In the experimental file do the folowing changes:
l 574:
switch_pe+=80;
l 594:
switch_pe -=350;
it seems to work better like this.
Regards,
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I added an "extra fast" mode. The quality is lower, but in some cases it
might be usefull, as an example if someone with a slow computer needs
realtime encoding from stdin
Regards,
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musicin.c
You apply them by hand? Don't you have "patch"?
I'm under win32
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,
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In fact this extra-speed trick is borrowed from the old xing encoders. It
was one of the secrets of theyr speed. Another is that they don't choose
what huffman table to use, they only got one for long blocks and one for
short ones.
Gabriel Bouvigne - France
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MP3
Just a clarification: I didn't meant that the fft you done is not optimal,
but that 1024 is perhaps not optimal
Gabriel Bouvigne - France
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- Message d'origine -
De : Gabriel Bouvigne [EMAIL PROTECTED]
À : [EMAIL PROTECTED
Anyone knows how much time is spent in each function?
Gabriel Bouvigne - France
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I remember seeing a 768 points FFT somewhere just before an adaptation of
KLT for mp3 encoding.
The problem is to remember where I saw it.
Regards,
Gabriel Bouvigne - France
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De : Mark Taylor [EMAIL
we'll be able to produce mixed blocks
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Does anyone plans to do something with the mpeg2-lsf, or can we remove it?
Gabriel Bouvigne - France
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doc.zip
Celeron 450mHz, 192M sdram
Faithless - Sunday 8pm
track 7 (God is a dj) - 481s
***
Lame pre11 (compiled with VC++)
-m s -f _ 141s _ 3.43x
-m f -f _ 147s _ 3.29x
-f _ 151s _ 3.20x
-m f _ 261s _ 1.85x
-m s _ 275s _ 1.76x
default settings _ 337s _ 1.43x
(finished==false)
{
try encoding;
if (over alwaysover_limit)
bitrate++;
if (there are 2 over =over_limit)
bitrate--;
else
finished=true;
}
this would allow real VBR with full bitrate range, and beeing faster than
current algorithm
Regards,
Gabriel
Do we really need byte reservoir when using vbr?
I'd like to hear opinions about it from various people, as I can't choose
the answer myself. I've got pros and cons about it in my head
Gabriel Bouvigne - France
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choose.
The other thing I'm quite sure is that they don't choose what huffmann table
to use, they only use a fixed one.
Gabriel Bouvigne - France
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- Message d'origine -
De : Mark Taylor [EMAIL PROTECTED]
À : [EMAIL PROTECTED
I was thinking about completely remove it when VBR
Gabriel Bouvigne - France
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- Message d'origine -
De : Mark Taylor [EMAIL PROTECTED]
À : [EMAIL PROTECTED]
Envoyé : lundi 7 juin 1999 00:46
Objet : Re: [MP3 ENCODER] VBR
I've got a paper about an enhanced psy model somewhere on my hard drive.
I'll search about it. I'll put it on my site as soon as I'll found where it
is.
Gabriel Bouvigne - France
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De : Mark Taylor [EMAIL
Why did you changed the min bitrate?
It won't change a lot the file size, but I don't understand why did you done
this.
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in
the header. Ok, it will not be recognized by some players, but they only
have to change a string comparison to be able to work.
We just have to drop a mail to the mp3 player programmers, and they'll
probably fix it. I really wouldn't like the see an "Xing" string in lame
Gabriel Bouvign
YES, that's wonderfull, please remove the ISO psy-model.
Gabriel Bouvigne - France
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- Message d'origine -
De : Mark Taylor [EMAIL PROTECTED]
À : [EMAIL PROTECTED]
Envoyé : vendredi 11 juin 1999 09:06
Objet : [MP3 ENCODER] lame
tails about
those algorithms may be found in [3].
[3] W. Press, S.A. Teukolsky, W.T. Vetterling, B. O. Flamery Numerical
Recipes, The Art of Scientific Computing Cambridge University Press, 1992"
Does anyone think that it could save time?
Gabriel Bouvigne - France
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qdesign announced an mp3 encoding library with the quality of FhG and the
speed of Xing.
Does anyone think it's possible?
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a few speed ups in iteration_loop
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loop.c
forget my last post, it seems that I've broken something
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I don't know if anyone reported it, but the new time calculation loop
doesn't work on win32.
sys/times.h doesn't exist under VC++, so I used the previous loop instead of
the current one.
