Add the following proto-type just above the main() function
int lame_decoder(lame_global_flags *gfp,FILE *outf,int skip);
and you should be set
Albert
http://www.cdex.n3.net/
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
- Original Message -
From: "Nathan D. Blomquist" <[EMAIL PR
That was written about a year ago by the authors not me.. I just notice
the new versions & report them.. Submit the change to freshmeat or tell
the lame guys to do it.
peace,
jolan
On Sat, 30 Sep 2000, Takehiro Tominaga wrote:
> Hello, jolan, from one of LAME developper.
>
> On the freshmeat.n
Dan Nelson wrote:
>
> In the last episode (Sep 28), Mark Powell said:
> > BTW The gcc 2.95.2 options are perfectly valid for FreeBSD as well as
> > Linux. Obviously :) Can they be copied into the FBSD section too?
> >
> > # these options for gcc-2.95.2 to produce fast code
> > # CC_OPTS = \
> >
On Wed, 27 Sep 2000, Dan Nelson wrote:
> In the last episode (Sep 28), Mark Powell said:
> > BTW The gcc 2.95.2 options are perfectly valid for FreeBSD as well as
> > Linux. Obviously :) Can they be copied into the FBSD section too?
> >
> > # these options for gcc-2.95.2 to produce fast code
> >
In the last episode (Sep 28), Mark Powell said:
> BTW The gcc 2.95.2 options are perfectly valid for FreeBSD as well as
> Linux. Obviously :) Can they be copied into the FBSD section too?
>
> # these options for gcc-2.95.2 to produce fast code
> # CC_OPTS = \
> # -Wall -O9 -fomit-frame-po
In the last episode (Sep 26), Joshua Bahnsen said:
> I don't have gtk-config on my system so when configure searches for
> it it returns no like it should, but then it tries to run the program
> "no", I assume to get the version # of gtk or something. So then in
> the makefile HAVEGTK is defined e
> I could not find the --voice page. is there a specific URL ?, - or which
> section is it in ?
>
http://www.multimania.com/bouvigne/lame/voice.html
--
Gabriel Bouvigne - France
[EMAIL PROTECTED]
icq: 12138873
MP3' Tech: www.mp3-tech.org
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3e
I could not find the --voice page. is there a specific URL ?, - or which
section is it in ?
Sorry to be a nuisance.
Eric
- Original Message -
From: Gabriel Bouvigne
To: [EMAIL PROTECTED]
Sent: Tuesday, September 12, 2000 10:10 AM
Subject: Re: [MP3 ENCODER] --voice
--voice was made be
Presets:
There are three blocks which can be selected via conditional compiling.
Block 2: use low fs
Block 3: do not use fsb21 if possible.
Differences are:
Block 2 Block
phon+ (0...4 kHz) fs=8 kHzfs=11 kHz
700 KB 61
Hallo Frank,
> :: Als Du zu -mf und -mj den Bitverbrauch über mehrere Titel
> :: verglichen hattest, so war LAME wahrscheinlich nicht mit
> :: meinem extra Code RH_VALIDATE_MS übersetzt? Eigentlich
> :: sollte nähmlich -mj weniger Bits verbraten als -mf.
> ::
> Die Korrelation heranzuziehen,
Hallo Robert,
::
:: > Kann man bei Layer III mehr Bits für das Differenzsignal als für das
:: > Summensignal verbraten?
::
:: Im Prinzip ja:
:: * bei CBR hat Mark allerdings eine reduce_side() routine
::eingebaut die dem Differenzsignal immer weniger Bits
::zugesteht als dem Summe
In the last episode (Sep 06), Frank Klemm said:
> May be evry (non trival function should get a history header?
>
> /*
> * Function: double sinc (IN double x);
> *
> * Purpose:calculates sin (pi x) / (pi x)
> *
> * Input:
> *
> * History:
> *Person_1 2000-0
In the last episode (Aug 27), Guybrush Threepwood said:
> At 09:01 PM 08/27/2000 +0100, you wrote:
> >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?
>
> I'm getting the same problem
>
> In the last episode (Aug 22), Jaroslav Lukesh said:
> > It should be maybe possible in wavelet transform, but not in discrete
> > cosinus. You should wait for wavelet encoder and decoder...
