Re: [MP3 ENCODER] normalization

2000-09-05 Thread Steve Lhomme

I'm currently doing a simple command-line program to do it, using
libsndfile. It will also trim files (remove blank in the beginning and end)
and adjust DC offset.

I'll publish the source when it reaches alpha state.

- Original Message -
From: "Francois du Toit" [EMAIL PROTECTED]
To: "Lame MP3 Encoder Mailing List" [EMAIL PROTECTED]
Sent: Tuesday, September 05, 2000 2:21 AM
Subject: [MP3 ENCODER] normalization


I want to implement a normalizing routine in one of my programs, can anybody
recommend one? It would be for 16 bit CD audio.

Would simply multiplying by a constant factor and rounding be good enough or
would the rounding errors cause some problems

sorry for the offtopic

Francois


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Re: [MP3 ENCODER] the -mx mode - different philosophy

2000-09-05 Thread David Balazic

Frank Klemm wrote:
 
 ::   -md is not documented.
 ::
 ::  What is it ? Does anyone made a dual channel mode?
 ::
 ::
 It's like -ms, but no bitrate balance is possible between the channels
 (maybe also for the main purpose, dual language, a disadvantage from the
 quality aspect) and also a hint for the decode to only decode one
 (selectable) channel.
 
 Main advantage is the lower CPU usage.

Just to clarify : -md is "dual channel" ?

David Balazic
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Re: [MP3 ENCODER] VBR, distortion, and CBR

2000-09-05 Thread David Balazic

Frank Klemm wrote:
 
 ::   This is a little bit problematic, because the number of distorted
 ::   bands does not tell you the weight of distortions you will get.
 ::   What do you think sounds uglier:
 ::20 distorted bands, each 0.1 dB
 ::   or
 :: 1 distorted band by 2 dB
 ::   ???
 ::
 ::  I would say that 20 distorted bands at 0.1dB is preferable, and assume
 ::  that this is the choice of lame's algorithms. Is this assumption wrong ?
 ::
 Yes and no.
 
 That depends on the distortion of the undistorted bands. Also undistorted
 bands have an distortion, it is negative.

You are saying : "has a distortion" != "is distorted" ?
 
 First you need a table of the probability to hear distortions:
 
 distortion  probability error noise
[d=dB]  [p=%]  [i]
 -1050.0   0.
 -3 50.8   0.0004
 -2.5   52.0   0.0025
 -2 53.2   0.0064
 -1.5   55.6   0.020
 -1 57.9   0.040
 -0.5   63.3   0.116
 -0.25  66.3   0.176
  0 70.2   0.281

isn't 0 dB distortion == no distortion ?
How can it be heared ?

 +0.1   72.2   0.336
 +0.2   74.2   0.422
 +0.3   76.1   0.504
 +0.4   78.0   0.593
 +0.5   80.0   0.706
 +1 87.1   1.28
 +1.5   93.7   2.34
 +2 97.2   3.65
 +2.5   99.0   5.43
 +3 99.8   8.4
 +3.5   99.9   9.5

[ snip ]

Sorry if this is an "reader IQ too low" error :-)

David Balazic
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Re: [MP3 ENCODER] normalization

2000-09-05 Thread mailing-steve

Addition :
I currently use a Windows shareware to do that. It's called 'WaveTrim' (or 'WavTrim')
and works fine.

I'm currently doing a simple command-line program to do it, using
libsndfile. It will also trim files (remove blank in the beginning and end)

and adjust DC offset.

I'll publish the source when it reaches alpha state.

- Original Message -
From: "Francois du Toit" [EMAIL PROTECTED]
To: "Lame MP3 Encoder Mailing List" [EMAIL PROTECTED]
Sent: Tuesday, September 05, 2000 2:21 AM
Subject: [MP3 ENCODER] normalization


I want to implement a normalizing routine in one of my programs, can anybody

recommend one? It would be for 16 bit CD audio.

Would simply multiplying by a constant factor and rounding be good enough or

would the rounding errors cause some problems

sorry for the offtopic

http://www..org
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Re: [MP3 ENCODER] Please stop breaking LAME... ;)

2000-09-05 Thread Jason Antony



Mark Taylor wrote:

 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)

Hey! Congrats, Mark! Give my regards to the new mother too. How are they
doing? I only wish I knew more about the lil one cos your website seems to
be currently down, along with www.sulaco.org But anyway take care and all
the best : )

Regards
Jason

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Re: [MP3 ENCODER] newbie question

2000-09-05 Thread Gabriel Bouvigne

 On www.r3mix.net, the following command line is recommended as the best
 option (size, speed, quality-wise):

 -V1 -h -mj -b128 -q1

 I was looking over the documentation that came with the zip file of LAME
 3.86b, and I couldn't find any mention of the "- q" command.

I didn't documented tha -q switch because it's a beta switch, and its
behaviour can change from a release to another. The final goal of this
switch will be to create a quality setting, going from -q9 (worst) to -q0
(best), but it's not finished yet.


 Can someone here explain? Does it apply only for VBR, or can it be used
with
 CBR as well?

It can be used with CBR as well

 Thanks.

 One last question, for the best quality at the expense of file size, is
"-b
 320 -h -k -m s" the best option?

This is likely to be the best quality setting, also some people (like me)
could argue that -m j would be able to give better quality.


Regards,

--

Gabriel Bouvigne - France
[EMAIL PROTECTED]
icq: 12138873

MP3' Tech: www.mp3-tech.org


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RE: [MP3 ENCODER] lame source C++ compatible?

2000-09-05 Thread alex . broadhead

Howdy,

 How to compile lame on a system where only a C++ compiler is
 available (the
 C compiler costs extra money)? Currently lame generates
 nearly uncountable
 errors with a C++ compiler.

I'm having a similar problem trying to compile LAME for my system, for which
only a BASIC compiler is available...  It seems like practically every line
generates an error or warning of some sort?!?  Can someone address this?

Thanks in advance,
Alex

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RE: [MP3 ENCODER] Frank's coherence [off topic?]

2000-09-05 Thread alex . broadhead

Howdy Gabriel,

 Am I the only one wich doesn't understand what Frank is telling?
 Can anyone explain a little?

You are not alone.  Once I get past his aggressively authoritative posting
style (which is a major disincentive to understanding for me), I find I
understand about half of what Frank posts.  (Some of which seems actually
both relevant _and_ useful.*)  This is not one of those postings.

I assume he has some theoretical basis for wanting to correlate
differentiated signals (rather than raw signals), and that this is somehow
relevant to LAME (I haven't grepped through the source, but I remember
hearing some discussion of a correlator being used for joint switching?),
but, since he mentions neither before reeling off a long tabulation of
statistics, it all seems pretty out of the blue to me, too.

Perhaps a better question would be:  Does anyone actually understand what
Frank is _saying_ (not tell), and would they care to enlighten the rest of
us?

