Lord Hasenpfeffer wrote:

Erwin Timmerman wrote:
Lord Hasenpfeffer wrote:

Despite all of this processing, all 8 final WAVs sound "normal" to me.
By the time I got to -40dB, I'd have expected the final MP3-WAV to
sound pretty crappy but it doesn't - at least not by any degree which I
would expect it to sound.

What does this tell you? Exactly.

It tells me that Normalize seems to treat WAVs with "kid gloves"
compared to Audacity. I've done some more tests in search of the noise
floor.

Do you know if Audacity does dithering, and if so, whether you've turned it on or
not?

If "Normalize" can reduce 90 dB in 16 bit and still produce something that even
resembles to a song, it must use dither. That by itself can make a huge
difference.

If you don't know that dither means:
http://www.pcrecording.com/dither.htm

All other reactions and conclusions are void if you didn't turn on dither in
Audacity (or if it doesn't have the option to do so).


With Audacity, I repeated (as suggested) a short segment of a WAV with 5
seconds between each segment. I then saved this original WAV to disk.
Then I highlighted all but the 1st segment, reduced amplification by
-5dB, hightlighted all but the 2st 2 segments, reduced by -5dB again,
highlighted all but the 1st 3 segments, reduced by -5dB again, and so
on.

This is not a good way of doing it. Either you're adding dither noise with every
step, or you're adding quantization noise with every step. That by itself will
make the more often processed wavs sound worse. To do this right you should
amplify each segment by itself with the amount needed. So don't select the whole
bunch, select only the segment that you're going to do, and then amplify with
-5dB, -10 dB, -15 dB, whatever is needed. Especially with very low level audio
the quantization or dither noise will become more and more audible, so it is
increasingly important to not add it cumulatively.


Meanwhile using just the starter WAV segment that I'd repeatedly
duplicated for the above test, I was able to drop the gain on it by as
much as -90dB using "normalize -g -90dB starter.wav" at a shell prompt,

Yeah, but giving the wavs a huge headstart by processing them only once.


http://www.mykec.com/mykec/images/20..._Me_Tender.png

Looks familiar.


I hope you will treat your generated mp3's the same way, as far as the
original copyright owners are concerned...

Absolutely. My MP3s stay with me. I have shared a few files with close
friends on occasion when the need has arisen to clarify or reinforce a
point brought up in conversation but never have I traded MP3s or posted
some online for all my friends to enjoy, etc..

OK, I wrote it just to make sure. You have to admit, putting names of all the
mp3's you have on a web site, saying you have a business with storing huge
amounts of mp3's on line, taking someone who talks about downloading mp3's
off-group, adds up a bit of suspicion. Good to see it was unfounded. Personally,
the last thing I would like to do is to help someone improve the quality of the
mp3's he's going to put on kazaa. But you've stated clearly enough that this
isn't the case. Thanks.

Greetings,

Erwin Timmerman

Man, Jonas, I am *so sorry* there isn't a way to correct so obvious a
mistake after one has clicked the Send button!

Just keep reading. Much brighter stuff still lies ahead!

I really do wish I hadn't have written what I did this time. What you
were actually trying to tell me didn't exactly click until about 10
seconds too late! I'm so sorry.

Myke

--

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Jonas Eckerman wrote:

How's that for a newbie?

Pretty good. :-)

Thanks. I *am* at least trying, y'see, to get all of this figured out
before everybody completely vacates the thread!

And you know what? It never occurred to me until late last night just
how much I'd forgotten the first rule of thumb every successful
programmer has to learn... Never enter into a den of engineers without
being fully dressed for battle! (And the same goes in reverse for
those on your side of the equation as well!)

If you *really* *wanna* *know* the *one thing* that's been seriously
flawed with both this thread - and the other one in the other newsgroup
before it - from the absolute very beginning, that's it! Hehehe!!! :-D

Maybe this is what Mr. Olhsson was trying to tell me all along!

The way I understood the docs, the final RMS level of the loudest parts of
the tracks combined will become -10dbFS.

What by his definition constitutes a "loudest part"? Where exactly does
he draw the line to even define what that is or what that means?

It would be instructive if you analyse a file you've told Normalize to
normalize (not in batch mode) to -10dBFS and see the average RMS of the
whole file really is -10dBFS or if, as I suspect, it is the loudest parts
of that file wich become -10dBFS.

Hmmm... Lemme post to you here what I get on my terminal display from
it. Let's just take out ol' Elvis Presley's "Burning Love" here for a
bit. Hold on just a sec... rip rip rip rip rip... OK!

The "-n" option causes Normalize to just read the peaks and levels of
the file and that's it. This is a 2 minute, 57 second song, btw....

[mykec@sillygoose Elvis2]$ normalize -n track19.cdda.wav

Computing levels...
level peak gain
-16.9302dBFS -1.0545dBFS 4.9302dB track19.cdda.wav

There. That took all of 2 seconds to scan.
Note how the level plus the gain = -12dBFS.

OK, now let's Normalize the file to my personally preferred level:

[mykec@sillygoose Elvis2]$ normalize -a -10dBFS track19.cdda.wav

Computing levels...
track19.cdda.wav 100% done, ETA 00:00:00 (batch 100% done, ETA 00:00:00)
Applying adjustment of 6.93dB to track19.cdda.wav...
track19.cdda.wav 100% done, ETA 00:00:00 (batch 100% done, ETA 00:00:00)

OK, there. All of 8 seconds were required to rescan and normalize this
particular WAV.

Now, let's repeat step 1 to see how things look now...

