[MP3 ENCODER] Moving on from MP3

[engdev]

MP3 - has some (I believe) serious design flaws which the LAME
 developers and others have been trying to eliminate

MP3Pro - Only really aimed at low bit rates

MP3plus - Looks like a good format... the developer *seems* to have
  taken the good stuff out of MP3 and added enhancements
  of his own, but it is not well known (or accepted)

WMA - Microsoft proprietry (a bad thing to start with) also aimed
  at low bit rates, but just happens to allow the use of higher
  rates. Can't convert back to WAV (*REALLY* bad limitation)

OGG - Seems like a good format, eats MP3 at low bit rates, but then
 falls badly as the bit rate is decreased further. There was some
 talk of OGG being in patent infringement (although that is *very*
 dubious, and is probably a rumor started by Thompson or FIIS
 to scare people away - and then they can get their hooks
 into you for MP3...)

AAC - I have not been able to find any encoders that can do anything
 like real-time. Decoders are also rare and usually linked to a
 particular encoder. (ie: the AAC streams produced by any given
 encoder are not necessarily compatible)

I think the LAME developers have done an excellent job of minimising
the flaws inherent in the MP3 specification, and further, I don't believe
they are developing MP3 to write a better encoder for *you* - after all,
what are they getting out of it?

Only thing I see is personal understanding and intellectual enhancement.

Owen.

BTW: This is not aimed at CG, but more to address the issue of
why we do not move on...

Christopher G wrote:

> MP3 Pro is only good for getting mediocre quality mp3s at low bitrates, and
> WMA files often have a time limit attached & cannot be transferred.
>
> How about we NOT move on?
>
> == Originally sent by [EMAIL PROTECTED] ==
> >Hi just some comments. I think mp3 is going obsolete pretty soon.why dno't we
> move on to some topics like mp3 pro  or wma?
>
> ___
> "May your spirit find the times & places to soar."
> - Chris
>
> Support "Project Majestic Mix: A Tribute to Nobuo Uematsu", the first fan-produced 
>VGM album to be released in America!
> http://www.majesticmix.com
>
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Re: [MP3 ENCODER] Re: MS Stereo

[Ivan Dimkovic]

> Good idea, but I'm afraid that doing only this it would still miss:
>
> *very short tones (less than 3 granules long)
> *tones rapidly changing of freqs (sweeps)
>
> But yes, doing forward and backward prediction is a good idea.
>

Well.. there are several alternative methods, check out the method
implemented in PEAQ (ITU-R 1387) and Frank Baumgarte's 'non linear' model.
Instead of computing tonality, these models perform exponential additions of
individual maskers, so the final effect is very similar to tonality
estimation (why? - because tones are built on individual "peaks" and noise
contains many similar spectral lines)

It is very important that one that implements this approach tunes-up the
"alpha" factor of smearing, so the pure noise and pure tone gives masking
powers according to Zwicker's data. I figured out that "alpha" factor
depends on window size and partition band median bark value.

I have tried this approach in the AAC encoder, but the problem of this model
is its speed - it requires lots of 'pow()' calculations in
spreading-function convolution process, and therefore it is not really
useful in real-time conditions. However, according to Baumgarte - it gives
much better masking estimation. However, FhG encoders do not use this.

Also, check out Anibal Ferreira's PhD thesis, where he described intra-frame
tonality estimation based on MDCT spectrum. I haven't tried this because it
was full of hard-core math, but one of these days I might try it :)

Best Regards,
Ivan Dimkovic


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[MP3 ENCODER] GPsycho Information

[John Matthews]

Hi,
 
I'm a Phd student looking at audio compression, 
though not particularly MP3.  
 
I've been looking through the GPSYCHO code - the 
following questions arose.
 
1) What is the difference between the two 
different psychoacoustic model functions?
 
2) Does anyone know exactly how the 
unpredictability measure is being calculated.
 
ie. numre=a1*b2;
    numim=(a1*a1-b1*b1) * 0.5 
etc.
 
