-Where do I get the latest source?
I have seen the source for 3.51 but not any of the beta..
-David Swigger
BTW:
Heres a logo I whipped up if anyone wants it..
-Original Message-
From: Robert Hegemann [SMTP:[EMAIL PROTECTED]]
Sent: Thursday, January 06, 2000 12:57 PM
Hello All,
Has anyone else had the problem of a small pop at the beginning
of playing an encoded file? I am just curious if it is a known problem.
BTW:
I am compiling and linking in Win98..
-David Swigger
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
sources in a .zip file
Both are windows only, but dont include the lame source.
You just have to drop the latest dist. into the OCX project
directory, and then build it..
-David Swigger
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Leftampfactor = (double)65535 / (double)LeftTotal ;
Reftampfactor = (double)65535 / (double)RightTotal ;
Now we go back through the wav file and multiply the ampfactor to all
the samples (left and right).
-David Swigger
-Original Message-
From: Mark Stephens [SMTP:[EMAIL PROTECTED
test
Hi!
I'm new here so please don't bash me too hard
;)
Anyway, just found this wonderfull encoder, LAME
:)
I've noticed that it seems to raise the dB level
very slighty in the encoded mp3 files, found this out by loading the original
.wav file into sound forge and the mp3 one, "clipping"
Quality (default value Normal):
With the LAME encoder, you can specify the output quality; thus you can trade
off encoding time against sound quality. The default (normal) is recommended,
since the high output option might cause ringing artifacts at lower bit-rates.
The voice quality is
sorry, hope this is better
- Original Message -
From: "Taupter" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 10, 2000 2:52 AM
Subject: Re: [MP3 ENCODER] -h (High quality causing ringing artifacts ?) ?
David,
Avoid HTML mail. Please send text-on
Don,
Is it just me or the mp3 produced with that setting (compared to
normal -b192) are somewhat LOWER in dB levels than normal -b192 mp3 ?
Either my erars are acting up or winamp is faulty! :) ..or i'm right ?
- Original Message -
From: "Don Melton" [EMAIL PROTECTED]
To: [EMAIL
Anything wrong with --vbr-new ? (buggy?)
Since the default VBR mode seems to be --vbr-old, one wonders...
Is there something wrong with the "new" mode, i've noticed it's a lot faster
... (version 3.85)
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
What is this -q1 parameter i've seen here on the list ?
is it for VBR,ABR, or CBR ?
is there -qx or only q1, tell me about it
are there any other undocumented parameters ?
if so, what do they do ?
And like i mentioned in another post, what's up with --vbr-old and --vbr-new
?
since --vbr-old is
I'de like to get a version with MP3x included, can some code wiz compile
such a version ?
i tried the 3.70 GTK version with the dll's but it never worked, it just
crashed.. :(
- Original Message -
From: "Robert Hegemann" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 11,
I own a couple of AKG- k70
Are these crap or are they considered "good" or even "pro" headphones?
i really haven't found any reviews yet... how well do they stand up to the
ones you guys are using ?
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Just noticed that when i use ABR (--abr 201 -b160 -h -mj %file %file for
example) the first frame is ALWAYS a 60kbit one, probably the xing header
since this "bug" does not occur when i use -t, now i know that this might be
a very small thing to bitch about, but i'm concerned about hw players not
Hey man, wtf is this ?
- Original Message -
From:
Fernanda
To: [EMAIL PROTECTED]
Sent: Thursday, July 13, 2000 8:13
PM
Subject: [MP3 ENCODER] Mulheres... Temos
que dar a resposta!
VERSCO MASCULINA:Chega ao caixa
automaticoDesliga o motor do carroTranca
I've noticed that the speed of GOGO is INSANE! (due to some massive ASM
optimations i guess)
Now, to the question, is any1 even optimizing LAME or is it still "under
development" (quality) and will it be ASM optimized for some serious speed
improvments, i know it fast now, very fast on my 850 p
Hope i'm not stepping on ppl toes for this...
