Re: Advantage of tape over MD?

[Myke Carter]

Teemu Torma wrote:
>
> Well, if you check some newer pop music, like Michael Jackson's
> Invincible, the average level is around -6dBFS, and peaks at 0dBFS.
> The sole purpose of this is to make them sound louder in radio than
> the other songs. But there's not much dynamic range left, even
> though it may not matter in this case.

Yes, I do believe this "follows suit" with my previous comments
regarding "Long View" from Green Day's 1994 CD, "Dookie". All of this
high level, low dynamic range stuff seems to be a common phenomenon of
with music of the '90s. Most of the "loudest CDs" I own seem to kick in
somewhere around 1994 which leads me to suspect that software similar to
"normalize" must have arrived on the mastering scene at some point
around that time - or maybe a few years earlier if one could afford to
buy it. While most CDs mastered prior to that time may have been
mastered "technically correctly", their sonic impact is nowhere near "up
to par" with "today's standards" because of the inexistence of such
software. My efforts to normalize the WAVs from my own CD collection
before encoding them to MP3 is merely an attempt to compensate for this.
And I'm able to derive stunning results from it nearly 100% of the time.

> It does not mean that the "professional" is right.

I've learned this! And, believe me, after running around for so many
years having so much blind faith in both the compact disc medium and the
music industry's ability to "get it right" with CD, I'm now learning
just what a fool I've been. When I personally can use nothing more than
a single application such as normalize and make my Eurythmics, "Sweet
Dreams (Are Made Of This)" CD sound nearly as if it's been "officially
digitally remastered" to my ears, I'm blown away!

> Check some other remasters, like Pink Floyd's The Wall.
> The average level for the album is around -18dBFS.

Yes, I am fanatical about Pink Floyd, "The Wall" and yet the generally
low levels of that particular CD - even with regard to the digitally
remastered version that I own - drives me *up* The Wall! And, in fact,
I'm very near to the time now when I will finally be normalizing it to
-10dBFS as well.

> Depends on what you want. If you want to listen a mixed collection
> of mp3 files without adjusting volume for each random song, normalize
> is one way of doing it, but increasing volume with it does harm the
> music (of course, with 128k mp3 does harm it too). Normalizing can
> never add anything, it can just take something away.

Y'know, you're a decent person with an obviously respectable point of
view, however, I just conducted some more tests with "normalize" and
both MFSL's older Ultradisc II and Capitol's 1994 digitally remastered
versions of "Dark Side Of The Moon" on CD. I think you'll be surprised
by how much sonic *improvement* the "normalize" application can provide.

I'm going to close this particular reply at this time, but keep an eye
out for my next post to this thread regarding these two Pink Floyd CD
reissues and how they compare. Before/after level readings and
screenshots will be provided!

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[cyrus the virus]

In article >, Myke Carter >
wrote:

> Teemu Torma wrote:
> >
> > Well, if you check some newer pop music, like Michael Jackson's
> > Invincible, the average level is around -6dBFS, and peaks at 0dBFS.
> > The sole purpose of this is to make them sound louder in radio than
> > the other songs. But there's not much dynamic range left, even
> > though it may not matter in this case.
>
> Yes, I do believe this "follows suit" with my previous comments
> regarding "Long View" from Green Day's 1994 CD, "Dookie". All of this
> high level, low dynamic range stuff seems to be a common phenomenon of
> with music of the '90s. Most of the "loudest CDs" I own seem to kick in
> somewhere around 1994 which leads me to suspect that software similar to
> "normalize" must have arrived on the mastering scene at some point
> around that time - or maybe a few years earlier if one could afford to
> buy it. While most CDs mastered prior to that time may have been
> mastered "technically correctly", their sonic impact is nowhere near "up
> to par" with "today's standards" because of the inexistence of such
> software. My efforts to normalize the WAVs from my own CD collection
> before encoding them to MP3 is merely an attempt to compensate for this.
> And I'm able to derive stunning results from it nearly 100% of the time.
>
> Myke

the tools have always been around to squash music into oblivion.. but
somewhere around the '93-'94 era, like you stated, it became "in" to do
it.

the sonic impact is there from older cd's still, turn up the volume
knob.

--
cyrus

[Lord Hasenpfeffer]

cyrus the virus wrote:
>
> the tools have always been around to squash music into oblivion..

Probably so. But please do not confuse my beloved "normalization" with
"squashing music into oblivion". I am still reeling from the shock of
having successfully whooped the ass of Mobile Fidelity Sound Labs. My
routine practice of normalization does not involve forcing peaks to
clip. It does not involve destroying dynamic range. It does not
involve the inadvertent addition of digital distortion to what was
originally a pristine, professionally-sanctioned waveform. All of what
you have read from others about my ill-advised use of normalize when
producing MP3s from WAVs from my CDs has been revealed to be an
overflowing crock of ****. And I can and will easily prove this in the
immediately near future.

> but somewhere around the '93-'94 era, like you stated,
> it became "in" to do it.

Alright. That's it! I'm blamin' Bill Gates! ;-D

> the sonic impact is there from older cd's still,
> turn up the volume knob.

This from you is ill-conceived too.

Stay tuned and I will *prove* to you soon that your assumption is 100%
totally incorrect.

We still have a lot to learn,

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Pete Carney]

You truly are clueless.

What you have with "normalize" is a compressor and expander in addition to
normalizing.

It really is unbelievable that you won't go read what the terms, clipping,
compression, expansion, dynamic range and amplitude actually mean.

While your little god "normalize" may be doing a wonderful job of smashing
or extending all your music into your preferred range of amplitude, it
doesn't mean it is appropriate or desirable to everyone, particularly the
engineer who mastered the CD in the first place.

There is a reason they call it art and not musak.

I'd be quite pleased if you'd go spray this sacrilege on the
www.homerecording.com BBS. There are some folks around that really do know
exactly what normalize does.

Cheers,
Pete

"Lord Hasenpfeffer" > wrote in message
...
>
http://www.mykec.com/mykec/images/2003-06-27_PF_DSOTM_MFSL_What_Were_They_Smoking.png
>
> In this corner...
>
> Pink Floyd, "Dark Side Of The Moon" (1973)
> Mobile Fidelity Sound Lab
> Ultradisc II ... "ORIGINAL MASTER RECORDING"
>
> And in this corner...
>
> Pink Floyd, "Dark Side Of The Moon" (1973)
> Mobile Fidelity Sound Lab
> Ultradisc II ... "ORIGINAL MASTER RECORDING"
> Simply "normalized" to -10dBFS by Lord Hasenpfeffer (Whoopeee!)
>
> What??? No clipping???
> What??? No compression???
> What??? No added distortion???
> What??? Dramatically *extended* dynamic range???
> What... were they smoking?!?!?!
>
> No way!!!
>
> Way!!!
>
> Game over,
>
> Myke
>
> --
>
> -================================-
> Windows...It's rebootylicious!!!
> -================================-
>

[Pete Carney]

I have looked at your screen shots. You have definitely limited or
compressed the peaks and expanded the range in addition to increasing the
amplitude. You are not clipping. That terms is not referring to what you
are doing Clipping is where it can't be dealt with by your system.
Completely different thing than limiting or compressing a wave. Clipping is
inadvertent trashing of the wave by a systems inability to deal with the
amplitude input.

If you can post two identical *.wav files of the exact time frame. One from
the original and the second from your remastered version. Something that
has a large range in the original, I can show you precisely what it is doing
to the wave. I can even post some screen shots of a few milliseconds of the
wave so you can see.

Your screen shots are not to any scale that can be seen at 72 dpi on even a
22" monitor. Look at the peaks and valleys of the wave over a few second
time frame rather than the whole song.

You can't make music better by doing any of this, you can only adjust it
such that you like the resulting amplitude better, or that your ear/brain
perceives it to be better. Someone else will differ with you. The point is
you are changing the artists work.

I do essentially the same thing with much of the live recordings I've
downloaded. See www.bluegrassbox.com for some spectacular shows in *.shn
format. I will never purport it to be better or any thing but sacrilege.
You are really stepping on some toes when you do so.

It's not bad, just not better. Don't make it out like they ripped you off.
That particular CD may be mastered at exactly the peak of the equipment used
at the time. It actually may sound significantly better on that equipment
the way it was mastered than the way you have it on your equipment.

People spend $10's of thousands of dollars on tube amps and such to
specifically reproduce the sound that older works were mastered with. Some
of them show a marked improvement on that equipment.

Cheers,
Pete

"Lord Hasenpfeffer" > wrote in message
...
> Pete Carney wrote:
>
> > You truly are clueless.
>
> And I've never even *been* to Seattle at that.
> Hmmm... How'd that happen?
>
> > What you have with "normalize" is a compressor and expander in addition
to
> > normalizing.
>
> OK???
>
> > It really is unbelievable that you won't go read what the terms,
clipping,
> > compression, expansion, dynamic range and amplitude actually mean.
>
> My best-guesses in a nutshell are:
>
> Clipping - flat-topping the peaks (at any level amplitude)
>
> Compression - supressing the amplitude
>
> Expansion - increasing the dynamic range
>
> Dynamic Range - the dB "distance" between the lowest and highest
> amplitudes in a recording
>
> Amplitude - the dB "distance" between a peak and the noise floor
>
> > While your little god "normalize" may be doing a wonderful job of
smashing
> > or extending all your music into your preferred range of amplitude,
>
> I notice that you haven't actually cited reference to anything in the
> screenshot which I provided at the start of this thread. Did you
> actually look at it or are you just shootin' off?
>
> > it doesn't mean it is appropriate or desirable to everyone, particularly
> > the engineer who mastered the CD in the first place.
>
> Since producing that screenshot, I've visually compared the waveform
> characteristics my own "normalized remastered edition" of the MFSL "Dark
> Side Of The Moon" WAV against that of the 1994 Capitol digitally
> remastered reissue and they're damn near identical in terms of amplitude.
>
> In other words, I've now improved MFSL's original WAV to such a degree
> that it now compares favourably with Capitol's digitally remastered one.
>
> Naturally this leads me to question why MFSL would bother to sell such
> an obviously weak product on a gold plated compact disc? I consider
> that to be a gross violation of the public's trust. Somebody's gotta
> call them on this - and I guess it won't be you who does it, huh?
>
> > There is a reason they call it art and not musak.
>
> Yes, I am a 20-year veteran recording artist in my own right. I am not
> new to art.
>
> > I'd be quite pleased if you'd go spray this sacrilege on the
> > www.homerecording.com BBS. There are some folks around that really do
know
> > exactly what normalize does.
>
> Why don't you go view my screenshot and tell me what the hell you think
> is so patently obviously wrong with what I've done to MFSL's original
> WAV - and then tell me what's obviously superior about selling an
> original recording such as theirs for $25+ to the public on a gold
> plated disc. They're CD sounded like **** to me 6 years ago when I
> bought it compared to Capitol's remastered reissue. And now I know why!
>
> Duh!
>
> Myke
>
> --
>
> -================================-
> Windows...It's rebootylicious!!!
> -================================-
>

[Martin Tillman]

On Fri, 27 Jun 2003 03:23:11 -0500, Lord Hasenpfeffer wrote:

> http://www.mykec.com/mykec/images/2003-06-27_PF_DSOTM_MFSL_What_Were_They_Smoking.png
>
> In this corner...
>
> Pink Floyd, "Dark Side Of The Moon" (1973)
> Mobile Fidelity Sound Lab
> Ultradisc II ... "ORIGINAL MASTER RECORDING"
>
> And in this corner...
>
> Pink Floyd, "Dark Side Of The Moon" (1973)
> Mobile Fidelity Sound Lab
> Ultradisc II ... "ORIGINAL MASTER RECORDING"
> Simply "normalized" to -10dBFS by Lord Hasenpfeffer (Whoopeee!)
>
> What??? No clipping???
> What??? No compression???
> What??? No added distortion???
> What??? Dramatically *extended* dynamic range???
> What... were they smoking?!?!?!

What the **** are YOU on?

You've completely munged the dynamics, that's just so damn obvious, and
you really have no idea that that is what you've done!! Normalise? My
arse.

Don't you comprehend the huge amount of talent, let alone real
understanding of the processes, that went into this? And you come along
with a bit of freeware an no knowledge and preach to us about how you
can magically transform this 'rubbish' into a work of art?

And, I'd hazard a guess that the peaks on the original are -6dBFS
(simply because that's typical of remastering from that period), and
your botch-up goes to odBFS, no matter what your mis-understanding
and/or mis-use of the software leads you to believe, and despite the
lack of any scale on your screenshot.

You really think this is better? You're insane.

How dare you speak as though you are some guru and mislead all those
with little knowledge and much gullibility.

You are now officially a ****wit.

[flint]

Goodness. I think you may be out of your league here. What you have done it
stupid and worthless in every possible way, except as an experiment to learn
what your software can do. ANY signal processing done to a digital audio
file is destructive in nature. Even changing the volume (as in this case).

Go read some more books on digital audio theory and practice, then consider
what you have done.

- Flint


"Lord Hasenpfeffer" > wrote in message
...
>
http://www.mykec.com/mykec/images/2003-06-27_PF_DSOTM_MFSL_What_Were_They_Smoking.png
>
> In this corner...
>
> Pink Floyd, "Dark Side Of The Moon" (1973)
> Mobile Fidelity Sound Lab
> Ultradisc II ... "ORIGINAL MASTER RECORDING"
>
> And in this corner...
>
> Pink Floyd, "Dark Side Of The Moon" (1973)
> Mobile Fidelity Sound Lab
> Ultradisc II ... "ORIGINAL MASTER RECORDING"
> Simply "normalized" to -10dBFS by Lord Hasenpfeffer (Whoopeee!)
>
> What??? No clipping???
> What??? No compression???
> What??? No added distortion???
> What??? Dramatically *extended* dynamic range???
> What... were they smoking?!?!?!
>
> No way!!!
>
> Way!!!
>
> Game over,
>
> Myke
>
> --
>
> -================================-
> Windows...It's rebootylicious!!!
> -================================-
>

[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message
> Yeah, that's right. So why doesn't Mobile Fidelity Sould Lab have one
too?
>
> I suppose their reasoning is that Pink Floyd fans are all a bunch of
> aging boomers who've got money to burn buying gold-plated CDs that sound
> like **** and don't know anything else beyond how to pop in a disc,
> sit back in a chair and toke just a little bit more.


Becuase MFSL have no business in dicking around with compression on the
source material. Their mission was to give as good as possible a quality
accurate transcription of the original master tape to CD. That includes
retaining the exact dynamic range, compression, EQ, etc that Alan Parsons
left the tape with. .

geoff

[flint]

You are comparing a CD mastered with a Peak of -6dB taken directly from Alan
Parson's master tape to one with a Peak of 0dB taken from the label's
re-mixed master tape. Aside from any differences the tapes may have in EQ
and mix, one is simply louder than the other. If you listen to any CD at one
level then compare it to the same CD with the volume turned up 6dB, the
louder one will sound better.

There have been hundreds of papers and articles written over the years
discussing this very point. With music, louder almost always seems better.

This is why it is so important to balance the levels to less than 0.25dB
when comparing the sound of electronics in ABX tests.

Forget everything you think you may have proven. Erase it from your head.
Then try listening to the Gold CD with the volume turned up 6dB.

- FLINT


"Lord Hasenpfeffer" > wrote in message
...
> flint wrote:
>
> > Goodness. I think you may be out of your league here. What you have done
it
> > stupid and worthless in every possible way, except as an experiment to
learn
> > what your software can do. ANY signal processing done to a digital audio
> > file is destructive in nature. Even changing the volume (as in this
case).
>
> So you disagree with Capitol's efforts at digitally remastering the Pink
> Floyd catalog too? Gee whiz... Their CDs sound phenomenal!
>
>
http://www.mykec.com/mykec/images/2003-06-27_PF_DSOTM_Capitol_1994_Remaster.png
>
> > Go read some more books on digital audio theory and practice, then
consider
> > what you have done.
>
> Would you suggest the same to Capitol?
>
> Myke
>
> --
>
> -================================-
> Windows...It's rebootylicious!!!
> -================================-
>

[George W.]

On Fri, 27 Jun 2003 17:06:39 -0500, Lord Hasenpfeffer wrote:

<blah, blah, blah>

Correct me if I'm wrong but this thread began in alt.audio.minidisc
regarding a simple question about the difference between compression
and normalization. In the context of minidisc where compilation discs
are made from various sources where levels are inconsistent due to the
different ways each track was produced normalization is an accepted
and practical way of balancing levels so that one is not always
turning the volume up or down. Hell, it's minidisc, often duped from
mp3's....at this point the integrity of the original has already been
well compromised.

That's very different from what you propose, and the fact that you've
generated replies in this group out of context due to crossposting to
two other groups hasn't helped make much sense out of it. Assuming
anyone's bother to read them.

[Lord Hasenpfeffer]

flint wrote:
> You are comparing a CD mastered with a Peak of -6dB taken directly from Alan
> Parson's master tape to one with a Peak of 0dB taken from the label's
> re-mixed master tape.

Correct.

> Aside from any differences the tapes may have in EQ and mix, one is simply
> louder than the other.

Fair enough.

> If you listen to any CD at one level then compare it to the same CD
> with the volume turned up 6dB, the louder one will sound better.

"The louder one will sound better"... I like that. I think you
understand me better than most when I say "my remastered WAVs sound
better than their source CDs". Because "the louder one will sound
better", yes, thank you.

Louder = more audible frequencies = more clarity in the upper and lower
ranges = livlier, more energetic sounding music = better MP3s = more fun
= more personal satisfaction = my original argument.

> There have been hundreds of papers and articles written over the years
> discussing this very point.

Obviously, then, many then still prefer "quiet", "dull" and "muddy"
sound, no?

> With music, louder almost always seems better.

Almost always, yes. Thank you.

> This is why it is so important to balance the levels to less than 0.25dB
> when comparing the sound of electronics in ABX tests.

Yes! Because by balancing the levels you eliminate significant
variables which would otherwise mask other less obvious differences.

I can't tell you how sweet it is whenever I encounter two copies of the
same song from two different CDs to simply be able to equalize their
levels (with "normalize") so that I can *then* discern which one of them
is mastered from a cleaner source. I do this constantly while creating
MP3s. Only the "winners" survive. All previously encoded MP3s which
fail in this competition are deleted forever - and I always keep notes
in my MP3 headers which remind me later about the CD from whence it came.

> Forget everything you think you may have proven. Erase it from your head.
> Then try listening to the Gold CD with the volume turned up 6dB.

Why don't I just make a digital copy of my 1994 Capitol remaster to an
Imation CD-R with the digital input gain turned down to -6dB?

In fact, I think I could even make that my standard practice for *every*
CD copy I create. And then if anyone complains about it, I'll just tell
'em to "pump up the volume" because "it's *supposed* to be that low".
How far d'ya think *that'd* fly? :) It's the same thing.

No harm, no foul,

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Teemu Torma]

Lord Hasenpfeffer wrote:

> flint wrote:
>> If you listen to any CD at one level then compare it to the same CD
> > with the volume turned up 6dB, the louder one will sound better.
>
> "The louder one will sound better"... I like that. I think you
> understand me better than most when I say "my remastered WAVs sound
> better than their source CDs". Because "the louder one will sound
> better", yes, thank you.
>
> Louder = more audible frequencies = more clarity in the upper and lower
> ranges = livlier, more energetic sounding music = better MP3s = more fun
> = more personal satisfaction = my original argument.

The argument has been whether increasing amplitude in the digital input is a
good thing, not whether louder sounds good. It is the amplifier that makes
the loudness to happen.

>> Forget everything you think you may have proven. Erase it from your head.
>> Then try listening to the Gold CD with the volume turned up 6dB.
>
> Why don't I just make a digital copy of my 1994 Capitol remaster to an
> Imation CD-R with the digital input gain turned down to -6dB?
>
> In fact, I think I could even make that my standard practice for *every*
> CD copy I create. And then if anyone complains about it, I'll just tell
> 'em to "pump up the volume" because "it's *supposed* to be that low".
> How far d'ya think *that'd* fly? :) It's the same thing.

It would be better thing in my mind. Limiting or even worse, clipping, the
high peaks by increasing the volume makes more harm than the slight amount
of added noise by doing the reverse. I have never seen even a "quiet"
remaster (in pop/rock arena) to not have it's peaks at very close to 0dBFS.

Teemu

[Lord Hasenpfeffer]

George W. wrote:
> On Fri, 27 Jun 2003 17:06:39 -0500, Lord Hasenpfeffer wrote:
>
> Correct me if I'm wrong but this thread began in alt.audio.minidisc
> regarding a simple question about the difference between compression
> and normalization.

You need no correction.

> In the context of minidisc where compilation discs are made from
> various sources where levels are inconsistent due to the different
> ways each track was produced normalization is an accepted and
> practical way of balancing levels so that one is not always
> turning the volume up or down.

This is true.

> Hell, it's minidisc, often duped from mp3's....

Not in my case. I always "dupe" mine from either normalized or "batch
normalized" WAVs (whichever is appropriate for the task at hand). My
WAVs are ripped straight from my own CDs. I also often create my own
MP3s from my own CDs for convenience purposes and for being better able
to judge the fidelities of like songs on different source discs.
Normalizing balances the levels in such cases so that differences in the
source qualities from which both CDs were mastered becomes more readily
apparent.

> That's very different from what you propose, and the fact that you've
> generated replies in this group out of context due to crossposting to
> two other groups hasn't helped make much sense out of it.

I was at first impressed to have found so many technical experts lurkin
about in a MiniDisc newsgroup! :) I don't know who's responsible for
the crossposting. Certainly not I. However, I am appreciative to have
found such an interesting and diverse cross-section of opinions
regarding what I've been using "normalize" to do with my WAVs. (All
references to "Liniots" and "****wits" and "trolls" aside, of course.)

> Assuming anyone's bother to read them.

Based upon some comments I've received from others in
alt.music.minidisc, quite a few of them over there seem to be paying
attention.

Hi everybody!!!

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

Teemu Torma wrote:

> The argument has been whether increasing amplitude in the digital input is a
> good thing, not whether louder sounds good.

Actually, my argument seems to have been munged into being a continual
defense of my preferential use of the Linux application named "normalize".

> It is the amplifier that makes the loudness to happen.

You score two points!

> Limiting or even worse, clipping, the high peaks by increasing
> the volume makes more harm than the slight amount of added noise
> by doing the reverse.

Point taken.

> I have never seen even a "quiet" remaster (in pop/rock arena)

Nor have I.

> to not have it's peaks at very close to 0dBFS.

Correct. And while many "older CDs" remain unavailable in digitally
remastered form, it is very sweet to at least be able to elevate the
loudnesses of older, quieter and, therefore, lackluster sounding
pop/rock CDs into the general range of more modern, remastered CDs.
While this obviously does not improve the fidelity of the original
recording, it certainly makes them much more of a pleasure to hear with
my computer.

I honestly believe that if a record label wanted to do it, it could
simply re-issue "normalized" versions of the CDs in its catalog, slap a
"Newly digitally remastered!" sticker on their wrappers and sell 'em to
a loud-hungry public - and, again, there would be dancing in the street.

Perception is everything.

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Dave Platt]

In article >,
Lord Hasenpfeffer > wrote:

>Actually, my argument seems to have been munged into being a continual
>defense of my preferential use of the Linux application named "normalize".
>
>> It is the amplifier that makes the loudness to happen.
>
>You score two points!
>
>> Limiting or even worse, clipping, the high peaks by increasing
> > the volume makes more harm than the slight amount of added noise
> > by doing the reverse.
>
>Point taken.
>
>> I have never seen even a "quiet" remaster (in pop/rock arena)
>
>Nor have I.
>
> > to not have it's peaks at very close to 0dBFS.
>
>Correct. And while many "older CDs" remain unavailable in digitally
>remastered form, it is very sweet to at least be able to elevate the
>loudnesses of older, quieter and, therefore, lackluster sounding
>pop/rock CDs into the general range of more modern, remastered CDs.
>While this obviously does not improve the fidelity of the original
>recording, it certainly makes them much more of a pleasure to hear with
>my computer.

Well, I've looked through the source code for "normalize", and
from what I see, all it's doing (in normal mode of operation) is a
simple volume-boost. It's not increasing the dynamic range of the
music signal at all... it's just lifting it up to a higher point in
the 16-bit digital number space curve. It may, in fact, actually
reduce the useful dynamic range of the signal a bit, if there are
peaks in the original signal which end up having to be gain-limited in
order to avoid clipping.