Any assistance would be greatly 
appreciated.
 
John. 
 
**John 
Matthews,Music Technology Group,Dept. of Electronics,University of 
York.**

RE: [MP3 ENCODER] Re: mp3encoder digest, Vol 1 #103 - 2 msgs

[Christopher G]

MP3 Pro is only good for getting mediocre quality mp3s at low bitrates, and 
WMA files often have a time limit attached & cannot be transferred.

How about we NOT move on?

== Originally sent by [EMAIL PROTECTED] ==
>Hi just some comments. I think mp3 is going obsolete pretty soon.why dno't we 
move on to some topics like mp3 pro  or wma?

___
"May your spirit find the times & places to soar."
- Chris

Support "Project Majestic Mix: A Tribute to Nobuo Uematsu", the first fan-produced VGM 
album to be released in America!
http://www.majesticmix.com

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[MP3 ENCODER] Re: BCP: CD ripping

[Alexander Leidinger]

On 28 Jan, Martin Faxér wrote:

> I personally don't think that you will have a problem with the output
> quality of GOGO, but if you do, I guess you can check out LAME (although
> GOGO is supposed to be based on it).

GOGO is based upon an older version of LAME, I don't know if they ported
recent quality improvements of LAME (maybe someone on the mp3encode
mailinglist is able to tell more, please don't forget to readd
[EMAIL PROTECTED] to the CC).
You may perhaps want to look at the history/ChangeLog of both projects
and decide yourself (the newest release (3.91, available in
${PORTSDIR}/audio/lame) of LAME has a lot of quality improvements
compared to the previous stable release (3.7x), it's easy to hear them
even for untrained ears).

Bye,
Alexander.

-- 
  ...and that is how we know the Earth to be banana-shaped.

http://www.Leidinger.net   Alexander @ Leidinger.net
  GPG fingerprint = C518 BC70 E67F 143F BE91  3365 79E2 9C60 B006 3FE7

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[MP3 ENCODER] 狂赚美元,举手之劳!!!

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[MP3 ENCODER] CD Ripper Beta Software: RipTrax

[John Lundy]

I am new to this list and hope that I am not breaking with list etiquette by
this post. Anyway, here goes...

I have been working on a CD Ripper for Windows for the past several weeks. I
have reached the point where it is at or very close to a beta code base. I
have set up a home page for the ripper which is named RipTrax. This is a
temporary name and may change in the future.

What I am looking for is some help checking out the program. Users, really,
that can try the program and provide feedback as to bugs, problems,
features, usability issues and general comments. Here is a brief description
of the program.

"RipTrax by John C. Lundy is a free Windows CD Ripper packed with features
and an easy to use interface. Options are provided to rip selected tracks
from an Audio CD to Raw or Wave formats as well as encoded using BladEnc or
Lame_Enc. RipTrax is the first known CD Ripper to provide access to the C2
Error codes which are output by most CD drives to display ripped image
quality. This provides the user with a clear visual indication of audio
distortion; or lack of it. RipTrax also supports CDDB queries and ID3 tags."

The url for RipTrax's home page is:

http://web2.airmail.net/jlundy/riptrax.htm

There is no documentation available yet but I hope that most standard
options are easy to configure without much help. You will need to have ASPI
installed on your machine. This is not a big constraint since Adaptec has
made this a publicly available tool. Just do a search for ASPI on Adaptec's
home page. If you experience problems, please feel free to contact me.

I have developed the software on both Windows 2000 and Windows XP so I am
particularly interested in feedback on OS compatibility issues with Windows
95/98/ME as well as the items outlined above.

Thanks you for your help.

John Lundy


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[MP3 ENCODER] Re: mp3encoder digest, Vol 1 #103 - 2 msgs

[Du Xiaonan]

Hi just some comments. I think mp3 is going obsolete pretty soon.why dno't we move on to some topics like mp3 pro  or wma?