> >
> > Or you should use ;-))
> >
> > lame --decode x.mp3 - | mp3enc31 -sti -of x.small.mp3 -esr 22
In the last episode (Aug 22), Jaroslav Lukesh said:
> It should be maybe possible in wavelet transform, but not in discrete
> cosinus. You should wait for wavelet encoder and decoder...
>
> Or you should use ;-))
>
> lame --decode x.mp3 - | mp3enc31 -sti -of x.small.mp3 -esr 22050 -qual 9 -bw 11
> > From: Mark Taylor [mailto:[EMAIL PROTECTED]]
> >
> > MP3 does not allow mid/side stereo to be turned on and off on a band by
> band basis (AAC does),
>
> Hmm, then what happens if a JS frame has both M/SS and IS enabled? In Layers
> I and II, IS can be enabled across several fixed bands, I gu
> From: Mark Taylor [mailto:[EMAIL PROTECTED]]
>
> MP3 does not allow mid/side stereo to be turned on and off on a band by
band basis (AAC does),
Hmm, then what happens if a JS frame has both M/SS and IS enabled? In Layers
I and II, IS can be enabled across several fixed bands, I guess something
> If you make a "grep RH_AMP *.c" over all C-sources you will find out,
> that it enables a different amp_scalefac_bands() in quantize.c and
> makes a little psymodel tweak in psymodel.c.
>
>
> Ciao Robert
Thank you for your quick replay.
Actually, most of the people might think 'thank you' mai
:: >Another point: I notice you changed the encoding status display to
:: >update every 2 seconds, rather than every 100 frames. Any reason for
:: >this? I prefer 100 frames because I like looking and nice round
:: >numbers, and it doesn't impact the speed. Also, to do the 2 second
::
::
Mark Taylor schrieb am Mon, 21 Aug 2000:
> The -V option is scaled from 0=best to 9=worst because
> originally, the value of 'n' was the number of bands for
> which distortion was allowed to be > masking.
> It is far too late to change this now!
In my opinion you could think of it now as
Frank Klemm schrieb am Mon, 21 Aug 2000:
> :: There are a lot of core dumps!
> ::
> Sorry. The results are not sooo bad. I switched to max quality via "-q 9".
> "-q 9" seem to be not very good tested.
in case of -q9, where we use no psy-model, some variables
namely pe_MS[][] were not initiali
>Another point: I notice you changed the encoding status display to
>update every 2 seconds, rather than every 100 frames. Any reason for
>this? I prefer 100 frames because I like looking and nice round
>numbers, and it doesn't impact the speed. Also, to do the 2 second
I also noticed this, an
> Date: Mon, 21 Aug 2000 13:52:48 +0200
> From: David Balazic <[EMAIL PROTECTED]>
> Content-type: text/plain; charset=iso-8859-2
> X-Accept-Language: en
> Sender: [EMAIL PROTECTED]
> Precedence: bulk
> Reply-To: [EMAIL PROTECTED]
> X-UIDL: V&"#!cKF!!#5S!!;+7"!
>
> Hi!
>
> I'm sending this messag
> :: There are a lot of core dumps!
> ::
> Sorry. The results are not sooo bad. I switched to max quality via "-q 9".
> "-q 9" seem to be not very good tested.
> File is to be found in "http://www.uni-jena.de/~pfk/resample.tar.gz".
>
> Every Coder uses other command line switches with another
:: It's an example I found in my instrument directory. Several files from
:: different sources (mostly Synth papers with CDs, i.e. Keys, Keyboards).
::
:: On is the of a shaker. Lame fails, also at rates of 320 kbit/s.
:: Lame 192 kbit/s sounds really bad. FhG (mp3encdemp31) is *much* better
I am sorry, but I accidentally deleted the email about the Win32 Version resource.
I was able to modify the source code (only added a file and changed the Makefile.MSVC)
to enable win32 users to have a version tab in the properties page.
Whoever wanted this I will be happy to email the new bina
> "M" == Monty <[EMAIL PROTECTED]> writes:
>> are there precedents of an piece of GPL upgrading to LGPL?
M> Sure! I changed all my own stuff from GPL to LGPL last year (I
M> decided I mostly agreed with Bill Joy ;-). Send the author a
M> letter; he may be willing.
but it
In the last episode (Aug 09), Nathan Blomquist said:
> I was wondering if anyone would be or knows anyone who would be
> interested in a 16-Bit DOS version of LAME? I have been able to
> produce these by using a DOS extender called WDOSX. This allows
> almost any Win32 console application to wor
Takehiro Tominaga schrieb am Die, 08 Aug 2000:
> > "R" == Robert Hegemann <[EMAIL PROTECTED]> writes:
>
> R> I just wanna let you know that it does not compile out of the
> R> box mainly because your provided mathinline.h interferes with
> R> the one installed in my Linux system.
