pretty much the best I could get but I know for example that
--nspsytune normally enables -X1, but -X3 sounds quite a bit better
although it is significantly slower... which isn't too big of a deal
to me. Also, I know that from earlier conversations --athlower isn't
perhaps the greatest
Note:
Both methods (thousands of functions and thousands of tags) are equivalent:
* use one function ( lame_ioctl() ) and thousands of constants
to tell this function what functionality is actually requested
* use thousands of functions (lame_x () ) to execute a
Hello... I've been lurking on this list for awhile now and I've
really started to become interested in some of the more advanced
aspects of lame such as the experimental modes and stuff. Basically
what I am trying to get out of lame is the highest possible sound
quality short of using
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Hi Frank,
1. FFT filters are strictly speaking no filters (they are not a LTI system),
so they have some nasty properties, which are more or less audible. The
audibility depends on the steepness of the filter. So high passes should
never be made by FFT filters. Never ever.
I was going through my backlog of email that still needs to be
answered, and I came upon Rob's Jun 21 message (below) about the
maximum value of the big_values range. As I read this message for the
third time, still not sure what to think about this issue, it occured
to me: This has to be the
What about the proposal to design and discuss a well designed lame API now
without implementation? AFAICS this takes at least 3 months, if you want to
have a durable and neat API, Mark don't want to change the API until summer
2001. May be should use this time to design all the structures
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Date: Mon, 02 Oct 2000 18:32:58 +0900
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Mark
True, it was the -t encode switch.
By the way, isn't "lame -?" -t disable writing wav header when
using --decode a bit misleading about this?
Yes, that is a little misleading: "-t" option is listed twice,
since it disables wav headers when decoding, and disables
Xing headers when
For decoder enthusiasts: 100hz bug fixed versions of Nitrane and their
upcoming Nitrane replacement are available at:
http://www.sulaco.org/mp3/winamp/winamp.html
Nullsoft sent me these .zip files, but I haven't tested them
yet. They are probably also available somewhere on www.winamp.com.
::
:: Question: When fs_in/fs_out is not representable by a:b with little a and b,
:: what do you like best:
::
:: [_] Lame rounds fs_in in a way, so fs_in/fs_out is representable by little
a:b
:: [_] Lame have a function to resample exactly any ratio
:: [_]
I think the mp3 encoding library should do what it is
supposed to do, encode pcm samples into mp3 frames.
BTW, Robert and everyone, how should we treat with mpglib and lame_mp3_XX
routines in frontend/get_audio.c and amiga_mpega.c ?
I think these routine should be another library,
Hi Robert,
This just seems like a lot of work for no real gain since all 4
libraries are samll and closely related. To build LAME we would need
a ./configure setup which builds 4 different libraries. so I guess I
just dont see any reason to seperate them.
Special applications, like embeded
OK, I think and plan to do so, too. My plan is to add 2 functions to the API,
named "lame_init_vbrtag" and "lame_finish_vbrtag".
1 int lame_init_vbrtag(int guessed_framenumber)
- for the initialization(malloc buffers and make gfp-vbrtag = 1, and so on)
- returns VBR/ID3 tag size.
On Sat, 30 Sep 2000, Heribert Maier wrote:
Welcome Michael!
I like your program very much. I have been using it for a year now and
it is *really* great! Summing up, you're a wizard at least on
ID3-tags. :-)
I second that. Never seen this great program before. Been very useful
:: As a value of 200 for BLACKSIZE showed an improvement in resampling, why
:: does is still got a value as low as 25?
::
Low pass, high pass and resampling code should be replaced by artefact-less
program code.
Which filters have artifacts?
low pass code is implemented with a very
I just separated frontend from library. but many things does not work
correctly yet (VBR histgram, etc), or even checked(many configure
option, etc).
Mark suggested that "library" code should be in the libmp3lame, but
before moving to it, I want to everyone to check my modifications.
As a value of 200 for BLACKSIZE showed an improvement in resampling, why
does is still got a value as low as 25?