Alex

*A lot of Frank's commentary seems to point to (possibly) radical areas for
improvement.  (Case in point - that rant about dithering, which would be
fairly difficult to incorporate [as dither noise shaping would have to be
integrated into the encoder distortion calculations as well as the decoder],
but potentially very useful.  Also, the bit about probabilistically
measuring the 'distortion of the undistorted bands' - I want to check out
that paper.)

But a similar amount seems clueless, useless, or counterproductive.  (For
instance, single-handedly taking on the task of making LAME C++ compatible,
when no one else has expressed any desire for it, and when it means massive
diffs against the original - no coder with any experience working on a major
project _with other people_ would ever do this lightly.)

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RE: [MP3 ENCODER] Frank's coherence [off topic?]

2000-09-05 Thread Robert Hegemann

[EMAIL PROTECTED] schrieb am Die, 05 Sep 2000:
 Howdy Gabriel,
 
  Am I the only one wich doesn't understand what Frank is telling?
  Can anyone explain a little?
 
 You are not alone.  Once I get past his aggressively authoritative posting
 style (which is a major disincentive to understanding for me), I find I
 understand about half of what Frank posts.  (Some of which seems actually
 both relevant _and_ useful.*)  This is not one of those postings.
 
 I assume he has some theoretical basis for wanting to correlate
 differentiated signals (rather than raw signals), and that this is somehow
 relevant to LAME (I haven't grepped through the source, but I remember
 hearing some discussion of a correlator being used for joint switching?),
 but, since he mentions neither before reeling off a long tabulation of
 statistics, it all seems pretty out of the blue to me, too.
 
 Perhaps a better question would be:  Does anyone actually understand what
 Frank is _saying_ (not tell), and would they care to enlighten the rest of
 us?
 
 Alex

Alex, if you remember Frank's post about DC offsets, there he attached
a little C program to calculate AC/DC offsets as well as a correlation
between left and right channels. (was around 00/08/05)

I plugged into LAME that correlation test for a single frame.
If you define RH_VALIDATE_MS at compile time this code gets active.
Actually the decision whether to use L/R or M/S coding is based
on masking relations. But sometimes LAME switches to M/S coding 
where L/R coding would be using fewer bits. The extra code you
can enable with above defined will try to get a rough estimation
on the correlation of left and right channels and will consider
the perceptual entropy to check if it would be better not to switch
to M/S coding.

Ciao Robert


PS: www.kraftwerk.de
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RE: [MP3 ENCODER] lame source C++ compatible?

2000-09-05 Thread Robert Hegemann

[EMAIL PROTECTED] schrieb am Die, 05 Sep 2000:
 Howdy,
 
  How to compile lame on a system where only a C++ compiler is
  available (the
  C compiler costs extra money)? Currently lame generates
  nearly uncountable
  errors with a C++ compiler.
 
 I'm having a similar problem trying to compile LAME for my system, for which
 only a BASIC compiler is available...  It seems like practically every line
 generates an error or warning of some sort?!?  Can someone address this?
 
 Thanks in advance,
 Alex

You have a BASIC compiler compiling C code??

OK, just a joke! I have no problems with GCC. Later on I will check
to see what the Intel Compiler will say. 


Ciao Robert


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Re: [MP3 ENCODER] newbie question

2000-09-05 Thread David


- Original Message -
From: "Gabriel Bouvigne" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 05, 2000 9:52 AM
Subject: Re: [MP3 ENCODER] newbie question


  On www.r3mix.net, the following command line is recommended as the best
  option (size, speed, quality-wise):
 
  -V1 -h -mj -b128 -q1
 
  I was looking over the documentation that came with the zip file of LAME
  3.86b, and I couldn't find any mention of the "- q" command.

 I didn't documented tha -q switch because it's a beta switch, and its
 behaviour can change from a release to another. The final goal of this
 switch will be to create a quality setting, going from -q9 (worst) to -q0
 (best), but it's not finished yet.


  Can someone here explain? Does it apply only for VBR, or can it be used
 with
  CBR as well?

 It can be used with CBR as well

Then i take it's good on all modes .. (CBR,VBR, and even ABR) i won't be
using it until it's final though :)



  Thanks.
 
  One last question, for the best quality at the expense of file size, is
 "-b
  320 -h -k -m s" the best option?

 This is likely to be the best quality setting, also some people (like me)
 could argue that -m j would be able to give better quality.


 Regards,

 --

 Gabriel Bouvigne - France
 [EMAIL PROTECTED]
 icq: 12138873

 MP3' Tech: www.mp3-tech.org


 --
 MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )


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[MP3 ENCODER] mpglib compiles once again...

2000-09-05 Thread Sigbjørn Skjæret

Right, just committed a bunch of (quick and ugly) fixes to mpglib so it should
once again compile without spewing errors .. it's probably still completely
broken though, I hadn't time to test.

Someone should take some time going over mpglib and clean it up abit (Frank,
since you broke it, why don't you fix it?)

Also possibly fixed a mono-decoding bug in layer1.c?!


- CISC

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RE: [MP3 ENCODER] Correlation mid/side

2000-09-05 Thread alex . broadhead

Howdy Robert,

 Alex, if you remember Frank's post about DC offsets, there he attached
 a little C program to calculate AC/DC offsets as well as a correlation
 between left and right channels. (was around 00/08/05)

I'm not sure I read those - DC offsets aren't particularly relevant to my
current efforts (real-time coding).

 I plugged into LAME that correlation test for a single frame.
 If you define RH_VALIDATE_MS at compile time this code gets active.
 Actually the decision whether to use L/R or M/S coding is based
 on masking relations. But sometimes LAME switches to M/S coding
 where L/R coding would be using fewer bits. The extra code you
 can enable with above defined will try to get a rough estimation
 on the correlation of left and right channels and will consider
 the perceptual entropy to check if it would be better not to switch
 to M/S coding.

Ah.  So LAME uses (or can use) multiple criteria for switching.

I'm in the process of adding mid/side to my encoder now, and have just put
in a first pass based on what the ISO spec 'specifies' (OK - suggests?
hints at?) in Appendix G.  Their switching criterion seems particularly
random and strange:  it is based on a comparison of the sum and difference
of the squared energies of the two channels:

sum(rl[i]^2 - rr[i]^2)  0.8 * sum(rl[i]^2 + rr[i]^2)

Summing 0=i512, where rl and rr are supposedly the energies (so that they
are squaring the energy?!?) of the FFT line spectra.  (I suspect that they
meant to have an absolute value around the difference term.)

I don't think this works very well, and I'm not sure why they thought it
would work (i.e. what its theoretical basis was), so I'm thinking of
substituting a correlator.  Do you know why one would correlate on the
differential instead of the signal?