Gabriel Bouvigne - France
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Did you noticed than using VBR, joint stereo files are bigger than regular
stereo ones?
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lame -b 32 -V 0 -m s track7.wav track7j.mp3 :150kbps
lame -b 32 -V 0 -m s -X track7.wav track7j.mp3 : 213kbps
It's perhaps a too high increase.
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Yes, it's better like this. I apologize for my mistake.
It seems that the ISO ATH is computed from this formula, as it gives the
exact same values as most of those mentionned in tables D1.
Gabriel Bouvigne - France
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- Message
I'm not sure that -X always contributes to reduce the artifacts. They may be
caused by something else, but the question is what?
Gabriel Bouvigne - France
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- Message d'origine -
De : Macik [EMAIL PROTECTED]
À : LAME Developers
I hope the server will survive to the slashdot effect ;-)
Gabriel Bouvigne - France
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- Message d'origine -
De : Frank Baumgart [EMAIL PROTECTED]
À : [EMAIL PROTECTED]
Envoyé : mardi 29 juin 1999 23:38
Objet : Re: [MP3 ENCODER
Just undef HAVEGTK instead of commenting all the gtk lines.
Gabriel Bouvigne - France
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De : Ryan Vanderwerf [EMAIL PROTECTED]
À : [EMAIL PROTECTED]
Envoyé : samedi 3 juillet 1999 05:07
Objet : [MP3 ENCODER
a lot of user wanting a speech option in mp3 encoders. There
was one with l3enc, but FhG dropped it in mp3enc.
What about short blocks? Are they really usefull with voice coding?
Regards,
Gabriel Bouvigne - France
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First of all, those parameters are quite illogical.
In VBR, the -b switch indicates the minimum bitrate. So he's encoding a vbr
file with a minimum bitrate of 256kpbs, wich is quite useless.
Regards,
Gabriel
- Original Message -
From: Mikhail M. Fedotov [EMAIL PROTECTED]
To: [EMAIL
to a 128kbs frame for the Xing header,
but I'd like to find out what Xing does first.
Xing engine seems to be unable to produce an mpeg2 VBR file. With a 22kHz
input file, it upsamples it to 44.1kHz when encoding.
Why not using an mpeg1 vbr header with mpeg2 vbr files?
Gabriel Bouvigne - France
I just wanted to notify that the voice option now only works (and is only
tuned for) with 44.1kHz samples.
Regards,
Gabriel Bouvigne - France
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De : Mark Taylor [EMAIL PROTECTED]
À : [EMAIL PROTECTED
.
As most of the Lame developpers are subscribed to this list, it would be
nice for them to say if they are opposed to the use of Lame in commercial
products. (because it'll be easier than to ask everyone if they agree)
Regards,
Gabriel Bouvigne - France
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,
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very rare, but sometimes
happens... (I only saw 1 bug with -O9 under gcc, and 1 internal compiler
error under both gcc and egcs during the past years)
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try
to use it, but it can decide to ignore it.
Yes please send me the ref offlist. In fact if we can probally take
this discussion off list unless other are interested in it.
I'm also interested in.
Gabriel Bouvigne - France
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Binaries are available from mp3tech.org.
Ps: Mark, I now think that you could put a link on the Website to these
binaries
Regards,
Gabriel Bouvigne - France
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I know it. What I meant is that Blade or 8Hz, as an example, are often
called "iso encoders" because they just use the iso ref code without any
quality change.
Gabriel Bouvigne - France
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D
continue also to provide full tgz.
Regards,
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About the html doc, I hope to be able to post a new one today. (but I must
finish another work related to mp3 first)
Regards,
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Here is an updated html doc.
Regards,
Gabriel Bouvigne - France
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html.zip
let me know.
Regards,
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of about 15% of the bitrate using this.
*Inter-channel masking: Is this used by any mp3 encoder?
What do you think about it?
Regards,
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incorporated in
mpeg-4 natural audio, it can be considered as a good (future) commercial
result.
Regards,
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Here is a little modification over the block switching to prevent switching
in the case where the big surge is from the first subblock to the next ones.
In this case, long block will only lead to a post echo of less than 20ms,
wich won't be heard.
Regards,
Gabriel Bouvigne - France
[EMAIL
about how .4054 could be optimal for any
value?