Regards,
--
Gabriel Bouvigne - France
Hi Gabriel,
Increasing BLACKSIZE only improves the sharpness of the lowpass
cutoff. For resampling, I dont think we need an
Should I start implementing this now, or wait after 3.87 release?
Hi CISC,
3.87 was released last night. It's mostly just to have a
last release before the code re-orginization that Takehiro
is going to do this week. That will involve splitting
the C code into a frontend and library
The first proposal is IMHO enough for everyday use, the second is for
architectures which didn't allow lossless transportation of information
between a void* and an integer type (in case you use ti_data to contain
the information instead of a pointer to the information).
Only "problem" I
Hi all,
I now have free format working in my decoder, MAD. If anyone would like to try
it -- get version 0.11.4b or later:
ftp://ftp.mars.org/pub/mpeg/
So far it seems to work with any free format bitrate I've created with LAME,
all the way up to 640 kbps.
The real reason I'm
Hi Frank,
A couple of comments/questiosn:
:: Also, every transition from two different size windows is lossy. The
:: MDCT is only lossless for overlapping windows of the same size.
::
Is this a problem of bad designed (asymmetric) window functions or a
problem of the MDCT (different
Hi Gaby
Why can we read in the litterature that humans got 25 CB but mp3 uses only
22?
let us try to get it in order:
bark scale is used by the spreading function
Bark 0 : 0-100 Hz, Bark 24: 15.5 - 20.4 kHz
masking is calculated for convolution bands
how about altering some of the mp3 specs themselves and creating a lame
specific mp3 variant?
are there any legal reasons not to do this? would the quality gain be
worth the effort?
The problem is that no player would them be able to play the files. MP3 is
an internationnal
A couple people have a few changes to make
this weekend, and then I'll put out 3.87 on monday.
I also want to compare and study various solutions to the
vbrtest.wav problem (Takehiro, Naoki and Robert all
have fixes!) but this probably wont be done before monday.
After that, Takehiro will start
1. go to transform sizes 1024 and 128
MP3 uses 576 and 192. When 576 is too low for tonal music and 192 too long for
percussions, then this is right. But a 1:8 ratio can create other problems.
Note that MD uses 128, 256, 512 and 1024 sample blocks.
Useful are block sizes from 1 ms ...
i'd like to take this opportunity to point out one of my pet peeves with
libmp3lame, and that is the lame_decoder() function takes a FILE* ... and
then helpfully closes it for you. so that you helpfully get a SIGSEGV
when you try to cleanup the files you open and close them again.
Is it theoretically possible to amplify the sound in a mp3 file without
reencoding it? What would be the quality loss of this operation?
Yes: modifying the global_gain field in each channel of each granule in a
Layer III frame has the effect of amplifying or diminishing the decoded
I would like to use the lame encoder in an embedded application. I have it
all up and running but I have had to make some alterations to the code base:
I also support this 100%. LAME has been slowly moving in
this direction, the key word being slow.
Some minor changes/suggestions:
I agree this 100%! I think we should make more simple mp3 encoding engine.
Compared with vorbis, what a messy code LAME is !
I want to make "example" or "frontend" directory and move lame.c and gtk*,
brhist.c and so on into it.
Any Ideas ?
I like "frontend". And also a "libmp3lame"
Takehiro Tominaga schrieb am Die, 19 Sep 2000:
Wow! great, Florian!
M All the various options (GTK, libsndfile, VBR historgram, mpeg
M decoding) are available through ./configure options. see
M INSTALL for details. By default it will try to install
M everything,
LAME now has a ./configure script! It is from Florian Bomers.
It still needs work, since it's only been tested on two systems.
./configure ; make ; make install
should install lame, lame.h, libmp3lame.a and libmp3lame.so
in /usr/local/{bin,lib,include}
If you have problems, it would be
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Date: Sun, 17 Sep 2000 19:45:10 +0200
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The declaration of
Gabriel Bouvigne schrieb am Son, 17 Sep 2000:
Well, CBR doesn't care about sfb21 distortions, VBR does
for MPEG-1. When using --noath the sfb21 still gets its
ATH value.