[mykec@sillygoose Elvis2]$ normalize -n track19.cdda.wav
Computing levels...
level peak gain
-10.1832dBFS -0.0244dBFS -1.8168dB track19.cdda.wav

And there you go. The level of the song is -10.1832dBFS.

Would you like to see some screenshots of this as well?

Myke


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Wow, Erwin! What a fascinating idea! I never thought of it.

As for the licensing is concerned... I think all a restaurant would need
is either a BMI or an ASCAP license and a way to track what songs were
played in the dining area or wherever. The publishing companies should
be able to handle everything else from there.

Are you an entrepreneur as well? Or was this just a lucky shot?

I like the way you think.

Myke


Erwin Timmerman wrote:
Lord Hasenpfeffer wrote:


And in case anyone still thinks I might be doing these hits this way
just as a hobby, that's only half correct. Once I've gone through all
of my CDs, I'll finally be ready to think more in terms of "high school
reunions" and "decade parties" - that is, if anyone'll have me to "do"
theirs. A lot of small communities don't have big budgets for big DJ
services at their small events and gatherings.


You could sell computers with your stuff installed (including the MP3's) to
restaurants, I bet. Of course that needs a lot of licensing going on, but it
might be worth checking out.

Good luck!

Erwin Timmerman



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Erwin Timmerman wrote:

Do you know if Audacity does dithering, and if so, whether you've turned it on or
not?

I have no clue. The version of Audacity I currently use is only v0.98
so the developer didn't even have it out of beta yet at the time I
downloaded and installed it - which was about a year or so ago now. It
seems to be sorely lacking in features for tests like these but for all
the kind of WAV editing I ever need to do, it's more than met my needs
so far. I'll look soon to see what version he's got it up to these days.

If "Normalize" can reduce 90 dB in 16 bit and still produce something that even
resembles to a song, it must use dither. That by itself can make a huge
difference.

If you don't know that dither means:
http://www.pcrecording.com/dither.htm

Thanks. I am familiar with the term "dithering" in only the graphical
image sense. GIF images use dithering to try and simulate
smooth-transitions (i.e. gradients) from one colour to another. But
with a palette of only 256 colours available, the results always pretty
pretty much suck.

All other reactions and conclusions are void if you didn't turn on dither in
Audacity (or if it doesn't have the option to do so).

This is just another thing I've got to look into then. Geez, that stack
gets taller by the minute it seems!

This is not a good way of doing it. Either you're adding dither noise with every
step, or you're adding quantization noise with every step.

Very well. I opted to do it that way because it was difficult to tell
where the 5 second breaks were located once the waveform is virtually
invisible! I'll try again and be more deliberate and careful. I've
got an idea for how to accomplish what you've said to do.

OK, I wrote it just to make sure.

You have to admit, putting names of all the mp3's you have on
a web site,

Oh, that's just a demonstration of one kind of thing I can do as a
web-programmer. Most of my work I do for other people. It's therefore
difficult to always come up with demonstration projects which are unique
to only me. That was just one thing I knew that I could do so I did it.
That's all. Plus the chronlogical arrangement of the lists is rather
unique and probably quite useful to a lot of people who're aware of its
existence on the web.

That page doesn't really look like much but there truly is a lot
behind-the-scenes which is required to make it happen, believe me!
One thing it does is demonstrate my skill when called to develop an
uncluttered, user-friendly, intuitive user interface. That's very
important in my business.

saying you have a business with storing huge amounts of mp3's on line,

That's in reference to a project I'm slowly but surely working on
building for with one of my clients. Any MP3s that go online for that
will definitely be files that he personally owns or legally controls for
the purpose of making it electronically available for download to the
public.

taking someone who talks about downloading mp3's off-group,

When did I do that?
I'm not familiar with that to which you're referring.

Personally, the last thing I would like to do is to help someone
improve the quality of the mp3's he's going to put on kazaa.

That's understandable. But I don't use or even have kazaa - and I
*won't* use or have it either. First off, I don't *need* it as I
already own a rather extensive collection of real CDs for source.
Secondly, there's a lot of effort involved in what I'm doing so the
*last* thing I'm going to want to do is start giving my stuff away.
Especially after I paid so much for the CDs in the first place and then
worked so hard to rip and encode them over such a long period of time.

It doesn't bother me that you've inquired about it either. We live in
very suspect times with regard to all of this and I completely
understand your concern.

Myke

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I was going to take you through a scenario that would put any ideas of LOUD
being a method to make better mp3s, but I just figured that A) it's a
holiday, and B) it's not worth the trouble, and C) everybody has been over
this in about 15 different ways in the first place.

Suffice it to say that an algorithm only does one thing, which is perform
the math. What's possible is that the parameters are variable, and thus the
overall output is variable dependant upon which parameters you choose. But
the algorithm won't perform different functions if the music itself is
louder or quieter. The results may well be different if extremes in volume
are used, but the perception of an algorithm doing something different due
to overall volume is plainly wrong.

But my original point was that it's not up to you to determine how someone
else's product, that you didn't have anything to do with during the
production phase, would sound better if you just did THIS or THAT. If
someone wants you to put up good, better, best mp3s, then it's not up to you
to change the product they give you and THEN make an mp3 of it. It's up to
you to go out and listen to all the possible products you could use in the
job and then make that purchase to give your clients THEIR PRODUCT in the
best mp3 form that you can. I said it before and I'm saying it again. You
are not a part of the creative process, and to alter their music to your own
tastes before you commit them to another process is destructive and is doing
a disservice to your clients. And it's not going to sit well for future
business because no one will be able to trust your judgement that THEIR
music is THEIR music and not subject to your whims and fancies. And if it
doesn't sit well with them, your reputation is going to be squat.