Also, does anyone know the basis for the ISO switching criterion?  Do they
really mean square energy (quadrupled magnitude)?  They give no hints as to
how to reconcile the mid/side samples with the right/left psychoacoustics in
the loop section of the encoder.  Computing psychoacoustics for the sum and
difference signals makes no sense to me, as one is never going to listen to
them and thus the psychoacoustic threshold figures are irrelevant, but the
alternative of trying to simultaneously allocate bit/noise for both channels
seems overly complicated/possibly impossible (oxymoron strikes!).  I'm
compromising right now by just calculating the distortion thresholds using
the L/R SMR and the M/S signal and bandwidths (and the loop distortions with
the M/S signal), but this seems like a pretty silly way to do things, and it
doesn't sound very good.  I'm trying not to plagiarize LAME (I still use the
ISO/ATT psych model, for one), but I gather that LAME does some sort of
mid/side psychoacoustic processing?

Thanks for any feedback/assistance,
Alex

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Re: [MP3 ENCODER] lame source C++ compatible?

2000-09-05 Thread Joshua Bahnsen

Now wouldn't a run at compile time mp3 encoder be awesome?? I actually know
BASIC, maybe I'll try it, I bet it will be just as good as the sample
nibbles code for QBASIC... Yeah, or maybe the NES emulator. ;-) It's REALLY
fast, yeah!

Josh
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 05, 2000 10:57 AM
Subject: RE: [MP3 ENCODER] lame source C++ compatible?


 Howdy,

  How to compile lame on a system where only a C++ compiler is
  available (the
  C compiler costs extra money)? Currently lame generates
  nearly uncountable
  errors with a C++ compiler.

 I'm having a similar problem trying to compile LAME for my system, for
which
 only a BASIC compiler is available...  It seems like practically every
line
 generates an error or warning of some sort?!?  Can someone address this?

 Thanks in advance,
 Alex

 --
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Re: [MP3 ENCODER] ABR 320?

2000-09-05 Thread Frank Klemm

::  would it be possible to encode an mp3 at ABR 320 using frames larger
::  than 320 kbps?
::  e.g. 440kbps , 560kbps
::  
560 kbps result in distortions.

551 kbps is the last without heavy artefacts.

-- 
Mit freundlichen Grüßen
Frank Klemm
 
eMail | [EMAIL PROTECTED]   home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641) 390545
sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany

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Re: [MP3 ENCODER] normalization

2000-09-05 Thread Frank Klemm

On Tue, Sep 05, 2000 at 02:21:45AM +0200, Francois du Toit wrote:
 I want to implement a normalizing routine in one of my programs, can
 anybody recommend one? It would be for 16 bit CD audio.
 
 Would simply multiplying by a constant factor and rounding be good enough
 or would the rounding errors cause some problems
 
Rounding cases problems, especially on high quality/low noise audio.

The simpliest way is to add 0.75 LSB triangle noise before rounding.

-- 
Frank Klemm

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Re: [MP3 ENCODER] normalization

2000-09-05 Thread Steve Lhomme

Even working with floats ???

What is the LSB in this case ?

- Original Message -
From: "Frank Klemm" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 05, 2000 9:23 AM
Subject: Re: [MP3 ENCODER] normalization


| On Tue, Sep 05, 2000 at 02:21:45AM +0200, Francois du Toit wrote:
|  I want to implement a normalizing routine in one of my programs, can
|  anybody recommend one? It would be for 16 bit CD audio.
| 
|  Would simply multiplying by a constant factor and rounding be good
enough
|  or would the rounding errors cause some problems
| 
| Rounding cases problems, especially on high quality/low noise audio.
|
| The simpliest way is to add 0.75 LSB triangle noise before rounding.

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Re: [MP3 ENCODER] lame source C++ compatible? (off topic)

2000-09-05 Thread Steve Lhomme

Nibbles ?

One of my favorite games !!

- Original Message -
From: "Joshua Bahnsen" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 05, 2000 7:31 PM
Subject: Re: [MP3 ENCODER] lame source C++ compatible?


| Now wouldn't a run at compile time mp3 encoder be awesome?? I actually
know
| BASIC, maybe I'll try it, I bet it will be just as good as the sample
| nibbles code for QBASIC... Yeah, or maybe the NES emulator. ;-) It's
REALLY
| fast, yeah!

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Re: [MP3 ENCODER] mpglib compiles once again...

2000-09-05 Thread Frank Klemm

::  Right, just committed a bunch of (quick and ugly) fixes to mpglib so it should
::  once again compile without spewing errors .. it's probably still completely
::  broken though, I hadn't time to test.
::  
::  Someone should take some time going over mpglib and clean it up a bit (Frank,
::  since you broke it, why don't you fix it?)
::
I had to present a lecture on Tuesday from 11:00 to 13:00.
So I had no time in the last 36 hours.

Layer I now seems to work.

Is there a simple Layer I coder out there to test Layer I decoding?
I've never have seen a Layer I file.

::  
::  Also possibly fixed a mono-decoding bug in layer1.c?!
::  
Fixed. First I tried to print out the Layer I/II/III file format (download
of Staroffice was necessary). Guessing is really not my strength.

-- 
Mit freundlichen Grüßen
Frank Klemm
 
eMail | [EMAIL PROTECTED]   home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641) 390545
sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany

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Re: [MP3 ENCODER] VBR, distortion, and CBR

2000-09-05 Thread Frank Klemm

::   ::   This is a little bit problematic, because the number of distorted
::   ::   bands does not tell you the weight of distortions you will get.
::   ::   What do you think sounds uglier:
::   ::20 distorted bands, each 0.1 dB
::   ::   or
::   :: 1 distorted band by 2 dB
::   ::   ???
::   ::
::   ::  I would say that 20 distorted bands at 0.1dB is preferable, and assume
::   ::  that this is the choice of lame's algorithms. Is this assumption wrong ?
::   ::
::   Yes and no.
::   
::   That depends on the distortion of the undistorted bands. Also undistorted
::   bands have an distortion, it is negative.
::  
::  You are saying : "has a distortion" != "is distorted" ?
::   
distorted bands:threshold  noise
undistorted bands:  0 = noise = threshold

So calculate: distortion[dB] = 20 dB * lg (noise/threshold)

distorted bands:distortion  0 dB
undistorted bands:  distortion  0 dB
zero distortion bands:  distortion = -oo dB (very unlikely that this happens)

Noise and threshold are voltages (no power, otherwise multiply by 10 dB).

The distortion = max ( x1, ..., xn) model implies an infinite sharp
transition of the distortion reception. But reality is far beyond from this
model.


::   First you need a table of the probability to hear distortions:
::   
::   distortion  probability error noise   voltage ratio
::  [d=dB]  [p=%]  [i]
::   -1050.0   0.  0.316 : 1
::   -3 50.8   0.0004  0.708 : 1
::   -2.5   52.0   0.0025  0.750 : 1
::   -2 53.2   0.0064  0.794 : 1
::   -1.5   55.6   0.020   0.841 : 1
::   -1 57.9   0.040   0.891 : 1
::   -0.5   63.3   0.116   0.944 : 1
::   -0.25  66.3   0.176   0.972 : 1
::0 70.2   0.281   1.000 : 1
::  
::  isn't 0 dB distortion == no distortion ?
::  How can it be heared ?
::
No: distortion == threshold.