Ok, but how about --athlower 100 ? Will it leads to an endless loop? That
could be a problem. I think
I added the stuff recently posted about making a libmp3lame.so to the
Makefile, but just now realized the following problem:
the LAME API is frozen. Different versions of LAME dont
require any code changes to the calling application, but
they do require the calling app to be recompiled since
Thanks - I'll revisit your website.
Another question :) Is there a preferred (or even mandatory)
sequence to command-line opions for Lame ? I remember how quirky
DOS could be in this respect
The only problem is if you use incompatiable options - LAME
does not check for
On 12 Sep 2000, Frank Klemm wrote:
RPC support, SMP support and distributed computing would be a very nice
thing for LAME and would be an outstanding feature.
Are there are plans and interests to support this? Not this year, but
starting with this things in spring next year, not earlier.
Frank, your are missing a third option:
PROJECT_MAINTAINER_DOES_NOT_CARE_AND DOES_NOT_WANT_THIS_CODE_IN_LAME
The display is updated every 50 frames. It is simple
and works well enough.
I want this type of code kept to an absolute minimum since this kind
of stuff really belongs in the front
I'm monitoring the encoding of material originally recorded in the
30's (now on CD). When I look at the MDCT and scale band info:
Left and Mid Channels closely match.
Right Channel shows very little energy.
Side Channel shows very little energy.
In CoolEdit LR "appear" to be almost
I'm not too fond of your new naming-scheme for defines to check if a include
has already been included (__VERSION_H__ f.ex.)...
Generally this scheme is only for internal compiler defines and system
includes, and normal projects should not use pre- and postceding underscores
in defines
Have someone measured the coding delay for FhG and lame ???
How many PCM samples must be feeded to achieve the first MP3 frame?
May be FhG have a longer delay to see more of it's "future"?
mp3enc3.1 high quality (default): encoding delay 1160
For other modes it is slightly different.
CBR VBR ABR
Can use different frame sizes no yes yes
File size depends also on
complexity of sourceno yes no
CBR and ABR adjust bits based on the PE. VBR adjusts the bits
based on the quantization noise.
I've decided to pass some waves thru cooledit to remove content above 16k.
I'm trying to use the frame analyzer to see the results after encoding.
I thought there were 21 scale factor bands. In the frame analyzer it
appears to show 22???
Hudson
--
There are 22 bands, but only 21
On Fri, Sep 08, 2000 at 06:48:49PM +0200, Steve Lhomme wrote:
Hum...
And if 19 is a magic value, why didn't you use the following ?
BLACKSIZE = 200
filter_l = (BLACKSIZE - 19)
| David: can you run the same test with a stencil 10x bigger?
| To do this, change:
|
|
Frank Klemm schrieb am Fre, 08 Sep 2000:
Now a list of "a" values. Sorry about using a little bit of math.
So see on the list below.
There are some stereo pieces of music with a "a" 1.667 or 0.600
1.25/1.92 Giora Feidman/Rabbi Chaim's Dance -- [15] Tishrei Saba.wav
Hello folks,
for some reason I have to convert something like 200 CDs to mp3 and
they should stay on 4CDs. An average bitrate of around 32kbps should
be o.k..
As lots of these CDs (old recordings) are actually mono, although it is
not specified on the cover, and as on a certain point
This could very will be a bug, but it my also just be the correct
resonce of the medium quality lowpass filter used by LAME.
For resampling, LAME is using a 19 point lowpass filter with
a Blackman window. The position of the window is adjusted
to do the resampling at the same time.
For a
Hi Everyone,
The LAME web site is down for over 24 hours now, as is my
email ([EMAIL PROTECTED]). So I haven't gotten any email
from the list or otherwise.
The web hosting company has 'lost' my account, but hopefully
this will be fixed in another day or so. In the mean time,
if you sent me
Mark Taylor schrieb am Mit, 06 Sep 2000:
So as promised, but delayed, LAME CVS mainline is reverted back
to as it was Aug 31. The previous version is in the 'pfk1'
branch. If you want it, you need to do:
cvs update -r pfk1
That branch has a lot of nice code improvements
Hi Everyone,
I haven't been keeping up with things for the last week, because
my wife and I just had our first baby :-) (baby's requisite website:
www.wildpuppy.com/baby)
Anyway, now LAME CVS fails all my test cases. This is normal since
small changes in just the order of operations will show
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
it would be a JS mode, but unlike the "-mj" mode it would not try to predict
anything, but just achieve optimal quality in an empirical way.