I'm not saying that your desire to do a good job for your clients isn't a
good thing. I'm saying that you have no rights, much less the experience,
to make judgement calls that affect your client's interests. Perhaps the
smarter thing to do so that you don't dig yourself into a grave is to simply
require that all product an artist wants put up on your website be submitted
as an mp3. And again, if you aren't working directly for clients that have
the right to make the determination of having their works put out on the web
in mp3 format, you're simply part of the problem and you won't find a
solution here. I can't make it any simpler.

And again, if you want the specifics on the algorithm, then by all means, go
to someone who does the coding and understands the principles. You're just
spinning all our wheels here and it doesn't appear that you've learned a
single thing. Take that as abusive if you want, but it's about as plain as
I can speak without coming down on you like some newbie that wants to tell
me about how I'm wrong because I love my Crest and he's right because he
love's his Behringer. The obvious answer to that conundrum is that both of
us are right, but I have more experience invested in my purchase. Nothing
hard nosed, nothing negative, just different perspectives and experience
levels.

--


Roger W. Norman
SirMusic Studio

301-585-4681




"Lord Hasenpfeffer" wrote in message
...
I've got to go back to work romorrow...

Duty calls! I understand.

For the past couple of days Myke, there have only been a few of us, and
quite truthfully, many of the regular and more experienced members of
this
forum never even showed their faces.

There has been way too much back and forth as it is, God forbid *more*
should join! Besides, why should I wish for *everyone* to know that I
know how to walk a wild pigeon?

For all intents and purposes, I do believe that the original subject
line killed the thread in the very beginning.

The subject line was carefully chosen and intended to serve as a
constant reminder of the topic at hand. Too many off-topic tangents in
the previous thread created mass confusion and no resolution. I was
told to expect a "train wreck" if I brought this topic here. If the
subject line "killed the thread" as you say, then I suppose it was
successful in averting that collision which had been prophesied.

Second, do yourself a favor and re-read the thread

Absolutely. The entire thing will soon be added to my private Usenet
archives "forever" as it is on Google.

paying close attention to those who only posted one to three times.

Very well.

Albiet that some of those were perhaps curt and brief, in most cases
(excepting some subject changes) more truth lays within those single
or small contributions than anywhere but the past few posts where you
seem to be getting a grip on your own software.

Those who speak the least oft say the most.

Third, it really is better to approach those who you must consider
as being more experienced, with questions rather than statements.

I believe you.

I've hung in here as an admitted MP3 novice hoping to pick up more info
without having to spend anywhere near as much time in research mode
as I actually did.

Time spent learning is never wasted.

Unfortunately, due to the train wreck (it takes much less here to make
people 'jump' the train than it may elsewhere) steady input from the
'pros' that you were seeking only _slightly_ reared it's head.

I would surely like to know your definition of a r.a.p. train wreck. As
heated and as occasionally vicious as things became at times in this
thread, this was a *smooth ride* compared to what happened previously in
the other newsgroup(s). I came to get escape from a train wreck in
hopes of actually discovering some truth - and I do believe we found it.
Even today, some hangers-on in the other groups still believe the
weirdest things and are behaving in the most rude and asinine ways. It
is that rudeness and stupidity which I came here to escape - and most
successfully I might add. We actually made progress in this forum
whether it is readily apparent or not at this time.

Lastly....
We are now but three or four voices.... some bickering, some flaming,
some searching, but the dozens upon dozens of people who could help
are simply not participating.

Oh well. If you don't vote, you can't bitch, I always say.

I guess this is as good of a time as any to be departing.

I've been thinking the same thing for myself, however, I don't recall
ever saying that I intended to vacate the newsgroup unless it was as
impossible to be received here as it was to be in the other groups.
Sure, we've had some stupid moments but none so pathetic here as what I
experienced "over there".

I know that I picked up enough (through research and the
few wise voices who shared a moment of their time) to have considered
it a positive holiday.

That's good to know. I do hope the week was able to afford you more
rest than I lost as a result of it.

I'm sorry Myke, that all you got here was a much smaller train wreck
than you got elsewhere, but I'm glad it wasn't obvious.

You're really confusing me here. "George W." is responsible for
prophesying the train wreck in this newsgroup. I accepted his challenge
and came here to defy both him and his opinion of the members of this
newsgroup.

Best of luck to you and thanks again for reminding me that this old
body and brain need be kept in better form, or they will fall behind in
the rapid changing of my questionable industry.

All the best to you and keep up the low-level goading! It's far more
respectable, appreciable and effective than flaming and all of that.

"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.....

See? The wreck was George's vision, not mine.

And who was it wrote the response?

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

Bob

Why, it was none other than (gasp!) the venerable *BOB CAIN*!!! ;-)

Nice to have "met" you David.

If you ever feel a need to goad, you know who to come to! *LOL*

Take care,

Over,

Myke

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Erwin Timmerman wrote:

Quite easy. Take the original wav. Trim to the part you want to use. Generate 1 (or 5,
or whatever suits you) second of silence. Then select the whole bunch, copy, and loop
paste the number of times you want to test. This will end you up with a wave file with
visually good definable "chunks" of music. Display the complete wave. Select the second
chunk, amplify by -5. Select the third chunk, amplify by -10 etc etc etc... As the left
over (not yet processed) chunks are still at full volume it is easy to select them.

Aside from the error in my attenuation method, this is how I did it.
The problem I had was knowing where the breaks between parts were while
I was trying to reamplify them.