I need a program that can record from line-in/mic to mp3 directly, any of
you guys know of such a program ? (would be nice if it could use LAME as an
encoder)
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VERY usefull for devices such as portable mp3 players.
- Original Message -
From: "Marc de Lange" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 27, 2000 5:26 PM
Subject: [MP3 ENCODER] Preservation a ID3 tags
Wouldn't it be useful if lame was able to automatically copy
Try doing a BINARY comparation to see
- Original Message -
From: "Wim" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 04, 2000 8:25 PM
Subject: [MP3 ENCODER] HQ/LQ switches in lame_enc.dll
I've encoded some wavefiles with Exact Audio Copy and Audiograbber using
this is strange... obiously something has changed
in 3.86...
C:\cdexbetalame.exe --abr 201 -b160 -h -mj
f:\temp.wav f:\templame385.mp3LAME version 3.85 (www.sulaco.org/mp3)Win32 binaries from
www.chat.ru/~dkutsanov/Using
polyphase lowpass filter, transition band: 20805 Hz - 21339
Hi guys, recently i've been thinking of doing a program that scans a dir,
reports the TOTAL time of all mp3 (MM:SS) and writes a file with that info.
Only problem is that i can't seem to find any info on this how to read this
things out of CBR/VBR files, any1 know where i can get such info ?
it display this info on the
(command-line) screen for each file. Not the whole thing in one file.
If you really want just this info on one file, you probably need to use
something in a library.
- Original Message -
From: "David" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sat
nfo on one file, you probably need to use
something in a library.
- Original Message -----
From: "David" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, August 12, 2000 2:06 AM
Subject: Re: [MP3 ENCODER] MP3 Programing question... (1 or 2 m's ?) :)
Thanks, but .. hmm guess
I'm currentlly using CDex (win32) as a ripper and lame.exe to encode on the
fly, but my question is:
Is ABR encoding supported by the dll ?
It does not show up on the list inside CDex, so i'm guessing it's not in
CDex yet, but ither way, is ABR avail from the DLL ? Or onlu CBR/VBR ?
Mark ?
Any
What is the point of free format ?
can winamp play it back ?
- Original Message -
From: "Steve Lhomme" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 18, 2000 11:13 PM
Subject: Re: [MP3 ENCODER] Free format
Doh !!!
Sorry for the lame question ;)
- Original Message
ok i'm asking again :)
what is freeformat ? (the point of it, why i should/should not use it
compared to "normal" mp3 ...)
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 21, 2000 2:40 PM
Subject: Re: [MP3 ENCODER] Free format
What is the point
- Original Message -
From: "Gabriel Bouvigne" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 05, 2000 9:52 AM
Subject: Re: [MP3 ENCODER] newbie question
On www.r3mix.net, the following command line is recommended as the best
option (size, speed, quality-wise):
Hi guys, i recently recorded a live broadcast from the radio to one big .mp3
file, now i need to split it up onto separate tracks, what tool can u
recommend for this task ?
I DO NOT want to go mp3 - wav - mp3
:)
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C:\cdexbetalame388.exe --abr 193 -h -mj -h -b 64 d:\01.wav 01mj.mp3
LAME version 3.88 MMX (beta 1, Mar 26 2001) (http://www.mp3dev.org/)
(Win32 binaries from http://dkutsanov.chat.ru/)
CPU features: i387, MMX (ASM used), SIMD
Using polyphase lowpass filter, transition band: 21517 Hz - 22050 Hz
Why not do somwthing like VBR to the lowpassfilter ?
i'll try to explain my idea here below :)
let's say there is a passage in a track that has wery few sounds above
13Khz, why not let the lowpassfilter adapt to it and cut at 13Khz where
there is no need for more ?
something like ALF, adaptive
Hi!
The new mpg123 V 0.59q fully supports VBR streams,
so work on VBR support in LAME can start :-)
Or is it started already ?
--
David Balazic , student
E-mail : [EMAIL PROTECTED] | living in sLOVEnija
home page: http://surf.to/stein
Computer: Amiga 1200 + Quantum LPS-340AT
with -V 0. Maybe the default min bitrate should be based on
the value of -V? 112kbs for -V0, and decreasing down to 32kbs
as -V is increased?