Decibel (dB) is a logarithmic unit. Zero distortion means -oo dB (oo stands
for infinite), tenth voltage -20 dB, half voltage -6 dB, equal voltage +/-0 dB,
double voltage +6 dB.

Do you know Graham Bell?

::   +0.1   72.2   0.336   1.011 : 1
::   +0.2   74.2   0.422   1.023 : 1
::   +0.3   76.1   0.504   1.035 : 1
::   +0.4   78.0   0.593   1.047 : 1
::   +0.5   80.0   0.706   1.059 : 1
::   +1 87.1   1.281.122 : 1
::   +1.5   93.7   2.341.189 : 1
::   +2 97.2   3.651.259 : 1
::   +2.5   99.0   5.431.334 : 1
::   +3 99.8   8.4 1.412 : 1
::   +3.5   99.9   9.5 1.496 : 1
::  
Search also for "masking_lower" in "lame.c".

-- 
Mit freundlichen Grüßen
Frank Klemm
 
eMail | [EMAIL PROTECTED]   home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641) 390545
sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany

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[MP3 ENCODER] Multi Pass MP3 Encoder

2000-09-05 Thread Frank Klemm

What about the idea to allow the user to code the file in two passes
for the very best quality by the extense of doubling the CPU time.

1st pass with "--hintfile":

WAV ---|  |---  MP3 (first pass quality)
| LAME |
|  |---  HINT


2nd pass with "--usehintfile"


WAV |  |---  MP3 (first pass quality)
| LAME |
HINT --|  |---  HINT (unused)


In the hint file for instance one byte per frame is stored:

Bit 7:4 Should MS or LR coding be used?
 0  no info 
 1  very strong buy for LR stereo   br(LR) = 0.545 br(MS)
 ...
 8  no differce br(LR) == br(MS)
 ...
15  very strong buy for MS stereo   br(LR) = 1.834 br(MS)

Bit 3:0 Bitrate demand for the best coding
 0  no info
 1  low bit rate demand  20% 
 ...
 5  normal bit rate demand   100%
 ...
15  very high bit rate demand   300%

MS/LR switching can be optimized in a seconds pass.
Also the bit reservoir can be better balanced.

You can empty the bit reservoir for an attack if there are following some
silent milliseconds of music 

5 5 5 8 10 3 2 4 4 3 4)
 ^
 empty bit pool here for best Q

But may be the attack is followed by a much more worse attack 

5 5 5 8 10 14 3 2 4 4 3 4
 ^  ^
 |  empty the bit pool here
 and not here

The same efect can be achieved by increasing the latency time of lame
so lame can see more of the music' future.


Proposal


Application feed samples via lame_encode_buffer() to lame.
lame collects the samples in an overlapping circular buffer°)
with an size of for instance 8192 samples. Every function
(gpsycho, lame encoder, MS/LR detection, ...) gets the same
big data block, but everyone are interested of another area
of the data:

   |--|
MS/LR   ^^^
gpsycho ^^
coder  ^
coder     (out of bit scenario)

-- 
Mit freundlichen Grüßen
Frank Klemm

°) circular overlapping buffer
   Two successive buffers are filled with the same data.

   | 1  2  3  4  5  6  7  8||1  2  3  4  5  6  7  8|
 || 

   | 9 10  3  4  5  6  7  8||9 10  3  4  5  6  7  8|
   || 
  
   | 9 10 11 12 13  6  7  8||9 10 11 12 13  6  7  8|
|| 

   Advantage: No wrapping
   Disadvantage:  Needs some 10^-2 MByte of RAM


 
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Re: [MP3 ENCODER] the -mx mode - different philosophy

2000-09-05 Thread Frank Klemm

[Charset iso-8859-2 unsupported, filtering to ASCII...]
::  Frank Klemm wrote:
::   
::   ::   -md is not documented.
::   ::
::   ::  What is it ? Does anyone made a dual channel mode?
::   ::
::   ::
::   It's like -ms, but no bitrate balance is possible between the channels
::   (maybe also for the main purpose, dual language, a disadvantage from the
::   quality aspect) and also a hint for the decode to only decode one
::   (selectable) channel.
::   
::   Main advantage is the lower CPU usage.
::  
::  Just to clarify : -md is "dual channel" ?
::  
Yes:

Bit 7,6 of the MPEG frame header:

00 LR Stereo frame(Stereo, Left + Right Channel, bit balancing possible)
01 MS Stereo frame(Stereo, Mid- and Side Signal)
10 Dual Channel frame (Mono, for instance dual language english/german)
11 Single Channel frame   (Mono)

Ist it right that Layer II MS-Frames are unable to NOT use intensity stereo
for f  fs/4 (for CD 11 kHz)?

-- 
Mit freundlichen Grüßen
Frank Klemm
 
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Re: [MP3 ENCODER] Correlation mid/side

2000-09-05 Thread Frank Klemm

::  
::  Also, does anyone know the basis for the ISO switching criterion?  Do they
::  really mean square energy (quadrupled magnitude)?  They give no hints as to
::  how to reconcile the mid/side samples with the right/left psychoacoustics in
::  the loop section of the encoder.  Computing psychoacoustics for the sum and
::  difference signals makes no sense to me, as one is never going to listen to
::  them and thus the psychoacoustic threshold figures are irrelevant, but the
::  alternative of trying to simultaneously allocate bit/noise for both channels
::  seems overly complicated/possibly impossible (oxymoron strikes!).  I'm
::  compromising right now by just calculating the distortion thresholds using
::  the L/R SMR and the M/S signal and bandwidths (and the loop distortions with
::  the M/S signal), but this seems like a pretty silly way to do things, and it
::  doesn't sound very good.  I'm trying not to plagiarize LAME (I still use the
::  ISO/ATT psych model, for one), but I gather that LAME does some sort of
::  mid/side psychoacoustic processing?
::  
1st step:

  - Calculate diffuse field corrected ear-drum SPL°):
L' = a(w) L + b(w) R,  R' = a(w) R + b(w) L w is a small omega, a(w),b(w) 
complex
Note: different for loudspeakers and head phones 
  (where a \approx 1 and b \approx 0)
Note: search for HRTF to get usable approaches for a and b.
Note: for very good results you must take into consideration that a and
  b changes depending on the temporal direct/indirect sound ratio
  - ignore in-brain talk over (ca. -60 dB @1 kHz, much lower than accoustic talk over) 
  - calculate threshold for the left and the right ear
  
2nd step:
  if frame can be coded in LR mode
- code L using L and threshold(L)
- code R using R and threshold(R)
  if frame can be coded in MS mode
- code M using M and 0.6...0.8 * min(threshold(L),threshold(R))
- code S using S, decoded coded M, threshold(L) and threshold(R)

3rd step:
  select MS/LR depending on:
- br(MS)/br(LR) ratio
- the past of the audio file
- bit pool fuel
- (the future of the audio file using one mechanism I mailed before)
- user options?

