> I raise to 99.99%. It's 0:43...0:47 of "I wan't tommorow" from the album
> "The Celt" performed by Enya. That explains why I haven't found it
> yesterday. The are a lot of similar spots in a lot of Enya albums.
"The Celts". :-)
> :: > BlackBird.wav: Also distortions in the original P
Is this the right address to post to the mp3encoder list? I didn't find
anything that mentioned the address to post to for this list, so if it is...
please respond!
Just so it's not a complete waste of a message, does anyone know if I can
get read access to the CVS server for LAME? (Whatever the
Hello Mark,
Tuesday, August 22, 2000, 9:18:22 PM, you wrote:
MT> Problem is, this is a lot of work and it is not clear that it would
MT> really improve things.
does it mean anthing if I say it will? :)
MT> The hard part is how do you tell if M/S gives
MT> better results than S? The only way is
Well I've got some modifications to nasm.h and Makefile.DJGPP that should do
the trick to get the MMX functions to work right, very simple, but it
works...
Makefile.DJGPP
nasm.h
I'm running the win32 compiles found on http://www.maindex.com/lame/ in
Windows 2000. After encoding is complete, I get a table showing bitrates and
percentages, but the linux versions I've compiled and ran always show and
update this table while encoding, frame by frame. The win32 version shou
>
> 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/mp3en
I think the only way is to find every frame, and analyse the header. It will give you
the frame length with the formula mentionned earlier
(sample_per_frame/sampling_frequency).
Otherwise, you'll be missing the VBR aspect of files (unless you can get the 'real'
bit rate).
- Original Messag
> > BIT_RATE = 1 means free format.
>
> This seems kinda confusing -- why use 1 to indicate free when MPEG uses 0?
0 is for 'reserved' (I'm not responsible of this part of the code).
> > I've checked the file in hexadecimal, and sometimes I get
> > frame header at 202 bytes from the previous i
> 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 distortio
> Why do the win32 versions not show on the fly bitrate analysis for vbr
> files?
They do. Try the one located on mp3-tech.org, and you'll see bitrate
analysis
Regards,
--
Gabriel Bouvigne - France
[EMAIL PROTECTED]
icq: 12138873
MP3' Tech: www.mp3-tech.org
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MP3 ENCODER mailing list ( htt
> > I had a problem with some files to compute the length in second of a
> > frame. I was using various Layer III format and is some cases the formula
> > "frame_time = samples / frame_per_second" wouldn't give me the right
> > length. I assume it might be comming from this LSF.
>
> How do you ca
Thanx, that was the info I was missing.
Now I get the track length on every example I had (some with free format too, some
with different sampling freq, some with different bit rates)
- Original Message -
From: "Mark Taylor" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, Aug
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
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
>
> 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 22
> Frank: if your point is that a compliant bitstream is generally not the
> best sounding one (where 'best' is defined by your taste), your point is
now
> well taken, and we can all read up on more effective interpolation schemes
> to our hearts' content. If, however, your point is that Rob was
Hello Gabriel,
Tuesday, August 22, 2000, 3:59:45 PM, you wrote:
GB> I'm not equiped for listening tests here (only an awe64)
GB> , but is the velvet
GB> problem the thing I'm hearing in the right channel? (or am I thinking I'm
GB> hearing something?)
I have $25 sb128 pci and $38 Sennheiser HD-4
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 -bw 11
Howdy,
> :: My purpose was to measure the computational accuracy of
> audio decoders in the
> :: manner described by ISO/IEC 11172-4.
> ::
> The result is a ISO/IEC 11172-4 compliant decoder. It's not
> so much better
> than a ISO 9001 certificate. A decoder with a bug fails the test.
> But not
Jaroslav Lukesh schrieb am Die, 22 Aug 2000:
> lame -h -m j -b 64 --voice --noshort --resample 22.05 --lowpass 11.025
> --lowpass-width 0 --mp3input x.mp3 x.small.mp3
why not lame --preset fm -h -k x.mp3 x.small.mp3 ???
but I would not use -k there.
> Note that --voice options sounds bet
Howdy Gabriel,
> First, please note that it has been a long time I didn't really looked
> inside of the Lame code, so I'll perhaps tell a few wrong statements.
> (btw, please could anyone explain me when to use the word "tell" and
> when "say"?)
Hmmm... That's actually kind of tricky. Neither
I'm not equiped for listening tests here (only an awe64), but is the velvet
problem the thing I'm hearing in the right channel? (or am I thinking I'm
hearing something?)
If this is the case, it seems to me that it's reduced in -m f, but the
stereo image is also changed by this switch.