How about fixing Lame to raise the bitrate as needed ?
I guess the problem is somewhere in the psycho-acustic model.
--
David Balazic , student
E-mail
bit rate )
--
David Balazic , student
E-mail : [EMAIL PROTECTED] | living in sLOVEnija
home page: http://surf.to/stein
Computer: Amiga 1200 + Quantum LPS-340AT
--
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
. Any volunteer ?
I just had a quick peek through mpg123 and didn't see anything about
de-emphasis. If players ignore it, why put it in the encoder?
Because it is part of the standard ?
--
David Balazic , student
E-mail : [EMAIL PROTECTED] | living in sLOVEnija
home page: http
of years. This flag
was removed in AAC and mpeg-4.
I was just going to ask Mark to remove this switch as it is totally
obsolete.
Unless you deal with pre-emphasised material ...
--
David Balazic , student
E-mail : [EMAIL PROTECTED] | living in sLOVEnija
home page: http://surf.to/stein
mpilers may behave differently.
--
David Balazic , student
E-mail : [EMAIL PROTECTED] | living in sLOVEnija
home page: http://surf.to/stein
Computer: Amiga 1200 + Quantum LPS-340AT
--
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...
I gave up any hope with them.
I attach their last mail.
Yes I know this is off topic, but I'm frustrated :-(
--
David Balazic , student
E-mail : [EMAIL PROTECTED] | living in sLOVEnija
home page: http://surf.to/stein
Computer: Amiga 1200 + Quantum LPS-340AT
8--- cut
can be
done in digital domain by adjusting sub-band
parameters.
Anyone have more info on how this is done/
how to actually code this?
Thanks,
Bill
--
Bill Eldridge
Radio Free Asia
[EMAIL PROTECTED]
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--
David Balazic , student
E-mail
From: David [EMAIL PROTECTED]
This is an example of peak normalization (correct me if I am wrong)..
We have a song in a (stereo 16bit)wavfile called mysong.wav
we scan the wav files and find these values..(they are fake of course)
MaxRightSample = 16000;
MinRightSample = -8000;
MaxLeftSample
to some more precise format in LAME,
then the normalization should be done after the conversion, because it
does not make it slower and it does have higher quality , and quality
is the point of the LAME project , no ?
DAVID BALAZIC [EMAIL PROTECTED] on 17.01.2000 14:41:05
Please respond
] http://www.cdex.n3.net)
Ross.
By normalizing 16 bit samples you loose quality,
due to rounding errors.
But maybe the loss can be reduced by doing the normalization
on when the samples are in more suitable format ( double or something ) ?
--
David Balazic , student
E-mail : [EMAIL PROTECTED
Ross Levis wrote:
I don't think this is related to LAME is it?
No , not really. Should have marked it as OFF-TOPIC ( I did now ).
David Balazic wrote:
And while you're on it , how about writing a driver ( for windows ) for
the SoundBlaster 64 PCI ( aka Ensoniq AudioPCI
Maybe you are feeding mono-44100 data into lame , but lame thinks it is
stereo ?
Jaroslav Lukesh wrote:
| Odeslatel: Mark Taylor [EMAIL PROTECTED]
|
| Can you try 3.66? There were a couple of glitches in 3.65 which
| I think are fixed now.
|
| Mark
No, is the same... :-(
Regards
for this, like -eh [extra high
quality].
But isn't --quality just a more detailde -h ?
--quality 0 -- no -h option
--quality 6 -- -h
--quality 9 -- your -eh option
and some in between
david balazic
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
two ( 1024
etc... )
b is short for byte ( I'm not sure about this )
bit should not be abbreviated , unless it is clear from context that it
is
bit and not something else ( byte for example ).
B is already used for Bell.
This document is on the web , but I forgot the URL :-(
david
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David Balazic wrote:
Matthew Lloyd wrote:
[ big snip ]
About the differences between Nitrane and other:
Did you try to compare the sound to the original ?