> "R" == Robert Hegemann <[EMAIL PROTECTED]> writes:
R> I just wanna let you know that it does not compile out of the
R> box mainly because your provided mathinline.h interferes with
R> the one installed in my Linux system.
umm,,, just remove the *MY* mathinline.h and delete the
Hi Takehiro!
> Hi all,
>
> I just made my latest snapshot with a "whole new psymodel".
> it uses always MAXNOISE and do not normalize the spread function.
> and it always uses mixed block, subblock gain, scalefactor scale.
>
> The VBR mode of this code passes the "vbrtest.wav", and even more,
>
Hello,
RV> finding: "--nspsytune" sounds _a lot_ worse than the normal psymodel.
RV> The graphs show a lower overall distortion amplitude, but there is
RV> this noise that I can even clearly hear upto V1 (didn't test V0).
I triple-checked this. Remember those noise graphs I made
(original-decod
Hello Mark,
Sunday, August 06, 2000, 11:34:04 PM, you wrote:
MT> I tend to agree with this, and I think we should disable
MT> scalefac_scale for now (it can still be enabled with -q1
MT> for testing)
after some re-consideration this seems wisest imo too. after some
reports of -q1 producing poor
Hi all,
I just made my latest snapshot with a "whole new psymodel".
it uses always MAXNOISE and do not normalize the spread function.
and it always uses mixed block, subblock gain, scalefactor scale.
The VBR mode of this code passes the "vbrtest.wav", and even more,
the file size will be almost
>
> Robert Hegemann wrote:
> >-h is always on by default if you use VBR modes like -v (default VBR mode),
> >--vbr-old or --vbr-new. You can switch to a lower quality setting as -h
> >(equals -q2) with -q3. But caution, the -q n switches are meant for internal
> >testings only, they are not doc
Hi Gaby!
> > For my ears, Takehiro's scalefac_scale feature will not give better
> > results.
> > The quality of many tracks is a little bit lower. I have found no
> > tracks
> > with better quality. The file size is reduced, but for my opinion, the
> > quality is more important. So i think, sett
> For my ears, Takehiro's scalefac_scale feature will not give better
results.
> The quality of many tracks is a little bit lower. I have found no tracks
> with better quality. The file size is reduced, but for my opinion, the
> quality is more important. So i think, setting this feature as defau
So, -q1 is theoretically the highest quality
setting? What does it alter in 3.85?
Begin Original Message
From: Jack Davis <[EMAIL PROTECTED]>
Sent: Thu, 03 Aug 2000 23:17:12
To: [EMAIL PROTECTED]
Subject: [MP3 ENCODER] Re: 3.86a, bug with -h ?
Robert Hegemann wrote:
>-h is
Robert Hegemann wrote:
>-h is always on by default if you use VBR modes like -v (default VBR mode),
>--vbr-old or --vbr-new. You can switch to a lower quality setting as -h
>(equals -q2) with -q3. But caution, the -q n switches are meant for internal
>testings only, they are not documented cos t
A 4th factor (kind of related to the others) is that the
chained CPU implementation generally means a greater delay
for Layer 3, meaning in practice more than a 1/2 second delay -
too much for doing a back-and-forth conversation, though
often fine for one-way transfers.
That's on equipment I had
On 28-Jul-2000 Bill Eldridge wrote:
> I thought the biggest issue with Layer 2 vs. Layer 3 is that
> Layer 2 holds together better over multiple generations.
I agree with Bill here. Layer 2 survives re-encoding much better than Layer 3.
Two other factors probably account for its widespread use
> On MP2's allegedly higher quality than MP3 at high bitrates, I can hardly
> ever tell the difference at such bitrates anyway. But looking at objective
> measures, LAME @ 320kbit/s gives significantly lower noise than any MP2
> encoder I've seen, even at 384kbit/s. However, I don't have access to
> From: Jaroslav Lukesh [mailto:[EMAIL PROTECTED]]
>
> I put package for cooledit (it works
> with cooledit
> 2000, confirmed by friend) from syntrillium at our corporate web at
>
> www.k-net.cz/!/lossaud.zip (564k)
>
> Here are MP3 decoder, source, MP2 encoder and decoder, [...]