This means, quite simply, that the effect of running a "whole album"
normalization such as you are doing is has a very specific and easily
describable effect:

It is _precisely_ the same as simply turning up the volume knob on
your computer or CD player by a few dB!

No other difference. No improvement in dynamics. No change in
frequency response or content. No improvement in the actual amount of
musical information present in the recording. None at all.

Well, that's not strictly true. The "normalize" program makes a
fairly common mistake. It's rescaling an audio signal by a
non-integer scale factor (which is OK), but it is *not* re-dithering
the signal when it does so. It's just rounding, and that's not OK.

By doing this, the normalization process is adding distortion -
it's adding a signal-correlated quantization noise.

It turns out that a very similar problem was likely the cause of many
of the complaints about "digital sound" during the early years of the
CD. Analog tapes were digitized using converters which didn't dither
the signal, and there's reason to believe that this probably
contributed somewhat to the "graininess" or harshness of many of the
early CD releases, and to the perceived loss of ambience and low-level
detail in some cases.

By using "normalize" on your CDs, you have re-created this error in
your resulting product.

If you really do want to gain-boost/normalize your CDs, I suggest two
things:

- Study up on digital recording theory and technology, so that you'll
understand that doing so adds precisely _no_ musical information,
and has no beneficial effect which couldn't be achieved by simply
turning up the CD player's volume control a bit.

- Use a better gain-alteration program - one which actually redithers
the signal after scaling it. This will ensure that the
normalization process doesn't add distortion.

>I honestly believe that if a record label wanted to do it, it could
>simply re-issue "normalized" versions of the CDs in its catalog, slap a
>"Newly digitally remastered!" sticker on their wrappers and sell 'em to
>a loud-hungry public - and, again, there would be dancing in the street.
>
>Perception is everything.

Marketing and reality often have little to do with one another.

If you want to market your "normalized" versions as superior, that's
your business... but please don't expect people who understand the
truth of the technology to agree that they are in any way superior to
the un-normalized versions.

--
Dave Platt > AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

[flint]

What I don't get is why you think that a louder CD is better than turning
the volume up on the amplifier. Why go through all this work of copying the
CD to your computer and processing it and burning a CD when all you are
doing is turning the volume up? My preamp has a remote control that I can
use to turn up the volume while sitting in my listening chair. It works
great and I don't get the nasty artifacts that ALL digital processing bring
to the party.

You keep calling the Gold CD inferior simply because the volume is a touch
lower. That is like saying a 60 watt light bulb in inferior to a 75 watt
bulb. Or a 25 watt soldering iron is inferior to a 50 watt soldering iron.
This is not a solid argument. Possibly the program you are using to turn up
the volume on the audio is also adding a form of distortion you really like
to listen to. Maybe it makes the music "crisper" and "sharper" which are two
ways a layman might describe subtle harmonic distortion.

Your argument is weak and you are conviced that you have solved the problem
with every older CD every pressed. Well more power to you. I hope you enjoy
all the work you have in front of you "improving" every CD ever made. But
why try to force the rest of us to agree with you?

If I enjoyed messing with my CDs by altering their sound with digital
processing, I would keep it to myself and enjoy it. But screwing with the
intended results of the producers is nothing I would be proud of.

- FLINT

[Bob Cain]

Lord Hasenpfeffer wrote:
>
> cyrus the virus wrote:
> >
> > the tools have always been around to squash music into oblivion..
>
> Probably so. But please do not confuse my beloved "normalization" with
> "squashing music into oblivion". I am still reeling from the shock of
> having successfully whooped the ass of Mobile Fidelity Sound Labs.

What is your measure of that whooping? Waveform pictures
tell nothing. Try level matched double blind listening
evaluations with a statistically signfigant number of
listeners.

> My
> routine practice of normalization does not involve forcing peaks to
> clip. It does not involve destroying dynamic range. It does not
> involve the inadvertent addition of digital distortion to what was
> originally a pristine, professionally-sanctioned waveform.

I'm afraid you are wrong. If you increase the RMS level of
a normalized piece with the use of limiting/compression you
are doing all the things you think you are not. I'm afraid
your ignorance of the technical apsects of the process is
showing and I say that with all due respect regarding any
other matter.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Lord Hasenpfeffer]

flint wrote:
> What I don't get is why you think that a louder CD is better than turning
> the volume up on the amplifier. Why go through all this work of copying the
> CD to your computer and processing it and burning a CD when all you are
> doing is turning the volume up?

OK. First of all, I do not go about ripping and normalizing my WAVs
only to then burn them back to CD-Rs. I own over 2,100 compact discs -
most but certainly not all of which belong in the pop/rock category. My
current task at hand is to rip and encode MP3s of every CD in my library
so that I can carry most if not all of it with me in only one or two
large Case Logic zipperbooksful of home-burned CD-Rs.

As I am doing this, I also make duplicates of each US Top 40 hit that I
encounter. These duplicate are then given unique filenames containing
year, month, date, peak, artist and title information so that as I go
about doing all this work, my "hits" will automatically be sorted
chronologically regardless of the order in which I've encoded any given
file.

Please click here: http://www.mykec.com/?page=AT40

Over time as more and more "hits" were added to my MP3 library I became
increasingly frustrated in how so many of them were either significantly
louder or quieter than each other due to their having been encoded from
so many different source CDs. Enter "normalize".

As soon as I began using "normalize", I also began noticing how "weak"
and "bad" nearly all of my previously encoded files were sounding in
comparison to the newer, "normalized" ones. Further experimentation
revealed that a significant number of my older, non-remastered CDs could
be "batch normalized" quite safely - usually by a good 5-8 decibels - in
turn making them sound (to my ears) using a single, typical level of
volume virtually as "strong" and as "good" as any of my 24-bit digitally
remastered CDs do. From this I've concluded that the simple process
which I've come to know as "normalization" (by way of this little
"normalize" application) almost certainly plays a significant role in
the digital remastering process - including, of course, all of the other
obvious elements (e.g. higher resolution source material, EQ'ing,
sometimes remixing, etc.) that also usually occurs which is beyond my
individual control.

> You keep calling the Gold CD inferior simply because the volume is a touch
> lower.

It's not just a touch lower. It's a lot lower. And it sounds bad
because of it. When I first bought Capitol's 1994 remaster, I couldn't
believe the difference between those two CDs. But now that the
amplification issue is within my ability to digitally correct, I think
it might be a fun exercise in boredom now to re-evaluate the relative
fidelities of the source materials used to produce these 2 CDs once
their levels are safely equalized.

> Your argument is weak and you are conviced that you have solved the problem
> with every older CD every pressed.

At least as far as my common pop/rock CDs are concerned, I am thrilled
now to be able to create MP3s from all of the older ones which sound
approximately equally as loud as my newer remastered discs, yes.

Think of it this way... Instead of just taking 15-20 songs from various
individual CDs and having to equalize their levels prior to burning them
to a new "mix-CD", I am over time creating a "mix-LIBRARY" of more than
2,100 full-length albums. So, just as it is a "good thing" for the
songs of a mix-CD to all share relatively similar levels - it is also a
"good thing" for all of the MP3s created from my older CDs to have
relatively similar levels with those that are made from my remastered
discs. Before I discovered "normalize", this was for me not possible.
How's that for an explanation?

> But why try to force the rest of us to agree with you?

In nearly every discussion somebody always has to say something like
that. I don't understand why, because I personally don't care what you
or anyone else might choose to do with your time/life/music/whatever -
unless, of course, you/they have some valid information that could help
me with mine.

> If I enjoyed messing with my CDs by altering their sound with digital
> processing, I would keep it to myself and enjoy it.

As a musical/noisical recording artist in my own right, I do not believe
in recreational file-sharing. Nearly 100% of everything I've ever done
with my personal music library has never been heard by anyone other than
my wife, my boss and my closest friends and/or relatives.

> But screwing with the intended results of the producers is nothing
> I would be proud of.

Well, if it means having to choose between (1) once again enjoying an
older CD that's been "normalized" to -10dBFS or (2) stashing it away
forever in a drawer somewhere because I think it sounds like ****, I
think I'll choose the former.

Because until last night when I finally decided to do these tests with
both of "Dark Side Of The Moon" CDs, my MFSL disc hasn't seen the light
of a laser even once in the past 5 years. What good is it to have a CD
that you know you'll never play again? I'm actually surprised that I
hadn't already taken it to a pawn shop or something to get rid of it
before now. After all this, though, I'm really glad I didn't!

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[flint]

If your goal is to make a bunch of MP3s with similar average levels, then
you have succeeded and I applaude you. MP3 encoding is distorting your audio
way beyond what even the cheapest normalizing software could do.

I never accused you of sharing music, I only acused you of creating a huge
stink in this group by sharing (in words) what you are doing and bragging
about it like you are inventing something.

I get the impression all you want to do is solve a simple problem with a
simple solution - great.

It was the subjective comments about a recording merely because it was not
recorded at a level you like that bothered me so much.

- FLINT

[Lord Hasenpfeffer]

Bob Cain wrote:
>
> What is your measure of that whooping?

1 (one).

> Waveform pictures tell nothing.

I was accused of clipping, limiting, compressing, and reducing the
dynamic range of my original WAVs by "normalizing" individual files or
"batch normalizing" whole groups of related files with the Linux
application known as "normalize".

My waveform pictures *do indeed* reveal that not one of these
accusations are true. Every single one of them is *false*.

> Try level matched double blind listening
> evaluations with a statistically signfigant number of
> listeners.

Such "listening evaluations" are worthless when it comes to proving that
I've neither clipped, nor limited, nor compressed, nor reduced the
dynamic range of MFSL's original WAV by "normalizing" it to an average
target level of -10dBFS.

You simply misunderstand my purpose in presenting the screenshots!

> I'm afraid you are wrong. If you increase the RMS level of
> a normalized piece

Stop right there... If I were to do this with "a normalized piece" then
yes, you would be correct - but the WAVs from my personal CD collection
which I subject to this particular treatment are not already normalized!

I'm not just bulldozing my way through my entire CD collection and
normalizing everything in sight. No way! I *always* scan the levels
and peak readings of every file from every CD before I even touch them
with "normalize" and then make my judgement calls from there. Most
modern, standard CDs (i.e. 1994-present) are already perfectly fine.
Virtually every 24-bit remastered CD I've encountered is too. But a
majority of the older CDs I own (i.e. 1983-1993) need a *significant*
degree of "normalization" assistance if they're to sound anywhere near
as good as the newer ones at roughly the same volume setting on my
amplifier.

This is my whole point, my argument, my *beef* with the entire system
that for so many years had me believing that if I bought a CD I was
getting "the best there was to get" - which was a complete and outright
*lie*.

> I'm afraid your ignorance of the technical apsects of the process
> is showing and I say that with all due respect regarding any other
> matter.

That's fair enough, I suppose, given your previous misinterpretions of
just what it is that I'm doing with the WAVs I've been ripping from my
CDs. At least you haven't called me a "Liniot", a "****wit" or a
"USENET troll".

Thank you very much,

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

flint wrote:
> If your goal is to make a bunch of MP3s with similar average levels, then
> you have succeeded and I applaude you.

Thank you! Applause accepted.

I check my work frequently by actually listening to the "normalized"
MP3s I've created because I *am* concerned greatly about not
deliberately contributing to (what I perceive as being) the poor
fidelity that's already a part of my life.

> MP3 encoding is distorting your audio way beyond what even the
> cheapest normalizing software could do.

Yes, and so does my local news/talk AM radio station. (To which I just
have to say, "Aw, damn," before going on about my usual daily affairs.) <g>

One nice thing, however, about my US Top 40 hits collection. I do spend
the money and take the time to burn the WAVs for *those* to CD-Rs before
I delete them. This is because I am already aware that one day
something better than MP3 is going to come along and make me feel the
need to do it all over again. At least my practice of saving the WAVs
will save me a lot of time in the future since I will not have to go
about re-ripping them all over again. But for now, I find the
*convenience* of my MP3s due to the compactness of their filesizes to be
an overwhelming asset in their favor over sonic purity when it comes to
listening to my music wherever "turntables with moonrock needles" cannot
be located.

> I never accused you of sharing music,

No, I didn't mean to imply that you did. I was simply providing a more
illuminated view of what I do actually do with my files (i.e. keep them
to myself) since it seems that you would do the same.

> I only acused you of creating a huge stink in this group by sharing
> (in words) what you are doing and bragging about it like you are
> inventing something.

For a common guy like me to finally have a means via freeware to do what
I do with my CDs, WAVs and MP3s it *is* similar to having invented
something. Sure, the tools that I'm using to do what I'm doing aren't
unique to me - but have you ever met anyone else in your life who's
actually decided to do something like what I'm actually doing?

For me, I think it *is* something to get excited about because every
other person I've ever met who was into ripping and encoding MP3s does
nothing but rip and encode MP3s - because it's a relatively easy process
for them to learn and do. I take it a giant step further, however,
because if I'm going to bother doing all this work, I'm damn sure going
to do everything I can to achieve the most consistently superior results
that I am able to achieve.

It's not a matter of bragging at all. It's a matter of attempting to
wake a few others up to an otherwise unheard of, unseen reality. If I
choose to share with a friend an MP3, they get from me something that
almost always sounds extremely good. Meanwhile, because they know
nothing about "normalize", if/when they reciprocate, I get something
that usually sounds pretty crappy in comparison. So, yeah, naturally
I'm going to promote my view. But promotion of my view and bragging
about it aren't the same.

Now as for the "huge stink" in this group, I'll credit that to all those
who believe they've found valid reasons to label me a "Liniot", a
"****wit" and a "USENET troll" while I've gone about simply attempting
to defend my integrity as a man who actually does care more than most
about the fidelity of the music to which he listens on a regular basis.

> I get the impression all you want to do is solve a simple problem with a
> simple solution - great.

Yes. I'd also "really like it a lot" if more people would normalize
their damn MP3s before making them. Because, trust me, all of the MP3s
I made from my "older CDs" prior to my discovery of "normalize" sound
like **** to me and I know I will eventually have to spend even more
time recreating them than I at first believed I would. <sigh>

> It was the subjective comments about a recording merely because it was not
> recorded at a level you like that bothered me so much.

Understood. However, regardless of its "mission", I still believe that
MFSL either (1) should keep any ****ty CDs they produce off the market
or (2) print the damn (lousy) peak/level readings for each of their CDs'
tracks clearly somewhere on the backside of their packaging so a guy
like me can see exactly what he's getting before he lays down some heavy
bucks for it. IIRC, Capitol's 1994 Remaster only cost me about half of
what that MFSL disc did ... and I enjoy it twice as much!

Over,

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[flint]

> ......I still believe that
> MFSL either (1) should keep any ****ty CDs they produce off the market
> or (2) print the damn (lousy) peak/level readings for each of their CDs'
> tracks clearly somewhere on the backside of their packaging so a guy
> like me can see exactly what he's getting before he lays down some heavy
> bucks for it. IIRC, Capitol's 1994 Remaster only cost me about half of
> what that MFSL disc did ... and I enjoy it twice as much!
>
> Over,
>
> Myke

Myke -

1. MFSL is out of business. They lost their market when people stopped
caring about carefully produced recordings of the original master tapes.

2. There is nothing deceiving or wrong about selling a CD with peaks
at -6dB. In the early days of CD production (and many experts still agree),
the goal was to set the levels so the noise floor (point of inaudibility)
3dB or so above digital zero. As long as the peaks were all well below 0dB,
it was a good thing.

- FLINT

[flint]

> This is my whole point, my argument, my *beef* with the entire system
> that for so many years had me believing that if I bought a CD I was
> getting "the best there was to get" - which was a complete and outright
> *lie*.
>

The average level of the signal on a CD will not affect the "sound" of that
signal. All you are doing is turning the volume up and declaring it "Better"
than a "defective" original. Just because the CD isn't as loud as you prefer
doesn't make it sound any worse. Who lied to you about the recordings. I can
assure you that the MFSL CDs sound as good or better than the originally
released CD made from the original tapes mastered for LP production.

- FLINT

[Bob Cain]

Lord Hasenpfeffer wrote:
>
>
> This is my whole point, my argument, my *beef* with the entire system
> that for so many years had me believing that if I bought a CD I was
> getting "the best there was to get" - which was a complete and outright
> *lie*.

Myke, I'm afraid you've completely lost me. I have no idea
what you want that a volume control doesn't equate to. I
can't imagine what you think might be missing. Best and
truth have nothing whatsoever to do with level in any
universe I've visited recently.


Later,

Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[flint]

> Meanwhile, what really scares me the most as a consumer is my fear of
> the possibility that much of what is being and has already been passed
> off as "digitally remastered stuff" to a sonically illiterate public is
> really nothing more than the "same old stuff" with a simple,
> "normalization" applied to boost the amplitude.

Okay, lesson two:

The many of the original CDs were produced from the master tapes used to
make LPs. In fact, there is a design in the CD format called de-emphasis,
which removes the RIAA curve used for making LPs from the CD signal so CD
producers could save time getting CDs out. If you ever see the "EMP"
indicator on your CD display, you are playing a CD made from a tape mastered
for LP.

When people like MFSL and others proved there was a huge market for using
the original master tapes (those the producer loved the most), they began
"re-mastering" from the original multi-track recordings. However, they did
not intend to change the sound from the original intent of the artists or
producers, they just wanted to create a CD that most closely resembled what
they heard when they mastered the tapes in the first place.

Along the way, the industry decided that using the noise floor as the
reference for the loudness of a recording was not the best idea. They
instead switched to using 0dBFS as their reference for the highest peak in
the recording. In the past couple of years they have started chopping off
the peaks with limiters and compressors and a thing called an "distresser"
to get the average level as high as possible. The goal being to have the
loudest sounding CD on the market, thus better sounding on the radio
(remember my comments about the perception of louder music?).

Also, I recently purchased some "remastered" CDs where they have not
attempted to replicate the original sound the artists and producers got.
Instead they have used different effect processors and added serious
compression to make the classic music sound similar to the current trendy
crap.

But you see, tastes have changed and they no longer primarily produce music
for LP.

- FLINT

[Lord Hasenpfeffer]

flint wrote:
>
> The average level of the signal on a CD will not affect the "sound" of that
> signal.

You and I might understand this but Joe Sixpack doesn't - and in my
little MP3-makin' world - Joe Sixpack, unfortunately, is *everywhere*,
whether he's welcome there or not...

If Joe were ever to care enough to actually sit down and compare 2
different versions of the same music on CD, he's more than likely going
to want to compare them at the same level of volume in order to
eliminate that variable.

If the two discs are not mastered at the same level of volume and all
Joe does is simply listen to each of them, side-by-side, one right after
the other, I guarantee you he's going to pick the louder one and
consider the quieter one to be "inferior" if not outright "defective".

And if you try to tell him that they're really the same thing and all he
has to do to make them sound equally well is crank up the volume just a
little bit more, I guarantee you he'll look you straight in the eye and
say, "But I shouldn't *have* to crank up the volume if it really is the
same as that other one over there."

Perception is key.

And if he really wants to buy it right then, Joe will take the
"superior" CD and may even also perceive you to be a liar.

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[flint]

> You and I might understand this but Joe Sixpack doesn't - and in my
> little MP3-makin' world - Joe Sixpack, unfortunately, is *everywhere*,
> whether he's welcome there or not...
>

WE cannot change the world all at once.

I only respond to these to help YOU to understand your misconceptions of
audio levels. I don't care about Joe Sixpack. Not only would he not care
about this, but he wouldn't be running Linux on his PC.

YOU claimed the MFSL CD was inferior, so I wanted to correct your
understanding of why you felt this way.

YOU claimed to have been lied to by MFSL, so I wanted to correct your
perception.

However, your goal is to make a mix tape that doesn't jump around in
loudness. To accomplish this goal you have found a perfectly acceptable
solution. Good for you.

I was bothered by the effect YOUR comments would have on any Joe Sixpacks
that might be reading this.

This is not a case of anyone trying to rip us off with crappy sounding
recordings. This is a case of the preferences of the mastering houses when
they set levels for CDs over time. Today's music sounds louder than music
from the past. There are two reasons: 1. Switching from using the noise
floor as a reference for CDs to using the highest dynamic peak as a
reference. 2. Compressing the crap out of music. The dynamic range of modern
pop/rock music is so limited, they could use a 10 bit (or smaller) digital
signal to capture all of it. ARGH!!!


- FLINT

[Bob Cain]

Bob Cain wrote:
>
> Lord Hasenpfeffer wrote:
> >
> >
> > This is my whole point, my argument, my *beef* with the entire system
> > that for so many years had me believing that if I bought a CD I was
> > getting "the best there was to get" - which was a complete and outright
> > *lie*.
>
> Myke, I'm afraid you've completely lost me. I have no idea
> what you want that a volume control doesn't equate to. I
> can't imagine what you think might be missing. Best and
> truth have nothing whatsoever to do with level in any
> universe I've visited recently.

Scratch that. Based on your last couple of posts I do
understand that you are just trying to make the levels of
your tracks the same.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[flint]

"Lord Hasenpfeffer" > wrote in message
...
> > Along the way, the industry decided that using the noise floor as the
> > reference for the loudness of a recording was not the best idea.
>
> Why not? Were others besides me calling them too often on the phone to
> voice their complaints? :)

Why not? When CDs hit the market, there was a fear that they would be too
loud and damage speakers and equipment (remember the Telarc 1812 Overture
with "digital canons'?). So, they chose to go with lower peaks and not lose
any sound in the noise floor. Later, someone decided that philisophy was
bunk since everyone had better speakers and electronics now. Once a few CDs
came out with peaks at 0dBFS, every was forced to follow suit or sound
quieter. This was also the same period when they started re-issuing older
CDs as remastered.

>
> > Also, I recently purchased some "remastered" CDs where they have not
> > attempted to replicate the original sound the artists and producers got.
> > Instead they have used different effect processors and added serious
> > compression to make the classic music sound similar to the current
trendy
> > crap.
>
> Namely? I'm interested in knowing if I too have some of these
> "remasters" so that I might be better able to conduct a few more Joe Keg
> sound experiments.

I won't say which CDs, as I do not want to taint anybody's enjoyment of the
music.

> Kinda makes you wonder now if all the "collector's edition vinyl" that's
> still being pressed these days are all mastered "with emphasis" from
> masters originally intended for making CDs. :)

Modern LPs are a niche market and the master tapes are very carefully
produced specifically for the LP transfer. With all the signal processing
available today, thee LPs sound amazingly better than the old ones.

- FLINT

[Lord Hasenpfeffer]

flint wrote:

> I only respond to these to help YOU to understand your misconceptions of
> audio levels. I don't care about Joe Sixpack.

Yeah, but if you happened to find yourself *working* for him sometime
you just might. (And I'm not gonna say anything else about that! *LOL*)

> Not only would he not care about this, but he wouldn't be running Linux
> on his PC.

I'm not gonna say anything else about *that* either! :-DDD
(You're starting to hit just a little too close to home there, flint!
It's makin' me nervous!)

> YOU claimed the MFSL CD was inferior, so I wanted to correct your
> understanding of why you felt this way.

And you have - quite handily, I might add. However, the music industry
obviously considered it to be bad practice to use the noise floor as a
peak reference point too, somewhere down the line. Otherwise they
wouldn't have changed it. And that's not in defense of clipping for the
sake of loudness either. I just don't see anything wrong with taking
more willful advantage of the larger range for amplitudes that CDs make
available. If it's really only a matter of turning it up or turning it
down, and you're not doing anything that harms the original dynamics,
and the range is right there waiting for you to use it - I say, by all
means, do.

> YOU claimed to have been lied to by MFSL, so I wanted to correct your
> perception.

Well, yeah, I guess they did deliver on their promise to provide me with
an approximately "ORIGINAL MASTER RECORDING", so to speak - but even
with the noise floor being used as their peak reference level, they
still had plenty of opportunity to make better use of all that available
range - and didn't for no apparent reason.

Could they have still been "excuseable" from an industrial perspective
if they'd used an even quieter signal which ultimately forced me to have
to turn it up to damn near eleven?

> However, your goal is to make a mix tape that doesn't jump around in
> loudness.