>From: [EMAIL PROTECTED] 
>Reply-To: [EMAIL PROTECTED] 
>To: [EMAIL PROTECTED] 
>Subject: mp3encoder digest, Vol 1 #103 - 2 msgs 
>Date: Tue, 29 Jan 2002 07:33:35 +1000 (EST) 
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>When replying, please edit your Subject line so it is more specific 
>than "Re: Contents of mp3encoder digest..." 
> 
> 
>Today's Topics: 
> 
> 1. lame 3.90 win32 - processed filesize (Andrey Glazunov) 
> 2. Re: MS Stereo (Mark Taylor) 
> 
>--__--__-- 
> 
>Message: 1 
>From: "Andrey Glazunov" <[EMAIL PROTECTED]>
>To: <[EMAIL PROTECTED]>
>Date: Sat, 26 Jan 2002 04:47:24 +0300 
>Subject: [MP3 ENCODER] lame 3.90 win32 - processed filesize 
>Reply-To: [EMAIL PROTECTED] 
> 
>This is a multi-part message in MIME format. 
> 
>--=_NextPart_000_0072_01C1A624.8BF690F0 
>Content-Type: text/plain; 
> charset="koi8-r" 
>Content-Transfer-Encoding: quoted-printable 
> 
>Hi All ! 
> 
>i try to use lame mp3 encoder (and decoder) to create a sort of media = 
>archiver for old enough game's databases which currently saves sounds in = 
>.wav ... 
> 
>but when i try to encode and then decode sample .wav file i.e. .wav -> = 
>.mp3 -> .wav (VBR) 
>i get strange result - at the output .wav the size of 'data' block = 
>exceeds the size of the 'source' .wav data at about 2..3% ... why ? can = 
>i get its sizes equal by any way ?=20 
> 
>i take 'data' block from my game's database and have to insert exact = 
>block (in bytes) after decompressing or database's format would crashes = 
>:( ... i.e. i need a way to keep the size of data samples of the same = 
>size in bytes as in the 'source' .wav (database) ...=20 
> 
>can anyone comment this situation ? 
> 
>--=_NextPart_000_0072_01C1A624.8BF690F0 
>Content-Type: text/html; 
> charset="koi8-r" 
>Content-Transfer-Encoding: quoted-printable 
> 
> 
> 
> 
>
 
>

 
> 
> 
>
Hi All !
>
 
>
i try to use lame mp3 encoder = >(and decoder)=20 
>to create a sort of media archiver for old enough game's databases which = 
> 
>currently saves sounds in .wav ...

>
 
>
but when i try to encode and then = >decode=20 
>sample .wav file i.e. .wav -> .mp3 -> .wav  = 
>(VBR)

>
i get strange result - at the = >output=20 
>.wav  the size of 'data' block exceeds the size of the 'source' = 
>.wav data="20" 
>at about 2..3% ... why ? can i get its sizes equal by any way ? = 
>

>
 
>
i take 'data' block from my = >game's database=20 
>and have to insert exact block (in bytes) after decompressing or = 
>database's=20 
>format would crashes :(  ... i.e. i need a way to keep the size of = 
>data="20" 
>samples of the same size in bytes as in the 'source' .wav = 
>(database)  ...=20 
>

>
 
>
can anyone comment this situation = > 
>?