Isn't
Mark Taylor schrieb am Die, 25 Jul 2000:
> -X: picks which of 7 algorithms will be used to determine if
> one noise 'signature' sounds better than another.
well, 0...7, seem to be 8 different ones ;-)
> -X should now be considered a permanent option. -Y and -Z change
> >from release to r
Hi.
On Tue, 25 Jul 2000, Mark Taylor wrote:
>> >Thanks for finding this. This was a subtle & rare bug in CBR,
>> >triggered by the -k option.
>> Can you tell me when this bug was introduced? I listened to some of my recent
>> encodings and found this blip in about every 20th track.
>Were they al
Hello Mark,
MT> I would guess it was introduced in 3.85: it requires a combination of
MT> scalefac_scale (defaulted in 3.85, before that enabled with -Y ), and
MT> not using enough low pass filtering for the amount of compression.
If I believe the history log and my own tests, scalefac_scale was
>
> >(-k, by the way, is *always* a bad idea. It overrides LAME's
> >default lowpass filters. It will cause ringing and twinking
> >at 128kbs)
>
> Are there examples for this? When I did some listening tests I noticed the
> missing frequencies but nothing else. Does the FhG encoder still do th
>
> Hi Mark,
>
> On Mon, 24 Jul 2000, Mark Taylor wrote:
> >Thanks for finding this. This was a subtle & rare bug in CBR,
> >triggered by the -k option.
>
> Can you tell me when this bug was introduced? I listened to some of my recent
> encodings and found this blip in about every 20th track.
Just a quick question, when cdex refers to "high
quality" that "may produce pinging" they are not using
the -k option are they? Also this bug only applies to
VBR right?
Regards,
Abe Corrie
--- Arne Zellentin <[EMAIL PROTECTED]> wrote:
> Hi Mark,
>
> On Mon, 24 Jul 2000, Mark Taylor wrote:
> >Th
Hi Mark,
On Mon, 24 Jul 2000, Mark Taylor wrote:
>Thanks for finding this. This was a subtle & rare bug in CBR,
>triggered by the -k option.
Can you tell me when this bug was introduced? I listened to some of my recent
encodings and found this blip in about every 20th track.
>(-k, by the way,
Hi Arne,
Thanks for finding this. This was a subtle & rare bug in CBR,
triggered by the -k option.
(-k, by the way, is *always* a bad idea. It overrides LAME's
default lowpass filters. It will cause ringing and twinking
at 128kbs)
Note to developers and -X6 fans:
The problem is in frame 232
Hello Mark,
Wednesday, July 19, 2000, 5:14:31 PM, you wrote:
MS> The site is dead for me also ... 768K DSL from Ohio USA. Murphy's law has
MS> been strong this year.
MS> mark stephens
yes it seems. thanks for letting me know. So, this afternoon both
sites are down. (1st time in 4 months)
any
Thanks to all who offered advice
Eric
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
On Sun, Jul 09, 2000 at 03:40:29PM -0700, Ralph Giles wrote:
> On Sun, 9 Jul 2000, Don Melton wrote:
>
> > Thanks. This fix is also in but in a slightly different way. I modifed
> > the actual tagging routines in lame.c to check for Ogg Vorbis output
> > rather than making the client code respo
On Sun, 9 Jul 2000, Don Melton wrote:
> Thanks. This fix is also in but in a slightly different way. I modifed
> the actual tagging routines in lame.c to check for Ogg Vorbis output
> rather than making the client code responsible, so we only have to do
> the test in one place rather than in al
Thanks. This fix is also in but in a slightly different way. I modifed
the actual tagging routines in lame.c to check for Ogg Vorbis output
rather than making the client code responsible, so we only have to do
the test in one place rather than in all clients.
On Fri, Jul 07, 2000 at 01:08:55AM
OK, the patch (with several modifications) is in the CVS tree.
On Thu, Jul 06, 2000 at 08:26:22PM -0700, Ralph Giles wrote:
> On Thu, 6 Jul 2000, Don Melton wrote:
>
> > Cool! I didn't even think to add vorbis support when I landed the id3v2
> > stuff this week. Nice idea, Ralph!
>
> Thanks.