Actually, my goal is to make a mix *library* of over 2,100 full-length
recordings - but the principle remains relatively unchanged.

By doing this the way I am, I *should* in the future be able to grab
just about any of the MP3s I've made since discovering "normalize" and
play them in any "mix" I want without ever having to ride the pump.

> To accomplish this goal you have found a perfectly acceptable
> solution. Good for you.

I hope it could be good for a lot more people than just me, though, too.

Things really do sound better to me now. :)

> I was bothered by the effect YOUR comments would have on any Joe Sixpacks
> that might be reading this.

I can see a little bit of sense in your being concerned about that, yes.

> This is not a case of anyone trying to rip us off with crappy sounding
> recordings. This is a case of the preferences of the mastering houses when
> they set levels for CDs over time.

However, you still haven't attempted to explain to me *why* the change
occurred ... and until you or somebody else does that, I'm left prone to
believing that enough people in the industry finally started to realize
that what they were doing sucked eggs (i.e. a lot of helpless people
were getting ripped off!) *LOL* :-D

This is, mind you, the exact same reason why I stopped only ripping and
encoding - and started normalizing in between the two as well! I've
still got too many of those older 2001/2002-vintage MP3s hangin' around
my hard drive and damn near every single one of them *sucks eggs*. :-)

> Today's music sounds louder than music from the past.

Not after I get ahold of it.

> The dynamic range of modern pop/rock music is so limited, they could
> use a 10 bit (or smaller) digital signal to capture all of it. ARGH!!!

Yes, I'm sure you're right, but then one day Joe Sixpack would surely
get wind of it and think it was defective.

Myke

--

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[Lord Hasenpfeffer]

Bob Cain wrote:
>
>> This is my whole point, my argument, my *beef* with the entire
>> system that for so many years had me believing that if I bought a
>> CD I was getting "the best there was to get" - which was a complete
>> and outright *lie*.
>
> Myke, I'm afraid you've completely lost me. I have no idea what you
> want that a volume control doesn't equate to. I can't imagine what
> you think might be missing. Best and truth have nothing whatsoever
> to do with level in any universe I've visited recently.

Do you think perhaps I may have been already feeling a bit "too ripped
off" by the RIAA after 23 years of constant bombardment with a line of
Pink Floyd LPs, then Pink Floyd CDs, then Pink Floyd remasters, then
Pink Floyd 24-bit remasters, and now next I'm sure it'll be Pink Floyd
DVD-As??? Y'know? It's like, c'mon. Honestly, when you bought that
133MHz Pentium processor PC, did you really believe that there wasn't
already a fully developed 500MHz Pentium III processor "waitin' in the
wings" somewhere just being held back from release until everybody'd
been ripped off by being made to buy cheap-ass 350s??? Please.

You don't think all this skepticism in me just suddenly came down with
yesterday's rain, now do ya? :)

Myke

--

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[flint]

"Lord Hasenpfeffer" > wrote in message
...
> .... I just don't see anything wrong with taking
> more willful advantage of the larger range for amplitudes that CDs make
> available. If it's really only a matter of turning it up or turning it
> down, and you're not doing anything that harms the original dynamics,
> and the range is right there waiting for you to use it - I say, by all
> means, do.

You are not taking advantage of anything. You are not improving the sound.
All you are doing it turning up the volume. If you weren't also compressing
these songs as MP3s, I would deride you fro adding distortion in the proces
of "normalizing" the signal. ALL DIGITAL PROCESSING IS IN NATURE
DESTRUCTIVE.

> Well, yeah, I guess they did deliver on their promise to provide me with
> an approximately "ORIGINAL MASTER RECORDING", so to speak - but even
> with the noise floor being used as their peak reference level, they
> still had plenty of opportunity to make better use of all that available
> range - and didn't for no apparent reason.

They made full use of the dynamic range. There is nothing missing from their
CD. Every single sound that was on the original master tapes is on that CD.
All the quiet stuff and all the loud stuff. It was not compressed or
limited. It was not hindered in any way while producing the CD. They only
chose not to push the peaks to 0dBFS.

>
> Could they have still been "excuseable" from an industrial perspective
> if they'd used an even quieter signal which ultimately forced me to have
> to turn it up to damn near eleven?

It doesn't matter how high you turn up your amp. If you like listening to
music at a certain level, then turn it up until the music is as loud as you
want it. There will be no more distortion than with the later release of the
CD.

> I hope it could be good for a lot more people than just me, though, too.

You are adjusting the levels of MP3s so they are similar. If your goal was
to discuss this, then your subject line should have been "Normalizing audio
for consistent loudness". That is a different discussion and wouldn't have
gotten so many people ****ed off. This isn't about what is "better" or
"superior." This is about your preference that the music, whether new or
old, have a similar level when you make a mix collection.

- FLINT

[Lord Hasenpfeffer]

flint wrote:
>
> When CDs hit the market, there was a fear that they would be too
> loud and damage speakers and equipment

Ok, ok. So when the CD was first rushed to market (the high-end market)
the homework still hadn't been handed in to be graded. I'll buy that.

> (remember the Telarc 1812 Overture with "digital canons'?).

Yes, I do remember that, as a matter of fact. Unfortunately, I never
got to hear it. What were the final results? Did anyone ever report in
as having destroyed their Polks with one of those things? I never found
out about it one way or the other. But, man, were those the days!

> Later, someone decided that philisophy was bunk since everyone had
> better speakers and electronics now.

Except all the $50G tube-ampers of course. :)

> Once a few CDs came out with peaks at 0dBFS, every was forced to
> follow suit or sound quieter. This was also the same period when
> they started re-issuing older CDs as remastered.

Lemme guess, this was sometime around 1993/1994, no?

> I won't say which CDs, as I do not want to taint anybody's enjoyment
of the
> music.

Oh, c'mon. You can tell *me*! :)

I'll Pink Floyd just lost their shirts 'cuz of me.

> Modern LPs are a niche market and the master tapes are very carefully
> produced specifically for the LP transfer.

Well, that's good to know because I actually buy a few of those every
now and then - if I like the group well enough.

> With all the signal processing available today,
> thee LPs sound amazingly better than the old ones.

Hmmm... I personally wouldn't know about that because, while I *open*
them to look at all the pretty pictures in side, I never play them -
'cuz I also buy the CD version as well.

Thanks,

Myke

--

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[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message

>
> Y'know sump'm, I think I've finally identified a major "missing link" in
> our discussion here.

No, you haven't 'identified it'; it is extremely well-known.

MP3 is 'brute force' merely by virtue of it's encoding rate being
user-selectable, almost universally to highly detrimental values - exactly
those that excite you so much by their 'small file-size'.

However, I guess you'll manage to turn it around and totally contradict
yourself yet again, ending up claiming it is actually a Good Thing,
especially if put through a particular command line application in your OS
of choice.

geoff

[Lord Hasenpfeffer]

Troll wrote:
> "Lord Hasenpfeffer" > wrote in message
>
>>Y'know sump'm, I think I've finally identified a major "missing link" in
>>our discussion here.
>
> No, you haven't 'identified it'; it is extremely well-known.

But it hasn't been brought up yet in our discussion.

Low amplitudes are certainly something to be avoided when recording to
MiniDiscs because they'll undoubtedly cause the ATRAC compression
filters to remove the weakest, most susceptible frequencies that are
present in the soundsource.

> MP3 is 'brute force' merely by virtue of it's encoding rate being
> user-selectable, almost universally to highly detrimental values -
> exactly those that excite you so much by their 'small file-size'.

For general listening purposes, 192KBps and even 128KBps MP3s are well
beyond adequate.

And by way of your deliberate misinterpretation of my use of the term,
"brute force", it is clear that you have depleted your potential for
injecting meaningful contributions into this thread.

> However, I guess you'll manage to turn it around and totally contradict
> yourself yet again, ending up claiming it is actually a Good Thing,
> especially if put through a particular command line application in your OS
> of choice.

Well, my normalized MP3s do unquestionably sound better than those which
are not. I listen to them all the time. When they play in random
shuffle mode, it's patently obvious which ones have and which ones have
not been normalized.

It seems to me that if the older method of measuring peaks vs. the newer
method of measuring peaks is real, what sense does it make to create
collections of MP3s from CDs which hail from both eras? Tis best to
normalize the old and leave the new one alone for a superior balance lf
loudnesses across-the-board.

Of course, you still don't believe certain frequencies can become too
weak to be heard at lower amplitudes while others remain less affected.

Well, I just conducted a test. I put on my Capitol 1994 Remastered CD
of Pink Floyd, "Dark Side Of The Moon" and turned the volume knob all
the way down - and son of a gun, I couldn't hear *any* of the
frequencies that are recorded on that disc! Although thanks to you I
wasn't fooled by this. I knew beyond all doubt that even though I
couldn't hear them, those frequencies were still on that gold disc -
safe and sound.

Myke


--

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Windows...It's rebootylicious!!!
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[Stewart Pinkerton]

On Fri, 27 Jun 2003 15:59:08 -0500, Lord Hasenpfeffer
> wrote:

>I have no disagreement with you here about the nature of the music
>itself. My initial argument was that "normalize" has enabled me to
>frequently produce 128KBps MP3s which sound better to my ears and brain
>than do the original source CDs from which the original WAVs were
>ripped. That in and of itself is the point I'm *trying* to prove by all
>of this. Everyone keeps telling me I'm full of **** when I say this and
>I know that I am not!

No, everyone keeps telling you that you're full of **** because you
claim to have 'whopped the ass of MFSL', when all you have done is
shove the signal through a meatgrinder that makes it sound 'better' to
*you*. Not 'better' in any absolute sense, just the way *you* like it.


> > The point is you are changing the artists work.
>
>No, I'm changing the work of Mobile Fidelity Sound Lab - which I
>personally believe should *not* be possible. I paid them in exchange
>for something I should not be able to improve on my own - yet that is in
>no way what I received - by any stretch of the imagination.

HTF do you know? Do you have access to the original master tapes? Do
you understand what MFSL was set up to *do*?

>And my experience is such that I have learned that a *majority* of the
>other standard, commercial compact discs within my personal collection
>are mastered just as horribly as my MFSL Ultradisc II version of "Dark
>Side". And for this I am *thankful* to have "normalize" on my side. It
>has truly proven to be a "magic bullet" in my arsenal for improving the
>sound of my CDs before I encode them to MP3.

If that's what floats *your* boat, then fine. Just don't give us all
this crap about how you have produced a 'superior' sound with your
scrunched and squeezed multi-processed MP3..................

>They sold to me (and obviously an untold number of other people) a
>****ty WAV on a gold-plated disc at a very high price. Have you a
>better term for this than I?

The closest possible approach to the original master tape.........

>Y'know, I'd really like to believe you but at this point without no
>reasonable explanation from someone who was actually on the MFSL staff
>at the time this disc was produced, I simply cannot. Their CD sounds
>like mud. After "normalizing" it to -10dBFS, however, I have made it
>"come alive" on my desktop.

Um, I'm not sure that produce something that sounds good and loud on
desktop speakers was entirely the intention of the MFSL staff......
--

Stewart Pinkerton | Music is Art - Audio is Engineering

[Martin Tillman]

On Sat, 28 Jun 2003 04:59:13 GMT, flint wrote:

> The many of the original CDs were produced from the master tapes used to
> make LPs. In fact, there is a design in the CD format called de-emphasis,
> which removes the RIAA curve used for making LPs from the CD signal so CD
> producers could save time getting CDs out. If you ever see the "EMP"
> indicator on your CD display, you are playing a CD made from a tape mastered
> for LP.

OK, just to take this OT even more...

Emphasis in relation to CD refers to HF lift applied to the source
material when mastering to CD, which is then removed on replay, in an
attempt to increase the perceived noiselessness of the system. A silly
idea that as far as I'm aware was never used - that is, there are no
commercial CDs that use emphasis. OK, bold statement, but I bet there
aren't many, and none are from the last 15 years or so.

This emphasis was not intended to be applied to the master. In the same
way, RIAA EQ was never applied to the master ('Master' meaning the tape
that left the mixing room as being the final product). Have you seen
the RIAA curve? It is so severe that is would be unlistenable to, so
any idea that removing this EQ in order to master to Cd is wrong.

However, ignoring what kit the end product is to be used on when mixing
and mastering is silly, so recordings made when 33rpm vinyl was king
took into account the limitations of the medium, in the same way as
mixing for the cinema, mixing for TV (which is what I do), mixing for CD
and mixing for shellac, all take into account the end conditions. So,
the original mix for Dark Side of The Moon would sound different to a
remix done today by the same people, probably. That's OK by me, but
bunging it through some 'normaliser' (which, as evidenced by the
screenshots, also buggers the dynamic range (so it isn't actually a
'normaliser anyway, by definition) ), is absolutely not on.

[Martin Tillman]

On Fri, 27 Jun 2003 17:06:39 -0500, Lord Hasenpfeffer wrote:

> Some yesterday were telling me that by using "normalize" to
> boost the amplitude of my original WAV that I was reducing the original
> dynamic range - which I've now proven is obviously not the case.

Eh? You blind too?

Not only have you reduced the dynamic range (which normalisation doesn't
do, so you are not normalising), in some places you have actually
INVERTED the dynamic range!!!

To give you a clue(!), look at the peaks around 11 minutes and 24.5
minutes. In the original the peaks are higher at 24.5 than at 11.
You've made the peaks at 11 higher than at 24.5!

[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message

> Low amplitudes are certainly something to be avoided when recording to
> MiniDiscs because they'll undoubtedly cause the ATRAC compression
> filters to remove the weakest, most susceptible frequencies that are
> present in the soundsource.

And moreso with MP3, which you delight in listening to extensively.

> > MP3 is 'brute force' merely by virtue of it's encoding rate being
> > user-selectable, almost universally to highly detrimental values -
> > exactly those that excite you so much by their 'small file-size'.
>
> For general listening purposes, 192KBps and even 128KBps MP3s are well
> beyond adequate.

128 is defintie insufficient. 192 is seldom-used, 160 more common, and much
better than 128 though still audibly inferior to uncompressed (datawise).

> And by way of your deliberate misinterpretation of my use of the term,
> "brute force", it is clear that you have depleted your potential for
> injecting meaningful contributions into this thread.

As you like...

> > However, I guess you'll manage to turn it around and totally contradict
> > yourself yet again, ending up claiming it is actually a Good Thing,
> > especially if put through a particular command line application in your
OS
> > of choice.
>
> Well, my normalized MP3s do unquestionably sound better than those which
> are not. I listen to them all the time. When they play in random
> shuffle mode, it's patently obvious which ones have and which ones have
> not been normalized.

'Better' to you being 'louder'. Although barely perceptably.

> It seems to me that if the older method of measuring peaks vs. the newer
> method of measuring peaks is real,

What new and old methods of measuring peaks ? There has always been one
consistent method.

> Well, I just conducted a test. I put on my Capitol 1994 Remastered CD
> of Pink Floyd, "Dark Side Of The Moon" and turned the volume knob all
> the way down - and son of a gun, I couldn't hear *any* of the
> frequencies that are recorded on that disc!

I have little confidence in your abiity to hear any subtleties at all, let
alone identify or describe them. Describing your playback chain might help.

geoff

[Geoff Wood]

"Martin Tillman" > wrote in message


> and mixing for shellac, all take into account the end conditions. So,
> the original mix for Dark Side of The Moon would sound different to a
> remix done today by the same people, probably.

But none of these variations are remixes (apart from the 5:1 toys). We are
talking about different *masters* of the same mix. Some EQed, some maybe
'restored', some (such as MFSL ?) left pure but treated scrupulously, as
with kid-gloves.

geoff

[Martin Tillman]

On Sun, 29 Jun 2003 00:26:52 +1200, Geoff Wood wrote:

>> and mixing for shellac, all take into account the end conditions. So,
>> the original mix for Dark Side of The Moon would sound different to a
>> remix done today by the same people, probably.
>
> But none of these variations are remixes (apart from the 5:1 toys). We are
> talking about different *masters* of the same mix. Some EQed, some maybe
> 'restored', some (such as MFSL ?) left pure but treated scrupulously, as
> with kid-gloves.

Yes, I'm fully aware of that.

To make it crystal clear, my point is that I'm fully prepared to accept
DSOTM sounding different to the original master IF it is REMIXED by Alan
Parsons and/or Pink Floyd. Anything else is a barstardisation.

[Bob Cain]

Lord Hasenpfeffer wrote:
>
> Yes, I do remember that, as a matter of fact. Unfortunately, I never
> got to hear it. What were the final results? Did anyone ever report in
> as having destroyed their Polks with one of those things? I never found
> out about it one way or the other. But, man, were those the days!

Almost twenty years ago I blew out one of the tweeters on my
Warfdale's with Flim And The BB's "Tricycle" CD, one of the
earliest. I had no idea what dynamic range meant until
then. :-)


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Lord Hasenpfeffer]

Stewart Pinkerton wrote:

> You clearly don't understand what's happening here. Pumping up the
> level on CD will have *no* effect on masking,

You clearly don't understand that I've never even *hinted* that it
would. My focus in this regard is in relation to preventing problems
associated with the Absolute Threshold of Hearing (which, incidentally,
sounds to me as if it could be the title of a Pink Floyd bootleg).

> and excess levels will clip, whether on CD or MD.

But there are no excess levels which clip in my normalized "remaster".
This is clearly evident in the screenshot which you didn't see.

> Data compression is used on MD because it
> *has* to be, not because it's a good idea.

Given the actual purpose for the existence of MD in relation to those
who use MD, it is a good idea. Your perception of the purpose and
usefulness of that format is obviously restricted within the confines of
a very small box. MiniDisc isn't just for music. And audiophiles have
for years been known to get snooty with regard to certain *cassette
tapes* which are far worse than MiniDiscs in *all* respects.

> Why are you using MDs as a source in the first place?
> CDs are fundamentally superior.

You're thinking strictly in academic terms here; certainly not practical
ones.

> ATRAC is widely regarded as superior,

That's nice to know because my extended experience with MiniDisc
recording and my "gut instinct" both seem to confirm this.
Unfortunately, it is also my experience with encoding my own MP3s from
my own CDs that have me believing that "louder is better".

> but they both work in the same basic way.

Are you sure of this? In terms of psychoacoustic filtration of sounds
which are deemed "too weak to be heard" are they the same?

Myke

--

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[Lord Hasenpfeffer]

Martin Tillman wrote:

> However, ignoring what kit the end product is to be used on when
> mixing and mastering is silly,so recordings made when 33rpm vinyl was
> king took into account the limitations of the medium, in the same way
> as mixing for the cinema, mixing for TV (which is what I do), mixing
> for CD and mixing for shellac, all take into account the end
> conditions.

Just as I am attempting to take into account my own "end conditions" by
normalizing an original MFSL wav ... which, btw, I will agree is
probably just fine for being heard straight from the CD with the volume
cranked up. I do believe that so many years ago when I first listened
to the 1994 Capitol remaster which more closely resembles my own
"normalized MFSL" WAV, I pulled a Joe Sixpack and compared both discs at
the same volume setting in an effort to eliminate that variable while
attempting to compare the fidelity of each disc in relation to the
others. Naturally, the Capitol disc "won" and the MFSL CD was relegated
to the dungeon.

> So, the original mix for Dark Side of The Moon would sound different
> to a remix done today by the same people, probably. That's OK by me,
> but bunging it through some 'normaliser' (which, as evidenced by the
> screenshots, also buggers the dynamic range (so it isn't actually a
> 'normaliser anyway, by definition) ), is absolutely not on.

Yes, I do believe it's quite clear now that the li'l Linux app named
"normalize" can be made to do more than just normalize in a pure sense -
which, btw, is not to imply that it *always* does more either.

Myke

--

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[Lord Hasenpfeffer]

Martin Tillman wrote:

> Eh? You blind too?

Yeah, I'm blind.

> Not only have you reduced the dynamic range (which normalisation doesn't
> do, so you are not normalising), in some places you have actually
> INVERTED the dynamic range!!!
>
> To give you a clue(!), look at the peaks around 11 minutes and 24.5
> minutes. In the original the peaks are higher at 24.5 than at 11.
> You've made the peaks at 11 higher than at 24.5!

*This* is actually more along the lines of the form of analysis I was
hoping to garner by way of posting the screenshots. Thanks.

I believe I see what you're seeing, however, given the obviously compact
visual nature of that screenshot, every peak you see in the image is not
necessarily being rendered *exactly* as it in reality may be.

Such comparisons will certainly require a more exploded view for more
accurate analysis.

My purpose in posting that particular screenshot was to illustrate the
bogus nature of the charges being hurled against me for introducing
clipping, limiting, compression and reduced dynamic range after having
merely adjusted its amplitude +4.5dB (which brings its loudness more in
line with that of Capitol's 1994 remastered edition.)

You can see my full-width screenshot of Capitol's remastered waveform
here if you still think it might be worth a look as far as comparing
their botch-job to mine:

http://www.mykec.com/mykec/images/2003-06-27_PF_DSOTM_Capitol_1994_Remaster.png

Myke

--

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-================================-

[Lord Hasenpfeffer]

Geoff Wood wrote:
> "Lord Hasenpfeffer" > wrote in message
>
>> Low amplitudes are certainly something to be avoided when recording to
>> MiniDiscs because they'll undoubtedly cause the ATRAC compression
>> filters to remove the weakest, most susceptible frequencies that are
>> present in the soundsource.
>
> And moreso with MP3,

Are you just saying that? Or do you actually *know* that MP3
compression and ATRAC compression are both the same with regard to their
penchant for discarding underpumped freqs? I admit that while I am
fairly familiar with the processes MiniDisc recorders employ to reduce
the amount of data being stored, I am less familiary with MP3's method.

> which you delight in listening to extensively.

I delight in listening to my MP3s when they are the most practical means
available to me for enjoying the music in my CD collection, yes. You
seem to have a problem with that. Remember, I own 2,100+ CDs. Hauling
them with me everywhere I go for the sake of being a sonic snot would
not only be a logistical nightmare, it would be physically impossible.
Now if you're gonna be a troll, take it somewhere else, please.

> 128 is defintie insufficient.

Insufficient to what end?

> 192 is seldom-used,

Pop E. Cock. I encode *all* of my full-album length MP3s at 192KBps.

> 160 more common

Who gives a rip about what's "more common"? I have a monumental task at
hand to be accomplished for *me*, not for the plebes! The last thing I
intend to do is "munge" (as you say) my entire project by employing some
inadequate yet "more common" bitrate. Geez.

> and much better than 128

When I transfer the cassette tapes of the radio I recorded throughout
the 1980s to CD-RW and them rip-n-encode them to MP3, I use 128KBps and
all the clarity and sonic beauty of the hiss from the master tape is
still in there to be fully enjoyed right along with the music.

> though still audibly inferior to uncompressed (datawise).

Agreed, but we're not really discussing that now are we? Nor have we
really ever been. And the only time - as far as I can see - that I've
ever been rightfully put in my place throughout this entire thread is
when I actually got sidetracked and misled into forgetting my original
purpose. My primary goal here is not to discuss what I can do to
produce the best compact discs because that's not what I do. I've
already purchased the commercial CDs and I always respectfully return to
them as my initial sources for material as needed. Nevertheless, a
majority of the compact discs I own were not mastered with subsequent
MP3 encoding practices in mind, hence the gross inadequacy of the nature
of my MFSL Pink Floyd CD in relation to the mission at hand.

>> Well, my normalized MP3s do unquestionably sound better than those which
>> are not. I listen to them all the time. When they play in random
>> shuffle mode, it's patently obvious which ones have and which ones have
>> not been normalized.
>
> 'Better' to you being 'louder'. Although barely perceptably.

'Better' to me being 'louder' because I believe - though I'm not
absolutely certain of it - that fewer of the frequencies were discarded
during the encoding process as a result of their amplitudes having
previously been increased via "normalize".

> What new and old methods of measuring peaks ? There has always been one
> consistent method.

The original practice of measuring appropriately optimum peak levels in
"up from 0dB" fashion vs. "down from 0dBFS" fashion as is apparently
done today. You're obviously not reading every single post in this
thead - and given the gargantuan size of it now, I can't say that I
blame you.

