me, then i have to skip this valid frame, is this correct?
Right. In the worst case, you need to skip 255 frames.
> Can anybody suggest me a pseudo-algorithm that considers both problems
> (resyncing & back-referring of "main data")?
Sorry, I've ever wrote an mp
gt; lame -s 16 -b 24 SPM4854.wav SPM48524.mp3
>
> encodes an mp3 at 8 kHz and 24 bits. I have sox-can I use that to
> help me make a file with a frequency of 16 kHz?
"-s" is the option to indicate input file frequency, not for output
file. Use "--resample 16" instea
;atid=300290&aid=1379187&group_id=290
It is the patch for the trunk, but there's the same problem on 3.97beta2.
I just fixed it on the branch (rev 1.23.4.1 of id3tag.c).
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From: Takehiro Tominaga <[EMAIL PROTECTED]>
Subject: Re: [mp3encoder] bug in id3v2 ??
Date: Tue, 06 Dec 2005 00:23:16 +0900 (JST)
> > i'm using LAME 3.98 under WinXP EN SP2 and it crashes when i use ID3v2 genre
> > higher than 9. The folowing examples differs
the bug report on sf.net.
Thank you for your valueable bug report.
> Is it possible to have track number as a string, i.e. "01" is "01" and NOT
> just "1" ?
Sorry, but this is not possible with current LAME (version 3.xx).
On experimental branch, it is possible
ke everything went
> fine. The diffs are below:
Thanx. fixed in CVS.
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kHz, 1 channel, when it's really
> 44 kHz, 2 channel (which mp3info confirms).
It may be because your mp3 file (Track12.mp3) is damaged.
The decoding part of LAME is not main function and poor to handle such files.
Anyway, it could be fixed. Can you send the file to me ? No need to send
whole o
e.h is always used when compiling libmp3lame and lame.exe.
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needed on such compiler. What will happedn when you remove the
include statement and compile it ?
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this audio file
> format so that it can convert the wave file to mp3?
Use libsndfile enabled lame.
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> However, I noticed that the executable found at
> http://rarewares.org/dancer/dancer.php?f=2 is not
> only smaller (184 KB) but also slightly faster at
> encoding MP3s.
I think the binary is compressed by UPX or something.
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ly integer number.
I think it may be a small change, but do not good for 3.x branch.
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E3.93 uses 22.05kHz as sampling rate when --abr 56.
LAME3.96 uses 24.00kHz as sampling rate when --abr 56.
check "-s 22" option with lame 3.96.1.
If the problem is solved, the player you used may have a problem
with 24kHz mp3.
If not, I have no idea.
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rimental") does it, too.
But it uses subband filter output instead of fs/4 high-pass filter.
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have seen some code where the upper bound is set according to the last
> scalefactor band. (i.e. 522 in case of MPEG2 22.05 kHz).
>
> could you please explain this?
I think it is the "DIRTY" hack to reduce the complexity of decoding.
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ge 865 is for Denmark/Norway, #850 is for (most of) Western Europe,
> as far as I remember)
>
> This has worked very well for several hundred files now!
>
> Thanks again for your help.
Great digest !
Could you allow us to these instructions to t
s.
> The same goes for ID3 tags. At best they get converted (my guess is that the
> MSB gets stripped off).
That is probabry the problem of your shell (like DOS-prompt).
I have no problem with "my" national character (kanji and kana and hira-gana),
on my linux box.
ingw" or "mingw32".
I've just committed the experimental patch to my experimental branch.
See configure.in revision 1.83.2.34.
But... I have no environment to test it.
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arrays -mtune=athlon-xp -MT
> > mpglib_interface.lo -MD -MP -MF
> > .deps/mpglib_interface.Tpo -c mpglib_interface.c
> > -DDLL_EXPORT -DPIC -o .libs/mpglib_interface.o
contains "-I../mpglib/". Ummm, really strange.
FYI: I can successfully compile them on cygwin with the option
dling such large data may lead a bug.
And, there're these bugs in LAME4.
If you think it is easy, why don't you write it yourself and send us
a patch :) ? We always welcome such contribution.
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t; A note about the default value for "scale" in --longhelp or switchs.html
> would be nice.
use "--verbose" option to display the value of scaling.