Maybe you could figure out that way where that
spectral components should be.
And you forgot to test mpg123 !
( and xaudio too
'Y' ? 1E21 #
zetta is 'Z' ? 1E24 # this two I got from that ISO draft document I
can't find anymore
david balazic
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that I get. Any comments?
mph is wrong too :-)
some also use kph for km/h
and ksps ( kilo samples per second )
those ( including kbps ) are all "unofficial" as far as I can tell.
P.S.: "p" stands obviously for "per" , which means (more-or-less) the
same as "/
Me neither !
Jaroslav Lukesh wrote:
sorry for test, but I do not receive any message for 2 days.
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On the site you recommend WinAMP for the listening,
while just recently a winamp bug was discussed on this
list, which add corruptions during playback of mp3s !
The bug is present in winamp 2.62 ( latest as of this writing ),
as reported on this list.
david balazic
George wrote:
Hi
Shawn Riley wrote:
David wrote-
On the site you recommend WinAMP for the listening,
while just recently a winamp bug was discussed on this
list, which add corruptions during playback of mp3s !
The bug is present in winamp 2.62 ( latest as of this writing ),
as reported on this list
extended Mp3 player. But I accept
ideas.
What do you mean most extended ? You only need simple playback for the
tests , no ?
I think you should put more attention to the procedure of decoding and
playback.
david
- Original Message -
From: "David Balazic" [EMAIL PROTECTED]
it doesn't support stdin
,
but then the responsible should be shot on sight ! :-) )
Hmm, or maybe loose95 doesn't support pipes ? I know NT4 does.
david balazic
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
ds to high-frequency echoes. (Some kind of phantom image of the true
sound at lower freqs).
Hmm , can you explain why this happens ?
Doesn't seem logical to me, but I'm no mp3/mdct expert.
Dag dag,
Segher
david balazic
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st band is for 38 Hz, as someone
recently mentioned, I think at least freq to 50 Hz would be affected )
david balazic
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Kimmo Mustonen wrote:
On Mon, 29 May 2000, David Balazic wrote:
if ( You are saying : "Adding this assert() fixes the problem" )
then {
This means that you are triggering some compiler bug.
At least it looks like that to me.
Rationale : The added code doesn't
it work , there is no such info on the page,
but it is sure an interesting idea !
David Balazic
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orking yet.
Mark
So the there will be ogg ( vorbis ? ) support in lame ?
Input, output or both ?
Isn't this bloat ?
After all lame is supposed to be an MP3 encoder ...
David Balazic
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at
http://www.mail-archive.com/mp3encoder@geek.rcc.se/msg02512.html
David
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Mark, what about some clarification ?
David Balazic
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, but then it was a few years
ago now and time maybe playing tricks.
The ISO sources are avaliable at
ftp://ftp.tnt.uni-hannover.de/pub/MPEG/audio/mpeg2/software/technical_report/dist10.tar.gz
from year 1997 on and will probably stay there for a while.
david balazic
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MP3 ENCODER mailing list ( http
years ago.
Is there some real audio engineer on the list ?
David Balazic
Mark Taylor wrote:
RE:
http://www.kenwoodcorp.com/i/topframes/press_sdrive_2620.html
Any comments?
--
Dmitry Boldyrev
Subband Software, Inc.
http://www.subband.com
This just made it int
is nothing to repeat ( no previous frame ) :-)
David
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David Bridson wrote:
That seems to be the world upside down. Adapting an encoder in
order
to fit a non-mp3-compliant player is no good move I think. Best is
to
contact the mpg123 author?
I don't think it's just mpg123 - I have problems playing same file
in
CL-Amp under BeOS with VBR
thing -
Bass sounds are much clearer on headphones. To get even a reasonable
level of
Bass on my stereo, I have to boost the bass. With headphones, there's
plenty
of bass, even with bass-boost turned off.
David
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15khz.
This would be strange behaviour for a station
specialising in classical music !