---
for cbr: encode each set of samples to both a M/S and a S frame and
take the one with least amount of introduced distortion.
Why do the win32 versions not show on the fly bitrate analysis for vbr
files?
Get Your Private, Free E-mail from MSN Hotmail at http://www.hotmail.com
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/
From: [EMAIL PROTECTED]
Date: Mon, 21 Aug 2000 06:40:53 -600 GMT
What is the point of free format ?
can winamp play it back ?
Well, Winamp crashes when I try to feed him with a free format file (which is
correct). It may be the input I'm using (in_mpg123.dll) . But it may be Winamp,
From: "David" [EMAIL PROTECTED]
Date: Mon, 21 Aug 2000 18:42:28 +0200
ok i'm asking again :)
what is freeformat ? (the point of it, why i should/should not use it
compared to "normal" mp3 ...)
it is a fixed, but arbritrary bitrate. No reason to use it
unless you want an exact bitrate
:: 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
Date: Mon, 21 Aug 2000 13:52:48 +0200
From: David Balazic [EMAIL PROTECTED]
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X-UIDL: V"#!cKF!!#5S!!;+7"!
Hi!
I'm sending this message to both lame
Date: Mon, 21 Aug 2000 16:44:17 +0530
From: Soyeb [EMAIL PROTECTED]
What's the testing methodologied adopted by LAME encoder.
Does mpeg talk anything on that?
rgds
soyeb
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Are you talking about bitstream complience, or
All:
I've been doing some tests on the accuracy of decoder implementations.
More specifically, I have carried out the tests for MPEG audio decoder
compliance spelled out in ISO/IEC 11172-4. The results are interesting, and I
want to share them:
From: [EMAIL PROTECTED]
Date: Sun, 13 Aug 2000 13:36:03 -0400
So, do we all believe that fraunhofer encoders
still edge out lame at low bitrate cbr?
Recently, I heard from a company that works with lots of radio
stations in the US was evaluting several codecs at 128kbs CBR. They
did
Which encoding engines are being used in MusicMatch and Realaudio Jukebox?
I seem to remember some discussion here about the Fraunhoffer encoder in
MusicMatch being crippled (not bitrate limited but missing some "features").
Which program has the better encoder? Can either one use LAME as
Date: Fri, 11 Aug 2000 20:37:00 +0200
From: Roel VdB [EMAIL PROTECTED]
Hi,
I don't know how hard it is to get lame converted to a win32 codec,
but I think it would have it's uses.
Any win32 people wanting to give it a shot?
I believe Vorbis now has an ACM wrapper, so starting with
Date: Thu, 10 Aug 2000 10:11:35 +1000
From: engdev [EMAIL PROTECTED]
I assume that most users of LAME are using the Linux OS, as the GTK
extensions are based around (I believe) X-windows. I don't use Linux,
and am stuck with windoze. Are there any utilities around to run the
frame
this is strange... obiously something has changed in 3.86...
C:\cdexbetalame.exe --abr 201 -b160 -h -mj f:\temp.wav f:\templame385.mp3
LAME version 3.85 (www.sulaco.org/mp3)
Win32 binaries from www.chat.ru/~dkutsanov/
Using polyphase lowpass filter, transition band: 20805 Hz -
Date: Sun, 30 Jul 2000 20:03:22 +0200 (CEST)
From: Alberto García [EMAIL PROTECTED]
I noticed this encoding a 2 seconds wav from the end of a song:
lame --abr 192 -h 1.wav 1.mp3
lame --decode 1.mp3 2.wav
The output from the decoding was:
input:1.mp3
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
What about an option "adjust-level-for-psycho-model", which increases the
level for the threshold computation, so low level music is coded with more
bits. To my mind low level pieces of music with a turned up volume control
are coded with too less bits. lame is coding for a full scale SPL of
I have briefly tried the "--voice" mode and the "normal" mode when
encoding a purely voice signal (with background noise) at 8kbps, and
have been very impressed with the difference. I would like to compress
the signal more... but 8 is as low as it goes.