>> Well, I just conducted a test. I put on my Capitol 1994 Remastered CD
>> of Pink Floyd, "Dark Side Of The Moon" and turned the volume knob all
>> the way down - and son of a gun, I couldn't hear *any* of the
>> frequencies that are recorded on that disc!
>
> I have little confidence in your abiity to hear any subtleties at all, let
> alone identify or describe them. Describing your playback chain might help.

That, Geoff, was a *real joke*. Remember how you said "some people have
a sense of humour" after I balked at your move to label me a "Liniot" in
"humourously" derogatory fashion?

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Bob Cain]

"George W." wrote:
>
> Since you're still crossposting why not add rec.audio.pro to the list
> and see what they think?
>
> I like watching a good train wreck.....

YES! ROTFLOL! On second thought, my tolerance for gore may
not be up to the spectacle. :-)


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message

> I delight in listening to my MP3s when they are the most practical means
> available to me for enjoying the music in my CD collection, yes. You
> seem to have a problem with that. Remember, I own 2,100+ CDs. Hauling
> them with me everywhere I go for the sake of being a sonic snot would
> not only be a logistical nightmare, it would be physically impossible.
> Now if you're gonna be a troll, take it somewhere else, please.

Would not a reasonable approach be selecting a wallet of , say, 12 CDs to
take with you for the day. Or do require instant access to 2100 x
(average)12 songs ?


geoff

[Lord Hasenpfeffer]

Martin Tillman wrote:

> To make it crystal clear, my point is that I'm fully prepared to accept
> DSOTM sounding different to the original master IF it is REMIXED by Alan
> Parsons and/or Pink Floyd. Anything else is a barstardisation.

I understand your thinking for the past and the present but it is
totally impractical for the future.

Is George Martin's attempt at adding orchestration to John Lennon's
original cassette demo of "Grow Old With Me" a *******isation or a
labour of love?

Myke

--

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-================================-

[Lord Hasenpfeffer]

Bob Cain wrote:

> Almost twenty years ago I blew out one of the tweeters on my
> Warfdale's with Flim And The BB's "Tricycle" CD, one of the
> earliest. I had no idea what dynamic range meant until
> then. :-)

Wow!

Almost fifteen years ago I melted a crossover with an amp more powerful
than the one I should have been using at the time. I had no idea what
exponential wattage meant until them. :-)

--

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Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

George W. wrote:
> Since you're still crossposting why not add rec.audio.pro to the list
> and see what they think?
>
> I like watching a good train wreck.....

Hey, that sounds like a great place to go for some really *serious* talk
about how best to prep a WAV for MD / MP3! I bet there's a *lot* of
"professional MP3 encoders" over there! Thanks for the tip! :)

Myke

--

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[Lord Hasenpfeffer]

Geoff Wood wrote:

> Would not a reasonable approach be selecting a wallet of , say, 12 CDs to
> take with you for the day.

What are you, my mother?

> Or do require instant access to 2100 x (average)12 songs ?

More is better than nothing.
Nothing is better than more.

Myke

--

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Windows...It's rebootylicious!!!
-================================-

[Martin Tillman]

On Sat, 28 Jun 2003 15:41:32 -0500, Lord Hasenpfeffer wrote:

>> To give you a clue(!), look at the peaks around 11 minutes and 24.5
>> minutes. In the original the peaks are higher at 24.5 than at 11.
>> You've made the peaks at 11 higher than at 24.5!
>
> *This* is actually more along the lines of the form of analysis I was
> hoping to garner by way of posting the screenshots. Thanks.
>
> I believe I see what you're seeing, however, given the obviously compact
> visual nature of that screenshot, every peak you see in the image is not
> necessarily being rendered *exactly* as it in reality may be.

The resolution of the screenshot is entirely adequate to clearly
demonstrate what I wrote. You've destroyed the dynamic range.

> Such comparisons will certainly require a more exploded view for more
> accurate analysis.

No, they won't.

>
> My purpose in posting that particular screenshot was to illustrate the
> bogus nature of the charges being hurled against me for introducing
> clipping, limiting, compression and reduced dynamic range after having
> merely adjusted its amplitude +4.5dB (which brings its loudness more in
> line with that of Capitol's 1994 remastered edition.)

Well, you've most certainly shot yourself in the foot - big time!

[Lord Hasenpfeffer]

Martin Tillman wrote:

> The resolution of the screenshot is entirely adequate to clearly
> demonstrate what I wrote. You've destroyed the dynamic range.

Compare the dynamic range of my WAV to that of Capitol's 1994 digitally
remastered edition and then tell me I've "destroyed the dynamic range".
If I have then they have too. Better write 'em a nasty letter right
now and 'em about it. Better alert the entire Pink Floyd fanbase too
while you're at it. Capitol's destroyed the dynamic range!!!

>> Such comparisons will certainly require a more exploded view for more
>> accurate analysis.
>
> No, they won't.

Oh, OK, fine. I was going to provide you with one but since you're so
certain it'll do no good, I won't bother.

>> My purpose in posting that particular screenshot was to illustrate the
>> bogus nature of the charges being hurled against me for introducing
>> clipping, limiting, compression and reduced dynamic range after having
>> merely adjusted its amplitude +4.5dB (which brings its loudness more in
>> line with that of Capitol's 1994 remastered edition.)
>
> Well, you've most certainly shot yourself in the foot - big time!

Is that right? Well, I challenge you then to demonstrate where the
clipping, limiting, compression and/or reduced dynamic range I've
contributed to it is located. Otherwise, cierre la boca, kapisce?

Myke

--

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Windows...It's rebootylicious!!!
-================================-

[Bob Cain]

Lord Hasenpfeffer wrote:
>
> Bob Cain wrote:
>
> > "George W." wrote:
>
> >> Since you're still crossposting why not add rec.audio.pro to the list
> >> and see what they think?
> >>
> >> I like watching a good train wreck.....
>
> > YES! ROTFLOL! On second thought, my tolerance for gore may
> > not be up to the spectacle. :-)
>
> Laugh it up.

Sorry, Myke, I just know what they do to people who come
armed only with confidant, self congratulatory speculation
(not that the best of them don't do a fair bit of that
themselves :-)

>
> I spent time today browsing around and reading FAQs, etc. to find out
> more information about the psychoacoustic model employed by MP3 and it
> is indeed similar to MiniDisc's ATRAC compression scheme in that it
> removes *not only* masked frequencies but also those frequencies which
> are predetermined to be "too quiet to be heard" by common human ears.

I suggest you run your hypothesis on the effect of absolute
level on the encoding process by the developers of the Lame
MP3 codec. Who knows, they may validate you. I've never
seen that issue addressed and can think of a lot of reasons
why and means by which that would be factored out of the
process, but that is speculation on my part. There is a
mailing list they monitor that you can find out more about
at:

http://minnie.tuhs.org/mailman/listinfo/mp3encoder

See you there, I hope. I've got to admit I sense a desire
in you to learn and understand but I find your technique of
leading with the chin to be a bit mystifying. :-)


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message
>
> While the MFSL CD of Pink Floyd's, "Dark Side Of The Moon" may be just
> fine for playback directly from the CD it is entirely inappropriate for
> being encoded as an MP3 because it amplitudes are too low to drive a
> majority of its frequencies above their Absolute Thresholds of Hearing.


Read the FAQs harder. A gain of 4dB is going to make diddley-squat
difference to the 'Absolute Threasholds of Hearing". Or to your encoding.

The difference (if any) would be totally insignificant compared to the audio
damage inflicted by the mere act of encoding in the first place.

geoff

[Lord Hasenpfeffer]

Bob Cain wrote:
>
> Sorry, Myke, I just know what they do to people who come
> armed only with confidant, self congratulatory speculation
> (not that the best of them don't do a fair bit of that
> themselves :-)

I hear ya. :)

> I suggest you run your hypothesis on the effect of absolute
> level on the encoding process by the developers of the Lame
> MP3 codec. Who knows, they may validate you.

I will take a stab at it at my earliest convenience, believe me.

> I've never seen that issue addressed

Neither have I. Everyone else I know simply rips and encodes to create
their MP3s ... like swiping a brush down the middle of a canvas and
calling it a "mashterpiece". This is how I used to do it too. Then
about 8 months ago, I discovered this "normalize" application for Linux
and thought it'd be worth investigating. Then I noticed an immediate
and stunning improvement in the sounds of my MP3s after using it.

Unfortunately, most everyone here appears to spend most of their time
pondering the delicacies of their high-definition masters and scoffing
at MP3, carte-blanche, instead of perhaps contemplating ways to
incorporate the best of both formats into their lives. This in turn
leads to no healthy communication about such matters which in turn
causes nobody to learn this technique until perhaps somebody like me
comes bumbling along and inadvertently hijacks a thread by mentioning it! :)

> http://minnie.tuhs.org/mailman/listinfo/mp3encoder
>
> See you there, I hope.

Thanks. I need some sleep first, though! :)

> I've got to admit I sense a desire in you to learn
> and understand but I find your technique of leading
> with the chin to be a bit mystifying. :-)

There's a word for it but I don't remember right now what it is.
Oh, yeah I do! It's called "personality". :-D

Myke

P.S. I'm a Taurus.

--

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Windows...It's rebootylicious!!!
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[Lord Hasenpfeffer]

Geoff Wood wrote:
> "Lord Hasenpfeffer" > wrote in message
>
>>While the MFSL CD of Pink Floyd's, "Dark Side Of The Moon" may be just
>>fine for playback directly from the CD it is entirely inappropriate for
>>being encoded as an MP3 because it amplitudes are too low to drive a
>>majority of its frequencies above their Absolute Thresholds of Hearing.
>
> Read the FAQs harder. A gain of 4dB is going to make diddley-squat
> difference to the 'Absolute Threasholds of Hearing". Or to your encoding.

Well, tell ya what. I'll just leave that kind of risky behaviour up to
you. You don't give a damn about either MP3 or MiniDisc recording
quality anyway, so I don't even know why you bothered to respond to me
again. You know what I'm trying to do with my CDs and my MP3s and you
obviously have no more experience with any of my projects than I do with
any of yours, so... Go shave your beard, Trolliot.

What about the Christopher Cross, "Another Page" CD I mentioned early on
in the other thread?

That "older, quieter" disc required a good 13.33dBs of pumpitude before
I was able to do anything reasonable with it. But I guess that's just
fine for you in your "just turn it up a little" CD-only world, isn't it?

Here are the images:

http://www.mykec.com/mykec/images/All_Right_Before.png
http://www.mykec.com/mykec/images/All_Right_After.png

And here are two MP3 samples in a zip archive:

http://www.mykec.com/mykec/audio/All_Right.zip

Eat that!

Myke

--

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Windows...It's rebootylicious!!!
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[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message

> What about the Christopher Cross, "Another Page" CD I mentioned early on
> in the other thread?
>
> That "older, quieter" disc required a good 13.33dBs of pumpitude before
> I was able to do anything reasonable with it. But I guess that's just
> fine for you in your "just turn it up a little" CD-only world, isn't it?
>
> Here are the images:
>
> http://www.mykec.com/mykec/images/All_Right_Before.png
> http://www.mykec.com/mykec/images/All_Right_After.png

That is not the CD - it is one track. That is the level that the producer
has decided it appropriate for that track, sitting well with the other
tracks to make up the whole album.

It is not clear from the graphic whether (presumably for the purpose of a
compilation) you have (conventionally) normalised it to 0dB and avoided
clipping, or have done your '-10 thang', which could well have driven the
peaks into clipping if additional compression was not applied.

geoff

[Lord Hasenpfeffer]

Geoff Wood wrote:

>> Here are the images:
>>
>> http://www.mykec.com/mykec/images/All_Right_Before.png
>> http://www.mykec.com/mykec/images/All_Right_After.png

> That is not the CD - it is one track.

Yes, but entire CD is just as "bad" as that track.

I was working only with that individual track for a separate purpose
than my usual "whole album" purpose at the time that screenshot was made.

> That is the level that the producer has decided it appropriate for
> that track, sitting well with the other tracks to make up the whole
> album.

Yeah, any particular idea *why* the producer might have decided to go
with such a low level? I've ripped a lot of WAVs from a lot of discs in
the past 2 years and that one CD stands alone as having the lowest
average level of them all. And it's not an MFSL disc either so you can
rest assured even when you pump up the volume, it sounds *baaaaaaad*.

Check out the before/after MP3s in the zip file I provided too.
Hear the difference.

> It is not clear from the graphic whether (presumably for the purpose
> of a compilation) you have (conventionally) normalised it to 0dB and
> avoided clipping, or have done your '-10 thang', which could well
> have driven the peaks into clipping if additional compression was not
> applied.

Those are older screenshots which predate our conversation. They were
taken at least a couple if not several months ago. There was no reason
to add all the pretty pictures and words to those images at the time I
placed them online.

And to answer your question, my '-10 thang' was exactly what I did to
create the difference you see between those two images.

Meanwhile, I just ripped-and-encoded the tracks from the digitally
remastered CD version of an album I can't stand; KISS, "Animalize".

Average level for it was -9.45dBFS.

The math says that's +0.55dB *louder* than what *I* would have given it
had I done the '-10 thang' to it so, it's pretty clear to me that what
I'm doing on my own is still fairly consistent with what passes for an
industry standard today ... at least as far average pop/rock pumpitude
is concerned.

Myke

--

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Windows...It's rebootylicious!!!
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[Martin Tillman]

On Sat, 28 Jun 2003 20:27:51 -0500, Lord Hasenpfeffer wrote:

> Martin Tillman wrote:
>
>> The resolution of the screenshot is entirely adequate to clearly
>> demonstrate what I wrote. You've destroyed the dynamic range.
>
> Compare the dynamic range of my WAV to that of Capitol's 1994 digitally
> remastered edition and then tell me I've "destroyed the dynamic range".

You both have, judging by the screenshots, though I can't tell what they
sound like...

> If I have then they have too. Better write 'em a nasty letter right
> now and 'em about it. Better alert the entire Pink Floyd fanbase too
> while you're at it. Capitol's destroyed the dynamic range!!!

They certainly have. I know which one I want to listen too.

Ironic, innit, that we had a crap system called vinyl with little
dynamic range, that was, thankfully, replaced by another system,
digital, with enormous dynamic range - more than you really need under
most circumstances - and what do most of the record producers then foist
on an unsuspecting public? Music so squashed it dies.


>>> My purpose in posting that particular screenshot was to illustrate the
>>> bogus nature of the charges being hurled against me for introducing
>>> clipping, limiting, compression and reduced dynamic range after having
>>> merely adjusted its amplitude +4.5dB (which brings its loudness more in
>>> line with that of Capitol's 1994 remastered edition.)
>>
>> Well, you've most certainly shot yourself in the foot - big time!
>
> Is that right? Well, I challenge you then to demonstrate where the
> clipping, limiting, compression and/or reduced dynamic range I've
> contributed to it is located. Otherwise, cierre la boca, kapisce?

I can't show you any clipping or limiting, but, to repeat for the third
time, 11 minutes and 24.5 minutes.

Just skirting over the largest thread I've ever seen on usenet in nearly
10 years, tape vs. MD, it is patently obvious that your 'normalize' is
perfectly capable of doing more than normalising, it'll limit and clip
too if you tell it to. Obviously, you are entering values into it that
will cause it to do this, because you certainly aren't just normalising.

[Stewart Pinkerton]

On Sat, 28 Jun 2003 06:00:30 -0500, Lord Hasenpfeffer
> wrote:

>> MP3 is 'brute force' merely by virtue of it's encoding rate being
>> user-selectable, almost universally to highly detrimental values -
> > exactly those that excite you so much by their 'small file-size'.
>
>For general listening purposes, 192KBps and even 128KBps MP3s are well
>beyond adequate.

I think we have discovered the fatal flaw right there. If you really
think that 128kB/sec MP3 is 'well beyond adequate', then we can safely
dismiss any further opinions you might have on sound quality.......

>Well, my normalized MP3s do unquestionably sound better than those which
>are not. I listen to them all the time. When they play in random
>shuffle mode, it's patently obvious which ones have and which ones have
>not been normalized.

That doesn't make them better, dude, it just means that *you* prefer
that sound. Heck, there's people out there who actually prefer tube
amps and vinyl!
--

Stewart Pinkerton | Music is Art - Audio is Engineering

[Stewart Pinkerton]

On Sat, 28 Jun 2003 14:28:08 -0500, Lord Hasenpfeffer
> wrote:

>Stewart Pinkerton wrote:
>
>> You clearly don't understand what's happening here. Pumping up the
>> level on CD will have *no* effect on masking,
>
>You clearly don't understand that I've never even *hinted* that it
>would. My focus in this regard is in relation to preventing problems
>associated with the Absolute Threshold of Hearing (which, incidentally,
>sounds to me as if it could be the title of a Pink Floyd bootleg).

Which part of 'no effect on masking' did you fail to understand? The
threshold of hearing is dependent on sound levels, not on closeness to
0dB FS on the CD, IOW just crank up the amplifier a little.

>> and excess levels will clip, whether on CD or MD.
>
>But there are no excess levels which clip in my normalized "remaster".
>This is clearly evident in the screenshot which you didn't see.

Sure I did, and just how do you think you can 'normalise' to a higher
average level *without* modifying the transfer curve, i.e. applying
compansion and/or limiting?

>> Data compression is used on MD because it
>> *has* to be, not because it's a good idea.
>
>Given the actual purpose for the existence of MD in relation to those
>who use MD, it is a good idea. Your perception of the purpose and
>usefulness of that format is obviously restricted within the confines of
>a very small box. MiniDisc isn't just for music. And audiophiles have
>for years been known to get snooty with regard to certain *cassette
>tapes* which are far worse than MiniDiscs in *all* respects.

You clearly don't understand the basic principles I'm talking about
here. Lossy compression is *never* a good idea, it's just *necessary*
if you want to get a certain amount of information into a storage
medium which is too small to hold it.

>> Why are you using MDs as a source in the first place?
> > CDs are fundamentally superior.
>
>You're thinking strictly in academic terms here; certainly not practical
>ones.

There's nothing 'practically' superior about MDs, they're simply
*convenient* .

>> ATRAC is widely regarded as superior,
>
>That's nice to know because my extended experience with MiniDisc
>recording and my "gut instinct" both seem to confirm this.
>Unfortunately, it is also my experience with encoding my own MP3s from
>my own CDs that have me believing that "louder is better".

Believe that if you like, it's not applicable to anyone else.

>> but they both work in the same basic way.
>
>Are you sure of this? In terms of psychoacoustic filtration of sounds
>which are deemed "too weak to be heard" are they the same?

Essentiallythey are - I'd have to do some digging to determine the
details. Indeed, *all* modern codecs are based on the same
psychoacoustic principles. Basically, they have to be!
--

Stewart Pinkerton | Music is Art - Audio is Engineering

[Stewart Pinkerton]

On Sat, 28 Jun 2003 17:12:34 GMT, Martin Tillman
> wrote:

>On Sun, 29 Jun 2003 00:26:52 +1200, Geoff Wood wrote:
>
>>> and mixing for shellac, all take into account the end conditions. So,
>>> the original mix for Dark Side of The Moon would sound different to a
>>> remix done today by the same people, probably.
>>
>> But none of these variations are remixes (apart from the 5:1 toys). We are
>> talking about different *masters* of the same mix. Some EQed, some maybe
>> 'restored', some (such as MFSL ?) left pure but treated scrupulously, as
>> with kid-gloves.
>
>Yes, I'm fully aware of that.
>
>To make it crystal clear, my point is that I'm fully prepared to accept
>DSOTM sounding different to the original master IF it is REMIXED by Alan
>Parsons and/or Pink Floyd. Anything else is a barstardisation.

How about the 5.1 SACD? :-)

--

Stewart Pinkerton | Music is Art - Audio is Engineering

[cyrus the virus]

In article >,
Lord Hasenpfeffer > wrote:

> cyrus the virus wrote:
> >
> > the tools have always been around to squash music into oblivion..
>
> Probably so. But please do not confuse my beloved "normalization" with
> "squashing music into oblivion". I am still reeling from the shock of
> having successfully whooped the ass of Mobile Fidelity Sound Labs. My
> routine practice of normalization does not involve forcing peaks to
> clip. It does not involve destroying dynamic range. It does not
> involve the inadvertent addition of digital distortion to what was
> originally a pristine, professionally-sanctioned waveform. All of what
> you have read from others about my ill-advised use of normalize when
> producing MP3s from WAVs from my CDs has been revealed to be an
> overflowing crock of ****. And I can and will easily prove this in the
> immediately near future.
>

i wasn't commenting on you, but the green day album and just about any
other rock album after that.

the normalization thing depends on the process. from the few times and
few apps i've used that use normalization, it scans the entire track for
the loudest point and raises the volume level so that loudest point is
at zero. there is no sort of clipping/compressing/limiting going on.


> > but somewhere around the '93-'94 era, like you stated,
> > it became "in" to do it.
>
> Alright. That's it! I'm blamin' Bill Gates! ;-D

he is of course to blame for anything gone bad :)

>
> > the sonic impact is there from older cd's still,
> > turn up the volume knob.
>
> This from you is ill-conceived too.

how so? if all you're doing is raising the loudest point in a song to
digital zero, thats basically turning up a volume knob (all else being
equal). but if there is some type of compressing/limiting going on, that
is a different animal.

>
> Stay tuned and I will *prove* to you soon that your assumption is 100%
> totally incorrect.
>

hopefully, this thread is getting a bit tiring.

> We still have a lot to learn,
>
> Myke

--
cyrus

[cyrus the virus]

In article >,
Lord Hasenpfeffer > wrote:

> flint wrote:
> >
> > The average level of the signal on a CD will not affect the "sound" of that
> > signal.
>
> You and I might understand this but Joe Sixpack doesn't - and in my
> little MP3-makin' world - Joe Sixpack, unfortunately, is *everywhere*,
> whether he's welcome there or not...
>
> If Joe were ever to care enough to actually sit down and compare 2
> different versions of the same music on CD, he's more than likely going
> to want to compare them at the same level of volume in order to
> eliminate that variable.
>
> If the two discs are not mastered at the same level of volume and all
> Joe does is simply listen to each of them, side-by-side, one right after
> the other, I guarantee you he's going to pick the louder one and
> consider the quieter one to be "inferior" if not outright "defective".
>
> And if you try to tell him that they're really the same thing and all he
> has to do to make them sound equally well is crank up the volume just a
> little bit more, I guarantee you he'll look you straight in the eye and
> say, "But I shouldn't *have* to crank up the volume if it really is the
> same as that other one over there."
>
> Perception is key.
>
> And if he really wants to buy it right then, Joe will take the
> "superior" CD and may even also perceive you to be a liar.
>
> Myke

and this perception is exactly why the loudness wars even started.
volume and quality is really trivial when the music actually affects
you.

--
cyrus

[Lord Hasenpfeffer]

StArSeEd wrote:

> Yep, the only effects peak-based normalization will have on the playback of a CD are raising its
> volume and raising its effective S/N ratio ('effective' referring to the noise from the analog
> components of the stereo system itself, not from the CD; lower volume on the knob = lower system
> noise, generally speaking). The dynamic range (and, therefore, "punch") will remain the same.

However, I believe it is foolish to not "normalize" a WAV prior to
encoding it to MP3 - or transferring it to MD - because the frequencies
in a less-than-normalized WAV stand much greater chance of being
discarded entirely from the sound by the lossy compression algorithm
employed during the encoding process.

If the amplitude of a given frequency in an unmodified WAV is "too quiet
to be heard" it will be removed by the ATRAC and MP3 lossy compression
schemes. However, if normalizing the WAV causes the amplitude of that
frequency to be boosted enough that it then becomes audible, it will be
retained.

This is why I normalize before I encode. If you're just talking about
boosting amplitude for subsequent playback of the WAV from CD, fine.
There is no advantage to that. However, if you're attempting to boost
the loudness of the WAV so that more of frequencies in the recording are
likely to survive the compression filters, then your resulting louder
MD/MP3 will sound better. I don't understand why this is being so hotly
debated here. It's all pretty obvious to me.

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Martin Tillman]

On Sun, 29 Jun 2003 17:23:44 -0500, Lord Hasenpfeffer wrote:

> StArSeEd wrote:
>
>> Yep, the only effects peak-based normalization will have on the playback of a CD are raising its
>> volume
[snip]
> However, I believe it is foolish to not "normalize" a WAV prior to
> encoding it to MP3 - or transferring it to MD - because the frequencies
> in a less-than-normalized WAV stand much greater chance of being
> discarded entirely from the sound by the lossy compression algorithm
> employed during the encoding process.