I suppose I should make 2 copies of the original and use one of them as
a guide for knowing where the breaks between segments are actually
located in the one that I'm actually working with.

Myke

--

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"Lord Hasenpfeffer" wrote in message
...
Wow, Erwin! What a fascinating idea! I never thought of it.

As for the licensing is concerned... I think all a restaurant would need
is either a BMI or an ASCAP license and a way to track what songs were
played in the dining area or wherever. The publishing companies should
be able to handle everything else from there.

Actually there are zillions of companies already doing exactly that !

geoff

Roger W. Norman wrote:

I was going to take you through a scenario that would put any ideas of LOUD
being a method to make better mp3s, but I just figured that A) it's a
holiday, and B) it's not worth the trouble, and C) everybody has been over
this in about 15 different ways in the first place.

Sounds perfectly reasonable to me.

The results may well be different if extremes in volume
are used, but the perception of an algorithm doing something different due
to overall volume is plainly wrong.

When you say "algorithm" here, are you referring to ATH filtering during
lossy encoding? After conducting my own first test, I'm very
*inclined*, though still not absolutely *convinced*, that Geoff's
initial assertion that +4dB or +5dB isn't enough of a difference to make
a difference, lossy or not.

The only reason I ever considered higher amplitudes to be "helping" with
regard to lossy compression is that the effects of ATH-based filtering
are directly determined by the amplitudes of the frequencies being
filtered or retained.

To me, the relationships in play between a frequency's amplitude and its
ability to survive passing through an ATH filter seem quite obvious and
direct - but much larger differences in amplitude than those which I'm
creating are required in order for the difference in fidelity to be
appreciable. Geoff said this. My tests appear to confirm this. I
would have been more willing to accept Geoff's message earlier had his
delivery not been seasoned with so much emotion and "out of the blue"
speculation.

But my original point was that it's not up to you to determine how someone
else's product, that you didn't have anything to do with during the
production phase, would sound better if you just did THIS or THAT. If
someone wants you to put up good, better, best mp3s, then it's not up to you
to change the product they give you and THEN make an mp3 of it. It's up to
you to go out and listen to all the possible products you could use in the
job and then make that purchase to give your clients THEIR PRODUCT in the
best mp3 form that you can. I said it before and I'm saying it again. You
are not a part of the creative process, and to alter their music to your own
tastes before you commit them to another process is destructive and is doing
a disservice to your clients. And it's not going to sit well for future
business because no one will be able to trust your judgement that THEIR
music is THEIR music and not subject to your whims and fancies. And if it
doesn't sit well with them, your reputation is going to be squat.

To this end I simply consider how I would feel if someone decided to
alter my own music in similar ways, therefore, you are quite correct in
terms of reputation preservation and the like.

With my particular project in mind, however, the assumption that I am
not a part of the creative process will not always be accurate. In many
cases, depending on the financial state and the technical savvy
possessed by the client being served, I may indeed be *entirely*
responsible for the decisions required to produce the original recording
from which all subsequent MP3s will result.

I'm not saying that your desire to do a good job for your clients isn't a
good thing. I'm saying that you have no rights, much less the experience,
to make judgement calls that affect your client's interests.

Because my role would not always be limited to just "webmaster" and "MP3
encoder", the truth of this should actually fluctuate on a
client-by-client basis. Nevertheless, I clearly understand what you're
saying and why you're saying it, and your advice is sound.

Perhaps the smarter thing to do so that you don't dig yourself into
a grave is to simply require that all product an artist wants put up
on your website be submitted as an mp3.

The current plan already allows for this, however, we will also need to
make "MP3 production" available as an additional service. A lot of
people don't understand how MP3s are made and may also not have the
resources available to create them on their own.

And again, if you aren't working directly for clients that have
the right to make the determination of having their works put out on the web
in mp3 format, you're simply part of the problem and you won't find a
solution here. I can't make it any simpler.

An essential part of the plan is to be certain that the
owners/maintainers of the website have complete legal authority to make
available the files which are placed online for distribution to the
public. If said authority is not obtained, the files in question simply
will not be placed online.

And again, if you want the specifics on the algorithm, then by all means, go
to someone who does the coding and understands the principles.

Of course.

You're just spinning all our wheels here and it doesn't appear that you've
learned a single thing.

Sure I have. There's just been so much to take into consideration that
I've not had time yet to absorb everything that's been said and suggested.

Take that as abusive if you want, but it's about as plain as
I can speak without coming down on you like some newbie that wants to tell
me about how I'm wrong because I love my Crest and he's right because he
love's his Behringer. The obvious answer to that conundrum is that both of
us are right, but I have more experience invested in my purchase. Nothing
hard nosed, nothing negative, just different perspectives and experience
levels.

Your input has not been perceived as being anything even remotely
resembling abusive or hard-nosed commentary. Honestly, I'm confused as
to why you think it may have been. It's all perfectly reasonable and
wholly understandable on this end.

Thank you!

Myke

--

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Erwin Timmerman wrote:

Well, the one doing the actual copying (you in this case, if you are preparing the
system for the given restaurant) has to pay a license fee as well.

Yes, I'd say that, um,... You, sir, are corrrrrrect! (a la Ed MacMahon) :-)

I'm not an entrepeneur. I have plenty of ideas, but never take actions to make
them into money.

Been there!

I have a nice day job.

As they say, "Necessity is the mother of invention."

For example: I don't consider my playing or my music
releaseworthy. Yet I see people actually making money
with stuff I wouldn't dare to release (quality-wise,
not music-style-wise as that is just a matter of taste).
They are doing something right that I am doing wrong.