The "nomal" mode renders the voice
I've encoded some wavefiles with Exact Audio Copy and Audiograbber using
Lame_enc.dll (3.85). I measured the time needed for encoding at 128 CBR (joint)
stereo. The measured times for encoding at HQ were the same as for encoding
at LQ.
Furthermore, the measured times Exact Audio Copy
Another question:
Is there any tool to analyze the number of SI, MS and LR frames in a MP3?
Frank, you just need a GTK enabled version of lame :-)
run lame -g on the mp3 file, scroll to the end, and then
click 'show' under the 'stats' pull down menu.
It shows the info you want, and any
1) With my encoder (64kbps stereo CRC), every fricative is almost painful to
listen to, as the pink noise bursts end up being narrow band filtered (due
to lack of bits - only the MDCT coeffs closest to the pole are making it
into the bitstream), and there are occasional weird high frequency
I'm writing a little code snippet to read MPEG headers and report
the info contained in them. After eliminating invalid headers by excluding
bad/invalid bits and combinations, it still recognizes a lot of
headers that aren't actually headers. Are there any other methods
for eliminating
Hello,
I believe Robert said that mp3 frames overlap 50%, then would it be
sufficient to init some values using _only_ the previous frame in
order to play the next ok?
So: I want good playback, starting with frame N, would it suffice to
load up frame N-1, and then start playing @
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Date: Fri, 28 Jul 2000 21:02:58 +0200
From: Roel VdB [EMAIL PROTECTED]
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Hi.
On Die, 25 Jul 2000, Roel VdB wrote:
best use "lame --decode", as it's the only correct decoder around
(check that decoding site) (but it makes no difference here)
What is wrong with mpg123 -w ? Or was "only" meant in a way that encompasses
only the windows-world?
I
(-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 this
Could anyone please tell me the difference in the header for the lame dll versus the
bladeenc dll.
Having seen the almost fourfold speed increase of lame I definitely want to continue
with lame.
From a previous posting from Albert Faber I understand that the interface is very
similar to
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Date: Mon, 24 Jul 2000 17:04:17 BST
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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
As a side note, I'm getting a blip inbetween songs, is this because of
headers and such? I would think that since it's raw PCM it shouldn't do
that... Basically, it would be the same as doing:
bash# (lame --decode in1.mp3 lame --decode in2.mp3) | lame -b 56k out.mp3
...and getting a
In http://privatewww.essex.ac.uk/~djmrob/mp3decoders/index.html
there's a comparison of mp3 decoders. Lame is one of the three that passed
the test (the other two are Winamp 2.22 and Ultra Player). But mpg123 is
not tested. What about it? Does anyone know any decoding problem with
I think it is certain that this problem is caused by noises
concentrated on pure tones.
A noise on a single MDCT coefficient increases as it's
amplitude increases. This is because quantized values are
actually values powered by 3/4.
According to the theory of psychoacoustic, it is a
Hi All-
I've been playing around with the latest version of GoGo-no-Coda. In the
setup portion of the program the user can disable psycho-acoustics. The
program says that when encoding at 128 kbs, quality of encoding is
improved with psycho-acoustics enabled. The implication given
Date: Tue, 11 Jul 2000 14:37:35 -0700
From: "Chris Haynes" [EMAIL PROTECTED]
We know that LAME now has roughly the same quality of MP3Enc 3.1. But,
as far as I am concerned, full huffman search hasn't been implemented
on LAME yet. I've noticed LAME 3.8x produces better quality than 3.70,
I've got another question, and I thought it'd be better to start a new
thread with it. There are some controversy with the quality and bugs
of LAME VBR mode. Would this affect the ABR mode, too? Which is safer
to use (though ABR is called the "safe VBR mode!): 192 CBR or 192 ABR?
