I regret to inform you that Peter Gubanov, Elecard co-founder, was
killed in an avalanche on 2nd March, 2007.
(I am only becoming aware of this now)
Peter provided us with the DirectShow filter implementation for Lame.
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at
http://mpg123.orgis.org/lame-resampling.shtml
I'd be thankful if someone could comment on that.
Well, this could possibly be a bug of the resampling.
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an engineernig point, it simply makes no sense. When this was
decided, it made sense considering a financial/strategic point of view.
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Takehiro Tominaga a écrit :
I just fixed it on trunk (The experimental branch is not affected because
the tagging code is changed at all to support UTF8).
Should I back port to 3.97 branches ?
Yes, please
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are not requiring it.
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had a look at configMS.h, but not at machine.h
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with libsndfile handling the input.
In this case it would work.
Another solution would be to open you file in an audio editor and save
it as uncompressed pcm wav file. (it might even work with the basic
sound recorder shipped with windows)
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personal
(or more) systems can also encode several
tracks in parallel with several instances of Lame. You will have some
nice speed boost, with zero quality reduction.
Regards,
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Gabriel Bouvigne
www.mp3-tech.org
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with
some encoding modes (still compliant) we are using in separated stereo.
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into the header of the file. If you use a
player able to read those info, the decoding will be gapless.
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,
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LAME4 experimentaly supports the preserve TAGS when transcoding/encoding
but it is experimental feature.
In the meantime, you could use a program that is copying tags between
two files.
I know that such a program exists, but I do not remember its name.
Regards,
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,
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/R/M
channel, y for S channel
I am sorry, but Lame does not support equalization settings. Those
settings are only related to psychoacoustic maskings.
Regards,
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[EMAIL PROTECTED] wrote:
My question is: Is the ATH Type 3 curve and the noise shape Type 2 better
when using with -q0 and -V0 then the current one?
You would better keep the default settings
Regards,
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is not relevant in this case, as the fees are
regarding mp3 technology, and not about a specific mp3 implementation.
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purpose of
creating some streams, he/she doesn't have to acquire a license.
The DISTRIBUTOR of the encoder should obtain the license, but not the
end-user.
Otherwise every MusicMatch user should acquire a license...
Regards,
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Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3
) --abr 172
regards,
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, but you can check the
following:
*lame website (gpsycho related pages)
*source code comments
*http://gabriel.mp3-tech.org/lame/diagrams/
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Kristian Hermansen wrote:
You were correct. The decode was the problem and here is the output:
--snip--
I uploaded the problematic portion of the mp3 file on
http://gabriel.mp3-tech.org/lame
On decoding this file Lame is crashing (access violation)
Regards,
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stream. I'll try to check
that in my source that I got from Fraunhofer.
Lame supports 24bits wav files fine, and is able to encode in vbr.
(to be sure that there was no bug I just tryed, it works perfectly)
regards,
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to wait for lame v4 to have it in the dll. In the
meantime, there is the possibility to use the static libmp3lame.
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to the compiler, with gcc intrinsics not
usable with VC or ICL, and VC intrinsics not usable with gcc?
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on technology implementation, but we
can not do anything regarding potential patent fees.
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was thinking you all doing that for fun and i notice it's more to have
stuff and to complain a bit too have them..
We are working on Lame for fun, and are not complaining. You are the
only one to complain here.
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computer just to port Lame on it.
In the meantime, I think that the command-line version should work just
fine on your new system.
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of your problem, like wich
settings you are using.
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you should use add --scale to your encoding.
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, or does it sometimes manage to
encode them
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bitrate, how would increasing the
default PE increase the number of allocated bits. The allocated bits are
constrained by the bitrate when using cbr.
Regards,
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some diagrams to http://gabriel.mp3-tech.org/lame
I made them about 1 year ago, so them might not be up-to date.
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pass noise will have a very low tonality value
(idealy 0).
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mp3 technology.
We are saving you the cost of technology implementation, but are not
able to provide you a license agreement on the general mp3 technology,
are we do not own the patents.
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.
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.
The same patent point would also apply to Mpglib, Mad, the Linux kernel,...
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Gabriel Bouvigne wrote:
I am writing them a mail asking for clarification about this.
Issue solved, next version will include a mention regarding Lame in the
about box.
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the mp3 coding scheme.
But the winamp redistribute the lamedll.dll in it's shout source dsp plugin setup
package.
Nullsoft has a licence agreement covering the mp3 patents:
http://mp3licensing.com/licensees/index.asp
Regards,
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*correct the offsets if needed in the pcm stream
*encode it again in full stereo
*do your cooking/mixing between the original mp3 and the new frames that
you can grab from the new mp3.
Regards,
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than with a C++ dll because C is
not ABI-dependant like C++, and so you have no problem when using
different compilers.