If it's gon
> > From: Mark Taylor [mailto:[EMAIL PROTECTED]]
> >
> > MP3 does not allow mid/side stereo to be turned on and off on a band by
> band basis (AAC does),
>
> Hmm, then what happens if a JS frame has both M/SS and IS enabled? In Layers
> I and II, IS can be enabled across several fixed bands, I gu
> From: Mark Taylor [mailto:[EMAIL PROTECTED]]
>
> MP3 does not allow mid/side stereo to be turned on and off on a band by
band basis (AAC does),
Hmm, then what happens if a JS frame has both M/SS and IS enabled? In Layers
I and II, IS can be enabled across several fixed bands, I guess something
Hello Gabriel,
Tuesday, August 22, 2000, 12:43:07 PM, you wrote:
GB> First, please note that it has been a long time I didn't really looked
GB> inside of the Lame code, so I'll perhaps tell a few wrong statements. (btw,
GB> please could anyone explain me when to use the word "tell" and when "say
| Like many people on this list (I imagine) I have a fairly substantial
| collection
| of good quality (-v --preset studio) MP3s encoded with lame. I'd love to
be
| able to be able to make them smaller to squash more musiconto my portable
| MP3 player.
|
| Is it possible to have lame digitally re
> If you make a "grep RH_AMP *.c" over all C-sources you will find out,
> that it enables a different amp_scalefac_bands() in quantize.c and
> makes a little psymodel tweak in psymodel.c.
>
>
> Ciao Robert
Thank you for your quick replay.
Actually, most of the people might think 'thank you' mai
> From: Stephen Kennedy [mailto:[EMAIL PROTECTED]]
>
> Is it possible to have lame digitally resample an mp3 on a
> frame-by-frame
> basis? Although granted, the quality wouldn't be as good as a 'slow'
> encoding,
> I'd imagine this could be extremely fast!
Hmmm, I doubt it. The psychoacoustic
> :: FYI, the strategy I took with MAD was to provide full (28-bit)
> :: internal precision output from the decoder library API and let
> :: the application decide how to scale or resample it to however
> :: many bits of precision it wants.
> ::
> I want 86 dB (CCIR). That can be achived with
Hi Gaby!
> > I'd be slow, but it'd be the best possible quality for cbr and best
> quality/size
> > for vbr. (as far as I can see)
>
> not so slower, if both sets of data are already computed
Psychoacoustics are calculated for stereo and mid-side
stereo frames. Then a decision is made using mas
Hello 'lamers',
Like many people on this list (I imagine) I have a fairly substantial
collection
of good quality (-v --preset studio) MP3s encoded with lame. I'd love to be
able to be able to make them smaller to squash more musiconto my portable
MP3 player.
Is it possible to have lame digitall
First, please note that it has been a long time I didn't really looked
inside of the Lame code, so I'll perhaps tell a few wrong statements. (btw,
please could anyone explain me when to use the word "tell" and when "say"?)
> If I understand correctly, the "-mj" is evaluating if a frame
> qualifie
:: >Another point: I notice you changed the encoding status display to
:: >update every 2 seconds, rather than every 100 frames. Any reason for
:: >this? I prefer 100 frames because I like looking and nice round
:: >numbers, and it doesn't impact the speed. Also, to do the 2 second
::
::
::
:: FK> May be mpg123 in october. I try to convince Michael Hipp to use my
:: FK> ultra-fast round-to-nearest-integer routines°°). If he uses these
:: FK> routines, it becomes very easy to use other quantization routines by
:: FK> simple replacing 2 routines by another and recompiling. Cur
Hi,
After a whole lot of testing and listening it came to me: "-mj nor
-ms" are optimal quality-wise.
* -ms unnecesarely wastes bits most of the time
* -mj has M/S making too much unnecesary mistakes:
If I understand correctly, the "-mj" is evaluating if a frame
qualifies for M/S coding beforeh
> Hi, there.
>
> Could someone explain to me what this new option called 'RH_AMP'
> of LAME CVS version means? Makefile for gcc tells me just
> 'special noise calculation.'
> However, I have no idea about what exactly this means. Oh, well...
>
> Thank you in advance.
>
> Keeshond.
If you
> From: "Shawn Riley" <[EMAIL PROTECTED]>
>
> Alex wrote:
>
> > Thus, I surmise that there are no 405 byte frames - you just aren't
> > tracking syncs correctly.
>
> Does that mean that each frame has to have a whole number of bytes
Yes, Layer III works with 8-bit "units".
> therefore that the s
In my opintion VBR encoding can be done with an ACM driver (as you might
know, vorbis atm is only VBR). The most severe problem with my ACM
driver atm is vorbis' block size, which is rathar large compared to
MP3 (about 4K, MP3 has blocks around 512B).
If you are interested in the sources, the lat
Alex wrote:
>Thus, I surmise that there are no 405 byte frames - you just aren't
tracking syncs correctly.
Does that mean that each frame has to have a whole number of bytes, &
therefore that the sync-word is only allowed to start on the first bit of
any byte?
Shawn
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MP3 ENCODER mailing lis
Hi, there.
Could someone explain to me what this new option called 'RH_AMP' of LAME CVS
version means? Makefile for gcc tells me just 'special noise calculation.'
However, I have no idea about what exactly this means. Oh, well...
Thank you in advance.
Keeshond.
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MP3 ENCODER mailing list ( h
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