Give me strength...

>
> If the amplitude of a given frequency in an unmodified WAV is "too quiet
> to be heard" it will be removed by the ATRAC and MP3 lossy compression
> schemes.

What? You are a troll.

> However, if normalizing the WAV causes the amplitude of that
> frequency to be boosted enough that it then becomes audible, it will be
> retained.

Troll, troll, troll. I've spotted you, hehehehe!

>
> This is why I normalize before I encode. If you're just talking about
> boosting amplitude for subsequent playback of the WAV from CD, fine.
> There is no advantage to that

Hey, you've finally seen the light?

> However, if you're attempting to boost
> the loudness of the WAV so that more of frequencies in the recording are
> likely to survive the compression filters, then your resulting louder
> MD/MP3 will sound better. I don't understand why this is being so hotly
> debated here.

It wasn't.

> It's all pretty obvious to me.

Yeah right.

Can someone please put this troll straight, I haven't got the energy
myself.

Good grief...

[StArSeEd]

Lord Hasenpfeffer:
> StArSeEd wrote:
> > Yep, the only effects peak-based normalization will have on the playback of a CD
> > are raising its volume and raising its effective S/N ratio ('effective' referring
> > to the noise from the analog components of the stereo system itself, not from the
> > CD; lower volume on the knob = lower system noise, generally speaking). The
> > dynamic range (and, therefore, "punch") will remain the same.
> However, I believe it is foolish to not "normalize" a WAV prior to encoding it ...
> If the amplitude of a given frequency in an unmodified WAV is "too quiet to be
> heard" it will be removed by the ATRAC and MP3 lossy compression schemes. ...
> However, if you're attempting to boost the loudness of the WAV so that more of
> frequencies in the recording are likely to survive the compression filters, then
> your resulting louder MD/MP3 will sound better. I don't understand why this is
> being so hotly debated here. ...

It's being hotly debated because you keep saying the same thing, over and over, to every single
post, regardless of whether or not the other posters are talking about premastering waves for
MP3/ATRAC encoding or not. In the case of my response above, I wasn't. Neither was the poster
to whom I replied. You're either skim-reading, thereby missing the important words (in my reply
above, "playback"), or you have a very short memory.

Here's some food for thought: bit allocation. If the 'inaudible' frequencies are
masked/dropped by the encoder, would there not be more bits allocated to those which ARE
audible, leading to a higher overall quality? This is, after all, the principal which led to
the use of psychoacoustic filters and masking and frequency cutoffs in the first place. As a
test of bit allocation vs. frequency response, try feeding LAME the -k (keep all frequencies;
disable filtering) parameter (using a bitrate which will make the differences more obvious, such
as 128k). If the encoder cannot allocate enough bits to the barely-audible frequencies to make
them sound good, the encoder should filter them out and allocate more bits to the REST of the
audio to make IT sound better. This is exactly why BladeEnc MP3s sound like utter **** -
inadequate filtering of the frequencies to which BladeEnc cannot allocate enough bits required
to retain some semblance of their original form. The results? High-frequency
ringing/twinkling/sparkling/whatever you want to call it. Early versions of ATRAC suffered from
a similar phenomenon.

-StArSeEd
--
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IRC EFnet #smashing_pumpkins
Email:
ICQ UIN: 1711589

[StArSeEd]

Martin Tillman:
> Lord Hasenpfeffer:
> > If the amplitude of a given frequency in an unmodified WAV is "too quiet
> > to be heard" it will be removed by the ATRAC and MP3 lossy compression
> > schemes.
> What? You are a troll.

Actually, he is correct.

> > However, if normalizing the WAV causes the amplitude of that
> > frequency to be boosted enough that it then becomes audible, it will be
> > retained.
> Troll, troll, troll. I've spotted you, hehehehe!

Correct again, two for two.

> > This is why I normalize before I encode. If you're just talking about
> > boosting amplitude for subsequent playback of the WAV from CD, fine.
> > There is no advantage to that
> Hey, you've finally seen the light?

The minor S/N ratio boost is an advantage for noisier systems. Two for three.

> > However, if you're attempting to boost the loudness of the WAV so that
> > more of frequencies in the recording are likely to survive the compression
> > filters, then your resulting louder MD/MP3 will sound better.

BZZZT! Incorrect. Two for four.

> > I don't understand why this is being so hotly debated here.
> It wasn't.

Sure it is, it wouldn't have gone on this long if it wasn't.

> > It's all pretty obvious to me.
> Yeah right.

What may seem obvious to him is in reality an imagined manifestation of his failure
to grasp the "bigger picture". Just because more of the source audio is audible in
a resulting MP3 does not mean it will sound "better" - unless the encoder can allocate
enough bits to those previously-missing signals (usually high-freq signals, which, I
can assure you, require many more bits to sound decent than do lower-frequency signals),
it will actually sound worse, and, more often than not, as in the case of BladeEnc,
*much* worse.

> Can someone please put this troll straight, I haven't got the energy myself.

How was that?

> Good grief...

R.I.P. Charles Schulz.

-StArSeEd
--
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Email:
ICQ UIN: 1711589

[Lord Hasenpfeffer]

Martin Tillman wrote:

>> This is why I normalize before I encode. If you're just talking about
>> boosting amplitude for subsequent playback of the WAV from CD, fine.
>> There is no advantage to that
>
> Hey, you've finally seen the light?

It's about ****ing time that you've finally realized that I understand
this and that CD-audio quality after normalization has nothing at all to
do with my hypothesis.

So now that YOU have finally seen the light, quit calling me names,
crawl back under your bridge and stop bothering the goats.

Myke

--

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-================================-

[Lord Hasenpfeffer]

StArSeEd wrote:

> Here's some food for thought: bit allocation. If the 'inaudible' frequencies are
> masked/dropped by the encoder, would there not be more bits allocated to those which ARE
> audible, leading to a higher overall quality? This is, after all, the principal which led to
> the use of psychoacoustic filters and masking and frequency cutoffs in the first place. As a
> test of bit allocation vs. frequency response, try feeding LAME the -k (keep all frequencies;
> disable filtering) parameter (using a bitrate which will make the differences more obvious, such
> as 128k). If the encoder cannot allocate enough bits to the barely-audible frequencies to make
> them sound good, the encoder should filter them out and allocate more bits to the REST of the
> audio to make IT sound better. This is exactly why BladeEnc MP3s sound like utter **** -
> inadequate filtering of the frequencies to which BladeEnc cannot allocate enough bits required
> to retain some semblance of their original form. The results? High-frequency
> ringing/twinkling/sparkling/whatever you want to call it. Early versions of ATRAC suffered from
> a similar phenomenon.

Thank you for moving this conversation forward instead of wallowing
around in the same old mud of "just turn the volume knob".

First off, a lot of people here are making the mistake of assuming that
I use "normalize" to push RMS levels to Full Scale. I am not doing
anything even *close* to that nor would I ever.

Secondly, others seem to be stuck with believing that I am using
"normalize" because I want to create "louder = better" sounding CDs.
That too is absolutely false. I *never* create/burn normalized audio
CDs unless I'm making a "mix" disc - a topic that's only marginally
related to this thread at best.

You are obviously on a "right track" as far as I and this thread are
concerned and I appreciate your "food for thought".

As for everyone else who's still operating under stupid false
assumptions about my hypothesis and, therefore, at their wits end about
what they think I'm saying and doing, just go away because you've
obviously got nothing positive to contribute here!

Myke

--

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[Yosah Akbah Muhammed]

Lord Hasenpfeffer > wrote in message >...
> ...
> Unfortunately, most everyone here appears to spend most of their time
> pondering the delicacies of their high-definition masters and scoffing
> at MP3, carte-blanche, instead of perhaps contemplating ways to
> incorporate the best of both formats into their lives.

Personally I think MP3 is really becoming obselete. It was great back
when hard drives were tiny & expensive, but now they are huge & cheap.
Using a lossy compression on audio would make sense only if you valued
your hard drive space more than quality. But since the hard drive
space issue is virtually non-existant do we really need to compress
audio to the tiniest possible space?

The only reason mp3 is so popular now is not because of the quality it
offers but because the size makes it ideal for bootlegging (this is
considering that bootlegging via mp3 takes place at 192k CBR, while a
320k CBR and even some VBRs are significantly higher quality).

If you do not own a cd-burner or have very limited hard drive space
then mp3 might still be viable for you. Everyone else should check out
loss-less compression or just archive everything as WAV files. I hate
to kill my favorite music by carving it up via mp3 (or insert any
other lossy audio compression scheme here).

This rant courtesy of the repressed child inside whom always kicks me
when I play an mp3m.

[Lord Hasenpfeffer]

Yosah Akbah Muhammed wrote:

> Personally I think MP3 is really becoming obselete.

I do not choose MP3 simply for Playin' Kewl Toonz Dood.
I have professional reasons for using it.

Besides, the principles addressed by my hypothesis are not limited to
MP3 encoding. If my assertions are correct, they are true for all forms
of psychoacoustically-based lossy compression algorithms. I personally
prefer MiniDisc's ATRAC to MP3 but use both for various reasons.

> But since the hard drive space issue is virtually non-existant do
> we really need to compress audio to the tiniest possible space?

I will continue compress until hard drive space is large enough to
accommodate 2,100+ CDs' worth of uncompressed audio data! :)

> If you do not own a cd-burner

I own a cd-burner.

> or have very limited hard drive space

I do not have very limited hard drive space.

> I hate to kill my favorite music by carving it up via mp3

Compact discs cannot be uploaded to websites.
Uncompressed WAVs are impractically large.

Myke

--

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[Bob Cain]

Lord Hasenpfeffer wrote:
>
> Yosah Akbah Muhammed wrote:
>
> > Personally I think MP3 is really becoming obselete.
>
> I do not choose MP3 simply for Playin' Kewl Toonz Dood.
> I have professional reasons for using it.
>
> Besides, the principles addressed by my hypothesis are not limited to
> MP3 encoding. If my assertions are correct, they are true for all forms
> of psychoacoustically-based lossy compression algorithms. I personally
> prefer MiniDisc's ATRAC to MP3 but use both for various reasons.

Are you planning on testing that hypothesis as has been
suggested?


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Pete Carney]

Here's a GIF animation of your All_Right_Before.mp3 and After files. I
think you'll see exactly what is going on with "Normalize".

http://www.c3di.com/images/All_right.gif

>
> Eat that!
>

The first image shows the original at the location of the max peak in the
file. Then the second image is proper normalization in CoolEdit 2000 of the
exact same time frame. Then the third is what the command line program
"Normalize" does to the file.

Definitely a bit of limiting going on there, but only at a couple of places
in the whole file. The perceived volume level of the properly normalized
file and the After "Normalize" file is significant in my opinion.

If better is having all your files RMS at about the same level, then this
file is definitely better. If anyone were to suggest that this "Normalize"
file sounds "better" I would have to call them "foolish" :-)

"Normalize" is not just a Unix command line program. It is available for
most platforms including Windows.

It is definitely doing some limiting where the peaks above 0 dB would clip.
Even the documentation mentions it, but not exactly what it is doing. It
would be intersting to see the exact code, but I'm too lazy to download it
and look at what it is doing.

Cheers,
Pete



"Lord Hasenpfeffer" > wrote in message
...
> > Read the FAQs harder. A gain of 4dB is going to make diddley-squat
> > difference to the 'Absolute Threasholds of Hearing". Or to your
encoding.
>
> Well, tell ya what. I'll just leave that kind of risky behaviour up to
> you. You don't give a damn about either MP3 or MiniDisc recording
> quality anyway, so I don't even know why you bothered to respond to me
> again. You know what I'm trying to do with my CDs and my MP3s and you
> obviously have no more experience with any of my projects than I do with
> any of yours, so... Go shave your beard, Trolliot.
>
> What about the Christopher Cross, "Another Page" CD I mentioned early on
> in the other thread?
>
> That "older, quieter" disc required a good 13.33dBs of pumpitude before
> I was able to do anything reasonable with it. But I guess that's just
> fine for you in your "just turn it up a little" CD-only world, isn't it?
>
> Here are the images:
>
> http://www.mykec.com/mykec/images/All_Right_Before.png
> http://www.mykec.com/mykec/images/All_Right_After.png
>
> And here are two MP3 samples in a zip archive:
>
> http://www.mykec.com/mykec/audio/All_Right.zip
>
> Eat that!
>
> Myke
>

[Bob Cain]

Pete Carney wrote:
>
> Here's a GIF animation of your All_Right_Before.mp3 and After files. I
> think you'll see exactly what is going on with "Normalize".
>
> http://www.c3di.com/images/All_right.gif

Now, that's cool and certainly makes the point. How do you
make a GIF animation?


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message
...
> StArSeEd wrote:
>
> > Yep, the only effects peak-based normalization will have on the playback
of a CD are raising its
> > volume and raising its effective S/N ratio ('effective' referring to the
noise from the analog
> > components of the stereo system itself, not from the CD; lower volume on
the knob = lower system
> > noise, generally speaking). The dynamic range (and, therefore, "punch")
will remain the same.
>
> However, I believe it is foolish to not "normalize" a WAV prior to
> encoding it to MP3

Foolish not to normalise maybe, but defintiely not to "normalise" - your
version which includes applying the compression that you seem to acknowledge
as a Bad Thing while similtaneiously explaining that you prefer it .

geoff

[StArSeEd]

Bob Cain:
> Pete Carney:
> > Here's a GIF animation of your All_Right_Before.mp3 and After files. I
> > think you'll see exactly what is going on with "Normalize".
> > http://www.c3di.com/images/All_right.gif
> Now, that's cool and certainly makes the point. How do you
> make a GIF animation?

Alchemy Mindworks' GIF Construction Set is probably the most popular oldschool GIF animator.. at
least it used to be. Microsoft made one too at some point, though as I recall, it wasn't quite
as full-featured as the former.

http://www.mindworkshop.com/alchemy/gifcon.html

*looks at the page*

Jesus, that program's come a LONG way since.. a long time ago.

-StArSeEd
--
dchub://tsphub.dyndns.org:1979
IRC EFnet #smashing_pumpkins
Email:
ICQ UIN: 1711589

[Lord Hasenpfeffer]

Bob Cain wrote:

> Are you planning on testing that hypothesis as has been
> suggested?

Planning on? Yes.

Done it yet? No.

It all I can do just to keep the discussion on-track and troll-free
right now!

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

Pete Carney wrote:

> The first image shows the original at the location of the max peak in the
> file. Then the second image is proper normalization in CoolEdit 2000 of the
> exact same time frame. Then the third is what the command line program
> "Normalize" does to the file.

With what parameters?
Normalize doesn't just do one thing and nothing else.

> Definitely a bit of limiting going on there

Yes, "normalize" will do that if it must in order to obey the parameters
specified by the user. You can even use "normalize" to reduce the
amplitude of your file rather than boost it if you want.

> but only at a couple of places in the whole file. The perceived volume
> level of the properly normalized file and the After "Normalize"
> file is significant in my opinion.

Significant in what way?

> If better is having all your files RMS at about the same level, then this
> file is definitely better.

Which file? You're working with three files there.

> If anyone were to suggest that this "Normalize" file sounds "better"
> I would have to call them "foolish" :-)

My main concern is to determine whether or not using "normalize" to
boost the amplitudes of low-RMS-level WAVs will cause fewer frequencies
in the recording to be axed on MP3's ATH chopping block as I believe
they would be in the WAV's default state for being "too low to be heard".

> "Normalize" is not just a Unix command line program. It is available for
> most platforms including Windows.

I use "normalize" under Linux. If "normalize" is also available for
Windows, great. So is "Mozilla". So is "The Gimp". So is StarOffice
and OpenOffice. Right on down the line. Makes no difference to me.

> It is definitely doing some limiting where the peaks above 0 dB would clip.

Well, if it *has* to limit the peaks in order to meet a user-specified
demand, great. I'd rather it do that than clip 'em off. Meanwhile, in
my "normalized" WAV of MFSL's "Dark Side Of The Moon", there is no
limiting present at all even at the greatest peak. There is only
textbook normalization there. And for causing that to happen, I'm a
*heathen*. But Capitol's version's OK 'cuz everybody's heard of them.

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

Bob Cain wrote:
>
> Now, that's cool and certainly makes the point. How do you
> make a GIF animation?

Well, y'know, I've already uploaded a screenshot of that maximum peak in
my Pink Floyd MFSL CD at

(1) it's original level
(2) at my +4.5dB personally preferred level and
(3) at a rudely pumped up +10dB level where the limiting actually
becomes visible.

Maybe I should animate those as well.

I just assumed that everybody could look up, look down and then look
straight ahead and see the differences - 'specially since they're lined
up perfectly straight one on top of the other.

Doh!

Myke


--

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[Lord Hasenpfeffer]

Geoff Wood wrote:

> Foolish not to normalise maybe, but defintiely not to "normalise" - your
> version which includes applying the compression that you seem to acknowledge
> as a Bad Thing while similtaneiously explaining that you prefer it .

Duhhh..........................

Doh!

Myke

--

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[Lord Hasenpfeffer]

Geoff Wood wrote:
> "Martin Tillman" > wrote in message >
>
>>What? You are a troll.
>>
>>>However, if normalizing the WAV causes the amplitude of that
>>>frequency to be boosted enough that it then becomes audible, it will be
>>>retained.
>>
>>Troll, troll, troll. I've spotted you, hehehehe!
>
> The real sad thing is that I don't think he is .....

Truth really hurts, don't it?

Myke


--

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Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

StArSeEd wrote:

> Alchemy Mindworks' GIF Construction Set is probably the most popular oldschool GIF animator.. at
> least it used to be. Microsoft made one too at some point, though as I recall, it wasn't quite
> as full-featured as the former.

No, no, no... All you need is The Gimp. It's like having Alchemy and
PhotoShop in one package - and it's free 'cuz it's open source. And
it's available for both Linux and Windows.

http://www.gimp.org

Get it today and never screw with another stupid commercial graphics app
unless your job requires you to have to do so. And you don't even have
to pay a dime to prevent those stupid annoying Shareware windows from
popping up every time you open it up - 'cuz they're never there to start
with... 'cuz it's FREE!

Myke

--

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[Lord Hasenpfeffer]

Geoff Wood wrote:
> "Yosah Akbah Muhammed" > wrote in message
>
> MP3 to 'archiving', is like chopping 10% off all your $ notes and putting
> them in a draw, while throwing the rest away and calling it 'saving'.


Is that a Geoffact or just an opinion?

Myke

--

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[Stewart Pinkerton]

On Mon, 30 Jun 2003 03:22:54 -0500, Lord Hasenpfeffer
> wrote:

>Pete Carney wrote:
>
>> The first image shows the original at the location of the max peak in the
>> file. Then the second image is proper normalization in CoolEdit 2000 of the
>> exact same time frame. Then the third is what the command line program
>> "Normalize" does to the file.
>
>With what parameters?
>Normalize doesn't just do one thing and nothing else.

Weren't you claiming that this was *exactly* what it did with the MFSL
disc? That it just increased *every* sample until the peaks were just
below 0dB FS, with no other effect *whatever*?
--

Stewart Pinkerton | Music is Art - Audio is Engineering

[Martin Tillman]

On Mon, 30 Jun 2003 03:22:54 -0500, Lord Hasenpfeffer wrote:

> Meanwhile, in my "normalized" WAV of MFSL's "Dark Side Of The Moon", there
> is no limiting present at all even at the greatest peak. There is only
> textbook normalization there.

Wrong. Explain your levels at 11 min and 24.5 min with respect to the
original.

> And for causing that to happen, I'm a *heathen*.

Yep. Even moreso because you claimed adamantly that you were not doing
it (distorting the dynamic range).

> But Capitol's version's OK 'cuz everybody's heard of them.

No one has said as far as I recall.

[Martin Tillman]

On Sun, 29 Jun 2003 18:37:05 -0500, Lord Hasenpfeffer wrote:

>
>>> This is why I normalize before I encode. If you're just talking about
>>> boosting amplitude for subsequent playback of the WAV from CD, fine.
>>> There is no advantage to that
>>
>> Hey, you've finally seen the light?
>
> It's about ****ing time that you've finally realized that I understand
> this

You certainly demonstrated that you absolutely did not believe this
earlier. You claimed your normalised DSOTM was better than the
original. Of course, what you did wasn't just normalisation...

> and that CD-audio quality after normalization has nothing at all to
> do with my hypothesis.

Previously...

Message-ID: >

> It amazes me that I can rip a WAV directly from an older commercial CD,
> "normalize" it to 2dB beyond zero (i.e. -10dBFS) and then encode from it
> an MP3 that sounds dramatically better than its own CD source.

Which means, if this is true:

> >>> If you're just talking about
> >>> boosting amplitude for subsequent playback of the WAV from CD, fine.
> >>> There is no advantage to that

meaning that this is the equivalent up twiddling the knob, then you're
saying, given the exact same source, but at different levels, you can
make two dramatically different mp3s!! This is just pure nonsense.

I suggest you restate exactly what you think you are doing and why,
because your story has changed several times over the last few days.

[Martin Tillman]

On Sun, 29 Jun 2003 02:02:09 -0500, Lord Hasenpfeffer wrote:

> What about the Christopher Cross, "Another Page" CD I mentioned early on
> in the other thread?
>
> That "older, quieter" disc required a good 13.33dBs of pumpitude before
> I was able to do anything reasonable with it.

> And here are two MP3 samples in a zip archive:
>
> http://www.mykec.com/mykec/audio/All_Right.zip
>
> Eat that!

Pity we can't see the original, or that you didn't encode to a higher
bitrate. However, despite that, there is very little doubt in my mind
that your 'after' has limiting applied to it in order to make it sound
louder.

Yep, the only difference between them is that one is louder. Muppet.

All I had to do to make them sound the same was to add 2.5dB to the
quieter one (and let it clip in a few places). I could post a
compilation of the two and challenge you to spot the joins...

[Arny Krueger]

"Stewart Pinkerton" > wrote in message


> Which part of 'no effect on masking' did you fail to understand? The
> threshold of hearing is dependent on sound levels, not on closeness to
> 0dB FS on the CD, IOW just crank up the amplifier a little.

Point of order...

The threshold of hearing as usually given *is* a sound level, and therefore
can't possibly be dependent on a sound level.

[KikeG]

(Stewart Pinkerton) wrote in message >...

> ATRAC is widely regarded as superior, but they both work in the same
> basic way.

Despite that, good quality encoders such as Fraunhoffer and LAME are
clearly superior to latest versions of "standard" ATRAC. It's easy to
verify throwing at them some medium and hard-to-encode samples such as
the ones at PCABX pages.

[Stewart Pinkerton]

On 1 Jul 2003 08:32:18 -0700, (KikeG) wrote:

(Stewart Pinkerton) wrote in message >...
>
>> ATRAC is widely regarded as superior, but they both work in the same
>> basic way.
>
>Despite that, good quality encoders such as Fraunhoffer and LAME are
>clearly superior to latest versions of "standard" ATRAC. It's easy to
>verify throwing at them some medium and hard-to-encode samples such as
>the ones at PCABX pages.

AAC and MP3 *are* Fraunhoffer algorithms................

Agreed that AAC is likely the best of the bunch.
--

Stewart Pinkerton | Music is Art - Audio is Engineering

[Lord Hasenpfeffer]

Martin Tillman wrote:

> Previously...
>
> Message-ID: >
>
>> It amazes me that I can rip a WAV directly from an older commercial CD,
>> "normalize" it to 2dB beyond zero (i.e. -10dBFS) and then encode from it
>> an MP3 that sounds dramatically better than its own CD source.

This is true. It does amaze me. Only now I know more about why this is
the case. The original CD source is "unnormalized" in my sense of the
term. The MP3 is made from a normalized WAV. In in general, most
people (including me) tend to believe that louder is better ... because
with loudness comes clarity.

This remains... an MP3 made from an older, quieter, unnormalized WAV
sounds poor to me compared to an MP3 made from a normalized one at the
same level of volume. I frequently listen to my MP3s in random shuffle
mode. Without "normalization", "remastered MP3s" sound are louder and
clearer sounding that "unremastered MP3s". Therefore, it is useful for
me to normalize older WAVs so that all of my MP3s have a nice, even
loudness. If I don't the older MP3s sound like crap in comparison the
newer "remastered" ones.