For VB programmers, a quick Google search returned this:
http://www.zetnet.co.uk/rad/dll.html
Regards,
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personal page: http://gabriel.mp3
is not the case of all Lame developpers.
*we (at least I) are not so sure that using COM is easier than to link
with a C dll for a beginner.
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is using a tonality estimation. Tonality can be estimated by using
peak detection method (Gpsycho) or predictability (NSpsytune).
The tonality estimation controls the ammount of masking.
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personal page: http://gabriel.mp3-tech.org
a specific (but fully mp3 compliant) plain
stereo configuration used by Lame.
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,
Takehiro is doing a lot of work on the experimental branch.
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and the MP3 file has
the scratch at the beginning. I don't know what am I doing wrong.
Perhaps you considered the wav header to be sound data.
Perhaps you forget to write vbr header/info tag
Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org
. If it is
indeed broken, I would think it obvious for the LAME crew to either fix it
or circumvent it. Thanks for the heads up.
This was fixed in 3.93.1, as mentionned in the history.html
Regards,
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, we do not plan to use sub-versions. We had to in
this case, but we would really like to avoid this.
Anyway, your patch is there, and if in the future we encounter a case
where sub-versions are needed, we will consider it, although our intent
is to avoid such situations.
Regards,
--
Gabriel
Charlie Lenahan a crit :
For those that have ripped all their CD's to wav w/ LIST tags, like those used by the Audiotron.
This patch will set the ID3 tags when converting a wav = mp3
Thank you for your submission. This should be included soon into Lame.
Regards,
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. This will be included soon.
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the tagging functions to properly display
and return an error, I am ok with it.
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encoded files.
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mode and mono.
I think that some people on this list tryed to port lame code to fixed
point. Perhaps they might give you some indications.
Regards,
Gabriel Bouvigne
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compiling it under visual studio .net (aka visual c++
7), you will just speed things up by about 5% compared to visual c++6.
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datatype.
You might also want to reduce the allowed dynamic range. Anyway, you will
probably have to reduce quality in some way in order to reduce processor
requirements.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org
MPEG2 Layer 1 do differ from
MPEG1 only in its multichannel feature; audio encoding for a channels are same
for MPEG1 and MPEG2.
ISO recommends to use psycho
model 1 for both layer 1 2.
Regarding Layer 3, this
uses entirely different audio representation. You may have to
I got the Layer-2 code from
Lame.
You probably
means that you downloaded dist10.tzg, as Lame doesn't include layer II
code
But the problem with the ISO code is that, It doesnot
work for MPEG2-Layer1.
You've got a
good point here. I am assuming that you are interested in the low
for
it.
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Gabriel Bouvigne
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dos you can can test the value of ERRORLEVEL in a batch file. I am not
sure, but I think that this might work.
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Gabriel Bouvigne
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--scale-r 0 -m m (for discarding right)
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get has nothing
to do with Lame. Lame is an audio codec, and your message is complaining
about a video codec missing.
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-? lameopts.txt 21
It should redirect stderr (output 2) to stdout (output 1) and redirect
stdout to lameopts.txt.
It works under cmd (NT5 shell), but I do not know about command.com under
DOS (is it really DOS or command.com from win95/98?)
Please report status.
Regards,
Gabriel Bouvigne
www.mp3
it means is the technology itself features this gap. All you can do is
to store the gap size along with the stream so the decoder can remove it.
Lame stores the delay, so a decoder could use this info to totally remove
the gaps.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
personal page: http
. This is probably old code from
someone who though that perhaps Lame could handle that using external
libraries, like for vorbis. This code should probably be cleaned up.
Btw for layer II you could try TooLame.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3
hai there, does anyone know what is the time complexity of MP3 encoder
algorithm such as Lame or the Fhg? i mean what is the big O number? is it
O(n), O(n2) or else? could someone tell me how to measure this up? thanx
alot
Encoder complexity is roughly linear related to the number of input
)
It is the common behaviour of win32 console applications. You have either to
use the short name, or to put the filename between double quotes.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
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- Original Message -
From: Sycotic Smith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 04, 2002 11:07 PM
Subject: [MP3 ENCODER] non-portable code in lame.h - compiling on cygwin
[PATCH]
I have been trying to include lame as an encoder for the mplayer/mencoder
your wmp version and your
operating system version.
Regards,
Gabriel Bouvigne
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artifacts
in
the upper midrange.
I'd suggest you to use the 96 kbps preset (--preset 96) with Lame 3.93.
About Shoutcast I think that we should ask to have different option in the
Shoutcast dsp plugin.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org
. for MEncoding?
It will be included in the 3.93 libmp3lame.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org
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better...
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work.
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personal page: http://gabriel.mp3-tech.org
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processors. Some code that is working fine
on Intel/Amd is not working on such processors.
You can add --noasm mmx to your command line. This will disable MMX
optimizations and allow you to run Lame.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org
I'm using described setting, is there anyone who could produce better
result for GSM speech, please?