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LAME4 experimentaly supports the "preserve TAGS when transcoding/encoding"
but it is experimental feature.
I don't know this feature will be imported to LAME3.x. IMHO, it will reduce
the stability of LAME and is not good to do so.
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>Why does Lame's CBR and ABR modes reduce the amplutude of the sound data
>by 0.44 dB (RMS) while VBR "tries" to preserve it and "succeeds". Can I
>turn it off? Thanks.
use "--scale 1.0" option with ABR/CBR encoding.
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t + XING-VBR header ...
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all) sound cards generated is not exactly
32000Hz.
Usually, you cannot synchronize the clock generator on your sound card
by NTP. That's why it is "10hours + 4seconds".
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quot;OFFICIAL" binary.
>
> Takehiro wanted to say: "There isn't" instead of "There's".
OPS, you are right, Alex.
Thanx correcting.
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gt; by re-encoding over 15.0 GB of MP3 (1400 files total). I found a
> problem with one file in particular. The link below will lead you to a
Thanx, but many of them are fixed, IMHO. Try the latest 3.96 beta.
TIA.
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e on icc on windows). No gcc support and no none-windows
support for it (unless you break the license ...)
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d of intrisics.
Now the situation is changed about availability of intrisics, but the
rest of them are not changed much, IMHO.
Anyway, if someone writes the asm code with intrisics, it will be welcomed
if it runs fine, of course.
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From: rocko <[EMAIL PROTECTED]>
Subject: [mp3encoder] safe to use NASM-SSE in LAME ?
Date: Thu, 19 Feb 2004 11:07:56 +0100
> Can anybody tell me if i am safe with using the NASM-SSE now ?
No, it's not safe.
I think we should remove it at least from the release package.
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rontend/get_audio.c) line 1458
> Patch 3: function decodeMP3_clipchoise() (mpglib/interface.c) line 345
Thanx. committed.
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t; do you think there is need to open the output stream also with fopen64 ?
>
> I think it would be better.
Output may be bigger than input (when with --decode), so it is important.
But, by default, the output of decoding is RIFF-wave which does not support
over 2GB So we have to check t
I think BIG endian is not problem. I can compile/run LAME on MacOSX and
SPARC Solaris successfully and they're also big endian.
>>Code: bad PC value.
Is it Ok ???
Is it really in FFT ?
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roblem,
Or everyone think it was fixed.
As for me, I am not much interested in it and am not willing to fix it.
> It would be a great project to work on, if there is demand for it. I
> will try to find the time to help, if needed.
Do you write the code ?
If yes, it sounds great. Let
LAME tag when
> decoding an MP3 file? Does it depend on what encoder was used for encoding that
> song?
Yes, it depends.
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his is from LAME, but these words are common for
many application.
And I found the below string in the binary,
"fatal error. MAXFRAMESIZE not large enough."
This is in the lame/libmp3lame/bitstream.c, but they are also common
strings, I afraid.
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ntly, LAME's ACM, DirectShow, and DLL are quite buggy and noone
want to maintain. IMHO, we have to drop them from LAME if the situation
is continued ... at least from LAME4 release.
So, is there anyone willing to help us ?
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Takehiro TOMINAGA // may the source be with you!
spsycho uses predictability and nspsytune uses peak/avg ratio
as tonality.
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ch] nor part2_length[ch].
And as you say, you should consider in scfsi[scfsi_band] (or
scfsi[scfsi_band][ch]) value when you decode scalefactor value.
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h
free to send
> enhancements to [EMAIL PROTECTED]
But I don't need anything except hydrogenaudio.org about this region.
IMHO, r3mix is already obsolete and brings only wrong information...
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mp3
t the beginning of the MP3
> file.
>
> Can anybody help me to solve this problem?
check you do not treat the header (RIFF-wav file header) as sound data and
encode it into mp3.
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mp3encode
ettings with appropriate low/high
> filters. I would like to stay in the 32-64kB/s range.
use --preset abr 40 or so.
LAME's VBR is usually inferior to ABR/CBR at these bitrate.