On Thu, 6 Jul 2000, Ralph Giles wrote:
> Oops, on further investigation those 128 bytes are an id3 tag, so we
> need an interlock of some sort. I'll take a look at this later.
The attached patch should protect against id3 tags in ogg files. I now get
a zero-length output file.
-r
--
[EMAIL P
On Thu, 6 Jul 2000, Don Melton wrote:
> Cool! I didn't even think to add vorbis support when I landed the id3v2
> stuff this week. Nice idea, Ralph!
Thanks. :-)
> This is probably a stupid question, but are there docs online about the
> comment header (he asks without even looking at the vor
Hi Tonic!
> >It looks like your LAME version 3.85 is an alpha version and not
> >a released beta. In LAME we have currently two versions of VBR code:
> >1 the "old" VBR code
> >2 the "new" VBR code by Mark
>
> No this is beta I do not DL any alfa
> I DL it on the day of release of 3,85 from rusi
> David Balazic wrote:
> >
> > Yeah , but what if someone in Germany wants to use it ?
> > He can not , unless he breaks the law.
>
> Isn't that the same with LAME and even an MP3 decoder? All
> you people using LAME in the US and Germany are currently
> breaking the law :-).
>
> Erik
If you only
> > I'm looking for a DOS/Win cmd line or even GUI util to do some or all
> > of the following:
you could also have a look at http://mp3renamer.de (it's english, too). This
program (called MIR) has tons of features.
Tilman
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
In the last episode (Jun 13), Dan Bridges said:
> I'm looking for a DOS/Win cmd line or even GUI util to do some or all
> of the following:
>
> Given: ARTIST_ -_ ALBUM_ -_ 01_ -_ TITLE.MP3
> use a cmd line like RENMP3 ARTIST_ -_ ALBUM_ -_ 01_ -_ TITLE.MP3 3 4
> to produce 01 - TITLE.MP3 or 0
> > b: if using the pre-computed tree, the same codebook is always used?
>
> I'm not sure what you mean... The pre-computed tree is just an
optimization to
> shorten search time on a given codebook. Each codebook has a custom tree
> generated for it.
I was thinking that you were using a differen
> Does that means that:
>
> a: if encoding time is not important, it should be possible to use a full
> brute search
> instead of the pre-computed tree?
Yes. Brute forcing the LSP fit is the most beneficial part and actually not
likely to increase encoding time a huge amount (not a factor of 2
> Specifically, Vorbis encodes its output using vector quantization; rather
than
> brute forcing the encoding process (by trying each of several hundred
codeword
> possibilities looking for closest fit), it uses a Monte-Carlo trained
> partitioning tree to find best fit in a few comparisons rather
> My first test of vorbis however was a bit disappointing. Testcase
> was gspi*.wav from SQAM. It had a strange sounding tremolo
> effect, but I consider vorbis still alpha. Perhaps something was
> temporarly broken 8´\
Not temporarily broken, but you did get bit by a shortcoming already correct
MoiN
On Tue, Jun 06, 2000 at 10:30:36PM -0300, Leonardo Stern wrote:
> But what about the player ???
> Anyone have a compiled vorbis encoder / player ?
The xmms input plugin seems to work. After compilation just "cd
xmms; make" and copy the lib*.so into /usr/lib/xmms/Input/.
My first test of vo
> > Perhaps the makefile was modified when I decompressed it or somewhen
else.
> > But Make tells me this "separator" error which disappears when I insert
> > tabs. And I am definately sure I use the cygWin "Make" because the djgpp
> > make tells me a completely other bunch of error messages.
>
>
> Perhaps the makefile was modified when I decompressed it or somewhen else.
> But Make tells me this "separator" error which disappears when I insert
> tabs. And I am definately sure I use the cygWin "Make" because the djgpp
> make tells me a completely other bunch of error messages.
Now I got d
In the last episode (Jun 01), Tilman Sauerbeck said:
> > The Makefile should be correct as-is; you shouldn't have to fiddle with
> > tabs. Try editing your path to only include the cygwin binary
> > directory and run make again.
>
> Perhaps the makefile was modified when I decompressed it or som
> The Makefile should be correct as-is; you shouldn't have to fiddle with
> tabs. Try editing your path to only include the cygwin binary
> directory and run make again.
Perhaps the makefile was modified when I decompressed it or somewhen else.