Regards,
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Anyone has some experience with cvs and win32?
Where to find it?
Regards,
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on it (according to their
website). If this is the case, it would be nice to share code with us.
Any comment from anyone working for Proteron
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Perhaps it's time for a 3.58 release of source code?
Regards,
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Layer II VBR is ISO compliant. However, players are not required to support
it (unlike layer III players).
Regards,
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- Message d'origine -
De : Patrick De Smet [EMAIL PROTECTED]
À : [EMAIL PROTECTED
Should we use libm library in linking ( -lm) or not ?
There is no difference in CBR in quality (only speed).
In VBR there is difference in byte by byte comparing,
and libm is about 13% slower.
Perhaps a stupid question, but what is libm ?
Regards,
Gabriel Bouvigne - France
[EMAIL
384, free, 384, 192, free, free, etc ?
Free bitrate isn't allowed to be mixed with any other bitrate.
Gabriel Bouvigne - France
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subband, then the just higher,...
Or perhaps an assymetrical process, as the very high frequencies are less
important than the lowest ones, in term of quality.
What do the people on this list think about such a bit allocation process
starting from the middle subband?
Regards,
Gabriel Bouvigne
rame1."
Gabriel Bouvigne - France
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- Message d'origine -
De : Takehiro Tominaga [EMAIL PROTECTED]
À : [EMAIL PROTECTED]
Envoyé : mercredi 15 décembre 1999 11:25
Objet : Re: [MP3 ENCODER] short blocks patch
uum,,, why no o
pass filter would be nice, as no one is able to ear
such frequencies, so why encoding them?
Regards,
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adequate equipment this is noticable. It
will make most music have a more muddy and boomy sound.
I was speaking about a 15Hz HIGH-PASS filter, and not about any LOW-PASS
filter.
Regards,
Gabriel Bouvigne - France
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FhG encoders, like Audioactive
or Producer, wich are better quality than l3enc 2.7x, but a little lower
than mp3enc 3.1.
Regards,
Gabriel Bouvigne - France
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Here's an update over the html doc.
Mark (or someone else), could you please put it in the cvs, as I don't have
cvs access during holidays.
Regards,
Gabriel Bouvigne - France
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html.zip
Here is the updated html doc for the 3.59 release.
Could anyone please put it in the cvs?
Btw the link to the html doc wich is on the LAME page is broken.
Regards,
Gabriel Bouvigne - France
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(temporary down, alternative url is http
oft time-domain filtering",
I'll have to update the documentation!
To my mind "soft time-domain filtering" refers to true time domain filter
instead of the "MDCT decimation". (here it is, you've got a name for this)
Regards,
Gabriel Bouvigne - France
[EMAIL PROTECTED]
i
totally backward compatible with the options
controlled by the switch.
Just a personnal opinion...
Gabriel Bouvigne - France
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While trying to find a sample where different block types for each channel
could help, I saw that this option (-d) was always enable. It the -d switch
now obsolete?
Regards,
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(temporary down, alternative url
,
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able
Code Generation:
Pentium Pro
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Now Proteron also got a windows encoder.
So the questions are:
is it also based on Lame like n2mp3?
if this is the case, why is this company earning money from other's work?
Regards,
Gabriel Bouvigne - France
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(temporary down
with and without Lame JS in order to see if
surround info is preserved or not?
Regards,
--
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available MP3 players?
Such high bitrates are allowed, but are required to be constant bitrate
only.
About the players, I don't know as mp3 players are not required to be able
to play free bitrates, and I never seen any encoder producing free bitrate.
Regards,
Gabriel Bouvigne - France
[EMAIL
.
Regards,
Gabriel Bouvigne - France
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,
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d by
Xing.
Regards,
--
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Using Lame 3.70 I saw that it accepts a strange thing:
using an highpass freq higher than the lowpass one
Of course the result is pure silent.
Regards,
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od version of this document. Mine is
incomplete and most tables are completely messed.
Regards,
--
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it's only an option, then it's ok.
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of their
encoders, there is an option for never using padding, always using it,
or complying to the standard.
Regards,
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Adam Whitehead a écrit :
Hi,
I don't know if this is what you're looking for, but my doc
contains the following:
authors:
Dipl.-Ing. Martin Sieler
Dipl.-Ing. Ralph Sperschneider
Issue: 1.2
04.03.97
Is this what you're after?