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mp3encod
From: "Rich Koziol" <[EMAIL PROTECTED]>
Subject: Re: [mp3encoder] LAME setting for speech encoding
Date: Tue, 12 Aug 2003 18:24:43 -0400
> Thanks Takehiro,
>
> On 9 Aug 2003 at 20:55, Takehiro Tominaga wrote:
>
> > From: "Rich Koziol" <[EMAIL PRO
ted to Title of ID3 tag.
* "ITRK" tag, which is an extension used by Turtle Beach's Audiotron,
is converted to Track Number of ID3 tag.
- It works only when libsndfile is not used.
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t ix^4/3 to be
* as close as possible to x^4/3. (taking the nearest int would mean
* ix is as close as possible to xr, which is different.)
That's all.
adj43 is initialized to fit the requirement.
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nt result.
By profiling the LAME, FFTs (for long and short) takes only about 4.5%.
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ucing what
> that was doing in later LAMEs?
-q0 on 3.88 is almost same with -q2 on LAME3.90 to LAME3.93.1,
and -q3 on LAME3.94 alphas.
And they are the default setting of each of them.
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to SIMDed.
>>3.8% sqrt
You may speed up the sqrt() by changing sqrt(x*sqrt(x)) to some polymorph.
Probabry this mail will help you.
http://sourceforge.net/mailarchive/forum.php?thread_id=1209312&forum_id=5500
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) for MPEG2/2.5.
And it works fine now (with my "little" tests).
The fix is already done in my experimental branch.
Any comments ? If it is right answer to fix the problem, I will
backport to the main branch.
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EG1.
http://cvs.sourceforge.net/cgi-bin/viewcvs.cgi/lame/lame/Attic/takehiro.c.diff?r1=1.69&r2=1.70&sortby=date
I searched the mail archive arroud these days, but I cannot find out
the reason at the moment :(
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Takehiro TOMINAGA // may the source be with you!
ng external libraries, like for vorbis. This code
G> should probably be cleaned up.
I think this has been already cleaned up.
pcm.c is no longer the part of LAME code.
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output:
:
M> Segmentation fault
It because Chris.mp3 seems have an ID3 tag and lame does not support it.
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>>>>> "G" == Gabriel Bouvigne <[EMAIL PROTECTED]> writes:
G> (we really need a good FAQ)
Seems there's tech-FAQ compiled by Mark.
http://lame.sourceforge.net/tech-FAQ.txt is it.
But I agree it is too much "technical"....
Does anyone willing
>>>>> "C" == Chetan Krishnamurthy <[EMAIL PROTECTED]> writes:
C> Does lame support layer-1 and layer-2.
No.
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http:
CVS.
Because the "warned" code is not called by the default frontend(lame
itself), I think this is not fatal bug of LAME and we don't need to
release the newer version, IMHO.
Does anybody disagree and want to release the new one ?
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ith sox, then pass the wav file through lame.
D> sox -bU -r 8000 -t raw 44.pcm -uw 44.wav
Or build LAME with libsndfile enabled.
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ecause it cause an internal compiler error for many files.
My test is done on my Duron 1GHz, Linux2.4.20, 384MB memory.
I don't have and don't know about the P4 box, but at least for me,
icc7 was not good as I expected...
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> If LAME made --alt-preset 96 or something thereabouts the default, then
this
> would solve the problem in SHOUTcast and would also obviate the need for
> --alt-preset 96.
>
> In other words, why make "crappy" the default?
Changing default setting needs concensus. And at least for me, I disagree
and so on), you cannot.
And except something which we cannot emulate, you can see the rest of
setting by using "--verbose" option, like
lame --alt-preset standard --verbose in.wav out.mp3
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tion is in-place (that is, the input data is
r> over-written by the output data)
yes, you are right. the calculation is done by in-place algorithm.
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l[i] * (1.0 / ( 1L << (8 * sizeof(long) - 16)));
in_buffer[1][i] = buffer_r[i] * (1.0 / ( 1L << (8 * sizeof(long) - 16)));
Yes, one character patches. change two "1" into "1L".
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> Why are there mp3 files in which the value of part2_3_length is zero?
> e.g. http://php.s3.to/bbs/up/img/216.png
> Isn't the part2_3_length length of main data per 1granule, 1channel?
Because the frame contains totally no sound (the frame is digital silence).
j> speak) mp3s.
Your patch will be used by (L)GPL-piracy vendor and they will love you.
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