But Make tells me this "separator" error which disa
> > Extraneous text after "ifeq" directive (this message appears for every
> > "ifeq" in Makefile)
> > and
> > Missing separator (line 237)
> I don't know what to say; I downloaded 3.70 myself and compiled it
> without errors. Be sure that the "make" you're running is cygnus'
> make, and not djg
> What version of cygwin, what version of Lame, and what errors do you
> get? I just compiled the CVS version with cygwin 1.1 with no errors at
> all.
okay, thanks. I'll check my installation of CygWin (I'm new to this, perhaps
I didn't setup CygWin correctly).
Tilman
--
MP3 ENCODER mailing li
In the last episode (May 31), Tilman Sauerbeck said:
> Hi everyone,
> I just downloaded the LAME source code and tried to compile using the
> CygWin software (CygWin is a port of the GNU tools to Windows). As
> "make" failed when reading the makefile I wondered if you have to
> make any changes to
Hello r3mix.net,
Wednesday, May 31, 2000, 1:40:37 PM, you wrote:
rn> Hello ,
rn> Warning: Unable to connect to ORACLE (ORA-12154: TNS:could not resolve service
name) in /usr/htdocs/mydomain-reds/index.php3 on line 241
rn> Fatal error: Call to unsupported or undefined function () in
/usr/htdoc
Hello Kimmo,
On 29-May-00, you wrote:
> Usually the compiler (SAS/C at least) on m68k side can automatically check
> and/or increase the stack size on the fly, if required and configured. But
> I don't know if this works on PPC side or not. Anyone who knows better?
I would prefer, if the User w
Hello Leonardo,
Monday, May 22, 2000, 11:07:07 PM, you wrote:
LS> How do you count frames of each bitrate ?
LS> I want to do some stats myself (with diferentes settings and lame versions)
LS> :o)
win32: musicutter: http://macik.homepage.com
click on mp3 file, and then "statistics - simple"
>>
How do you count frames of each bitrate ?
I want to do some stats myself (with diferentes settings and lame versions)
:o)
> the stats about this:
>
> 3.80
> 32 - 0 - 0,0%
> ||| 128 - 17066 - 54,1%
> 160 -
Hello Robert,
Sunday, May 21, 2000, 9:05:19 PM, you wrote:
RH> Hi Roel,
RH> the speed drop in Version 3.83 is because of the noise calculation
RH> in scalefactor band 21 (at 44.1 kHz samplerate it represents the
RH> frequency range from 16 to 22.05 kHz). If there is some noise in that
RH> scal
Hello Takehiro,
Sunday, May 21, 2000, 6:41:25 PM, you wrote:
TT> I wrote;
>>>I added a new scalefac_scale algorithm to CVS version of LAME,
>>>which enables with -Y option.
TT> Sorry, -Y option is used for Mark's new VBR routine
TT> so I changed "new scalefactor_scale algorithm" from -Y to
Roel VdB schrieb am Son, 21 Mai 2000:
> RV> To be clear, I don't mind, but am just curious:
> RV> Lame 370 gave me 1.7 speed on my Cel400@450 (dkutsanov)
> RV> Lame 383 gives me 1.0 speed on my Cel400@450 (dkutsanov)
> RV> [Lame 384 -Z gave me 1.7 speed on my Cel400@450 (dkutsanov)]
>
> sorry sho
RV> To be clear, I don't mind, but am just curious:
RV> Lame 370 gave me 1.7 speed on my Cel400@450 (dkutsanov)
RV> Lame 383 gives me 1.0 speed on my Cel400@450 (dkutsanov)
RV> [Lame 384 -Z gave me 1.7 speed on my Cel400@450 (dkutsanov)]
sorry should have been -Z : 0.7 speed :)
--
Best regards,
I wrote;
>>I added a new scalefac_scale algorithm to CVS version of LAME,
>>which enables with -Y option.
Sorry, -Y option is used for Mark's new VBR routine
so I changed "new scalefactor_scale algorithm" from -Y to -Z option.
---
Takehiro TOMINAGA // may the source be with you!
--
MP3 ENC
Please ignore my previous message. I found the problem.
Ross.
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
MoiN
On Tue, May 16, 2000 at 09:23:38PM +0200, Ingo Saitz wrote:
> http://www.geocitites.de/brabanzu/BarbaraAnn.gz
Hmm, well, it should have been written
http://www.geocities.com/babanzu/BarbaraAnn.gz
Ingo *ducks*
--
I am the "ILOVEGNU" signature virus. Just copy me to your
Do you mean: what is EAC?