Hmm-hmm-hmm... Whaddaya think it is (or might be)?

Myke

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Jonas Eckerman wrote:

What by his definition constitutes a "loudest part"? Where exactly
does he draw the line to even define what that is or what that means?

That's a good question. Here's two quotes from the documentation:

I'm going to spend a little more time on this before posting any real
reply to it. This is indeed kinda weird - and probably commands an
email to the author of the program.

Meanwhile, I'll grab that WAV and see what happens...

Thanks for the link.

Myke

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Jonas Eckerman wrote:

Download this file:
http://vvv.truls.org/temp/RMS-test.wav

Run it through Normalize and tell us what numbers Normalize showed.

These are the values shown by CoolEdit's statistics:

--8--
window = 50ms:
0dB = FS Sine Wave
Minimum RMS Power: -31.47dB
Maximum RMS Power: -6dB
Average RMS Power: -22.35dB
Total RMS Power: -15.79
--8--
window = 50ms:
0dB = FS Square Wave
Minimum RMS Power: -34.48dB
Maximum RMS Power: -9.01dB
Average RMS Power: -25.36dB
Total RMS Power: -18.8
--8--

[mykec@sillygoose mykec]$ normalize -n RMS-test.wav
Computing levels...
level peak gain
-9.0106dBFS -6.0002dBFS -2.9894dB RMS-test.wav

Looks to me like we're talking *Maximum* RMS power here.
Without really thinking too much about it, I'd venture to guess that
that's why the limiting, if any, which I've seen in my "improvements" to
my original WAVs is minimal at best compared to what would be occurring
otherwise if Average RMS Power levels were being considered.

And your assessment is?

Myke

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Geoff Wood wrote:


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"Lord Hasenpfeffer" wrote in message
...
Geoff Wood wrote:


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Sorry, I don't understand this rash of posts such as that quoted in full
above. Are they accidental posts, or some sort of childish snub ?


geoff

Lord Hasenpfeffer wrote:

P.S. By the way, how's the ol' tolerance for gore holding up these days?
;-)

Whaddaya t'ink, I'm some kinda pigeon or sumpm'? :-)


Bob
--

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

A. Einstein

Looks to me like we're talking *Maximum* RMS power here.

Yes, it does.

We should note that the wav file I created was a very simple file though,
and very different from music.

Without really thinking too much about it, I'd venture to guess that
that's why the limiting, if any, which I've seen in my "improvements" to
my original WAVs is minimal at best compared to what would be occurring
otherwise if Average RMS Power levels were being considered.

Exactly.

That was why I thought you didn't really want your music normalized to an
average -10dBFS RMS. And now we see that I probably was correct in this.

The sound you like is closer to the reasult of normalizing to a maximum of
-10dBFS RMS. Huge difference.

Regards
/Jonas

On Sat, 05 Jul 2003 13:04:29 GMT, Jonas Eckerman wrote:

As I can see, the relative loudness of all the tracks are not well
preserved.

I don't know what that picture shows, but according to the documentation of
Normalize it's "batch" mode is meant specifically for preserving the
relative loudness of all tracks. In this mode it treats all the tracks as
one long audio stream.

But that's not what he said he did. He said he normalised each track
individually, when he apparently whipped the arse of MSFL by
'remastering' DSOTM, though he didn't mention that he didn't use batch
mode when he originally made this claim.


While normalising any track to 0dbFS is harmless in itself, doing what
you've done to DSOTM (normalise and potentially limit in order to make
the RMS level of each individual track -10dbFS), is truly idiotic.

And, our little limited test indicates that Normalize does not do that. It
seems that with Myke's settings it normalizes the tracks, seen as one
single stream, to a *maximum* (or close to maximum) -10dBFS RMS.

OK, I admit to not reading all the posts in the multiple threads about
this issue, and to skim reading many of the others, but Normalize (sic),
as used by Myke in recent times, but perhaps not at the moment, will
limit or clip, depending on user settings, when normalising to an RMS
value if the file contains such samples. Are you saying that Normalize
(sic) will normalise to a lower RMS value than specified by the user if
the file contains samples that would otherwise cause clipping or
limiting? If so, fine, but that's not how Myke was using it when he
whipped the arse of MSFL - I specifically recall him mentioning the
limit option around that time.

Jonas Eckerman wrote:


That was why I thought you didn't really want your music normalized to an
average -10dBFS RMS. And now we see that I probably was correct in this.

I thought dBFS was a peak voltage indication reading and in contrast,
RMS - root mean square - was a mathmatical average power reading. It
would seem to me that dBFS and RMS don't belong in the same measurement.

I never dreamed this thread would go this far...


AT

The sound you like is closer to the reasult of normalizing to a maximum of
-10dBFS RMS. Huge difference.

Regards
/Jonas

But that's not what he said he did.

Well... That's one of the main problems with this thread. Myke hasn't been
clear on either the terminology or how his Normalize app works. So a lot of
the time he has said he did one thing while in actuality he did something
else.

OK, I admit to not reading all the posts in the multiple threads about

Wich is quite understandable. This thread has been extremely convoluting
and confusing at times.

At least we have actualluy gotten through a lot of it, and some things have
become clearer in recent posts. For example what Normilize actually does.
It hasn't become clear, but a lot clearer than it was at the start of the
thread. :-)

will limit or clip, depending on user settings, when normalising to an
RMS value if the file contains such samples. Are you saying that
Normalize (sic) will normalise to a lower RMS value than specified by
the user if the file contains samples that would otherwise cause
clipping or limiting?