> 
>--=_NextPart_000_0072_01C1A624.8BF690F0-- 
> 
> 
> 
>--__--__-- 
> 
>Message: 2 
>Date: Fri, 25 Jan 2002 13:44:13 -0700 
>From: Mark Taylor <[EMAIL PROTECTED]>
>To: [EMAIL PROTECTED] 
>Subject: Re: [MP3 ENCODER] MS Stereo 
>Reply-To: [EMAIL PROTECTED] 
> 
> 
> > 
> > Hi all, 
> > 
> > Just to inquire more about MS Stereo. I am confused about how the block is 
> > decided between long and short blocks. 
> > 
> > In the previous algorithm, the blocks are decided based on LR thresholds 
> > after L3psycho_anal. Is that true? Can't we use the final LR/MS thresholds ie 
> > fthr for the block decision. 
> > 
> > One more thing on bmax(b) from 13818-7 MS Stereo section. Is that applied to 
> > both short and long blocks? In lame 3.21, I notice that it is use on the 
> > scalefactorbands instead of the partitions. So when the standard states to 
> > calculate the LR/MS and final thresholds, it actually means the LR/MS ratios in 
> > the after calculating the PE. Is that right? 
> > 
> > Will appreciate all assistance. 
> > 
> > Hann Long 
> > NUS Student 
> > 
> 
>Keep in mind that the IS

Re: [MP3 ENCODER] segmentation fault in lame 3.91

[Mark Taylor]

> hi,
> 
> I get this segfault whenever I try to encode with lame 3.91:
> 
> Current directory is /home/felix/src/lame-3.91/frontend/
> GNU gdb 19990928
> Copyright 1998 Free Software Foundation, Inc.
> GDB is free software, covered by the GNU General Public License, and you are
> welcome to change it and/or distribute copies of it under certain conditions.
> Type "show copying" to see the conditions.
> There is absolutely no warranty for GDB.  Type "show warranty" for details.
> This GDB was configured as "i686-pc-linux-gnu"...
> (gdb)  set args ~/music/eigene/test.wav
> (gdb) r
> Starting program: /home/felix/src/lame-3.91/frontend/lame ~/music/eigene/test.wav
> LAME version 3.91  (http://www.mp3dev.org/)
> Resampling:  input 44.101 kHz  output 44.1 kHz
> Using polyphase lowpass  filter, transition band: 15115 Hz - 15648 Hz
> Encoding /home/felix/music/eigene/test.wav
>   to /home/felix/music/eigene/test.wav.mp3
> Encoding as 44.1 kHz 128 kbps j-stereo MPEG-1 Layer III (11x) qval=5
> Frame  |  CPU time/estim | REAL time/estim | play/CPU |ETA 
>  0/65 ( 0%)|0:00/0:00|0:00/0:00|   0.x|0:00 0 0 404 
>404


It is a bug in the resampling routine - I guess it was never tested
when resampling from 44101 Hz to 44100 Hz!

Quick workaround:  edit the .wav header and change the samplerate
from 44101 to 44100.  That's only an error of 1 second for every
12 hours of music.   Or check the program you used to make the test.wav,
since I'm sure the intended samplerate was 44100?

Temporary fix:  I'm going to modify the code so that it doesn't
bother to resample if the input and output rates agree to 4 digits.  
We should fix this bug, but what should really be done is replace the
resampling code with a well designed resample + lowpass + highpass
system.  

Mark







Mark

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[MP3 ENCODER] segmentation fault in lame 3.91

[Felix Natter]

hi,

I get this segfault whenever I try to encode with lame 3.91:

Current directory is /home/felix/src/lame-3.91/frontend/
GNU gdb 19990928
Copyright 1998 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
Type "show copying" to see the conditions.
There is absolutely no warranty for GDB.  Type "show warranty" for details.
This GDB was configured as "i686-pc-linux-gnu"...
(gdb)  set args ~/music/eigene/test.wav
(gdb) r
Starting program: /home/felix/src/lame-3.91/frontend/lame ~/music/eigene/test.wav
LAME version 3.91  (http://www.mp3dev.org/)
Resampling:  input 44.101 kHz  output 44.1 kHz
Using polyphase lowpass  filter, transition band: 15115 Hz - 15648 Hz
Encoding /home/felix/music/eigene/test.wav
  to /home/felix/music/eigene/test.wav.mp3
Encoding as 44.1 kHz 128 kbps j-stereo MPEG-1 Layer III (11x) qval=5
Frame  |  CPU time/estim | REAL time/estim | play/CPU |ETA 
 0/65 ( 0%)|0:00/0:00|0:00/0:00|   0.x|0:00 0 0 404 404
<906> 30 30 388 404
<454> 162 162 290 398
<728> 491 491 2 398
<497> 677 389 142 398
<680> 528 528 2 394
<531> 43 43 376 400
<778> 434 434 86 394
<544> 7 7 394 398
<797> 819 531 2 394
<535> 25 25 382 398
<722> 536 536 0 396
<536> 6 6 392 396
<782> 85 85 342 398
<550> 197 197 268 396
<719> 25 25 382 398
<579> 12 12 388 396
<751> 84 84 340 396
<555> 7 7 392 396
<723> 468 468 2 394
<477> 0 0 398 398
<791> 62 62 358 398
<554> 6 6 396 400
<728> 823 535 2 398
<539> 61 61 362 398
<782> 534 534 2 394
<538> 14 14 390 398
<752> 562 562 2 398
<562> 96 96 334 398
<755> 6 6 394 398
<527> 80 80 350 398
<799> 432 432 86 398
<539> 7 7 392 396
<859> 5 5 392 396
<645> 6 6 396 400
<843> 118 118 318 398
<605> 6 6 396 400
<938> 11 11 388 396
<632> 6 6 394 398
<877> 27 27 380 396
<666> 7 7 392 396
<832> 11 11 388 396
<614> 6 6 396 400
<900> 5 5 396 400
<663> 7 7 396 400
<913> 36 36 374 398
<651> 6 6 396 400
<844> 26 26 382 398
<662> 7 7 394 398
<891> 5 5 396 400
<648> 6 6 392 396
<891> 6 6 394 398
<660> 16 16 390 398
<885> 42 42 374 398
<630> 6 6 396 400
<862> 20 20 386 398
<631> 6 6 396 400
<873> 12 12 390 398
<677> 6 6 396 400
<902> 460 460 84 396
<570> 5 5 396 400
<932> 6 6 390 394
<672> 14 14 388 396
<824> 261 261 230 398
<617> 6 6 396 400
<949> 5 5 392 396
<642> 0 0 398 398
<904> 7 7 394 398
<648> 0 0 398 398
<888> 18 18 382 394
<609> 22 22 386 398
<897> 39 39 374 398
<668> 
Program received signal SIGSEGV, Segmentation fault.
0x8063137 in fill_buffer_resample (gfp=0x81216c8, outbuf=0x40158a54, 
desired_len=1152, inbuf=0x8126ce0, len=1152, num_used=0xbfff8834, ch=0)
at util.c:662
(gdb) bt
#0  0x8063137 in fill_buffer_resample (gfp=0x81216c8, outbuf=0x40158a54, 
desired_len=1152, inbuf=0x8126ce0, len=1152, num_used=0xbfff8834, ch=0)
at util.c:662
#1  0x8062a83 in fill_buffer (gfp=0x81216c8, mfbuf=0xbfff883c, 
in_buffer=0xbfff8844, nsamples=1152, n_in=0xbfff8834, n_out=0xbfff8838)
at util.c:531
#2  0x805c1fe in lame_encode_buffer_sample_t (gfp=0x81216c8, 
buffer_l=0x8126ce0, buffer_r=0x8127ee8, nsamples=1152, 
mp3buf=0xbfffad0c 
"\016Ë渼\227\213%'¼\fÃ\eø^Ñ°²ÑÕÚNc\032\203'¤ Ëµjm]\006ܤÜæCKjN[L¹\232Oü\004çæl\035\237¾2î\231\034|÷¹9çÅ\037\037épôå4ｿ\034¿lÿu\221¹»ÿû\222dî\017óµcC\001æ3rPLH\200\004cn\016!\217\016\006\230ÍÁD1b",
 mp3buf_size=16384)
at lame.c:1491
#3  0x805c5e7 in lame_encode_buffer_int (gfp=0x81216c8, buffer_l=0xbfff890c, 
buffer_r=0xbfff9b0c, nsamples=1152, 
mp3buf=0xbfffad0c 
"\016Ë渼\227\213%'¼\fÃ\eø^Ñ°²ÑÕÚNc\032\203'¤ Ëµjm]\006ܤÜæCKjN[L¹\232Oü\004çæl\035\237¾2î\231\034|÷¹9çÅ\037\037épôå4ｿ\034¿lÿu\221¹»ÿû\222dî\017óµcC\001æ3rPLH\200\004cn\016!\217\016\006\230ÍÁD1b",
 mp3buf_size=16384)
at lame.c:1651
#4  0x8049cf2 in lame_encoder (gf=0x81216c8, outf=0x81218a0, nogap=0, 
inPath=0xb0a4 "/home/felix/music/eigene/test.wav", 
outPath=0xb874 "/home/felix/music/eigene/test.wav.mp3") at main.c:416
#5  0x804a45b in main (argc=2, argv=0xbcc4) at main.c:745
(gdb) 