It's a CD ripper.
PAC is a Perceptual Audio Codec.
Ross.
Jeremy Hall wrote:
> whatis pac?
>
> _J
>
> In the new year, [EMAIL PROTECTED] wrote:
> > Why the lenght of the encoded music with LAME isn't the
> same of the original
> > track? EAC can detect the offset at
whatis pac?
_J
In the new year, [EMAIL PROTECTED] wrote:
> Why the lenght of the encoded music with LAME isn't the same of the original
> track? EAC can detect the offset at the beginning of the track and fix it, but
> the problem remains at the end of the file. Is it possible to correct this
>
Howdy,
> -Original Message-
> From: Adam Whitehead [mailto:[EMAIL PROTECTED]]
> Sent: Wednesday, May 10, 2000 6:14 AM
> To: [EMAIL PROTECTED]
> Subject:
> Sorry for this not totally MP3-encoder-related mail, but I
> was wondering
> if anyone had some pointers to web sites, books etc. I
Hello Jaroslav,
try resubscribing. You should get a message then saying that you're
already/still subscribed.
Regards
--
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__ __ / / / / Better than ever!
\ \ \ \/ / / /
\ \ \/ / / / 730 kKeys RC5 (210MHz)
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Hello Jaroslav,
On 05-May-00, you wrote:
> All are killed by I-LOVE-YOU virus :-)
The Amiga certainly not!
Regards
--
__ __
/ / / /
/ / / /
/ / / /
__ __ / / / / Better than ever!
\ \ \ \/ / / /
\ \ \/ / / / 730 kKeys RC5 (210MHz)
\ \/ /
Hello Pierre,
On 28-Apr-00, you wrote:
> I use WinAmp (2.62) to play my mp3 files encoded by lame (3.70), and I find it
> rather bad. The mp3 files are not faulty, since they sound good with the
> Windows Media Player. But Windows Media Player doesn't have playlists, so I'd
> like to use WinAmp
Hello Shawn,
On 25-Apr-00, you wrote:
> reproduce them). I'm in serious doubt as to whether frequencies >13kHz really
> contribute any musicality to the tracks on CDs.
Wut? Then you obviously never heard of Heavy or Trash Metal, needs much
Bandwidth or it will sound like a muffeled Sock.
Or hav
> No, this is really a bug. I got bitten by it three or four times until
> I just started using "- - < infile > outfile" to work around it. Like
> this:
>
> $ ls -l infile.mp3
> -rw-rw-rw- 1 dan dan 2745849 Apr 20 20:48 infile.mp3
> $ lame -h -a -b3 2 infile.mp3 -
> Could not find "2".
> $
> Any ideas?
Sorry, that was a mistake in the pipe, lame was innocent.
Felix
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
In the last episode (Apr 20), Monty said:
> > A bug report that arrived in my inbox -
> > Seems like Mitya needs a "do not overwrite" mode.
>
> His shell already offers one, as does UNIX file permissions. Do we
> really need...
>
> "Are you sure? [y/N] y"
>
> (Now if LAME is doing something pu
Just to contrast with the last sample, here's another pre-echo related
problem where LAME is reported to do better than FhG:
Mark
> From: [EMAIL PROTECTED]
> Date: Thu, 6 Apr 2000 18:13:39 +0100
> Content-Type: text/plain;
> charset="iso-8859-1"
> X-UIDL: g"#"!cHY!!C+G!!6=`"!
>
>
>
Here's something interesting about German patent law:
--- Start of forwarded message ---
Date: Thu, 3 Feb 2000 12:32:31 +0100
From: Patrick Goltzsch <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
CC: [EMAIL PROTECTED]
Subject: Re: lame and patents
Though my original answer might be more c
Actually you can extend a Win95/98 DOS window to 43 or 50 lines which then
allows scrolling up. It is in the Properties/Screen tab. It doesn't completely
help because there are more than 50 lines printed. It goes back as far as
"--lowpass freq".
I have the same problem. I have become rather s
.
mark stephens
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Cavallo de Cavallis
Sent: Friday, February 04, 2000 2:22 PM
To: [EMAIL PROTECTED]
Subject: Re: [MP3 ENCODER] Re: highq mode
Wow what a f**kin useful remark !! so stop compiling lame for
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