-- 
Mit freundlichen Grüßen
Frank Klemm

°) SPL = Sound preasure level
   
 
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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Robert Hegemann @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] presets
Dated: 01 Sep 00  09:48:58
[processed here:  05 Sep 00 14:12:44]

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From: Robert Hegemann [EMAIL PROTECTED]
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Frank Klemm schrieb am Don, 31 Aug 2000:
 ::  b)  LAME's filter is not much useful for highpass filtering
 ::
 External filters are available. For integration the f*cking "signed short
 int" format must be replaced by "float" or "double".
 
 I've heared (by listening) that the lame FIR high pass filter is not very
 well. Extern 4th order butterworth filter are working much better. They also
 can be used for Hifi highpass filtering (fu=30 Hz for records).


LAME has FIR filters? I thought it works on the polyphase filter bank 
where you have 32 equidistant controlers.

Good filters would be nice to have, go for it...


 ::  c)  when you resample and want to lowpass, try to get rid off sfb21!
 ::  
 Can you explain this?

long blocks have 22 scalefactor bands 0...21. But for sfb21 there are
no masking ratios and it will not be transmitted. So you cant color
that band, means you cannot adjust the quantizer stepsize for it.
(short blocks: 13 bands sfb12 will not be transmitted). 

I meant, if you want a resulting bandwidth of x kHz, try to select
a outsamplerate where this does not interfere with sfb21 (short sfb12). 


Ciao Robert


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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Holger Dors @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re[2]: [MP3 ENCODER] RazorLame 1.1.0 released
Dated: 01 Sep 00  10:24:45
[processed here:  05 Sep 00 14:12:45]

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From: Holger Dors [EMAIL PROTECTED]
To: Steve Lhomme [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

Hello Steve,

 Yep, and I personnaly use a lame EXE with libnsdfile support, so not
 only wav files are supported !
if you feel a bit adventurous, take a look at RazorLame.dat and look
for the section "InputFileTypes". You can add more file types there,
so if you know your LAME supports more, add them there! (I'd suggest
making a copy of the file though; if you ruin this file RazorLame
won't work properly any more)

Regards,
  Holger Dors   mailto:[EMAIL PROTECTED]


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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Robert Hegemann @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] Encoding 44.1 kHz as 48 kHz sounds not so very
well
Dated: 01 Sep 00  11:21:26
[processed here:  05 Sep 00 14:12:43]

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From: Robert Hegemann [EMAIL PROTECTED]
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Frank Klemm schrieb am Fre, 01 Sep 2000:
 Why a quadratic interpolation is used for upsampling instead of a sinc
 interpolator?
 
 -- 
 Frank Klemm


I can't answer your question, but where is upsampling from 44.1 to 48 kHz 
good for?


Ciao Robert


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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Roel VdB @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re[3]: [MP3 ENCODER] RazorLame 1.1.0 released
Dated: 01 Sep 00  03:18:24
[processed here:  05 Sep 00 14:12:47]

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From: Roel VdB [EMAIL PROTECTED]
To: Holger Dors [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

Hello Holger,

Thursday, August 31, 2000, 2:31:48 PM, you wrote:

HD It's hard to come up with a good option dialog for RL, given the vast
HD amount of options LAME has to offer. I think it's a good idea to put
HD those settings in categories, however, I agree, it can get a bit
HD unclearly...

I think this is not good way of handling things.  If there are too
much options, there are too much options.  If you decide to catalog
some of them as "advanced", some as "expert" and other with another
subjective label, it only complicates things even further.

30 options in 1 screen are less complicated than 30 options scattered
over 5 screens.

What I suggest, since lame will continuously develop and new options
will be added, is to simply handle things like this:

- 1 screen (oversight  clarity)
- in top, a (greyed out or so) continuous display of current command
line.  I never understood why enabling this is an option.
- only a limited set of "core" options. stereo mode, bitrate, encoder
mode, delete after, and the most common flags.  Just so you have a nice
oversight of
options.
- a box to add non-common command line options.
- a flag to make lame only use that box, instead of adding the content
to current selected options.

This is, basically, the way lamebatch did things.  In the long run
this seems to be the only managable way of constructing a frontend
imho: You simply see what's set, and you don't need to look 3 windows
back for some setting you also might need.

then, why no multithread? it'd be nice to update the to-be-encoded
list in one window while the encoding-box is somewhere else...

I had to press abort too often :(

just some suggestions ...

-- 
Best regards,
 Roelmailto:[EMAIL PROTECTED]


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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Mathew Hendry @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] lame source C++ compatible?
Dated: 02 Sep 00  02:48:11
[processed here:  05 Sep 00 14:13:39]

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From: "Mathew Hendry" [EMAIL PROTECTED]
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 From: "Segher Boessenkool" [EMAIL PROTECTED]

  How to compile lame on a system where only a C++ compiler is available
(the
  C compiler costs extra money)? Currently lame generates nearly
uncountable
  errors with a C++ compiler.

 Put

 extern "C" {
 }

 around everything? (Around a whole file is ok, I believe).

That only affects the linkage of functions (e.g. disables name mangling).

There's no easy way to do it in general. I've done it a few times for some
fairly large pieces of code, and it's no fun at all. Then there's the
maintenance...

Is the C compiler expensive? :)

-- Mat.


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[MP3 ENCODER] Message *bounced*

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From:  David Balazic @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] RazorLame 1.1.0 released
Dated: 01 Sep 00  09:54:38
[processed here:  05 Sep 00 14:12:45]

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From: David Balazic [EMAIL PROTECTED]
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Holger Dors wrote:
 
 Hello David,
 
  For VBR mode the max bitrate is set on one screen, while min bitrate
  on the other. Confusing.
 Yes, I know, and I'm open to any suggestions. But I'm not even
 supporting all switches now, e.g. free bitrates are still missing, as
 well as the presets aren't available. If someone does come up with a
 layout of a new option screen, I'd be willing to implement that in an
 upcoming version. (If I believe that it's better, of course, and it's
 not too hard to implement.)
 
  Also when ABR is selected, the max bitrate slider should be turned off.
 I disagree. According to the help, it's "NOT RECOMMENDED", but after
 all it is possible.
Oh, didn't know that.

But I noticed that the -B option is not present in the command line !
( according to the "Show LAME commands:" button ,
shouldn't that be "show used LAME options:" ? )

An idea for the layout : put the min and max VBR bitrate
slider on the same screen !
Why is the ABR bitrate not a slider ?
With the option of manual number entry maybe.

 Who am I to say to the user that he cannot do
 this when he knows that it's possible? ;-)
 
 Regards,
   Holger Dors   mailto:[EMAIL PROTECTED]
 
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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  [EMAIL PROTECTED] @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: [MP3 ENCODER] MLame Help Requested
Dated: 02 Sep 00  09:40:23
[processed here:  05 Sep 00 14:14:53]

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I'm encoding MP3 - MP3.  How do you keep MLAME from adding and extra 
".mp3" to the file name?