This has become a very boring conversation.

> meaning that this is the equivalent up twiddling the knob, then you're
> saying, given the exact same source, but at different levels, you can
> make two dramatically different mp3s!! This is just pure nonsense.

Do as you like.

> I suggest you restate exactly what you think you are doing and why,
> because your story has changed several times over the last few days.

My mistake: Providing no well-defined hypothesis in this thread.
My correction: Providing a well-defined hypothesis in my other thread.

So it's been done.

Myke

--

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[Lord Hasenpfeffer]

Arny Krueger wrote:
> "Stewart Pinkerton" > wrote in message
>
>
>
>>Which part of 'no effect on masking' did you fail to understand? The
>>threshold of hearing is dependent on sound levels, not on closeness to
>>0dB FS on the CD, IOW just crank up the amplifier a little.
>
> Point of order...
>
> The threshold of hearing as usually given *is* a sound level, and therefore
> can't possibly be dependent on a sound level.

To clarify:

An ATH chart shows that frequency Y is "inaudible" below amplitude X.

In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y
is discarded and does not become a part of the final MP3.

However, after "normalizing" the WAV by a factor of +3dB, frequency Y's
amplitude now becomes X+1dB, therefore, frequency Y is retained and
becomes a part of the final MP3.

Myke

--

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Windows...It's rebootylicious!!!
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[Dave Platt]

In article >,
Lord Hasenpfeffer > wrote:

>
>This is true. It does amaze me. Only now I know more about why this is
>the case. The original CD source is "unnormalized" in my sense of the
>term. The MP3 is made from a normalized WAV. In in general, most
>people (including me) tend to believe that louder is better ... because
>with loudness comes clarity.
>
>This remains... an MP3 made from an older, quieter, unnormalized WAV
>sounds poor to me compared to an MP3 made from a normalized one at the
>same level of volume. I frequently listen to my MP3s in random shuffle
>mode. Without "normalization", "remastered MP3s" sound are louder and
>clearer sounding that "unremastered MP3s". Therefore, it is useful for
>me to normalize older WAVs so that all of my MP3s have a nice, even
>loudness. If I don't the older MP3s sound like crap in comparison the
>newer "remastered" ones.

The real question, then, is "Where does the enhanced sense of clarity
come from?" Or, conversely, "Why do the MP3 encodings made from
lower-level CDs sound poorer to you, and why?"

I can see at least two possible hypotheses:

Hypothesis 1: the _relative_ amount of frequency content in the two
MP3 versions is identical, to within the inevitable low-level errors
implied by any coding system. Or, in other words, the lower-level
(duller-sounding-to-you) MP3 would sound identical to the higher-level
(clearer-sounding) MP3, if you simply turned up the volume a bit. The
differences in how you perceive the two aren't due to the actual
content of the MP3, only to the playback amplitude.

You might be perceiving the two as substantially different either due
to the well-known Fletcher-Munson effect: your ears' relative
sensitivities to bass and treble fall off faster than their
sensitivity to midrange, as the volume is reduced, and thus quieter
signals tend to sound as if they lack both bass and treble.

Or, it might be a masking and ambient-noise effect. If you typically
listen to MP3s in conditions of high ambient noise (office, car,
outdoors, etc.) then more of the lower-level-MP3 playback would be
drowned out by ambient noise, making it sound less clear.

The cure for this would be simply "Turn up the volume during playback."

Hypothesis 2: the MP3s encoded from lower-amplitude (non-normalized)
WAV inputs are actually, and significantly different than their
higher-amplitude cousins in ways other than just amplitude. Possibly
some frequencies are missing (excessive masking, or sounds below
threshold), possibly some frequency bands are less accurately encoded
and have a higher noise level.

This would suggest that the MP3 encoder you are using is less than
optimal. Possibly it has a poorly-set threshold detector, which is
sensitive only to the absolute signal level and not to the relative
levels. Possibly one of its other encoder or bit-allocation
algorithms is misbehaving, and is dedicating larger portions of the
available bit bandwidth to certain frequencies and is being forced to
discard other bands for lack of sufficient bit-reservoir in the
encoding.

The cure for this would be "Use a better encoder, or better settings
for the one that you have."

--
Dave Platt > AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

[flint]

> An ATH chart shows that frequency Y is "inaudible" below amplitude X.
>
> In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y
> is discarded and does not become a part of the final MP3.
>
> However, after "normalizing" the WAV by a factor of +3dB, frequency Y's
> amplitude now becomes X+1dB, therefore, frequency Y is retained and
> becomes a part of the final MP3.
>
AH.... I think I see the problem with your understanding of frequency
masking (as it was originally called).


The threshold of audibility is not a defacto amplitude. It is the amplitude
of a one frequency as compared to another frequency.

The codec works based on the concept that one loud tone will mask our
perception of another quieter tone which very close in frequency to the
louder tone.

If you have a tone of kHz at -5dB and another tone of 1.001KHz at -20dB,
that second tone may be removed from the signal since it is unlikely that
anyone could distinguish the presence of the quieter tone.

If the kHz tone goes away and the 1.001KHz tone is still at -20dB, the codec
will restore the 1.001KHz tone to the signal. as it will now become audible
since it is no longer masked by a louder, nearby tone.

If, however the second tone is several octaves away from the first, say
200Hz at -20dB, it will still be audible even though it is much quieter.
Since there is not another, louder tone nearby to mask it, will be audible.


- FLINT

[Bob Cain]

Lord Hasenpfeffer wrote:
>
> Arny Krueger wrote:
> > "Stewart Pinkerton" > wrote in message
> >
> >
> >
> >>Which part of 'no effect on masking' did you fail to understand? The
> >>threshold of hearing is dependent on sound levels, not on closeness to
> >>0dB FS on the CD, IOW just crank up the amplifier a little.
> >
> > Point of order...
> >
> > The threshold of hearing as usually given *is* a sound level, and therefore
> > can't possibly be dependent on a sound level.
>
> To clarify:
>
> An ATH chart shows that frequency Y is "inaudible" below amplitude X.
>
> In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y
> is discarded and does not become a part of the final MP3.
>
> However, after "normalizing" the WAV by a factor of +3dB, frequency Y's
> amplitude now becomes X+1dB, therefore, frequency Y is retained and
> becomes a part of the final MP3.

Have you checked with the Lame developers to see if this is
how the ATH is applied or if it is applied after the block
is normalized internally?


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Stewart Pinkerton]

On Tue, 01 Jul 2003 20:48:05 -0500, Lord Hasenpfeffer
> wrote:

>Arny Krueger wrote:
>> "Stewart Pinkerton" > wrote in message
>>
>>
>>
>>>Which part of 'no effect on masking' did you fail to understand? The
>>>threshold of hearing is dependent on sound levels, not on closeness to
>>>0dB FS on the CD, IOW just crank up the amplifier a little.
>>
>> Point of order...
>>
>> The threshold of hearing as usually given *is* a sound level, and therefore
>> can't possibly be dependent on a sound level.
>
>To clarify:
>
>An ATH chart shows that frequency Y is "inaudible" below amplitude X.
>
>In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y
>is discarded and does not become a part of the final MP3.
>
>However, after "normalizing" the WAV by a factor of +3dB, frequency Y's
>amplitude now becomes X+1dB, therefore, frequency Y is retained and
>becomes a part of the final MP3.

We are well aware of what you're talking about, but there are two
problems with this:

1) The reference is to *sound level*, not amplitude on the CD. As
previously noted, just turn up the volume.

2) Have you checked that this is how your MP3 encoder actually works,
rather than on time and frequency-proximate masking (which is a
completely different matter)?

--

Stewart Pinkerton | Music is Art - Audio is Engineering

[Stewart Pinkerton]

On Tue, 01 Jul 2003 20:38:11 -0500, Lord Hasenpfeffer
> wrote:

>Martin Tillman wrote:
>
>> Previously...
>>
>> Message-ID: >
>>
>>> It amazes me that I can rip a WAV directly from an older commercial CD,
>>> "normalize" it to 2dB beyond zero (i.e. -10dBFS) and then encode from it
>>> an MP3 that sounds dramatically better than its own CD source.
>
>This is true. It does amaze me. Only now I know more about why this is
>the case. The original CD source is "unnormalized" in my sense of the
>term. The MP3 is made from a normalized WAV. In in general, most
>people (including me) tend to believe that louder is better ... because
>with loudness comes clarity.

No, with loudness comes compression, and a *false* impression of
clarity given by the restricted dynamics of the reproduction. Radio
stations have been doing this for decades.


>This remains... an MP3 made from an older, quieter, unnormalized WAV
>sounds poor to me compared to an MP3 made from a normalized one at the
>same level of volume. I frequently listen to my MP3s in random shuffle
>mode. Without "normalization", "remastered MP3s" sound are louder and
>clearer sounding that "unremastered MP3s". Therefore, it is useful for
>me to normalize older WAVs so that all of my MP3s have a nice, even
>loudness. If I don't the older MP3s sound like crap in comparison the
>newer "remastered" ones.
>
>This has become a very boring conversation.

This is true................

Basically, mix 'em up any way *you* like, but please refrain from
coming on here claiming that you've discovered some wonder product,
and 'whopped the ass of MFSL', 'cos both statements are *wayyy* wide
of the mark. What you're talking about has been known for decades, and
is just the sort of barbarity that MFSL deliberately tried to avoid.

--

Stewart Pinkerton | Music is Art - Audio is Engineering

[KikeG]

(Stewart Pinkerton) wrote in message >...

> AAC and MP3 *are* Fraunhoffer algorithms................

First, whilst one could agree that Fraunhoffer did develop the MP3
format and its first implementation on their own , I think that's not
the case of AAC format, where other companies participated too in the
development.

Second, I was talking about encoders, or compressors. Both a encoder
and a decoder and its algorithms do not need to share any code from
Fraunhoffer, they just need to conform to the standard. LAME encoder
is said not to have a single line of code written from Fraunhoffer
developers, and is considered to be superior to Fraunhoffer encoders
at bitrates over 128 kbps.

> Agreed that AAC is likely the best of the bunch.

True. And ATRAC would be at the bottom.

[Lord Hasenpfeffer]

Dave Platt wrote:

> I can see at least two possible hypotheses:

Dave, I have drawn a picture. Please click the on the link provided
below and tell me which of your two hypotheses are best described by
what you see.

http://www.mykec.com/mykec/images/Good_MP3_Bad_MP3.gif

Myke

> Hypothesis 1: the _relative_ amount of frequency content in the two
> MP3 versions is identical, to within the inevitable low-level errors
> implied by any coding system. Or, in other words, the lower-level
> (duller-sounding-to-you) MP3 would sound identical to the higher-level
> (clearer-sounding) MP3, if you simply turned up the volume a bit. The
> differences in how you perceive the two aren't due to the actual
> content of the MP3, only to the playback amplitude.
>
> You might be perceiving the two as substantially different either due
> to the well-known Fletcher-Munson effect: your ears' relative
> sensitivities to bass and treble fall off faster than their
> sensitivity to midrange, as the volume is reduced, and thus quieter
> signals tend to sound as if they lack both bass and treble.
>
> Or, it might be a masking and ambient-noise effect. If you typically
> listen to MP3s in conditions of high ambient noise (office, car,
> outdoors, etc.) then more of the lower-level-MP3 playback would be
> drowned out by ambient noise, making it sound less clear.
>
> The cure for this would be simply "Turn up the volume during playback."
>
> Hypothesis 2: the MP3s encoded from lower-amplitude (non-normalized)
> WAV inputs are actually, and significantly different than their
> higher-amplitude cousins in ways other than just amplitude. Possibly
> some frequencies are missing (excessive masking, or sounds below
> threshold), possibly some frequency bands are less accurately encoded
> and have a higher noise level.
>
> This would suggest that the MP3 encoder you are using is less than
> optimal. Possibly it has a poorly-set threshold detector, which is
> sensitive only to the absolute signal level and not to the relative
> levels. Possibly one of its other encoder or bit-allocation
> algorithms is misbehaving, and is dedicating larger portions of the
> available bit bandwidth to certain frequencies and is being forced to
> discard other bands for lack of sufficient bit-reservoir in the
> encoding.
>
> The cure for this would be "Use a better encoder, or better settings
> for the one that you have."
>


--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

Stewart Pinkerton wrote:

>> To clarify:
>>
>> An ATH chart shows that frequency Y is "inaudible" below amplitude X.
>>
>> In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y
>> is discarded and does not become a part of the final MP3.
>>
>> However, after "normalizing" the WAV by a factor of +3dB, frequency Y's
>> amplitude now becomes X+1dB, therefore, frequency Y is retained and
>> becomes a part of the final MP3.
>
> We are well aware of what you're talking about, but there are two
> problems with this:
>
> 1) The reference is to *sound level*, not amplitude on the CD. As
> previously noted, just turn up the volume.

Where is the volume knob on my MP3 encoder?

> 2) Have you checked that this is how your MP3 encoder actually works,
> rather than on time and frequency-proximate masking (which is a
> completely different matter)?

I am absolutely certain that MP3 and ATRAC employ (1) masked frequency
and (2) Absolute Threshold of Hearing filtration methods.

Frequencies which are masked are removed AND frequencies which are too
quiet to be heard are removed.

My concerns lie NOT with masked frequencies. My purpose in normalizing
WAVs prior to encoding them is to assist them in successfully passing
through the A.T. of H. filter so that they end up as part of the final
MP3 rather than on the cutting room floor.

If the A.T. of H. filter prevents the under-amplified frequencies from
getting across and into the resultant MP3, you can playback that MP3 all
you want at whatever volume you want and those frequencies will still be
absent from the file.

Here's a picture: http://www.mykec.com/mykec/images/Good_MP3_Bad_MP3.gif

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

Stewart Pinkerton wrote:

> No, with loudness comes compression, and a *false* impression of
> clarity given by the restricted dynamics of the reproduction. Radio
> stations have been doing this for decades.

Radio stations have not been encoding MP3s for decades!

> Basically, mix 'em up any way *you* like, but please refrain from
> coming on here claiming that you've discovered some wonder product,
> and 'whopped the ass of MFSL', 'cos both statements are *wayyy* wide
> of the mark. What you're talking about has been known for decades, and
> is just the sort of barbarity that MFSL deliberately tried to avoid.

http://www.mykec.com/mykec/images/Good_MP3_Bad_MP3.gif

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message

> My concern with creating better MP3s from louder WAVs lies not with
> frequency masking but with filtration based upon what it known as the
> Absolute Threshold of Hearing.

For the benefit of those not following the r.a.p thread. The (now) famous
"Absolute Threshold Of Hearing" relates to signals at extremely low levels
that are disgarded in MP3 and ATRAC coding. Not levels like -1dB, or -10dB,
or -20dB, but in the order of -60dB and lower.

geoff

[Lord Hasenpfeffer]

Geoff Wood wrote:

> Radio stations have been *restricting dynamics* for decades.

http://www.mykec.com/mykec/images/Sunday_Bloody_Sunday_012.gif

0) Original WAV
1) "Normalize" (default -12dBFS)
2) "Normalize" (default -10dBFS)

Where is the evidence of this gawd-awful *restriction of dynamics* of
which you accuse me of wreaking against my WAVs?

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

Geoff Wood wrote:

> For the benefit of those not following the r.a.p thread. The (now) famous
> "Absolute Threshold Of Hearing" relates to signals at extremely low levels
> that are disgarded in MP3 and ATRAC coding. Not levels like -1dB, or -10dB,
> or -20dB, but in the order of -60dB and lower.

Well, thank God, somebody finally cited some hard numbers as opposed to
hard opinions regarding the Absolute Threshold of Hearing! Sheesh!!!
What was so hard about that? And what is your source for this
information? Or are you just picking them out of the blue?

-60dB in relation to what? Full Scale?

Of course this value fluctuates higher and lower depending on frequency
value. It's definitely *not* a straight line even though I drew it to
look like one in my "Good_MP3_Bad_MP3.gif".

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

Geoff Wood wrote:

> For the benefit of those not following the r.a.p thread. The (now) famous
> "Absolute Threshold Of Hearing" relates to signals at extremely low levels
> that are disgarded in MP3 and ATRAC coding. Not levels like -1dB, or -10dB,
> or -20dB, but in the order of -60dB and lower.

It's nice to see you talking more on the same level with me here.
Maybe we're finally beginning to see eye-to-eye just a little bit
better. Reading your quote above sure beats having to read about
how no frequecies magically or disappear as a result of the
"normalization" - which I've never said nor even implied.

http://www.mykec.com/mykec/images/Good_MP3_Bad_MP3.gif

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Lord Hasenpfeffer]

> how no frequecies magically or disappear as a result of the

should have been:

"how no frequecies magically appear or disappear as a result of the"

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Arny Krueger]

"Lord Hasenpfeffer" > wrote in message

> Geoff Wood wrote:

>> For the benefit of those not following the r.a.p thread. The (now)
>> famous "Absolute Threshold Of Hearing" relates to signals at
>> extremely low levels that are disgarded in MP3 and ATRAC coding.
>> Not levels like -1dB, or -10dB, or -20dB, but in the order of -60dB
>> and lower.

Well not really but kinda-sorta.

The usually-given threshold of hearing is the "0 dB" line on charts like
this one:

http://www.webervst.com/fm.htm

However, due to masking the just noticeable levels for reliable perception
in normal day-to-day circumstances are way far higher.

> Well, thank God, somebody finally cited some hard numbers as opposed
> to hard opinions regarding the Absolute Threshold of Hearing!
> Sheesh!!! What was so hard about that? And what is your source for
> this information? Or are you just picking them out of the blue?

For a real thrill, search google for "Fletcher Munson Curve".

> -60dB in relation to what? Full Scale?

Actually, the threshold of hearing is closer to 0 dB SPL @ 1000 Hz, and 0 dB
SPL is defined in terms of physical units relevant to acoustics (pressure
over area).

> Of course this value fluctuates higher and lower depending on
> frequency value.

Well doooh!

> It's definitely *not* a straight line even though I
> drew it to look like one in my "Good_MP3_Bad_MP3.gif".

I can't believe how much bandwidth and energy has been used up with this
tiring discussion when a few minutes of research with google would provide
most of the relevant facts.

[Arny Krueger]

"Lord Hasenpfeffer" > wrote in message

> KikeG wrote:
>
>>> Agreed that AAC is likely the best of the bunch.
>
>> True. And ATRAC would be at the bottom.
>
> You're implying that ATRAC is worse than MP3?

High bitrate MP3 can be really pretty good, expecially compared to low
bitrate ATRAC.

If you want to listen for yourself, please see:

http://www.pcabx.com/product/coder_decoder/index.htm

and

http://www.pcabx.com/product/mds-jb920/index.htm


You might want to start looking at this site starting with its home page at
www.pcabx.com .

[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message
...
> Geoff Wood wrote:
>
> > For the benefit of those not following the r.a.p thread. The (now)
famous
> > "Absolute Threshold Of Hearing" relates to signals at extremely low
levels
> > that are disgarded in MP3 and ATRAC coding. Not levels like -1dB,
or -10dB,
> > or -20dB, but in the order of -60dB and lower.
>
> Well, thank God, somebody finally cited some hard numbers as opposed to
> hard opinions regarding the Absolute Threshold of Hearing! Sheesh!!!
> What was so hard about that? And what is your source for this
> information? Or are you just picking them out of the blue?

Out of the blue.

> -60dB in relation to what? Full Scale?

Either FS or the wavfile peak level, which in most circumstance will be not
too dissimilar. As long as the peak isn't bizarrely low.

geoff

[Lord Hasenpfeffer]

Arny Krueger wrote:

> You might want to start looking at this site starting with its home page at
> www.pcabx.com .

Thanks Arny. I've bookmarked all three pages. I'll look at 'em closer
ASAP.

It's beginning to look a lot like Usenet... the one I used to know... :)

Myke


--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Per Stromgren]

On Wed, 02 Jul 2003 06:15:29 -0500, Lord Hasenpfeffer
> wrote:

>Arny Krueger wrote:
>
>> The usually-given threshold of hearing is the "0 dB" line on charts like
>> this one:
>>
>> http://www.webervst.com/fm.htm
>
>Ba-da-bing!! That's definitely right on track with what I've been
>trying to find.

God heavens! Where did you look? Really?

This is so basic for most of us around here, that it is hard to
envision somebody not knowing it. You cannot open any book, however
basic, on hearing that does not start of with these facts!

Do you hate books that much, Myke?
If not, your public library could probably supply you with a book that
takes you thru this a lot faster than it takes to ask all these
questions on Usenet.

Per.

[KikeG]

Lord Hasenpfeffer > wrote in message >...

> You're implying that ATRAC is worse than MP3?

Yes. And not only ATRAC3 long play modes at 132 kbps and below. Best
quality, standard 292 kbps ATRAC, is quite inferior to a LAME 3.90.2
200 kbps variable bitrate MP3, according to my blind tests: see first
and last post at http://www.hydrogenaudio.org/index.php?act=ST&f=32&t=3373&hl=atrac&s=31dca1739876fe98babb10a72975fb36

One of the ATRAC compression artifacts, the pre-echo of the castanets
sample, is blantantly evident. LAME 3.90.2, 3.90.3 or 3.91 with the
--alt-preset standard commandline are far better in this particular
sample, being transparent for many untrained people. Artifacts in the
other samples are more subtle, but LAME still does better in all of
them.

I'd dare to say that even a 192 kbps constant bitrate, joint-stereo
LAME mp3 is probably better than 292 kbps ATRAC.

[Lord Hasenpfeffer]

Per Stromgren wrote:

> Do you hate books that much, Myke?

I own a huge personal library of programming manuals persuant to my
purpose as a web-designer and Linux system/database administrator, etc..

Too many demands on my time for programming related needs throughout the
past 4 years have not left time for any other kinds of book-related
studies. My needs along these lines are very few and far-between.

> If not, your public library could probably supply you with a book that
> takes you thru this a lot faster than it takes to ask all these
> questions on Usenet.

Well, the chart isn't the answer to everything I'm trying to find. It's
just a good, solid clue towards the solution - much closer and a *lot*
more helpful than all that other sometimes obvious stuff that's been
being discussed which while possible true is completely beside the point
and tangential at best.

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Martin Tillman]

On Tue, 01 Jul 2003 20:38:11 -0500, Lord Hasenpfeffer wrote:

> Martin Tillman wrote:
>
>> Previously...
>>
>> Message-ID: >
>>
>>> It amazes me that I can rip a WAV directly from an older commercial CD,
>>> "normalize" it to 2dB beyond zero (i.e. -10dBFS) and then encode from it
>>> an MP3 that sounds dramatically better than its own CD source.
>
> This is true. It does amaze me. Only now I know more about why this is
> the case. The original CD source is "unnormalized" in my sense of the
> term. The MP3 is made from a normalized WAV.

Your definition of normalise is flawed, as is your practice of it.


> In in general, most
> people (including me) tend to believe that louder is better ... because
> with loudness comes clarity.

Louder is just louder. Depending on the circumstances louder is either
better or worse than quieter. Therefore, 'louder is better' is actually
totally meaningless.

>
> This remains... an MP3 made from an older, quieter, unnormalized WAV
> sounds poor to me compared to an MP3 made from a normalized one at the
> same level of volume.

I've proved you completely wrong with your Chris Cross example. Care to
listen to my proof?

> I frequently listen to my MP3s in random shuffle
> mode. Without "normalization", "remastered MP3s" sound are louder and
> clearer sounding that "unremastered MP3s". Therefore, it is useful for
> me to normalize older WAVs so that all of my MP3s have a nice, even
> loudness. If I don't the older MP3s sound like crap in comparison the
> newer "remastered" ones.

Faulty terminology, faulty logic, faulty use of the software.

Hint: There is nothing fundamentally wrong with true normalistion, nor
the desire to to have certain tracks sound more or less as loud as other
certain tracks, under certain circumstances. However, buggering the
dynamic range is totally wrong if you care about music, and even more
wrong when you don't realise you're doing it.

Hint 2: Do a search for 'Replay Gain'. If you can understand Replay
Gain you will be a better person. (Shock Horror! Replay Gain will
require that you normalise many of your tracks to around 6-10dB below
maximum!!!!!)