No.

I'm saying that Normalize will not normalize based on the average RMS of
the file. It normalizes based on the loudest parts of the file, wich of
course is a completely different thing.

Exactly what it counts as the loudest parts of the file is unclear. In a
very limited test we just did, the RMS level Normilize based it's operation
on was the same as CoolEdit showed as the maximum RMS of the file. That
file wasn't representational of music though (it was just a sine way, 1
second at -6dB and 9 seconds at -31.46dB).

The Normilize app does limit in order to avoid digital clipping if
necessary.

But obviously normalizing to a maximum -10dBFS RMS (or somthing similar) is
not nearly as brutal as normalizing to an average -10dBFS would be.

If so, fine, but that's not how Myke was using
it when he whipped the arse of MSFL - I specifically recall him
mentioning the limit option around that time.

Myke did say that he normalized to an average 10dBFS RMS with limiting
applied, but he wasn't. He normilized to something similar to a maximum of
-10dBFS with limiting applied. He simply didn't know how the Normalize
application really worked.

He did try to tell us that when he normalized a complete CD he did it in a
way that kept the relative loudness of the tracks, but as he thought that
kind of operation was implicit in the use of term "batch" he completely
failed to explain it to us in the beginning of the thread.

As the thread then became so large, the explanations and findings necessary
to understanding what Myke actually did became hidden in variuous branches
of it.

One revelation was finding the documentation for Normalize. It doesn't only
explain that Myke wasn't doing what he said he was doing, it also partly
explained why Mike's terminology was different from everyone else's.

An important quote from the docs:
--8--
Please note that I'm not a recording engineer or an electrical engineer, so
my signal processing theory may be off. I'd be glad to hear from any signal
processing wizards if I've made faulty assumptions regarding signal power,
perceived volume, or any of that fun signal theory stuff.
--8--

The rest is at:
http://www1.cs.columbia.edu/~cvaill/...ze/README.html

Regards
/Jonas Eckerman

Martin Tillman wrote:

I "batch Normalized" the entire set of 10 separate WAVs from that CD to
-10dBFS and then loaded then all into Audacity end-to-end so I could
then take the pretty picture.

Ah, so that's how you managed to bugger up the dynamics. Shame you
didn't respond to me when I pointed this out buggeration days and days
ago.

The dynamics are not "buggered up".

Correct me if my memory has failed me, but there aren't many gaps
between tracks on DSOTM. How do the jumps in level at the joins sound,
given that each track has almost certainly had its gain increased by a
different value from the one that precedes and succeeds it?

You need to read the thread before you post this kinda crap.

While normalising any track to 0dbFS is harmless in itself, doing what
you've done to DSOTM (normalise and potentially limit in order to make
the RMS level of each individual track -10dbFS), is truly idiotic.

You need to read the thread before you post this kinda crap.

Myke

--

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Jonas Eckerman wrote:

We should note that the wav file I created was a very simple file
though, and very different from music.

But the principles are the same.

Without really thinking too much about it, I'd venture to guess
that that's why the limiting, if any, which I've seen in my
"improvements" to my original WAVs is minimal at best compared to
what would be occurring otherwise if Average RMS Power levels were
being considered.

Exactly.

That was why I thought you didn't really want your music normalized
to an average -10dBFS RMS. And now we see that I probably was correct
in this.

Well, until these discussions began, I didn't know or even understand
the term "RMS" so all I knew based upon what I've been *hearing* is that
-10dBFS is usually a perfectly fine "level" for my tastes.

I believe it was Geoff who suggested that I start using the term "RMS
level" for clarification, so to please him I did. I didn't know until
you submitted your readings for that WAV yesterday that there was also
Maximum and Minimum RMS levels in addition to Average RMS! So, Jargon
strikes yet *again* and causes even more confusion and frustration.

The sound you like is closer to the reasult of normalizing to a
maximum of -10dBFS RMS. Huge difference.

Yeah, I guess *so*.

So, despite the inherent evil in my habitual tamperings with the sound
from a professionally technical point of view, does this "huge
difference" make what I've been saying all along seem just a little more
sensible and harmless (at least for my purpose in making "louder/better"
MP3s) than it ever did before? Because based upon all the listening to
my files that I have done, I've yet to create an aurally offensive,
pop/rock MP3 by going with my beloved -10dBFS setting - and I have
maintained this all along, despite all the rantings and condemnation.

Myke

--

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Roger W. Norman wrote:

OR, obviously, if the band doesn't have a problem with you taking a
CD of their product and doing whatever you want with it to make mp3s,
then you're fine.

Most if not all of the band's we're going to be working with are
unsigned and looking for a better way to obtain any exposure at all.
The extent of our production services would to be to assist in creating
affordable, amateur demos which would are better than *nothing* if
nothing is all they currently have by which to present themselves; a
"first steps" kinda thing at best. Once they have enough resources
available to afford hiring professionals, they can do so.

However, as you can see, I'm still coming up with somewhat different
scenarios because out of all these posts, you really haven't
specifically said what your goal was, whom you were working with
(titles, not people's names) and such.

I certainly understand the difficulty in your position in that regard.
It's a little hard for me to since I'm still working with the client and
using a lot of what's being discovered via this discussion as
information upon which to base more specific future decisions. A lot is
still "up in the air" with this project as for exactly what we're going
to be doing and how we're going to do it. Right now we have the basic
idea down and are working to formulate a strategy for implementation.

If we all look like we're talking off the topic, perhaps it's because
we haven't really had this type of specific information to work with
in formulating answers.