Brent
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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Steve Lhomme @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] Please stop breaking LAME... ;)
Dated: 02 Sep 00  09:04:54
[processed here:  05 Sep 00 14:14:54]

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From: "Steve Lhomme" [EMAIL PROTECTED]
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 Also it would be nice if you cast types that don't match the functions types
 into the correct one (or use the correct one to begin with) .. it's both the
 correct and nice way to do it...

I think that casting can be dangerous. It removes some compile errors than can
sometimes be an error ! So I think you'd better cast when you know what you're
doing, otherwise let the compile error...

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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Sigbj°rn Skjµret @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: [MP3 ENCODER] Please stop breaking LAME... ;)
Dated: 02 Sep 00  08:48:08
[processed here:  05 Sep 00 14:14:55]

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Please check that your code compiles and works ok before checking it in to
the CVS .. I've repeatedly had to go fix compile errors in the CVS now... :P

Also it would be nice if you cast types that don't match the functions types
into the correct one (or use the correct one to begin with) .. it's both the
correct and nice way to do it...

Also don't set ambigous types like "unsigned", use the proper full name,
"unsigned int".

And a problem I can't figure out right now is why LAME won't encode at all
anymore, it just creates an empty mp3 (that means it atleast opens it), and
then quits saying "Error writing mp3, disk full?" .. what have you done? ;)


- CISC

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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Josh Coalson @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] linux lossless encoder
Dated: 02 Sep 00  22:22:03
[processed here:  05 Sep 00 14:15:21]

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From: "Josh Coalson" [EMAIL PROTECTED]
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hello , considering that there isn't a monkeys audio for linux, I don't
supose there are any developers out there would like to take a look
a shorten to see if it could be improved to match monkeys audio
performance
, or perhaps even better? :)

well, once upon a time there was ogg squish, which was open
source, but the source has disappeared (at least I haven't been
able to find it).  I heard Monty is currently too busy with
vorbis to get to the lossless stuff right now, so in the meantime
I'm doing just what you mentioned.  time is short but when I get
something useful I will put up the code.

at least with lossless encoding it's not so bad to have an
interim solution... when ogg squish comes back at least there
will be no generation loss converting.

Josh

p.s. don't expect radically greater compression ratios though...
there is a limit with what you can do with lossless compression.
but the format I'm working on will support streaming and sample-
accurate seeking, which shorten doesn't.

_
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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Scott manley @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] linux lossless encoder
Dated: 02 Sep 00  21:36:17
[processed here:  05 Sep 00 14:15:30]

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 well, once upon a time there was ogg squish, which was open
 source, but the source has disappeared (at least I haven't been
 able to find it).  I heard Monty is currently too busy with
 vorbis to get to the lossless stuff right now, so in the meantime
 I'm doing just what you mentioned.  time is short but when I get
 something useful I will put up the code.
 
 at least with lossless encoding it's not so bad to have an
 interim solution... when ogg squish comes back at least there
 will be no generation loss converting.


You should search for a prgaram called 'Shorten' this is also a lossless
compressor for Unix (free for unix - but you pay for the windows
version)
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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Ingo Saitz @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] linux lossless encoder
Dated: 02 Sep 00  23:51:59
[processed here:  05 Sep 00 14:15:31]

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MoiN

On Sat, Sep 02, 2000 at 08:22:03PM +, Josh Coalson wrote:
 well, once upon a time there was ogg squish, which was open
 source, but the source has disappeared (at least I haven't been
 able to find it).

Well, it still seems to be there:

http://web.mit.edu/afs/sipb/user/xiphmont/ogg-98.10.tgz 

Be sure to enable lossless compression with -Q10 or -L
No, I didn't test the lossless compression it offers, yet. So
have fun playing :)

Ingo
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[MP3 ENCODER] Message *bounced*

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From:  Sigbj°rn Skjµret @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] Please stop breaking LAME... ;)
Dated: 03 Sep 00  07:51:43
[processed here:  05 Sep 00 14:15:57]

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::  Please check that your code compiles and works ok before checking it in
::  to the CVS ...
::
That doesn't help fully to solve the problem. Compiler and runtime libs are
different. So the check can only be done for one (or two) compilers.

Still, it's not fun when certain files fail to compile because of silly
errors (like conflicting prototypes). ;)

Currently my aim is, that the program is compilable with:
   gcc 2.95.2
   g++ 2.95.2

I run gcc 2.95.(3) and SAS/C (the latter is usually much more helpful on
warnings, and still much more forgiving on "errors")...

::  Also it would be nice if you cast types that don't match the functions
::  types into the correct one 
Casting is a bad thing. Correct the reason for the casting, not the warning
(or the error in C++) itself. See Ada Rationale. It is Ada related, but the
reasons are the same for every procedural language like C, Pascal, Ada, ...

Ofcos, as I said, the best thing is to use the correct one to begin with, but
if that isn't possible, atleast cast the value into the correct one.


- CISC

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[MP3 ENCODER] Message *bounced*

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From:  Eric Howgate @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: Re[4]: [MP3 ENCODER] RazorLame 1.1.0 released
Dated: 03 Sep 00  18:50:45
[processed here:  05 Sep 00 14:16:38]

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- Original Message -
From: Holger Dors [EMAIL PROTECTED]
To: Roel VdB [EMAIL PROTECTED]
Sent: Friday, September 01, 2000 12:01 PM
Subject: Re[4]: [MP3 ENCODER] RazorLame 1.1.0 released


 Hello Roel,

  30 options in 1 screen are less complicated than 30 options scattered
  over 5 screens.
 This can probably be disputed for very long!


  - 1 screen (oversight  clarity)
  - in top, a (greyed out or so) continuous display of current command
  line.  I never understood why enabling this is an option.
  - only a limited set of "core" options. stereo mode, bitrate, encoder
  mode, delete after, and the most common flags.  Just so you have a nice
oversight of
  options.
  - a box to add non-common command line options.
  - a flag to make lame only use that box, instead of adding the content
  to current selected options.
 The problem starts with the "most common flags". Who will define
 those? You've left out VBR, for a start. Similar, I know at least one
 person who was very happy about that the "Audio Processing" tab where
 you can set sampling frequency and filter options.

 Personally, I started RL because of the many options Lame has to
 offer, it's fun to play with them, and I wanted an easy interface. I
 admit that the options dialog is far from being perfect, but I think
 it's a good start.


  then, why no multithread? it'd be nice to update the to-be-encoded
  list in one window while the encoding-box is somewhere else...
 Actually, this is on my wish list. However, unlike LAME, I'm currently
 the only programmer in RazorLame (I hope this changes soon!), and the
 wishlist grows and grows. Additional, I do this in my spare time, so
 please be patient.



 Regards,
   Holger Dors   mailto:[EMAIL PROTECTED]


Perhaps the 'disputants' might care to take a look at a new front end for
Lame, available from
http://www.geocities.com/enlamer/

This has many of Lame's options in the one large user dialog, but I suspect
it will confuse the novice user (and it seems to be able to handle only
single files at present);  perhaps it is time for someone who fully
understands all the options/switches to settle down and write a
comprehensive  'Help' file for Lame, - I doubt many who use Lame with a
front-end or ripping utility bother to read the 'Usage' file that
accompanies the exe file and I don't think users of the dll file even see
it.