[Bob Cain]

Lord Hasenpfeffer wrote:
>
>
> Actually, flint, what you've just described *is* frequency masking which
> is only one of 2 different filtration methods employed by lossy
> compression methods such as MP3, AAC, MiniDisc, etc..

Just a minor terminology quibble, Myke. Filtration is what
is done to sewage. I'm an old E.E. with a DSP upgrade and
have never heard the word applied to signals. "Filtering"
is what you want.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Martin Tillman]

On Wed, 02 Jul 2003 04:46:35 -0500, Lord Hasenpfeffer wrote:

> Geoff Wood wrote:
>
>> Radio stations have been *restricting dynamics* for decades.
>
> http://www.mykec.com/mykec/images/Sunday_Bloody_Sunday_012.gif
>
> 0) Original WAV
> 1) "Normalize" (default -12dBFS)
> 2) "Normalize" (default -10dBFS)
>
> Where is the evidence of this gawd-awful *restriction of dynamics* of
> which you accuse me of wreaking against my WAVs?

Right in front of your eyes! It shows absolutely that you are
compressing the original. If you can't see it you really don't know
what you are looking for.


Clues:

Left channel peaks at .55 and .58 mins retain relationship to lower
peaks. No compression.

Left channel peak just before 2.5. Look at the slightly lower peak just
before that one on top of waveform. See how it gets closer to the value
of the higher peak on the two higher level waveforms. What do you think
we could call this phenomenon?

See also grosser example on left channel lower waveform at 4.37 and
4.55.

Thanks for providing such a classic example of how wrong you are.

While there is no scale, I can guess what it is, and this also shows
that the highest peaks in the original are pretty close to full scale,
perhaps a couple of dBs down, indicating that there is precious little
room for REAL normalisation to maximum level anyway.

[Lord Hasenpfeffer]

Bob Cain wrote:

> Just a minor terminology quibble, Myke. Filtration is what
> is done to sewage. I'm an old E.E. with a DSP upgrade and
> have never heard the word applied to signals. "Filtering"
> is what you want.

Hahahaha!!! As they say, "my bad!" :-D

Well here's what my gDict for Linux has to say about this issue:

Filtration Fil*tra"tion, n. Cf. F. filtration.
The act or process of filtering; the mechanical separation of
a liquid from the undissolved particles floating in it.

So, I guess we're both right! :)

The answer is clearly "yes and no"! ;-)

Myke

--

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[Lord Hasenpfeffer]

Martin Tillman wrote:

>> Where is the evidence of this gawd-awful *restriction of dynamics* of
>> which you accuse me of wreaking against my WAVs?
>
> Right in front of your eyes! It shows absolutely that you are
> compressing the original. If you can't see it you really don't know
> what you are looking for.

Yeah, I believe I'm on record as having stated there is some slight
*limiting* going on there to avoid clipping. Note however too that this
is the loudest song on the CD. All other songs would not suffer from
any limiting at all because of this +5dB boost in RMS level.

5 or so peaks being slightly limited in a single WAV does not offset the
advantages of boosting the average RMS level of the entire file. It is
a compromise I am perfectly willing to accept.

Myke

--

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[Martin Tillman]

On Wed, 02 Jul 2003 16:02:23 -0500, Lord Hasenpfeffer wrote:

> > Your definition of normalise is flawed, as is your practice of it.
>
> Yes, I know. It's grossly misdefined in my "normalize" manpage by the
> author of the program itself. That's why. So blame *me* *again* for my
> misunderstanding of the term, why dontcha?

Well, you keep using it, even now.

[Bob Cain]

Lord Hasenpfeffer wrote:
>
> Bob Cain wrote:
>
> > Just a minor terminology quibble, Myke. Filtration is what
> > is done to sewage. I'm an old E.E. with a DSP upgrade and
> > have never heard the word applied to signals. "Filtering"
> > is what you want.
>
> Hahahaha!!! As they say, "my bad!" :-D
>
> Well here's what my gDict for Linux has to say about this issue:
>
> Filtration Fil*tra"tion, n. Cf. F. filtration.
> The act or process of filtering; the mechanical separation of
> a liquid from the undissolved particles floating in it.
>
> So, I guess we're both right! :)
>
> The answer is clearly "yes and no"! ;-)

You just don't give up a point do you. :-) In the world of
processing signals, analog or digital, it's "filtering."
Period, point, dot.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message
...
> Martin Tillman wrote:
>
> >> Where is the evidence of this gawd-awful *restriction of dynamics* of
> >> which you accuse me of wreaking against my WAVs?
> >
> > Right in front of your eyes! It shows absolutely that you are
> > compressing the original. If you can't see it you really don't know
> > what you are looking for.
>
> Yeah, I believe I'm on record as having stated there is some slight
> *limiting* going on there to avoid clipping. Note however too that this
> is the loudest song on the CD. All other songs would not suffer from
> any limiting at all because of this +5dB boost in RMS level.

But you are diverting from your compilation scenario, where you normalise
each track to -10dB.

geoff

[Lord Hasenpfeffer]

Bob Cain wrote:
>
>> Filtration Fil*tra"tion, n. Cf. F. filtration.
>> The act or process of filtering; the mechanical separation of
>> a liquid from the undissolved particles floating in it.

> You just don't give up a point do you. :-) In the world of
> processing signals, analog or digital, it's "filtering."
> Period, point, dot.

Well, if it's any consolation to you, Bob, I *can* see it from your
perspective as well as from my own. :-)

I just thought it was funny to see that the first half of the first
definition I was able to locate seemed to support both your and my
preferred usages of the word - and my usage was listed first. :)

This really did make me laugh out loud. My wife had to ask me what was
so funny. Which then meant I had to *explain*... And god how I hate to
*explain* things!!! ;-)

Myke

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[Lord Hasenpfeffer]

Geoff Wood wrote:

> But you are diverting from your compilation scenario, where you normalise
> each track to -10dB.

You know, I really do wish that instead of everybody wanting to jump my
ass and call me sssttyyyeeewwwpid, they could have just asked me what I
meant whenever I used the term "batch normalize". Having been using
this application for such a long time, I didn't know it was confusing
anybody else.

Back to the Normalize man page we go:

MIX MODE

This mode is made especially for making mixed CD's and the like.
You want every song on the mix to be the same volume, but it
doesn't matter if they are the same volume as the songs on some
other mix you made last week. In mix mode, average level of all
the files is computed, and each file is separately normalized to
this average volume.


BATCH MODE

When operating on a group of unrelated files, you usually want all
of them at the same level, and this is the default behavior.
However, a group of music files all from the same album is generally
meant to be listened to at the relative volumes they were recorded
at. In batch mode, all the specified files are considered to be part
of a single album and their relative volumes are preserved. This
is done by averaging the volumes of all the files, computing a
single adjustment from that, and applying the same adjustment to
all the files. Some analysis is also done so that files with
volumes that appear to be statistical aberrations are not
considered in the average. This is useful if you have albums
(like many of the author's) in which there is one "quiet song"
that throws off the average.

Myke

--

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[Stewart Pinkerton]

On Wed, 02 Jul 2003 15:28:54 -0700, Bob Cain
> wrote:

>
>
>Lord Hasenpfeffer wrote:
>>
>> Bob Cain wrote:
>>
>> > Just a minor terminology quibble, Myke. Filtration is what
>> > is done to sewage. I'm an old E.E. with a DSP upgrade and
>> > have never heard the word applied to signals. "Filtering"
>> > is what you want.
>>
>> Hahahaha!!! As they say, "my bad!" :-D
>>
>> Well here's what my gDict for Linux has to say about this issue:
>>
>> Filtration Fil*tra"tion, n. Cf. F. filtration.
>> The act or process of filtering; the mechanical separation of
>> a liquid from the undissolved particles floating in it.
>>
>> So, I guess we're both right! :)
>>
>> The answer is clearly "yes and no"! ;-)
>
>You just don't give up a point do you. :-) In the world of
>processing signals, analog or digital, it's "filtering."
>Period, point, dot.

Perhaps in the US, but here in the UK both terms are applied. I've
certainly seen references to 4th order filtration, which is not I
believe to be found in sewage farms..... :-)

--

Stewart Pinkerton | Music is Art - Audio is Engineering

[Per Stromgren]

On Wed, 02 Jul 2003 09:52:14 -0500, Lord Hasenpfeffer
> wrote:

>Per Stromgren wrote:
>
>> Do you hate books that much, Myke?
>
>I own a huge personal library of programming manuals persuant to my
>purpose as a web-designer and Linux system/database administrator, etc..

Manuals are not what I mean by a "book"! That is probably your
"problem" right there. If one only reads problem-solving "books" (as
manuals tend to be) you will never get to the point where you can
solve problems all by yourself. As a university teacher, I know this,
believe me. You need a bit of theory that is not connected to a
product. The man page for the normalize program is a good case in
point: it gave you the wrong defintion of "normalize", at least one
that is never heard of outside that product.

>
>Too many demands on my time for programming related needs throughout the
>past 4 years have not left time for any other kinds of book-related
>studies. My needs along these lines are very few and far-between.

The discussion we have on usenet say something different to me. You
can't be *that* busy right now, you have written 517 postings to the
usenet in the last two weeks, over 35 per day. This according to
Google Groups, I do not know how they calculate cross-postings.

>
>> If not, your public library could probably supply you with a book that
>> takes you thru this a lot faster than it takes to ask all these
>> questions on Usenet.
>
>Well, the chart isn't the answer to everything I'm trying to find. It's
>just a good, solid clue towards the solution - much closer and a *lot*
>more helpful than all that other sometimes obvious stuff that's been
>being discussed which while possible true is completely beside the point
>and tangential at best.

As I have read your threads with interest, I have seen that people has
tried to give you some help in that they have pointed out some obvious
misconception of yours. But is only you that can do anything about
learning these things. We can only point it out. Sorry.

Per.

[Per Stromgren]

(This a follow-up to Myke, not Martin. Sorry for any confusion)

Myke wrote:
> In in general, most
> people (including me) tend to believe that louder is better ... because
> with loudness comes clarity.

Wasn't it Rainbow that said "Our single aim in life is to get louder!"
You can not discuss anything audio with people that has this as a
target.

Per.

[Lord Hasenpfeffer]

Stewart Pinkerton wrote:

> Perhaps in the US, but here in the UK both terms are applied. I've
> certainly seen references to 4th order filtration, which is not I
> believe to be found in sewage farms..... :-)

Now Stewart, didja really hafta go and add insult to injury like that? :-)

*LOL*,

Myke

--

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[Bob Cain]

Lord Hasenpfeffer wrote:
>
> Based upon what I interpret "(multiply PCM data)" to mean, it's not linear.
>

Mulitiplication by a constant is a linear operation.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Bob Cain]

Lord Hasenpfeffer wrote:
>
> This really did make me laugh out loud. My wife had to ask me what was
> so funny. Which then meant I had to *explain*... And god how I hate to
> *explain* things!!! ;-)

You wanna hear something funny, since that post in which I
mentioned sewage I've received two spams relative to home
sewage processing systems. :-)


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Lord Hasenpfeffer]

Richard D Pierce wrote:

> Yup, it makes it abundantly clear. Not a single reference text
> on signal processing principles or theory. Nothing, nada.
>
> The nonsense all makes perfect sense, now.

I've maintained from the start that I'm not an audio professional,
however, certain aspects of my work do force me cross over into that area.

I don't need any **** because of it.

Myke


--

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[Richard D Pierce]

In article >,
Lord Hasenpfeffer > wrote:
>Richard D Pierce wrote:
>
>> Yup, it makes it abundantly clear. Not a single reference text
>> on signal processing principles or theory. Nothing, nada.
>>
>> The nonsense all makes perfect sense, now.
>
>I've maintained from the start that I'm not an audio professional,

Your effort at maintaining that is really unnecessary, as this
fact has been abundantly clear from the outset.

>however, certain aspects of my work do force me cross over into that area.

Indeed that may be true. Certain aspects of my work force me to
cross over into areas where I might have sufficient knowledge.
When that happens, I don't pretend that my misinterpretations
represent sufficient knowledge.

>I don't need any **** because of it. >

Then why give it?

--
| Dick Pierce |
| Professional Audio Development |
| 1-781/826-4953 Voice and FAX |
| |

[Lord Hasenpfeffer]

StArSeEd wrote:

> So write your own LAME frontend and *make* it a knob.

Because that's not the kind of programming with which I'm professionally
or even personally familiar. I could probably pick it up with a little
practice but time so far will not allow it.

All languages share common concepts but just because I speak English
doesn't mean I can also speak Chinese.

> Yes. --scale 2.0 = +6dB. --scale 4.0 = +12dB.
> Likewise, --scale 0.5 = -6db and --scale 0.25 = -12dB.

Thank you. That helps a *lot*.

> Decibels are not linear.

Thanks again. I didn't know that. Why didn't Richard D. Pierce take
time to be *helpful* by telling me *this* instead of opting to delight
in ripping off my head and ****ting down my neck for not owning a single
reference text on signal processing principles or theory? Am I correct
to assume he's just here to be a thorn in other peoples' sides like I
always *expect* Usenet "people" to be? If so, I'm not *surprised*.

>> Based upon what I interpret "(multiply PCM data)" to mean, it's not linear.
>
> Now yer "Log"gin'.

Yeah, and I've got a "decent" mathematical background too so the
concepts here aren't exactly something I'd have too much trouble picking
up with a little practice. Although without a background in
engineering, logarithms have never been very useful in everyday life.

I'm much better with internet-based, creative/logical problem-solving
involving database structures and web-applications and user interface
designs, etc.

Myke

--

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[Lord Hasenpfeffer]

Richard D Pierce wrote:

> Amongst your "extensive library,"

Oh, *now* I'm *bragging* about my "extensive library" is it?

You obviously fail to see the reason why I posted that list.

Chill out.

Myke

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[Lord Hasenpfeffer]

Richard D Pierce wrote:

> Indeed that may be true. Certain aspects of my work force me to
> cross over into areas where I might have sufficient knowledge.
> When that happens, I don't pretend that my misinterpretations
> represent sufficient knowledge.

Man are you behind the curve on the development of this conversation or
what?

Your kind of crap is exactly why I packed up and moved the discussion
elsewhere.

We've actually arrived at some fairly decent conclusions as a result of
it too.

Get lost.

Myke

--

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[Richard D Pierce]

In article >,
Lord Hasenpfeffer > wrote:
>Richard D Pierce wrote:
>
>> Indeed that may be true. Certain aspects of my work force me to
>> cross over into areas where I might have sufficient knowledge.
>> When that happens, I don't pretend that my misinterpretations
>> represent sufficient knowledge.
>
>Your kind of crap is exactly why I packed up and moved the discussion
>elsewhere.

Bye bye!

>Get lost.

Kind sir, it would seem you've done that for us.

--
| Dick Pierce |
| Professional Audio Development |
| 1-781/826-4953 Voice and FAX |
| |

[Lord Hasenpfeffer]

>> Richard D Pierce wrote:


--

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[Geoff Wood]

"Bob Cain" > wrote in message
...
>
>
> Lord Hasenpfeffer wrote:
> >
> > This really did make me laugh out loud. My wife had to ask me what was
> > so funny. Which then meant I had to *explain*... And god how I hate to
> > *explain* things!!! ;-)
>
> You wanna hear something funny, since that post in which I
> mentioned sewage I've received two spams relative to home
> sewage processing systems. :-)


**** ! Really !


geoff

[Stewart Pinkerton]

On Thu, 03 Jul 2003 14:29:09 -0500, Lord Hasenpfeffer
> wrote:

>StArSeEd wrote:
>
>> So write your own LAME frontend and *make* it a knob.
>
>Because that's not the kind of programming with which I'm professionally
>or even personally familiar. I could probably pick it up with a little
>practice but time so far will not allow it.
>
>All languages share common concepts but just because I speak English
>doesn't mean I can also speak Chinese.
>
>> Yes. --scale 2.0 = +6dB. --scale 4.0 = +12dB.
>> Likewise, --scale 0.5 = -6db and --scale 0.25 = -12dB.
>
>Thank you. That helps a *lot*.
>
>> Decibels are not linear.
>
>Thanks again. I didn't know that.

WHAAAAAT? Oh phukkke, just forget everything..................

>Why didn't Richard D. Pierce take
>time to be *helpful* by telling me *this* instead of opting to delight
>in ripping off my head and ****ting down my neck for not owning a single
>reference text on signal processing principles or theory? Am I correct
>to assume he's just here to be a thorn in other peoples' sides like I
>always *expect* Usenet "people" to be? If so, I'm not *surprised*.

Jeez, maybe he gave you credit for knowing one of the most basic facts
in all of audio? Silly Dick.............

>>> Based upon what I interpret "(multiply PCM data)" to mean, it's not linear.
>>
>> Now yer "Log"gin'.
>
>Yeah, and I've got a "decent" mathematical background too so the
>concepts here aren't exactly something I'd have too much trouble picking
>up with a little practice. Although without a background in
>engineering, logarithms have never been very useful in everyday life.

Actually, not knowing that deciBels are logarithmic means that you
actually have no *useful* mathematical background to speak of in this
forum.................

>I'm much better with internet-based, creative/logical problem-solving
>involving database structures and web-applications and user interface
>designs, etc.

Oh, you mean you can design real kool websites with like animated
flames and stuff? Woweeeeee...................
--

Stewart Pinkerton | Music is Art - Audio is Engineering

[Lord Hasenpfeffer]

Stewart Pinkerton wrote:

> Oh, you mean you can design real kool websites with like animated
> flames and stuff? Woweeeeee...................

Yep. And the money's good too. Woweeeeee...................

Myke

--

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[Richard D Pierce]

On Thu, 03 Jul 2003 14:29:09 -0500, Lord Hasenpfeffer
> wrote:
>>> Decibels are not linear.
>>
>>Thanks again. I didn't know that.

>>Yeah, and I've got a "decent" mathematical background too so the
>>concepts here aren't exactly something I'd have too much trouble picking
>>up with a little practice.

Okay, so it's not the "decent" mathematical background that
s giving you trouble picking stuff up. So, what might you
specualte IS giving you these seemingly insurmountable
difficulties?

But, wait! The mere fact the assumption that:

F(x) = k * x

is a NON-linear operation IS INDEED A CLEAR AND OBVIOUS MISTAKE
even to someone who has a basic background in high school
algebra! And, oh by the way, the same person who made it through
that high-school algebra class that would know that that
multiplication by a constant is a LINEAR operation would also
have likely encountered logarithms as well.

>Although without a background in
>engineering, logarithms have never been very useful in everyday life.

Given that the sensitivity of the ear and the eye to changes in
intensity is very nicely approximated to a first degree by
logarithmic functions, yes, I suppose you COULD assume that
knowledge of such is not very useful in everyday life.

>I'm much better with internet-based, creative/logical problem-solving
>involving database structures and web-applications and user interface
>designs, etc.

Ah, okay, so you're not a real software engineer, then. That's
cool, it explains quite a bit.

--
| Dick Pierce |
| Professional Audio Development |
| 1-781/826-4953 Voice and FAX |
| |

[Richard Crowley]

"Lord Hasenpfeffer" wrote...
> --

Would that all his posts were as pithy.
Plonk to you, sir.

[Lord Hasenpfeffer]

Richard Crowley wrote:


--

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[Geoff Wood]

"Lord Hasenpfeffer" > wrote in message
...
> Richard Crowley wrote:
>
>
> --
>
> -================================-
> Windows...It's rebootylicious!!!
> -================================-


Is this '--' stuff your new skill ?

geoff

[Lord Hasenpfeffer]

Geoff Wood wrote:

> Is this '--' stuff your new skill ?

No.

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[StArSeEd]

Geoff Wood:
> Lord Hasenpfeffer
> > --
> Is this '--' stuff your new skill ?

It's actually '-- ', the standard delimiter for UseNet and e-mail signatures.

-StArSeEd
--
dchub://tsphub.dyndns.org:1979
IRC EFnet #smashing_pumpkins
Email:
ICQ UIN: 1711589

[Lord Hasenpfeffer]

StArSeEd wrote:
> Geoff Wood:
>
>>Lord Hasenpfeffer
>>
>>>--
>>
>>Is this '--' stuff your new skill ?
>
>
> It's actually '-- ', the standard delimiter for UseNet and e-mail signatures.
>
> -StArSeEd

Hehehe... Thanks. :) (And, yes, you're right.)

Myke

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[Lord Hasenpfeffer]

Stewart Pinkerton wrote:

>> Pete Carney wrote:
>>
>>> The first image shows the original at the location of the max peak in the
>>> file. Then the second image is proper normalization in CoolEdit 2000 of the
>>> exact same time frame. Then the third is what the command line program
>>> "Normalize" does to the file.
>>
>> With what parameters?
>> Normalize doesn't just do one thing and nothing else.
>
> Weren't you claiming that this was *exactly* what it did with the MFSL
> disc? That it just increased *every* sample until the peaks were just
> below 0dB FS, with no other effect *whatever*?

No way. I said nothing of the sort.

Myke

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[Lord Hasenpfeffer]

Martin Tillman wrote:
> On Mon, 30 Jun 2003 03:22:54 -0500, Lord Hasenpfeffer wrote:

>> But Capitol's version's OK 'cuz everybody's heard of them.

> No one has said as far as I recall.

Nobody seems to be *avoiding* it.

Myke

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[Lord Hasenpfeffer]

Martin Tillman wrote:

>> And here are two MP3 samples in a zip archive:
>>
>> http://www.mykec.com/mykec/audio/All_Right.zip

> Pity we can't see the original,

Pity ****. I've posted links to the screenshots to, so you *can* if
you'll just bother to look.

> or that you didn't encode to a higher bitrate. However, despite that,
> there is very little doubt in my mind that your 'after' has limiting
> applied to it in order to make it sound louder.
>
> Yep, the only difference between them is that one is louder. Muppet.

Duh. Buttweej.

Myke

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[Martin Tillman]

On Fri, 04 Jul 2003 19:55:58 -0500, Lord Hasenpfeffer wrote:

>> Yep, the only difference between them is that one is louder. Muppet.
>
> Duh. Buttweej.

So I'll take that to mean you won't attempt to identify where the joins
are in case that proves you wrong in public.

[Lord Hasenpfeffer]

Martin Tillman wrote:

> So I'll take that to mean you won't attempt to identify where the joins
> are in case that proves you wrong in public.

I'm so afraid of being proven wrong in public.

Myke

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[Martin Tillman]

On Sat, 05 Jul 2003 10:25:38 -0500, Lord Hasenpfeffer wrote:

> Martin Tillman wrote:
>
>> So I'll take that to mean you won't attempt to identify where the joins
>> are in case that proves you wrong in public.
>
> I'm so afraid of being proven wrong in public.

So will you take the challenge and identify the edits between the
original and your infinitely improved version of it?

[Lord Hasenpfeffer]

cyrus the virus wrote:

> the normalization thing depends on the process. from the few times and
> few apps i've used that use normalization, it scans the entire track for
> the loudest point and raises the volume level so that loudest point is
> at zero. there is no sort of clipping/compressing/limiting going on.

Some tests with Normalize being conducted in conjunction with some
CoolEdit Pro users in another newsgroup have revealed that Normalize
does this same thing. It's determines the Maximum RMS level and works
with/from that rather than the Average RMS level as everybody here was
thinking previously. That's why even when I push it 2dB "hotter" than
it would normally on its own by default, any limiting that occurs is
minimal at best.

Myke

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[Lord Hasenpfeffer]

cyrus the virus wrote:

> and this perception is exactly why the loudness wars even started.
> volume and quality is really trivial when the music actually affects
> you.

Well, I've always been against the idea of doing obvious harm to
existing recordings. When I settled for my preferred -10dBFS setting, I
was actually coming back down from even louder settings which I'd tried
and rejected because I could hear the difference in the music and felt
that I was "pushing too far".

I'm not into overloud music at all. I'm just not into underloud music
either. -10dBFS gives me what I consider to be a nice, happy
medium-loud volume that is in no way offensive to the listener.

Being a foreigner to the audio.tech community (and happily so, I might
add), I completely missed the "loudness wars" which I've come to hear so
much about lately.