For stereo you must not exceed 19 kHz , because that is the
position of the 'pilot' signal, needed for stereo decoding.
David Balazic
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
with
it?
Yes, don't even try to decode a VBR MP3 with CRC data with mpg123. Just
won't work. Don't forget that Winamp isn't flawless - there's a link to
a known bug on the LAME home page.
David
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VBR do not ENCODE the LSB signal correctly
( CEP is CoolEdit mp3 plugin )
They used lame 3.81beta
David Balazic
A 1khz signal, with amplitude 1, will be considered "analog silence"
by LAME VBR mode, and encoded at 32kbs. This is probably why
distortion was introduce
Yeah i mentioned the winamp MMX code produced noisier output
don't have proof but i could definetly hear a difference with headphones :)
dave brown
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On
Behalf Of David Balazic
Sent: 18 July 2000 11:51
To: [EMAIL
at least
fifteen minutes Plays everything what it exists in stock market n
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On
Behalf Of David
Sent: 24 July 2000 15:24
To: [EMAIL PROTECTED]
Subject: Re: [MP3 ENCODER] Mulheres... Temos que dar a resposta!
Hey man, wtf is
Any idea how I get LAME to compile with Vorbis support?
David
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Don't forget about Vorbis ( www.vorbis.com and www.vorbis.org )
David Balazic
Roel VdB wrote:
Hello,
A reporter I know is planning on writing a book about mp3. (benelux)
We all know the downsides of SDMI and consorts, so I was wondering, is
there a site or so that has a nice
t;-V0".
In my K6-2 300 MHz the difference is 31.695 seconds without MMX
vs. 30.176 seconds with MMX enconding a little wav (using -V0)
Are the outputs identical ?
WinAMP produces different output , with each code :
MMX / pentium / 486 ( I didn't test 3DNow! )
David Balazic
--
M
no!!! turn this mailing list into an audio appz war3z list ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Tim Ruddick
Sent: 07 August 2000 14:31
To: [EMAIL PROTECTED]
Subject: Re: [MP3 ENCODER] Mayah EditPro
On Mon, 7 Aug 2000, Jaroslav Lukesh
May I take it from these e-mails that most builds of LAME do not
incorporate
MMX enhancements? If that is the case, where can I find a build that
does
include MMX (or even 3D Now!) enhancements?
Cheers,
David
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Speaking of Vorbis, when (if) will Vorbis support (with no extra
downloads no
extra hassle) be built into the downloadable source code?
Cheers,
David
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From the mpg123 docs :
mpg123 supports all 3 layers, but layer I and II are not very tested.
libmpg123 supports layer III only.
David Balazic
Roel VdB wrote:
Hi,
The author of CoolPlayer (win32) is considering to put in mpg123
instead of the xaudio decoder. http://www.daansystems.com
David Balazic wrote:
From the mpg123 docs :
mpg123 supports all 3 layers, but layer I and II are not very tested.
libmpg123 supports layer III only.
And according to the CHANGES.CVS file on www.mpg123.de
layer i/II and VBR will be supported in the lib in the next
version 0.59s
P.S
doesn't state that the source code
must be distributed with the programme, but it must be made available
to anybody who wants it.
David
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such
a limited number of bit rates. I've seen VBRs made
with other encoders that don't rely just on a few
bit rates but precisely adjust the bit rate
according to the complexity of the music.
David
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Pierre Hugonnet wrote:
Hello,
not purely mp3, but...
I want to buy a soundcard, and am hesitating between the SoundBlaster 128 (~25$) and
the SounBlaster 1024 Live (~60$).
Main Usage:
- playing my mp3 collection (mainly 128kbs quality) on my (very standard) HiFi
equipement
-
| I don't think disk space is a problem these days.
| The (big) advantage of one large mp3file containing the entire album (like
| AiD suggests) is that players like winamp don't delay playback when a
track
| ends and another begins. When using seperate files, Winamp checks
playtime,
| ID3-TAG
into applications that record live audio. But that can be done
with Virtual Audio Cable from www.ntonyx.com
David Balazic
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i suppose you should have also a RIFF WAVE header in the file...
some wav/mp3 encoder/ripper do this
As previously mentioned, CDEx ( http://www.cdex.n3.net/ ) can do
this (with LAME support, no less) either from an MP3 or WAV.