Roel's site is certainly a good guide for achieving optimal performance with
music - but there are other uses, eg voice recording, not touched on there,
and only barely mentioned in the 'Usage'  file.

Eric







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[MP3 ENCODER] Message *bounced*

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From:  Joshua Bahnsen @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: [MP3 ENCODER] problems decoding with latest CVS version
Dated: 01 Sep 00  20:13:41
[processed here:  05 Sep 00 14:12:40]

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From: Joshua Bahnsen [EMAIL PROTECTED]
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When using the latest CVS code, I cannot decode mp3s any longer it gives
me this error

Bitrate 9 kbps not legal for 44100 Hz output sampling

This is a 192 kBits 44.1 Stereo mp3. I've tried many different files,
all with the same result.

Josh
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[MP3 ENCODER] Message *bounced*

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From:  [EMAIL PROTECTED] @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: [MP3 ENCODER] mlame usage without naming .mp3.mp3
Dated: 01 Sep 00  23:46:04
[processed here:  05 Sep 00 14:12:32]

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How do you get mlame to batch encode MP3-MP3 without adding a ".mp3" to 
every file; resulting in a file with ".mp3.mp3" as extentions?

Brent
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Re: [MP3 ENCODER] normalization

2000-09-05 Thread Frank Klemm

::  Even working with floats ???
::  
::  What is the LSB in this case ?
::  
The LSB is 1 for int's.

See "MPEG-2 AAC Stereo Verification Tests" and the item "Glockenspiel was
too high". May be this is more comprehensible.

::  | On Tue, Sep 05, 2000 at 02:21:45AM +0200, Francois du Toit wrote:
::  |  I want to implement a normalizing routine in one of my programs, can
::  |  anybody recommend one? It would be for 16 bit CD audio.
::  | 
::  |  Would simply multiplying by a constant factor and rounding be good enough
::  |  or would the rounding errors cause some problems
::  | 
::  | Rounding cases problems, especially on high quality/low noise audio.
::  |
::  | The simpliest way is to add 0.75 LSB triangle noise before rounding.

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[MP3 ENCODER] Coding history report for each (non trivial) function

2000-09-05 Thread Frank Klemm

May be evry (non trival function should get a history header?

/*
 *  Function:   double sinc (IN double x);
 *
 *  Purpose:calculates sin (pi x) / (pi x)
 *
 *  Input:  
 *
 *  History:
 *Person_1 2000-06-02: first version
 *Person_2 2000-06-12: zero cross error bug removed
 */

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[MP3 ENCODER] MGDIFF

2000-09-05 Thread Frank Klemm

May be the following file can be added. I use it do quick show differences
of two brances. Quick and dirty, can still be optimized. I named it MGDIFF
and it needs mgdiff.

#! /bin/bash

for i in {*,*/*,*/*/*,*/*/*/*}.{c,h}; do 
  file1=`pwd`/$i
  file2=`echo $file1 | sed s/lame/lame_old/`
  if diff -bwB --brief $file1 $file2; then
echo equal $i
  else
echo -e "\033[7mDifferent: $i\033[0m"
mgdiff -title "mgdiff $i" -geometry 1536x800 -args "-bw" $file1 $file2 
sleep 2
  fi
done


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Re: [MP3 ENCODER] Frank's coherence [off topic?]

2000-09-05 Thread Frank Klemm

::  
::  I plugged into LAME that correlation test for a single frame.
::  If you define RH_VALIDATE_MS at compile time this code gets active.
::  Actually the decision whether to use L/R or M/S coding is based
::  on masking relations. But sometimes LAME switches to M/S coding 
::  where L/R coding would be using fewer bits. The extra code you
::  can enable with above defined will try to get a rough estimation
::  on the correlation of left and right channels and will consider
::  the perceptual entropy to check if it would be better not to switch
::  to M/S coding.
::  
I've added a remark on this piece of code.

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[MP3 ENCODER] Layer III decoding has been broken between Mo and now

2000-09-05 Thread Frank Klemm

Decoding of Layer 3 doesn't work anymore.

Last checkout/commit/checkout in the night So/Mo worked, now it is broken.

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[MP3 ENCODER] Neues File

2000-09-05 Thread Frank Klemm

Ich benötige ein neues File für mlame/mlame.bat

mlame_corr.c

oder so. 

mlame soll (auf Wunsch) automatisch zwischen -ms, -mj und -mm umschalten.

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Re: [MP3 ENCODER] MGDIFF

2000-09-05 Thread Robert Hegemann

Frank Klemm schrieb am Mit, 06 Sep 2000:
 May be the following file can be added. I use it do quick show differences
 of two brances. Quick and dirty, can still be optimized. I named it MGDIFF
 and it needs mgdiff.
 
 #! /bin/bash
 
 for i in {*,*/*,*/*/*,*/*/*/*}.{c,h}; do 
   file1=`pwd`/$i
   file2=`echo $file1 | sed s/lame/lame_old/`
   if diff -bwB --brief $file1 $file2; then
 echo equal $i
   else
 echo -e "\033[7mDifferent: $i\033[0m"
 mgdiff -title "mgdiff $i" -geometry 1536x800 -args "-bw" $file1 $file2 
 sleep 2
   fi
 done

I always do 

diff -q lame-old/ lame-new/ | grep differ

this shows me all files that differ in lame-old and lame-new folders.

I prefer xxdiff over mgdiff to visualize differences :-)


Ciao Robert


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Re: [MP3 ENCODER] mpglib compiles once again...

2000-09-05 Thread Robert Hegemann

Frank Klemm schrieb am Die, 05 Sep 2000:
 Layer I now seems to work.
 
 Is there a simple Layer I coder out there to test Layer I decoding?
 I've never have seen a Layer I file.

Yes, get the ISO reference software dist10.tgz (ie at www.mp3-tech.org)


Ciao Robert

 
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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Christopher Wise @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] RazorLame 1.1.0 released
Dated: 01 Sep 00  01:12:21
[processed here:  05 Sep 00 14:12:48]

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From: Christopher Wise [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

On Thu, 31 Aug 2000, Roel VdB wrote: 
 btw: why are mp3's shown anyway when I choose "encode"?  I think only
 showing .wav by default would be better?

No, please don't change this. RazorLame can can be used to set up
re-encoding of mp3 files. This is very useful if you want to re-encode at
a lower bitrate for a Rio.

Thanks Holger for a great program. 

Now all I need is for lame to keep the ID3 tags when re-encoding.
Has anyone implemented this yet?