thanks,

-- 
Felix Natter

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Re: [MP3 ENCODER] [ot]patent issues and open source

[Gabriel Bouvigne]

> I think their letter to NetBSD's was more dangerous than usual letters for
> AAC that we all have :)

In fact I have both. In order to reduce costs, it seems that right now
they're sending letters for both ac3 and aac at the same time.

> Yes, and FhG and Thomson decided not to charge patent fees for free
decoders
> since 1996, and for AAC this is not true :)

They are now also friendly with encoders, but we're really getting off topic
here...

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Re: [MP3 ENCODER] [ot]patent issues and open source

[Ivan Dimkovic]

> > Slightly off-topic, but if I remember correctly, Dolby also demanded
from
> > NetBSD to cease distribution of AC3DEC software, which is a source-code
of
> > the AC-3 decoder implementation. In their warning letter (also signed by
> > Christy Bonner) they said that Dolby has all intellectual rights to AC-3
> and
> > that they do not tollerate any unlicensed distribution of the AC-3
> software.
>
> The same kind of letter than the one we (Menno, you, me, ...) received
>

It is little bit different, since AC-3 is entirely Dolby's IP and AAC is
basically mostly AT&T's and FhG's work, and Dolby is licensing adminstrator.

I think their letter to NetBSD's was more dangerous than usual letters for
AAC that we all have :)

>
> I really don't know what happened.
> Btw relations are much more "friendly" with Thomson than with Dolby.
>

Yes, and FhG and Thomson decided not to charge patent fees for free decoders
since 1996, and for AAC this is not true :)

-- Ivan


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Re: [MP3 ENCODER] [ot]patent issues and open source

[Gabriel Bouvigne]

> Slightly off-topic, but if I remember correctly, Dolby also demanded from
> NetBSD to cease distribution of AC3DEC software, which is a source-code of
> the AC-3 decoder implementation. In their warning letter (also signed by
> Christy Bonner) they said that Dolby has all intellectual rights to AC-3
and
> that they do not tollerate any unlicensed distribution of the AC-3
software.

The same kind of letter than the one we (Menno, you, me, ...) received

> Does anybody know what happened with this case? I do not use NetBSD, and
> therefore I can't verify does it contain AC-3 decoder source code anymore.

I really don't know what happened.
Btw relations are much more "friendly" with Thomson than with Dolby.

Regards,


Gabriel Bouvigne
www.mp3-tech.org


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Re: [MP3 ENCODER] [ot]patent issues and open source

[Gabriel Bouvigne]

> Where did you get the information that source code is fully
> legal?

The source code itself doesn't do anything. It's just a written description
of something. The description of a patent is publicaly available, as it's
the principle of patents. So another description is as legal as the patent
itself.

