if it has been
corrupted. As it uses bits for the crc, it will lower (a little) the quality
of CBR encoding, and will higher the filesize of vbr.
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also: isn't that "-35" not extremely harsh on the athlower?
I think that using a negative value with athlower is a bad idea.
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, but it's easy to fix in the decoder, now that
we know exactly what the problem is.
The advantage is that 8205 is 0x1FFE, which is not a valid syncword for
mpeg1-2. With this, audio resync on corrupted bitstreams would be easier for
the decoder
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? It seems that some decoders
report false durations, even for cbr encoding.
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Oh, oh no!
Do you want to start a contest of writing the dirties possible program?
main(_){float x,sin();for(_=0;__*_8;_+=_) ...
you forget the reverse access to arrays :-)
int a[5];3[a]=2;
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be upgraded soon, this is why there is only a
few mmx on Lame.
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results.
I compiled both releases (with and without mmx) with VC6, and the mp3 output
is bit identical on my Cel460/win98.
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If the mmx version of choose table is used for the compilation, what
will
happen on a non-mmx cpu?
it will crash, and it does so on my old P100.
That's bad. Would it be possible to compile both routines in the same
binaries, with an auto detection?
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), and after reducing to i mode starting from the
latest band (22th) down to a given freq, according to the number of bits you
need to save. According to the DTS whitepaper you can go down to 3-4kHz.
But it's perhaps a little too simplist.
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and,
therefore, useless.
Well, if you want to stay under win95/98, you can use FlaskMpeg to backup
your data, as it includes a built-in mpeg-2 decoder and got an avi output
plugin. It helped me a lot, as I now don't have to decompress first in a big
wav before re-encoding.
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be nice to have a --l3wav option in lame allowing to
output an mp3 with a wav header.
About the ACM codec, there is none yet. It will perhaps be easier to write
one when the ACM Vorbis codec will be finished, I only hope that vorbis guys
are not waiting the same thing about lame.
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.
Have you tried digging into audio patents? It would perhaps bring an idea.
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calefactor.
Is there a scalefactor for 16kHz in AAC? (Meno, are you listening?)
Also an off topic question for Robert: as you're german, is there a specific
knowledge about audio compression floating around in your university? (like
specialists, research or thesis)
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think that we must
follow it.
For creating a new standard, there is another
project called Vorbis.
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that there is a problem?
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in
Ansi C.
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--athlower 100 would
result in an incredible ammount of bits in the latest sfb. I'll test it and
post the results.
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.
This is why I think that trying to put the sfb21's ath value in both masking
and ath would be good, if possible. This way, using -v1 the sfb21 masking
would still be reduced, but it would be possible to also use --athlower with
some high values.
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not
provides full frequency response, but that's all.
Moreover your sample is not good for testing a psy model. Psy model is made
for modeling the human hearing, not an electronical analyser.
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The results of the modification of the resampling filter size are shown
here:
http://bboard.mp3.com/mp3/ubb/Forum1/HTML/003127.html
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decoding, I think that it would be better to repeat the latest valid frame
than just skip the invalid one.
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I've got a question: is there any program allowing to add a wav header to an
mp3 file?
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Gabriel,
Post the file somewhere on mp3-tech.org or other ftp site , so we can take
a
look at the problem
The file is on gabriel.mp3-tech.org
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e reaching the full potential of
vorbis, or beeing close to it.
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for
optimal quality/filesize at 96 kbps ?
I would add -h
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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
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not serious of course)
[curious thing is I never seem to offend any ladies on this list]
A good point. Is there any lady on this list. I think that a lot of people
subscribed to this list. Statistically, it would be strange that only men
subscribed.
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. But I don't remember which one is which one, and I'm not
sure if it works on every OS.
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,
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in
the benefit of --preset voice, but some people disagree about it.
If you're really interested about what was behind this option
when it wascreated, I still have a webpage about it.
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accurate.
a good thing.
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bits. (I don't know about the FhG vbr).
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there is a 320k frame, and that's why I
choose to limit to 256k.
This rectification seems important to my mind.
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choose something like lame --abr 32 -h -mm --noshort input.wav
output.mp3
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I just want to ask which layer2 encoder do you prefer or recommend??
(SOLOH or toolame?)
I'd recommend using Qdesign.
If you want a free one, I'd use toolame rather than Soloh, as toolame is
soloh+some improvements.
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MP3
bandwidth for LR stereo
//
// °) mostly prevent by automatic downsampling
Why using forced joint by default? I'm personnaly againt it, as it leads to
spacialization artifacts.
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,
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would need a lot of coding.
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filter be usefull for mp3?
I think I must be missing something about it.
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able to give better quality.
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-md is not documented.
What is it ? Does anyone made a dual channel mode?
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of the dll file even see
it.
I'm trying to do it in the html doc. I plan to add to it a page about the
basic options because there are now too much ones in the full switches page.
If you've got any suggestion about the html doc, please share them.
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doesn't understand what Frank is telling?
Can anyone explain a little?