Chris Wise

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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Estrunfus Maleficus @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: [MP3 ENCODER] linux lossless encoder
Dated: 02 Sep 00  01:42:57
[processed here:  05 Sep 00 14:12:13]

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From: Estrunfus Maleficus [EMAIL PROTECTED]
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hello , considering that there isn't a monkeys audio for linux, I don't
supose there are any developers out there would like to take a look
a shorten to see if it could be improved to match monkeys audio
performance
, or perhaps even better? :)

well thats part of my wishlist any volunteers up to the chalenge
P.S.: am I the only one that feels that development on grip as stagnated
wa before time?, I never saw the promissed rip first, encode
later function, any volunteers for that too?

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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Steve Lhomme @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] RazorLame 1.1.0 released
Dated: 01 Sep 00  08:23:11
[processed here:  05 Sep 00 14:12:46]

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From: "Steve Lhomme" [EMAIL PROTECTED]
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Yep, and I personnaly use a lame EXE with libnsdfile support, so not only wav
files are supported !

- Original Message - 
From: "Christopher Wise" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 01, 2000 1:12 AM
Subject: Re: [MP3 ENCODER] RazorLame 1.1.0 released


 On Thu, 31 Aug 2000, Roel VdB wrote: 
  btw: why are mp3's shown anyway when I choose "encode"?  I think only
  showing .wav by default would be better?
 
 No, please don't change this. RazorLame can can be used to set up
 re-encoding of mp3 files. This is very useful if you want to re-encode at
 a lower bitrate for a Rio.
 
 Thanks Holger for a great program. 
 
 Now all I need is for lame to keep the ID3 tags when re-encoding.
 Has anyone implemented this yet?

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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Holger Dors @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re[2]: [MP3 ENCODER] RazorLame 1.1.0 released
Dated: 01 Sep 00  12:49:32
[processed here:  05 Sep 00 14:12:41]

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From: Holger Dors [EMAIL PROTECTED]
To: David Balazic [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

Hello David,

 But I noticed that the -B option is not present in the command line !
congratulations, you found a bug! ;-) I'll fix this in the next
release.

 ( according to the "Show LAME commands:" button , shouldn't that be
 "show used LAME options:" ? )
Indeed, I'll add this to my to do list.

 An idea for the layout : put the min and max VBR bitrate slider on
 the same screen !
Yes, I thought about that myself recently.

 Why is the ABR bitrate not a slider ? With the option of manual
 number entry maybe.
I'm not sure if a slider is a good control for this; I use sliders for
the bitrates because normally only a discreet set of values are
available.


Regards,
  Holger Dors   mailto:[EMAIL PROTECTED]


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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Robert Hegemann @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re: [MP3 ENCODER] lame/CODING_STYLE
Dated: 01 Sep 00  11:24:59
[processed here:  05 Sep 00 14:12:42]

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From: Robert Hegemann [EMAIL PROTECTED]
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Frank Klemm schrieb am Fre, 01 Sep 2000:
 lame/CODING_STYLE, version 0.001 ;-)
 -
 This is the first try of a Coding Style:

notes on some points

 * Don't use tabulators (the character with the value '\t') in source code,
   especially these with a width of unequal 8. Lame sources are using
   different sizes for tabulators.

I don't like tabulators too

 * Functions should be not longer than 50 lines of code.

this is debateable

   Every function should only do ONE thing, and this should
   be done well.

every proggi thinks his functions are doing well, until he finds
out the flaws he introduced.

 * Document functions.

programmers don't document, you know ;-)
OK, your point is clear and I'll try to add more comments
(hopefully useful remarks..)

 * Don't use single 'short' variables to save storage.
   Short variables are especially on Pentium Class Computer much slower than
   int's. DEC alpha also hates short variables.
 
   Example:   float bla [1024];
short i;
  for ( i = 0; i  1024; i++ )
 bla [i] = i;  

I'm not so sure about shorts. Documents on the Intel compiler suggest
for example to put the index variables of nested loops in a struct
to improve cache performance, this way they would be in the same cache line.


Ciao Robert


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[MP3 ENCODER] Message *bounced*

2000-09-05 Thread O/T-Track+ 2.66b


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From:  Holger Dors @2:292/862
To:Sergey Sapelin @2:5020/1844.33
Subj.: Re[4]: [MP3 ENCODER] RazorLame 1.1.0 released
Dated: 01 Sep 00  13:01:09
[processed here:  05 Sep 00 14:12:42]

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From: Holger Dors [EMAIL PROTECTED]
To: Roel VdB [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

Hello Roel,

 30 options in 1 screen are less complicated than 30 options scattered
 over 5 screens.
This can probably be disputed for very long!


 - 1 screen (oversight  clarity)
 - in top, a (greyed out or so) continuous display of current command
 line.  I never understood why enabling this is an option.
 - only a limited set of "core" options. stereo mode, bitrate, encoder
 mode, delete after, and the most common flags.  Just so you have a nice
oversight of
 options.
 - a box to add non-common command line options.
 - a flag to make lame only use that box, instead of adding the content
 to current selected options.
The problem starts with the "most common flags". Who will define
those? You've left out VBR, for a start. Similar, I know at least one
person who was very happy about that the "Audio Processing" tab where
you can set sampling frequency and filter options.

Personally, I started RL because of the many options Lame has to
offer, it's fun to play with them, and I wanted an easy interface. I
admit that the options dialog is far from being perfect, but I think
it's a good start.


 then, why no multithread? it'd be nice to update the to-be-encoded
 list in one window while the encoding-box is somewhere else...
Actually, this is on my wish list. However, unlike LAME, I'm currently
the only programmer in RazorLame (I hope this changes soon!), and the
wishlist grows and grows. Additional, I do this in my spare time, so
please be patient.



Regards,
  Holger Dors   mailto:[EMAIL PROTECTED]


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Re: [MP3 ENCODER] Multi Pass MP3 Encoder

2000-09-05 Thread Greg Wooledge

Frank Klemm ([EMAIL PROTECTED]) wrote:

 What about the idea to allow the user to code the file in two passes
 for the very best quality by the extense of doubling the CPU time.

This has the obvious disadvantage of not being possible when reading
from a pipeline.  But as a user I would definitely use this option if
it were offered.

-- 
Greg Wooledge| "Truth belongs to everybody."
[EMAIL PROTECTED] |   Red Hot Chili Peppers
http://www.kellnet.com/wooledge/ |

 PGP signature


[MP3 ENCODER] New Decoder For Winamp

2000-09-05 Thread E. Zann

Hi folks,

Have you already seen this from Justin of Nullsoft ???

http://landoleet.org/~deadbeef/in_mp3_260_alpha7.zip

It is linked from http://www.fileclicks.com front
page, just scroll down until Today's Top News
Clicks section. Here what is says:

"New Decoder Plug-in For Winamp:

The new decoder plug-in for Winamp is coming along
good, download the alpha version 7 above. It's not
quite as fast as Nitrane, but is completely ISO
compliant. Justin says "there's plenty of room for
improvement". A few changes : bugfix: vbr headers
read when id3v2 tag is present now, downsampling
modes have better vis support, id3v2 writing
support, stream info box and more."

Greetings,
-E.
[EMAIL PROTECTED]


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