David
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masked, that would
not be masked in regular stereo mode.
This seems wrong to me, alltough is is possible that the brain would do
this masking itself.
So what is the comment of experts here ?
Forgive me if I posted non-sense :-)
David Balazic
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MP3 ENCODER mailing list ( http://geek.rcc.se
Erm, something screwy is happening to all my mail from this list. Here's a
sample. Just started happening about 2-3 hours ago! Any ideas on what's
going on?
- Original Message -
From: "O/T-Track+ 2.66b" "O/T-Track+=202.66b"@f854.n292.z2.fidonet.org
To: [EMAIL PROTECTED]
Sent: 26 August
After monday I got only two messages ( both on wednesday ) :
- [MP3 ENCODER] test please ignore , from Jan Penman
- [MP3 ENCODER] presets , by Robert Hegemann
David Balazic
Robert Hegemann wrote:
test please ignore
is this list dead?
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For VBR mode the max bitrate is set on one screen, while min bitrate
on the other. Confusing.
Also when ABR is selected, the max bitrate slider should be turned off.
Don't just blindly throw the CLI options on the screen , think too :-)
David Balazic
Holger Dors wrote:
Hello Roel,
I
all ( most ) compilers.
I.E. :
On 16 bit systems is int16
On 32 bit systems it is int32
On 64 bit systems ( alpha ) it is int64 ( unless tweaked to 32 bit for
bugward compatibility )
So for loop counters and like, "int" should be used.
David Balazic
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for the decode to only decode one
(selectable) channel.
Main advantage is the lower CPU usage.
Just to clarify : -md is "dual channel" ?
David Balazic
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1.28
+1.5 93.7 2.34
+2 97.2 3.65
+2.5 99.0 5.43
+3 99.8 8.4
+3.5 99.9 9.5
[ snip ]
Sorry if this is an "reader IQ too low" error :-)
David Balazic
--
M
, for which
only a BASIC compiler is available... It seems like practically every line
generates an error or warning of some sort?!? Can someone address this?
C++ is a superset ( more or less ) of C , so it should
handle C code.
David Balazic
--
MP3 ENCODER mailing list ( http://geek.rcc.se
on CDR. Based on what I
have heard on this list I would like to stick with CBR.
What about -h -b 192 -m j --lowpass 16 ?
Please note: I do have perfectionist tendencies. :)
If space is not critical, then either use CDA or CBR 320 :-)
David
--
MP3 ENCODER mailing list ( http://geek.rcc.se
3.81beta
David Balazic
Roel VdB wrote:
Hello,
If you haven't already, you should check out this decoder test page...
http://privatewww.essex.ac.uk/~djmrob/mp3decoders/intro.html
All tested, and on "objective sound quality" there is one more thing
to do :))
--
Best regar
of
10^10
| US $ per citicen.
Not only in germany, but in Czech too :)
But it's the americans fault ! :-)
For using those non-standard units !
David Balazic
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
with changing sample-rate.
But I could be wrong.
David Balazic
--
"Be excellent to each other." - Bill Ted
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
%
:: [100%] 99.5% = p +oo
::
:: Why not just allocate more characters , like [0.13%] ?
::
That's also possible.
And much easier to read/understand ( not need for the "decoding" table
:-)
David Balazic
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
data. My
question: is there a LAME CODEC which could encode audio into an AVI file
directly?
Cheers,
David
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
wrong, it will.
Therefore, the LAME guys are definitely waiting for the Ogg guys and the Ogg
guys are definitely waiting for the LAME guys :-)
David
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
from a PCM Wav file to an MP3 Wav file. The only
slight problem comes with larger AVIs where the file size of the AVI with
uncompressed data goes over 2 Gb rendering the file non-standard and,
therefore, useless.
David
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
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