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mind, the goal of -mj, so any change should be made into -mj
I'd be slow, but it'd be the best possible quality for cbr and best
quality/size
for vbr. (as far as I can see)
not so slower, if both sets of data are already computed
just my 2 centimes about it...
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gone in forced mode, it could be a toggling problem. Roel, could you
check this?
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it is that in english, it seems that a lot of things
like "in my mind", "in my opinion",... must be used in order to not beeing
perceived as beeing rude, but it's less evident in other languages and not
necessary a reflex for non-english people writing in english.
Regards,
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Why do the win32 versions not show on the fly bitrate analysis for vbr
files?
They do. Try the one located on mp3-tech.org, and you'll see bitrate
analysis
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here is the 3.86 doc
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html.zip
I'd like to find a free format encoded file.
Does anyone of you know where I could find one or an encoder that support
free
format (also known as free bitrate) ?
lame --freeformat -b yourbitrate in.wav out.mp3
Regards,
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MP3
as this player uses Xaudio
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treat an mp3 file correctly
whose mode(stereo - mono) is changing during one file.
Switching between any stereo modes (stereo, m/s, is, ms and is) is allowed,
but switching between stereo, mono and dual is forbidden by the standard.
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But isn't the ATH adjusted according to the
VBR scale?
It's right that a switch to manually adjust the ATH would be good, as I also
think that some low level pieces of classic music are encoded with too few
bits.
A rough solution is to completely disable use of the ATH with --noath
Regards,
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,
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2nd problem
~~~
Some of my CDs have a lot of DC offset. Other are well DC free. Examples:
What is DC?
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, it would be good to change a little the vbr scale, or
coefficients affecting the file size.
Is anyone ok with this or am I saying (telling? I never know wich one to
use) something wrong?
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w if the behaviour is the same
as --voice.
If it's doing the same, I'd suggest you to stop using --voice and start
using --preset voice instead.
I personnaly think that now the --voice switch should be removed, as there
is --preset voice.
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ic
by Kalah.
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is
brocken?
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.
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, but I don't want to use alphas, because to my mind
they are not targeted toward final users but only testers ans developpers.
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Taken straight from the cdex 1.30 beta1 help page, is this true with 3.85 ?
does it really cause ringing artifacts with high quality on ?
It has been right with a few betas in the past, but I think that now you can
use the -h switch safely.
Regards,
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it.
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I forget why the -B option was added, but it should not be
used under normal circumstances.
It was added because there are some decoder chips wich can't handle more
than 192k frames.
For (strange) cases like -b128 -B 128, why not made lame using cbr instead
of vbr?
Regards,
--
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), as many applications are unable to deal with v2 tags.
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ium
hacked FhG codec, in order to choose the stereo mode.
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producing 128k, how do we produce lower bitrates vbr? With
using -V 15?
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idn't succeded in obtaining an FhG license", or
something like this.
It seems that as long as an article is not written by some of us, there
always be some mistakes in it.
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Xing evolved from tompg, which I think is also a direct
descendent of the ISO code. The tompg source code is out there,
so you could check.
It's the second time that someone mentions the tompg sources as beeing
available. But does anyone knows where to find them?
Regards,
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the law are Dmitry, myself, and all the people downloading
the binaries from our websites.
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,
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quite sure you wouldn't gain 50% in re-using computations.
This is my opinion, perhaps others could tell us what they think about it.
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Mambo X
http://www.mambox.com
According to the sheet, it seems interesting. However it seems that there
are only a few ones manufactured, and you could wait 3 months or more before
getting your.
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Are the frequent cut-offs different at the higher bitrates, like 96, 128
and 192? (Yes admittedly I probably won't be streaming over 128kbit/sec,
but I'm curious.)
Yes they are. It would be silly to try encoding full frequency range at 96.
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...
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distortion ratio. However
such a thing would increase buffering complexity.
What do you think about it?
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Probably because of different compilers, and different
optimizations
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- Original Message -
From:
Ampex
To: [EMAIL PROTECTED]
Sent: Saturday, June 10, 2000 11:42
PM
same size
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description of the encoding process in the
vorbis doc
would save you time, removing a lot of questions like mine?
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rk doesn't have any against the use of the usage
file.
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up my copy.
Details can be found here : www.wotsit.org
Perhaps a -l3wav option could be added to Lame?
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new message board: as the old one crashed, loosing about 1 year of valuable
posts, I set up a new one. This one is more powerfull than before, allowing
to search or to further edit posts.
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Sorry, I posted to the wrong mailing list
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.
Or use tar!
Anselm
Winzip handles well the cr/lf conversion.
If you're using the tarball, there is something else to do, as the files
contain all the modification steps, but I don't know what.
Regards,
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Gabriel Bouvigne - France
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interest me more than
just using it.
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Gabriel Bouvigne - France
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In the big (too big?) l3psycho_anal function, in the "determin final block
type" section, is there a way to know if the frame is m/s or stereo? I mean,
is there something like blocktype[chn] to know it?
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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.
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iso switch?
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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
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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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
quot;fix" thing?, and
do you think that it should be in the html doc?
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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.
Regards,
--
Gabriel Bouvigne - France
www.mp3
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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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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