This is true - and I have been taking this into consideration all along.
Sometimes the replies are off-topic yet still useful for other
additional topics which relate to my other projects. Other times,
they're "dead on". The only times I risk losing my patience is when
people post unwarranted, rude comments stating that I'm an idiot, etc.
Those people are just looking to prove themselves superior and are not
here to help. The sooner they leave the discussion, the better.

I mean, over this thread I've recommended for you to talk with the
developers of the mp3 specs, the coders have been suggested by
others, talking with people whom are in the know about how to mix for
mp3, and even downright calling you on some perhaps misinformed
assumptions.

Unfortunately, the thread has kept me so occupied, however, that I
simply haven't had time enough in each day to do all of that - but I
will. I definitely have my homework cut out for me!

And, of course, getting called on our own misinformed assumptions
didn't help any.

As long as truth prevails in the end, it doesn't matter to me.

This thread has so far been a great example of miscommunications
throughout, or maybe that's just my perception.

No, you're quite right - and my experience with the other group made
this quite clear to me before I spawned this discussion with my
"hypothesis". I didn't want David to even begin looking at my
screenshots before I knew that everyone involved was "clear on the
terms" but he Googled them out anyway so there wasn't much I could do
about it.

But ifyou're working WITHIN the creative process, then that was
either my greatest misconstruence, or your worst presentation of
information! g

My sin of omission, perhaps? g

So still, what's it gonna be? Theory of perceptual encoding, or
pratical application? g

Hehe... Being not an audio professional, I generally weigh in on the
side of practical application because that's from the money's ultimately
going to come. However, understanding the theory is always helpful as
well when it comes to making decisions about things where practical
application says "either way is fine". In such cases, I always prefer
to go with what is more technically correct than "I don't know. Guess
I'll do this instead of that" - even if the difference is not readily
apparent. There's always a certain level of satisfaction in knowing
that you've "done the right thing" even if no one else can tell.

Thanks for all your input. It has not been wasted on me.

Myke

--

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Jonas Eckerman wrote:

As I can see, the relative loudness of all the tracks are not well
preserved.

I don't know what that picture shows, but according to the documentation of
Normalize it's "batch" mode is meant specifically for preserving the
relative loudness of all tracks. In this mode it treats all the tracks as
one long audio stream.

You are absolutely correct about that, Jonas.

And, our little limited test indicates that Normalize does not do that. It
seems that with Myke's settings it normalizes the tracks, seen as one
single stream, to a *maximum* (or close to maximum) -10dBFS RMS.

Thank you.

As other's have pointed out, this whole thread has been an excanple of
miscommunication. If we had had access to the docs for Normalize from the
start, the thread would probably have been very different.

I have noticed in both threads - though definitely more in the other,
previous one - that a lot of the miscommunication comes from people
posting immediately to older messages from when terms were still
misunderstood and problems were still unresolved. If they'd just keep
reading a bit before they spout off on an already resolved issue, they'd
see just how pointless some of their rantings and ravings have been.

Myke

--

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On Sat, 05 Jul 2003 10:36:00 -0500, Lord Hasenpfeffer wrote:

Martin Tillman wrote:

I "batch Normalized" the entire set of 10 separate WAVs from that CD to
-10dBFS and then loaded then all into Audacity end-to-end so I could
then take the pretty picture.

Ah, so that's how you managed to bugger up the dynamics. Shame you
didn't respond to me when I pointed this out buggeration days and days
ago.

The dynamics are not "buggered up".

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

To those reading this who haven't seen the screenshot, in the original
MFSL CD, the level at 11' is significantly below that at 24.5'. In the
arse whipping Hasenpfeffer remaster, the relative levels are reversed.

As there is no scale on the screenshot it is impossible to know real
values, but I'd guess that the level at 11' is 4dB below that at 24.5 on
the original, and 2dB up on the Hasenpfeffer remaster - a whopping 6dB
difference.

So, you don't feel like calling that buggering up the dynamics? Think
carefully before answering, you *are* in the company of professional
audio people, including me.


Correct me if my memory has failed me, but there aren't many gaps
between tracks on DSOTM. How do the jumps in level at the joins sound,
given that each track has almost certainly had its gain increased by a
different value from the one that precedes and succeeds it?

You need to read the thread before you post this kinda crap.

Did you not say that you normalised each wav individually, then joined
them up again?
------------------------------------------------------------------------
Message-ID:

Well, just so ya know... That screenshot of "Dark Side..." with the
clock, the cash, and the paper? Well, *that* is what happens when you
"batch Normalize" an entire set of WAVs from a single album -10dBFS.

I didn't run Normalize across the whole album as a single WAV.

I "batch Normalized" the entire set of 10 separate WAVs from that CD to
-10dBFS and then loaded then all into Audacity end-to-end so I could
then take the pretty picture.
------------------------------------------------------------------------

Hmm... You didn't... Serves me right for skimming.

So, explain the level discrepancy. Deny it, and I'll have to post a
damn screenshot, together with ugly graphics.

Jonas Eckerman wrote:

OK, I admit to not reading all the posts in the multiple threads about

Wich is quite understandable. This thread has been extremely convoluting
and confusing at times.

Yes it has. I've had a hell of a time just keeping up with it myself -
and all the rude interjections from people such as Martin who've *not*
worked as hard as I have to keep up with it have been extremely annoying
to say the least.

I'm saying that Normalize will not normalize based on the average RMS of
the file. It normalizes based on the loudest parts of the file, wich of
course is a completely different thing.

