of 134k frames is pretty good :-)
> Mark
this is definitely good, i could not have done it better :)
the thing that still seems to be estimated not so well is the CPU time.
1193046 hours would be about 136 years, that would be a great uptime ,)
regards
wernfried

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[MP3 ENCODER] bug in lame 3.91 (estim time)?

2002-01-29 Thread Haas Wernfried

hello,
when encoding really large files (cd-quality, wav, >500 megs) with lame
(via the mlame script, with encoder "nice -n 19" and 
options="--alt-preset standard -ms" the following happens:

everything starts fine:
LAME version 3.91  (http://www.mp3dev.org/)
Using polyphase lowpass  filter, transition band: 18671 Hz - 19205 Hz
Encoding test.wav to test.mp3
Encoding as 44.1 kHz VBR(q=2) stereo MPEG-1 Layer III (ca. 7.3x) qval=2
Frame  |  CPU time/estim | REAL time/estim | play/CPU |ETA
  9400/134282 ( 7%)|3:52/   55:26|4:38/ 1:06:23|   1.0546x| 1:01:44
128 [  84] %%
[..]

but after some time 
the cpu time / estim dissapears:
Frame  |  CPU time/estim | REAL time/estim | play/CPU |ETA
134280/134282(100%)|1193046 h/0:00| 1:07:06/ 1:07:06|   0.x|0:00
here the job is already finished, but 2 frames seem to be missing too (which
did not appear to be a problem at the end of the file btw).

my system:
suse 7.1 with kernel 2.2.18-suse (which is 2.2.18 with some patches applied by
suse such as reiserfs afaik)
gcc 2.95.2

maybe someone can test, if the same problem appears on other systems because
it is a bug in lame, or if just my system is broken.

thanks
wernfried

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[MP3 ENCODER] vbr in lame 3.91 vs 3.87

2002-01-16 Thread Haas Wernfried

hello,
my first message seems to be lost somewhere, so i am re-(and hopefully not
double) posting it:
i upgraded from lame 3.87 to lame 3.91 and i am a bit confused about the fact,
that lame is producing quite bigger files now:
lame -V 5 -b 96 -h -ms -k xxx.wav produced files with a bitrate of about
140, now they are > 160, files with -V 0 are much bigger (about 50% of the
frames are at 320) now, too. actually, there is nothing really wrong, i guess
-V 7 (or whatever) in 3.91 will produce as big files as -V 5 in 3.87, but i am
just wondering about this thing. does anyone have an idea about it, as there
have been some changes in vbr within the last versions.
thanks
wernfried haas


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[MP3 ENCODER] vbr in lame 3.91 vs 3.87

2002-01-16 Thread Haas Wernfried

hello,
i upgraded from lame 3.87 to lame 3.91 and i am a bit confused about the fact,
that lame is producing quite bigger files now:
lame -V 5 -b 96 -h -ms -k xxx.wav produced files with a bitrate of about
140, now they are > 160, files with -V 0 are much bigger (about 50% of the
frames are at 320) now, too. actually, there is nothing really wrong, i guess
-V 7 (or whatever) in 3.91 will produce as big files as -V 5 in 3.87, but i am
just wondering about this thing. does anyone have an idea about it, as there
have been some changes in vbr within the last versions.
thanks
wernfried

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