But I might be wrong as I'm not a lawyer.

Either Christina Bonner (current licensing coordinator for AAC ) or her
predecessor (who also send me a nice letter in the past) never contradicted
me on this point.

Regards,


Gabriel Bouvigne
www.mp3-tech.org
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Re: [MP3 ENCODER] [ot]patent issues and open source

[Ivan Dimkovic]

Gabriel,

- Original Message -
From: "Gabriel Bouvigne" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, February 04, 2002 8:44 AM
Subject: Re: [MP3 ENCODER] [ot]patent issues and open source

> You can definitively publish source code only.
> I received the same kind of letter from Dolby about AAC source code,
> explaining that they could even sue me.
> I responded back that source code is fully legal, and that I'll never
remove
> the source code. It was several months ago, and I didn't heard back from
> them.
>

Slightly off-topic, but if I remember correctly, Dolby also demanded from
NetBSD to cease distribution of AC3DEC software, which is a source-code of
the AC-3 decoder implementation. In their warning letter (also signed by
Christy Bonner) they said that Dolby has all intellectual rights to AC-3 and
that they do not tollerate any unlicensed distribution of the AC-3 software.

Does anybody know what happened with this case? I do not use NetBSD, and
therefore I can't verify does it contain AC-3 decoder source code anymore.

Best Regards,
-- Ivan Dimkovic


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Re: [MP3 ENCODER] [ot]patent issues and open source

[alexander lerch]

Hi,

Gabriel Bouvigne wrote:
>
> You can definitively publish source code only.
> I received the same kind of letter from Dolby about AAC source code,
> explaining that they could even sue me.
> I responded back that source code is fully legal, and that I'll never remove
> the source code. It was several months ago, and I didn't heard back from
> them.

I agree, if they do nothing anymore there is a high
possibility that they are not able to do anything.  But there
is also the possibility that they decided to start the war
later. 
Where did you get the information that source code is fully
legal?

Regards,
Alexander

-- 
alexander lerch
[EMAIL PROTECTED]

zplane.development
http://www.zplane.de
holsteinische str. 39-42
D-12161 berlin
 fon: +49.30.854 09 15.0
 fax: +49.30.854 09 15.5
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Re: [MP3 ENCODER] [ot]patent issues and open source

[Ivan Dimkovic]

> Same goes for me, I received a few of those letters, too. They did not
> complain anymore after removing the binaries only (while they said I had
to
> remove everything (even a program that stripped a few bytes from an AAC
file
> :-)))
>

According to Christina Bonner, licensing coordinator for AAC - Dolby takes
EVERY use of unlicensed AAC technology as "patent infringement" - even if
you are making your own AAC implemetion without their license :)  Of course,
this is their own view of things and probably it is a pure nonsense from the
legal point of view, but...

I would be very careful and ask for an advise of some very good lawyer, with
strong background in this kind of patents.

Since Alexander Lerch has problems with other company, maybe they have
different opinion on source code distribution, who knows.. and patent law in
Canada might be different from one in the U.S.

-- Ivan


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Re: [MP3 ENCODER] [ot]patent issues and open source

[Menno Bakker]

> > However, I do not provide binaries and I state in the readme
> > "Please note that parts of this software can be under patent
> > right restrictions and thus cannot be used freely in some
> > countries.", but I got a letter about patent infringements
> > which demands to stop all distribution of software.
>
> You can definitively publish source code only.
> I received the same kind of letter from Dolby about AAC source code,
> explaining that they could even sue me.
> I responded back that source code is fully legal, and that I'll never
remove
> the source code. It was several months ago, and I didn't heard back from
> them.

Same goes for me, I received a few of those letters, too. They did not
complain anymore after removing the binaries only (while they said I had to
remove everything (even a program that stripped a few bytes from an AAC file
:-)))

Menno
http://www.audiocoding.com
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