Lame --alt-preset 16 -m m
Gabriel Bouvigne
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. It is more an
example code with some potential bugs...
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likely with abr/cbr
With --alt-preset standard, I'm reaching about 1x on my Duron 750. But on my
laptop PIII 1GHz, I'm nearly encoding at an average 2.5x with the same
settings. For --alt-preset standard, you might want to try --alt-preset
fast standard
Regards,
Gabriel Bouvigne
www.mp3
is much lower than Lame.
Regards,
Gabriel Bouvigne
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.
Unfortunately it seems that the dll internal doc was not updated, and so so
programmers didn't called this finishing function.
In order to understand a few of the advantages, you could try using Encspot
on a such a file. Then select Lame tag on the right click menu...
Regards,
Gabriel Bouvigne
Could you point us to some specification on how to write/adapt a
decoder to do this? I'm guessing that somewhere in the Lame info
tag there's a field indicating how many samples/frames/granules to
omit from the start and end of the file - is that correct?
Hi Steve,
The problem was that the initial VBR tag (all zero's) was inserted into
the
MP3 file for CBR files, so when the beWriteVbrTag function was not called
at
the end of the encoding process you end up with an empty frame at the
beginning of the MP3 file.
It is on purpose that the INFO
be some padding
when in the last frame there is remaining space after audio encoding.
Moreover you could check the info tag which indicates encoder delay and
padding for the file in number of samples.
Bye,
Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org
mp3_tags_fid.
Also a suggestion: perhaps we sould write a valid xing/info tag with an
empty seek table at the beginning instead of just 0 padding? This way if
mp3_tags_fid is not called the entire information won't be present, be the
tag will still be valid.
Regards,
Gabriel Bouvigne
www.mp3
a bunch of zeros at the
start of the file does it get rude and display Not MP3.
Might be an indication that your player is broken. Compliant mp3 players
should skip unrecognized data.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org
outdated, I'd like the
generated files
to stay in the cvs repository.
Regards,
--
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I'm not sure, but I believe MPEG1 L1 and L2 only has frames
of 1024 samples.
1152
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option!!!
The curent problem is very unlikely to be a problem with small frames, so -b
or -F won't help testing.
Regards,
Gabriel Bouvigne
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--decode, Lame will ignore this
frame.
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by the so-called Painter Spanias formula,
wich is a great improvement.
Later it was replaced by another formula wich was improving for 15 kHz, but
produced quite similar results for lower freqs.
Have a look at utils.c, the ath curves are there.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
itself.
But I might be wrong as I'm not a lawyer.
Either Christina Bonner (current licensing coordinator for AAC ) or her
predecessor (who also send me a nice letter in the past) never contradicted
me on this point.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
from
them.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
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suggest you to have a look at references on the Lame website, I
added references to papers about this tonality estimation.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
- Original Message -
From: reinhard
To: [EMAIL PROTECTED]
Sent: Monday, January 28, 2002 10:58 AM
Subject: Re: [MP3
suggest you to have a look at references on the Lame website, I
added references to papers about this tonality estimation.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
- Original Message -
From: reinhard
To: [EMAIL PROTECTED]
Sent: Monday, January 28, 2002 10:58 AM
Subject: Re: [MP3
Happy new year to everyone.
--
Gabriel Bouvigne - France
[EMAIL PROTECTED]
mobile phone: [EMAIL PROTECTED]
MP3' Tech: www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org
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Here are the debugging info I have when I encode to 44kHz/96 kbps and I
get a
32kHz/96 kbps file/stream :
...
1161 3866.37668017 [288] Disable Resorvoir =1
Is this one on purpose? Seems strange to disable bit reservoir
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They don't give reasons why they use LAME. But I'm really curious to know.
I'm not sure if it's really legal to use LAME for commercial selling of
the
encoded files (because of the patents on MP3).
They're probably using Lame because it's saving money compared to FhG.
For sure it's legal to
have only 1
granule, so they are only 576 samples. The sample rate of MPEG-2 is half the
MPEG-1 sample rate.
Regards,
Gabriel Bouvigne
www.mp3-tech.org
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Is there a way to predict this timeshift? Does anyone know if it's a
variable based on encoding or constant?
It's because of the encoder delay. Now (3.90) Lame writes a kind of header
into the first frame, and one of the fields of this header is the encoder
delay.
Given that could a well written decoder strip it off and give time and
length coherency between source and result?
Yes, it's perfectly possible. However this tag is quite a new thing, and I
think that right now there is no encoder that uses (yet) this info.
(including me usually).
As an example, I only use lame -mj -V1 for my encodings, nothing else.
I use only the other switches for experimentation.
Do you think that the basic switches section is too complex?
Regards,
Gabriel Bouvigne
www.mp3-tech.org
URL?
http://golem.cs.berkeley.edu/~slingn/research/publications/mm_workload.htm
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