Who won? :)

Myke

--

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[Arny Krueger]

"Lord Hasenpfeffer" > wrote in message

> cyrus the virus wrote:
>
>> the normalization thing depends on the process. from the few times
>> and few apps i've used that use normalization, it scans the entire
>> track for the loudest point and raises the volume level so that
>> loudest point is at zero. there is no sort of
>> clipping/compressing/limiting going on.
>
> Some tests with Normalize being conducted in conjunction with some
> CoolEdit Pro users in another newsgroup have revealed that Normalize
> does this same thing. It's determines the Maximum RMS level and works
> with/from that rather than the Average RMS level as everybody here was
> thinking previously. That's why even when I push it 2dB "hotter" than
> it would normally on its own by default, any limiting that occurs is
> minimal at best.

Cool Edit Pro bases normalization on peak levels.

If you work in Cool Edit Pro's 32 bit mode, there is no limiting at any
reasonable level including many dBs above nominal FS. It's working in
floating point!

[Dick Pierce]

Lord Hasenpfeffer > wrote in message >...
> Stewart Pinkerton wrote:
>
> > Sure I did, and just how do you think you can 'normalise' to a higher
> > average level *without* modifying the transfer curve, i.e. applying
> > compansion and/or limiting?
>
> Well, after having read the Normalize F.A.Q. you should know that
> Average RMS isn't what's being "Normalized"; it's the *Maximum* RMS
> level that is.

This statement, all by itself, shows the patent absurdity of "Lords"
position, illustrating how technically inept he is. The terms
"average RMS" and "maximum RMS" are meaningless: RMS is RMS. There
is but a single RMS figure for a sognal computed over the interval.
There is no "maximum" or "avergae" RMS.

RMS means one VERY specific and well defined thing: the "root-mean-
square" value of a time-variant signal. It has a specific and fully
deterministic means of getting there. It's not even a definition
unique to signals, indeed, it is a STATISTICAL terms with the exact
definition of:

RMS = sqrt ( sum F(Xn)/n )

That's it, there ain't no "average RMS" there ain't no "maximum RMS,"
there's RMS.

And should a signal over an interval me normalized by some factor
k, ALL values will be multuplied by precisely the same factor k,
regardless of their original value. Normaization is simple:

F(x) = k * x

> I know *how* to make Normalize give me satisfactory results.

That's not clear, since you still have no clue what the normalize
process MEANS, and have no clue about the meaning of such fundamental
and, indeed, conceptually simple definitions like "RMS."

> If the Average RMS were being adjusted you would have a point, but it's
> not so you no longer do.

Since "average RMS" is a nonsense term, you have NO point.

[Bob Cain]

Lord Hasenpfeffer wrote:
>
> Stewart Pinkerton wrote:
>
> > Sure I did, and just how do you think you can 'normalise' to a higher
> > average level *without* modifying the transfer curve, i.e. applying
> > compansion and/or limiting?
>
>
> Time to reevaluate your attack strategy. Maybe this time you'll do some
> homework on your own before you go about believing an admitted amateur
> like me about such technical details.

His point stands regardless of whether the RMS level of the
file is determined from a peak region or averaged over the
whole file. The point is that to increase the level more
than than what would bring the largest instantaneous peak to
0 dB requires compression and/or limiting if clipping is to
be avoided.

>
> If the Average RMS were being adjusted you would have a point, but it's
> not so you no longer do. So go be rude to someone else for a change.

It is independant of that consideration. Try again.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Bob Cain]

Dick Pierce wrote:
>
> This statement, all by itself, shows the patent absurdity of "Lords"
> position, illustrating how technically inept he is. The terms
> "average RMS" and "maximum RMS" are meaningless: RMS is RMS. There
> is but a single RMS figure for a sognal computed over the interval.
> There is no "maximum" or "avergae" RMS.

Not quite true. In CEP, for example, one can specify a
window of evaluation and it will report the maximum RMS
value over that width window (50 ms is the default) within
the file.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Arny Krueger]

"Bob Cain" > wrote in message


> Dick Pierce wrote:
>>
>> This statement, all by itself, shows the patent absurdity of "Lords"
>> position, illustrating how technically inept he is. The terms
>> "average RMS" and "maximum RMS" are meaningless: RMS is RMS. There
>> is but a single RMS figure for a sognal computed over the interval.
>> There is no "maximum" or "avergae" RMS.

> Not quite true. In CEP, for example, one can specify a
> window of evaluation and it will report the maximum RMS
> value over that width window (50 ms is the default) within
> the file.

I see no conflict here. Piece said that there is but a single RMS figure
for a signal computed over the interval. What CEP does is chop the user's
interval into smaller intervals the size of the window, and then report RMS
values for selected windows.

I don't get why people are talking about RMS values in the context of CEP
and normalization when it is so clear that CEP bases normalization on the
largest magnitude any single sample in the user's interval.

[Dick Pierce]

Bob Cain > wrote in message >...
> Dick Pierce wrote:
> >
> > This statement, all by itself, shows the patent absurdity of "Lords"
> > position, illustrating how technically inept he is. The terms
> > "average RMS" and "maximum RMS" are meaningless: RMS is RMS. There
> > is but a single RMS figure for a sognal computed over the interval.
> > There is no "maximum" or "avergae" RMS.
>
> Not quite true. In CEP, for example, one can specify a
> window of evaluation and it will report the maximum RMS
> value over that width window (50 ms is the default) within
> the file.

No doubt, and that is entirely consistent with my statement:

"There is but one RMS figure for the signal computed over
the interval."

Please note the phrase "over the interval," congruent with the width
of the evaluation window.

Regardless of whether you use the qualifier "window width" or "over
the interval," there is one and only one RMS figure for the data.
If Lord HuffenPuffer is making statements about "average RMS" and
"maximum RMS," he is either utterly clueless as to the principles
involved or is being a deliberately destructuve troll.

[Bob Cain]

Arny Krueger wrote:
>
> I see no conflict here. Piece said that there is but a single RMS figure
> for a signal computed over the interval. What CEP does is chop the user's
> interval into smaller intervals the size of the window, and then report RMS
> values for selected windows.
>

Not sure what you mean by "interval" here. The Analyze
funtion runs a sliding window of a specifiable width over
the selected region and samples the RMS value within the
sliding window to report it's "Maximum" and "Minimum"
values.

> I don't get why people are talking about RMS values in the context of CEP
> and normalization when it is so clear that CEP bases normalization on the
> largest magnitude any single sample in the user's interval.

Unless you use it's "Group Normalize" function accessable in
2.x from the multitrack view. The "Analyze" function
available in track view has always had "Average", "Maximum",
"Minimum", and "Total" RMS values in its report. I'm not
sure the distinction between "Average" and "Total".


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Richard Crowley]

"Bob Cain" wrote ...
> Dick, have you actually used CEP's "Waveform Statistics"
> function (sorry, Arny, I called it "Analyze" in my response
> to you)? What you say is not correct. The function runs a
> window of a specifiable width over the overall region that
> is selected and remembers the maximum and minimum values
> seen in that window. There is a separate value reported
> which is the single value that you refer to which applies to
> the whole selected region.

50mS is the default interval.
I wonder if the average (RMS) of all the 50mS averages is the
same as the average over the entire file?

[Bob Cain]

Richard Crowley wrote:
>
> "Bob Cain" wrote ...
> > Dick, have you actually used CEP's "Waveform Statistics"
> > function (sorry, Arny, I called it "Analyze" in my response
> > to you)? What you say is not correct. The function runs a
> > window of a specifiable width over the overall region that
> > is selected and remembers the maximum and minimum values
> > seen in that window. There is a separate value reported
> > which is the single value that you refer to which applies to
> > the whole selected region.
>
> 50mS is the default interval.
> I wonder if the average (RMS) of all the 50mS averages is the
> same as the average over the entire file?

This has puzzled me. "Average" and "Total" are usually
pretty close but they do differ. I think, but am not sure,
that "Average" is just the average of all the windows while
"Total" is what we usually measure as the RMS level of a
region. I'm not really sure how to interpret the
difference.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Arny Krueger]

"Bob Cain" > wrote in message


> Arny Krueger wrote:

>> I see no conflict here. Pierce said that there is but a single RMS
>> figure for a signal computed over the interval. What CEP does is
>> chop the user's interval into smaller intervals the size of the
>> window, and then report RMS values for selected windows.

> Not sure what you mean by "interval" here. The Analyze
> function runs a sliding window of a specifiable width over
> the selected region and samples the RMS value within the
> sliding window to report it's "Maximum" and "Minimum"
> values.

The RMS measurement interval is the width of the window, right?

>> I don't get why people are talking about RMS values in the context
>> of CEP and normalization when it is so clear that CEP bases
>> normalization on the largest magnitude of any single sample in the
>> user's interval.

> Unless you use it's "Group Normalize" function accessible in
> 2.x from the multitrack view.

The Help file seems to say that Group Normalize still bases its operation
on peak values.

> The "Analyze" function
> available in track view has always had "Average", "Maximum",
> "Minimum", and "Total" RMS values in its report. I'm not
> sure the distinction between "Average" and "Total".

I can't see where the help file sheds any light on that question. My
understandings suggest that they should be the same, but they clearly aren't
often the same with real world signals.

BTW, thanks for the stimulus to read the docs, even if not all my questions
were answered.

[Browntimdc]

Bob Cain > wrote in
:

>
>
> Dick, have you actually used CEP's "Waveform Statistics"
> function (sorry, Arny, I called it "Analyze" in my response
> to you)? What you say is not correct. The function runs a
> window of a specifiable width over the overall region that
> is selected and remembers the maximum and minimum values
> seen in that window. There is a separate value reported
> which is the single value that you refer to which applies to
> the whole selected region.

So the program computes RMS within a window, shifts the window (with or
without overlap) and repeats. So each interval has one unique RMS value,
no? This would still agree with what Mr. Pierce is saying.

Tim

[Martin Tillman]

On 6 Jul 2003 14:21:29 -0700, Dick Pierce wrote:

> If Lord HuffenPuffer is making statements about "average RMS" and
> "maximum RMS," he is either utterly clueless as to the principles
> involved or is being a deliberately destructuve troll.

Which is more or less where we came in with the arse whipping he gave
MFSL ;-)

[Bob Cain]

Browntimdc wrote:
>
> Bob Cain > wrote in
> :
>
> >
> >
> > Dick, have you actually used CEP's "Waveform Statistics"
> > function (sorry, Arny, I called it "Analyze" in my response
> > to you)? What you say is not correct. The function runs a
> > window of a specifiable width over the overall region that
> > is selected and remembers the maximum and minimum values
> > seen in that window. There is a separate value reported
> > which is the single value that you refer to which applies to
> > the whole selected region.
>
> So the program computes RMS within a window, shifts the window (with or
> without overlap) and repeats. So each interval has one unique RMS value,
> no? This would still agree with what Mr. Pierce is saying.

Perhaps I misunderstood but my interpretation of what he
said was that the idea of a minimum and maximum RMS value
have no meaning. I'm just saying that it can be given a
meaning by assigning an RMS value to each point that is the
the RMS value of a window centered on that point. In that
case minima and maxima can be discussed and have intuitive
as well as utility value.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Lord Hasenpfeffer]

Martin Tillman wrote:

> CEP analyses a file by looking at specific time intervals (default 50ms)
> and calculating the RMS power for that interval. Someone might find
> that useful. It then reports the max and min values it found - meaning
> the max and min for the 50ms intervals. If you want to normalise your
> file based on the value of 50ms of it, please feel free.
>
> Then it takes all the RMS values for the file and averages them. This
> gives the average of all the values (just to be clear...). It also
> reports the total RMS power of the file, which is not the same as the
> average of all the values.

Normalize uses similar procedures and reports back its own Maximum RMS
level as the "level" of the song - and *that's* the level that it uses
as a guide for adjusting loudnesses, not the Average RMS level as
everyone has been assuming to be the case.

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[sgordon]

In rec.audio.tech Lord Hasenpfeffer > wrote:
: and the range is right there waiting for you to use it - I say, by all
: means, do.

But that is done by saturating the original recordings, long before
mastering. All the normalizing in the world is not going to increase
the dynamic range of the recording.

Scott

[Lord Hasenpfeffer]

sgordon wrote:
> In rec.audio.tech Lord Hasenpfeffer > wrote:
> : and the range is right there waiting for you to use it - I say, by all
> : means, do.
>
> But that is done by saturating the original recordings, long before
> mastering. All the normalizing in the world is not going to increase
> the dynamic range of the recording.

You are correct.

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Arny Krueger]

"Lord Hasenpfeffer" > wrote in message
...
> Martin Tillman wrote:
>
> > CEP analyses a file by looking at specific time intervals (default 50ms)
> > and calculating the RMS power for that interval. Someone might find
> > that useful. It then reports the max and min values it found - meaning
> > the max and min for the 50ms intervals. If you want to normalise your
> > file based on the value of 50ms of it, please feel free.
> >
> > Then it takes all the RMS values for the file and averages them. This
> > gives the average of all the values (just to be clear...). It also
> > reports the total RMS power of the file, which is not the same as the
> > average of all the values.
>
> Normalize uses similar procedures and reports back its own Maximum RMS
> level as the "level" of the song - and *that's* the level that it uses
> as a guide for adjusting loudnesses, not the Average RMS level as
> everyone has been assuming to be the case.

The CEP normalize function is based on peak levels - the sample in the data
being normalized with the largest magnitude.

[Bob Cain]

Arny Krueger wrote:
>
>
> The CEP normalize function is based on peak levels - the sample in the data
> being normalized with the largest magnitude.

Arny, CEP 2.x has a "Group Normalize" function available
from the multitrack view which works on multiple files and
utilizes RMS levels.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Lord Hasenpfeffer]

Arny Krueger wrote:

> As far as I can tell this is irrelevant to the discussion at hand which
> relates to 2-channel MP3s. I see it as just one of many red herrings that
> were dragged in, resulting in a complex and inconclusive discussion of a
> fairly simple problem.

And MiniDiscs...

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Bob Cain]

Arny Krueger wrote:
>
> "Bob Cain" > wrote in message
> ...
>
> > Arny Krueger wrote:
>
> > > The CEP normalize function is based on peak levels - the sample in the
> data
> > > being normalized with the largest magnitude.
>
> > Arny, CEP 2.x has a "Group Normalize" function available
> > from the multitrack view which works on multiple files and
> > utilizes RMS levels.
>
> As far as I can tell this is irrelevant to the discussion at hand which
> relates to 2-channel MP3s. I see it as just one of many red herrings that
> were dragged in, resulting in a complex and inconclusive discussion of a
> fairly simple problem.

Perhaps the thread is too long for your limited attention
span, and perhaps it has wandered about some, but initially
it was about normalizing groups of CD tracks prior to making
MP3's from them. The question of basing that on RMS levels
has been relevant from the beginning and I'm pointing out
how CEP can do that (which you said it couldn't.) Bad day,
Arny?


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[Arny Krueger]

"Paul Dormer" > wrote in message


> Normalizing a bunch of tracks together isn't a bad idea, but I would
> discourage newbies from normalizing individual tracks - software just
> doesn't understand the difference between ballads and trash metal..

IOW, normalizing is a poor choice as a means to match perceived loudness.

Matching perceived levels based on RMS values is generally more effective
than using peak levels. However, RMS measurements don't consider spectral
balance, which is very important to perceptions of perceived loudness.

[normanstrong]

Could I enter this discussion, or is private?

Today, rock and pop music is largely electronic and composed in the
mixing room. It's also compressed to the point of near death. How
one handles this is largely a matter of taste. Since it's already
been clipped, smashed and otherwise manhandled almost any method of
volume adjustment will work just fino.

In the case of classical music, I imagine a perfect recording is the
starting point--no compression, clipping, and the level of the
original is adjusted so that it has a full dynamic range, but as
little as possible wasted at the peak. This works well on the
original, but makes for some awkwardness in the release version. It's
entirely possible that a full symphony orchestra will have more than
one peak level: The apparent maximum output, and also the absolute
instantaneous peak that could easily run 10's of db above the apparent
maximum level, but only for a few milliseconds.

The human ear/brain is highly nonlinear. The shorter a peak is, the
less apt the brain is to give it full value. So, the solution is to
compress these very short peaks so that there's room left to bring up
the apparent level to where it sounds much the same as other
recordings.

A very good article on this subject is entitled "How can the headroom
of digital recordings be used optimally?" Written by Manfred Krause
and Holger Petersen, it's in the Journal of the Audio Engineering
Society, Vol 38, No.11, dated November 1990.

Norm Strong




"Bob Cain" > wrote in message
...
>
>
> Arny Krueger wrote:
> >
> > "Bob Cain" > wrote in message
> > ...
> >
> > > Arny Krueger wrote:
> >
> > > > The CEP normalize function is based on peak levels - the
sample in the
> > data
> > > > being normalized with the largest magnitude.
> >
> > > Arny, CEP 2.x has a "Group Normalize" function available
> > > from the multitrack view which works on multiple files and
> > > utilizes RMS levels.
> >
> > As far as I can tell this is irrelevant to the discussion at hand
which
> > relates to 2-channel MP3s. I see it as just one of many red
herrings that
> > were dragged in, resulting in a complex and inconclusive
discussion of a
> > fairly simple problem.
>
> Perhaps the thread is too long for your limited attention
> span, and perhaps it has wandered about some, but initially
> it was about normalizing groups of CD tracks prior to making
> MP3's from them. The question of basing that on RMS levels
> has been relevant from the beginning and I'm pointing out
> how CEP can do that (which you said it couldn't.) Bad day,
> Arny?
>
>
> Bob
> --
>
> "Things should be described as simply as possible, but no
> simpler."
>
> A. Einstein

[Lord Hasenpfeffer]

Paul Dormer wrote:

> Normalizing a bunch of tracks together isn't a bad idea, but I would
> discourage newbies from normalizing individual tracks - software just
> doesn't understand the difference between ballads and trash metal..

This is an excellent point. I remember when I got to my John Denver CDs
and Normalized them in my usual way. "Take Me Home, Country Roads"
sounded painfully wrong to my ears after that. Needless to say, I redid
those files at a much more conservative level.

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[George W.]

On Fri, 11 Jul 2003 23:19:42 -0500, Lord Hasenpfeffer wrote:

>Paul Dormer wrote:
>
>> Normalizing a bunch of tracks together isn't a bad idea, but I would
>> discourage newbies from normalizing individual tracks - software just
>> doesn't understand the difference between ballads and trash metal..
>
>This is an excellent point. I remember when I got to my John Denver CDs
>and Normalized them in my usual way. "Take Me Home, Country Roads"
>sounded painfully wrong to my ears after that. Needless to say, I redid
>those files at a much more conservative level.

Did you set your normalize value to 0 ? That's where Denver sounds
best.

You're welcome.

[Lord Hasenpfeffer]

George W. wrote:

> Did you set your normalize value to 0 ? That's where Denver sounds
> best.
>
> You're welcome.

Are you implying that John Denver's music sucks big green donkey dongs?

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Bob Cain]

Lord Hasenpfeffer wrote:
>
> Bob Cain wrote:
>
> > Perhaps the thread is too long for your limited attention
> > span, and perhaps it has wandered about some, but initially
> > it was about normalizing groups of CD tracks prior to making
> > MP3's from them.
>
> Actually, that's not entirely true - as I also do a lot of MiniDisc
> recording which is also lossy although to a much lesser extent than with
> common MP3s. That's why the thread in the other NG says "(With Lossy)"
> instead of "(With MP3)".

That's right, it was for ATRAC rather than MP3. Chalk it up
to my limited attention span. :-)


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

[John Albert]

While we're on the subject of "normalizing" (not responding to any previous
msg in particular)...

Can anyone recommend a freeware/shareware application for the Macintosh that
can do this?

Thanks,
- John

[Lord Hasenpfeffer]

Geoff Wood wrote:

> Almost any audio software should have this as one of it's primary features.

No, no... Normalize is in a class all by itself! :)

Myke

--

-================================-
Windows...It's rebootylicious!!!
-================================-

[Richard Kuschel]

>
>While we're on the subject of "normalizing" (not responding to any previous
>msg in particular)...
>
>Can anyone recommend a freeware/shareware application for the Macintosh that
>can do this?
>
>Thanks,
>- John
>

I think that Pro Tools Free can do this.
Richard H. Kuschel
"I canna change the law of physics."-----Scotty

[Richard D Pierce]

In article >,
Ken Bouchard > wrote:
>OK, since Nyquist was mentioned, I'll go ahead and ask this:
>According to Nyquists theorem, a waveform only has to be sampled at a rate
>TWICE it's highest frequency, right?

Wrong, it's more than twice.

>Hence, we sample most audio waves (for
>CD anyway) at 44.1 KHZ because thats about TWICE the highest frequency we
>humans can hear, right?
>OK, let's take a waveform that's got a sound in it with a frequency way up
>there like at 22 KHZ. (I've got REAL good hearing:-)
>How can that high frequency be faithfully reproduced if it's only being
>sampled a couple times?

Because if your waveform is at 22 kHz, any deviations from that
22 kHz sine wave MUST be at multiples of 22 kHz, and anything at
22.05 kHz and above MUST be eliminated by the prerequisite
filtering any sampling system must have.

Let's look at it more fundamentally. A 44.1 kHz system MUST have
its bandwidth limited to less than 22.05 kHz. This is simply the
correlary of saying that you must sample at more than twice a
waveform's frequency. If the bandwidth is limited to 22.05 kHz,
only one kind of 22 kHz waveform can possibly exist: a 22 kHz
sine wave. You can't have a 22 kHz square wave or triangle wave
or perfect tone burst because ALL of those waveforms have
components outside of the 22.05 kHz bandwidth that this system
has.

Given that, the number of points needed to PERFECTLY represent a
22 kHz SINE waveform with PERFECT fidelity is only slightly more
tha 2 per cycle. Only one sine wave can pass through those
points. Only one, and that's the original waveform. That's
essentially the non-methematical statement of the mathematical
proof of the sampling theorem.

Now, you are undoubtedly still wondering how this can be. If you
take these two points and "connect the dots," you don't get a 22
kHz sine wave. You get something else entirely. Well, if we were
to look at the spectrum of the signal represented by the
"connect-the-dot" waveform, we'd find that it would consist of
since components at 22 kHz and at many other frequencies, all
above 22.05 kHz. For example, you'd have a component at 22.1
kHz, one at 44 kHz, one at 44.2 kHz and so on and so forth. But,
all those extra components aren't permitted, since they all
violate the Nyquist criteria. So, as soonas we get rid of them
all, all but the 22 kHz component poof! we have our 22 kHz
waveform back again, just as it originally was.

So filtering is needed in two places: first, when we sample,
because Nyquist says we cannot sample wavforms equal to or
greater than the sampling frquency: That;s at the original A/D
stage. Second, when we convert back to the analog domain: the
extra components that result from the "connect-the-dots" or
"stair case" (or whatever representation) are outside the
Nyquist band and also must be eliminated.

It works just fine boith in theory and in practice.

--
| Dick Pierce |
| Professional Audio Development |
| 1-781/826-4953 Voice and FAX |
| |

[A E]

Harvey Gerst wrote:

> (Richard D Pierce) wrote:
>
> >Let's look at it more fundamentally. A 44.1 kHz system MUST have
> >its bandwidth limited to less than 22.05 kHz. This is simply the
> >correlary of saying that you must sample at more than twice a
> >waveform's frequency. If the bandwidth is limited to 22.05 kHz,
> >only one kind of 22 kHz waveform can possibly exist: a 22 kHz
> >sine wave. You can't have a 22 kHz square wave or triangle wave
> >or perfect tone burst because ALL of those waveforms have
> >components outside of the 22.05 kHz bandwidth that this system
> >has.
> >
> >It works just fine both in theory and in practice.
>
> And in practice, nothing above 11 kHz is gonna get through unscathed, unless
> it's a pure sine wave (and if it isn't, it will be after the filtering). If
> you're willing to go for just the first overtone, you can reproduce stuff
> accurately up to around a little above 7 kHz, but don't expect any more than the
> octave component. From around 5 kHz and down, you can start doing a credible
> job on sawtooth and triangle waves, but you hafta go a lot lower to do a square
> wave any justice. Doesn't mean that 44.1 sucks, but you should be aware of
> those limitations.

And all perfectly irrelevant because you will never hear that.

[Geoff Wood]

"A E" > wrote in message

> And all perfectly irrelevant because you will never hear that.
>

And is the same in any band-limited scenario ( not just digital) , as in
recording with real-world micophones, onto reel tapes, etc.

geoff