Can I
Hiya,
The polyphase filterbank has a delay of 511 (encoding + decoding together;
there's no such thing as a delay in _only_ en- or decoding).
Ciao,
Segher
Hi Segher,
How did you get 511? observations, or analysis of the algorithm?
I've spent quite a lot of time on this, because
Hi Mark,
Mark Right now, the spreading function
Mark is normalized so that (for example) convolving s3 with a constant
Mark function will not remove any energy and return the same constant.
Is there any reason why energy should be preserved here?
just a convention. s3[][] is
There is a relation between sound pressure level and sound intensity level
if there is a plain wave (in german "ebene Schallwelle")
L = 20*lg(p/p0) dB = 10*lg(I/I0) dB
p0 := 2E-5 Pa
I0 := 1E-12 W/m^2
1W = 1 Nm/s
for coherent sounds one uses the sound pressure
(coherent sounds:
Hi,
I have just commited changes to add --nspsytune option.
With this option, vbrtest.wav is encoded perfectly, and
encoded sound becomes more natural to my ear though encoded file size
increases.
--nspsytune command line option turns on following things
1. Addition of
I was under the impression that VBR mode was better...so I have been trying
to use it. But at this point I have absolutely no idea if what I'm encoding
is better than CBR mode or not. I have absolutely no idea if I'm using a
good set of options or not. I'll probably just wait until lame
When encoding 44.1 kHz audio to a 128kbit/sec mp3, lame by default cuts
off the high end with a transition band of 15115 Hz - 15648 Hz. (32 kHz
audio to 128kbit/sec mp3 has a slightly lower cut-off by default with a
transition band of 14065 Hz - 14452 Hz.)
If someone was encoding
After 8-9 hours of trying to grasp a fraction of the source (no
programmer, no psy-coding background, plain dumb :)) (in a futile attempt to find
the reason why
"-b" is misinterpreted (?)) this catched my eye:
qadjust=-2.5; /* start with -1 db quality improvement over quantize.c VBR
There is so much stuff constantly changing with lamenew options, VBR,
CBR, etc.. At this point I have totally lost touch with what mode is what
and which flags I should use. I sure hope you guys will start thinking
more about usability at some point
-steve
We do think
Hi Mark,
I would love to add this to the LAME distribution! And think
it is worth making the necessary changes to LAME.
A couple of questions:
Which version of LAME are you working with? A lot of work *has* been
done to make LAME re-entrant. This has not yet been tested or
debugged (maybe
Hello,
I just compiled the newest alpha, and was testing the new vbr mode.
I decided to like it (_a lot_), since it's 3x faster than --vbr-old (!!).
I did some reading up on the list, but I cannot find an answer to
some Q's, maybe someone can address them, thanks
1- What was the
Hello,
In order to give that idea I had last month about an "album-header",
in order to complete (imho) mp3 as a platform to encode all kinds of
cds seamlessly, as opposed to only non-live/mix cds given current
implementation, a chance, I'd like to get some information :) [yes, it
is
Is it just me, or is it virtually impossible to encode VBR files with the
new VBR code with switches like this "-X 6 -V 0 -q 1 -b 128 -B 128 -F"?
Because it takes like 4 hours per song to encode. With the old VBR code it
takes 4-6 minutes. Something has changed I take it?
Josh
It's
Vorbis by the way, is a VQ codec
...
I'm sorry, but I'm pretty stupid *wry smile*. What defines a VQ codec?
Ivo
VQ combines quantization and entropy coding into one step. It takes a
vector of real valued MDCT coefficients and replaces them with a
integer code word.
MPEG takes
I've compiled lame with the free Borland C++ compiler (for Windows).
Some code changes were necessary. In ts_process_time,
LARGE_INTEGER Kernel = { KernelTime.dwLowDateTime,
KernelTime.dwHighDateTime };
LARGE_INTEGER User = { UserTime.dwLowDateTime,
Hi all.
I found a bug of MS threshold adjustment, but I don't have a paper
describing the algorithm(*). so I want to someone to check this.
Hi Takehiro,
I put that code in a while ago, before we had the freq2bark() formula.
Since the scalefactor bands are supposed to be equally spaced
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