Yes, that's it
--
Gabriel Bouvigne - France
Segher Boessenkool a écrit :
At high frequencies, humans only decide direction of the sound by the
relative left/right intensity, not by phase difference.
So this idea seems like the intensity stereo. In this case, why not
implementing intensity stereo?
Regards,
--
Gabriel Bouvigne
The MP3 licencing terms are updated.
A new website about it is available: www.mp3licencing.com
Mark: Thomson, not Thompson (main Lame page)
Regards,
--
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al channel was made for using in dual channel transmissions or other
things like this.
In dual channel, each channel has to got exactly half of the bits.
In stereo, you're not constrained to 50% for each channel.
Regards,
--
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www.mp3-tech.org
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Shawn Riley a écrit :
2- Are some people saying Layer2 is actually better than Layer3 at the same
bitrates for some types of music? I wonder if quality could be improved by
switching layers midstream... Do MPEG standards support that?
I think that it's forbidden by iso
Regards,
--
Gabriel
he goal of this mail is not to discuss about patents validity, but to
propose a solution.
I'd like to hear what both people on this list and Proteron think about
this.
Regards,
--
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www.mp3-tech.org
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Mark Taylor wrote:
Latest version of lame, in CVS, should now decode free format bitstreams.
Where did you find free format bitstreams to test?
Regards,
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, ActiveMovie/MediaPlayer
6.4, Xing MPEG player : all failed (quite disappointing)
Regards,
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m valid bitrate for each
layer."
You're reading the mpeg-2 doc. That's why you get lower limits. But your doc
only indicates the maximum bitrate that decoders are required to support,
and does not mention a limit for encoders.
Regards,
Gabriel Bouvigne - France
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call
it mp4 to benefit from the confusion of the names. The format used is a
modified AAC (probably low profile, but not sure about this).
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Could someone please check this:
http://www.stud.uni-hannover.de/~andbusch/audiocoder.html
I don't know if it's interesting, as I don't understand the slightest
word of german.
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vdbj wrote:
(removed long mail)
Perhaps we could add an advanced option for adjusting the encoder delay?.
I know that if reduced to the max, it will lower the quality of the
first frame, but in some cases it could be a better choice.
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Gabriel Bouvigne - France
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that it will lower the quality of the first frame, but it's
very short. I don't remenber the exact of the quality reduction, but
Mark mentionned it once on this mailing list.
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, but
conducting to some differences from the output of other decoders.
However, there are some bugs in ae4 as Sonique 1.51 fails to the 100Hz test
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Gabriel Bouvigne - France
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quot;fix" thing?, and
do you think that it should be in the html doc?
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looking at this doc,
but it seems that I don't have it. Is it possible for you to send it to me?
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440 or 320? According to the ISO doc, it does not seems that there is
any upper limit to the bitrate.
(The only limit is perhaps that it seems that only Xing based decoders and a
few mpg123 based are able to decode it)
Btw, did anyone tryed freeformat on Xaudio or Mad?
Regards,
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Gabriel Bouvign
- Original Message -
From: Roel VdB [EMAIL PROTECTED]
To: Gabriel Bouvigne [EMAIL PROTECTED]
Sent: Sunday, May 28, 2000 4:06 AM
Subject: Re[2]: [MP3 ENCODER] Normalize audio input
Hello Gabriel,
Saturday, May 27, 2000, 2:44:36 PM, you wrote:
GB Good point here. If I'm not too wrong
kHz lowpass freq with 32kHz resampling is impossible.
So in this case I think that it produces the same results as -k (full freq
bandwidth). But once again, I personnally would avoid it.
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Btw, what is Mad?
Robert
Mad is a young integer only decoder, quite fast. I don't remember the url,
but there is some source code on my page.
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Yes, maybe -k would
be a
better idea all round anyway. Sounds fine on my sources.
Please apologize, you wrote 15.1, but my (tired?) brain interpreted as 16.1.
So you can continue to use your lowpass setting.
Regards,
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Gabriel Bouvigne - France
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MP3' Tec
iso switch?
Regards,
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Hello everyone!
What batch/GUI front-end does everyone use for lame under Windoze?
I personnally use Audiograbber with the command-line release of Lame.
I use Audiograbber because I'm used to, but it's a commercial software.
Otherwise, you could use Cdex
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Gabriel Bouvigne
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