Just *how* different it is, though, is what's got me bugged now. So far
you've said it's a "huge difference" - and as polite and honest as
you've been with me throughout this ordeal, I'm certainly willing to
believe you - but I'd still like something a little more specific to go
on before we call this whole thing a wrap.

Myke did say that he normalized to an average 10dBFS RMS with limiting
applied, but he wasn't. He normilized to something similar to a maximum of
-10dBFS with limiting applied. He simply didn't know how the Normalize
application really worked.

But I've got a much tighter grip on its reality now than I've ever had
before, thanks to you!

He did try to tell us that when he normalized a complete CD he did it in a
way that kept the relative loudness of the tracks, but as he thought that
kind of operation was implicit in the use of term "batch" he completely
failed to explain it to us in the beginning of the thread.

Gee it feels nice to be so completely understood for once!

As the thread then became so large, the explanations and findings necessary
to understanding what Myke actually did became hidden in variuous branches
of it.

I'll say they did!

One revelation was finding the documentation for Normalize. It doesn't only
explain that Myke wasn't doing what he said he was doing, it also partly
explained why Mike's terminology was different from everyone else's.

Bingo.

Please note that I'm not a recording engineer or an electrical engineer, so
my signal processing theory may be off. I'd be glad to hear from any signal
processing wizards if I've made faulty assumptions regarding signal power,
perceived volume, or any of that fun signal theory stuff.

Again, it all comes down to programmers vs. engineers. These two really
do need to learn how better to get along with one another - especially
as we all depend on each others specialized talents and abilities just
to make it through our daily lives.

I'm sorry I don't dream of logarithms in my sleep, but that's just the
way I'm made and I can't say I've ever felt a need to change in that
regard.

Myke

--

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Lord Hasenpfeffer wrote in
:

No, not at all. And in fact, I've done it already.

Please see my next reply to your message.

Yes, I replied before I saw that.

Do you have these files to listen to anywhere online?

Regards,
Mark

--
http://www.marktaw.com/

http://www.prosoundreview.com/
User reviews of pro audio gear

Mark T. Wieczorek wrote:

Do you have these files to listen to anywhere online?

The WAVs? Or The MP3s? Or both?

Unfortunately, I've deleted them already.

But I can make them again if you'll tell me what you'd like to examine.

Myke

--

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On Sat, 05 Jul 2003 19:58:49 -0500, Lord Hasenpfeffer wrote:

Martin Tillman wrote:

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

snip

So, explain the level discrepancy. Deny it, and I'll have to post a
damn screenshot, together with ugly graphics.

When you have that much data being packed into a single frame, it is not
practical to believe *everything* you see because the resolution of the
data being represented exceeds the resolution of the image.

Hmm... If that screenshot is distorted so much due to the scaling that
the points I'm talking about AREN'T so hugely different, then, well,
words fail me.

If you'd like, I will re-rip and re-batch-normalize the WAVs from that
CD and provide close-ups of those points in particular.

Yes, I'd like, (or, even better, email me mp3s of the two versions those
two areas so I can see for myself in CoolEditPro).

But haven't you still got your 'remaster'? Even the mp3 would do - it's
not going to look so different from the wav. I'd hate to think that a
recreation of your 'remaster' would be any different from the original
remaster - can you be absolutely certain of your original settings in
Normalize?

"Artie Turner" wrote in message
m...
Jonas Eckerman wrote:


That was why I thought you didn't really want your music normalized to
an
average -10dBFS RMS. And now we see that I probably was correct in this.

I thought dBFS was a peak voltage indication reading and in contrast,
RMS - root mean square - was a mathmatical average power reading. It
would seem to me that dBFS and RMS don't belong in the same measurement.

I never dreamed this thread would go this far...


Given that the RMS of a single cycle sinewave is 0.7071 of the peak voltage,
there is a simplistic relationship there ...


geoff

Geoff Wood wrote:
"Artie Turner" wrote in message
m...

Jonas Eckerman wrote:


That was why I thought you didn't really want your music normalized to

an

average -10dBFS RMS. And now we see that I probably was correct in this.

I thought dBFS was a peak voltage indication reading and in contrast,
RMS - root mean square - was a mathmatical average power reading. It
would seem to me that dBFS and RMS don't belong in the same measurement.

I never dreamed this thread would go this far...



Given that the RMS of a single cycle sinewave is 0.7071 of the peak voltage,
there is a simplistic relationship there ...

Then it's either redundant, misleading or both to have both RMS and dBFS
in the same reading, especially if you're talking about the subjective
loudness of program material and not sine waves.


geoff

Martin Tillman wrote:

Hmm... If that screenshot is distorted so much due to the scaling that
the points I'm talking about AREN'T so hugely different, then, well,
words fail me.

Many times with Audacity I've seen peaks which appear higher than they
really are because pixel resolution is only so good. Zooming in on the
waveform solves this.

If you'd like, I will re-rip and re-batch-normalize the WAVs from that
CD and provide close-ups of those points in particular.

Yes, I'd like, (or, even better, email me mp3s of the two versions those
two areas so I can see for myself in CoolEditPro).

We'll start with the screenshots.

But haven't you still got your 'remaster'?

Not the WAVs, no. If I want to hear CD-quality audio, I play the CD.

I have stated many times that my effort here with Normalize is to create
"better sounding" MP3s (and MiniDiscs on occasion where MP3Gain is useless).

Even the mp3 would do - it's not going to look so different from the wav.

Yeah, but what ****wit would send you an MP3 in a discussion like this?

I'd hate to think that a recreation of your 'remaster' would be any
different from the original remaster - can you be absolutely certain
of your original settings in Normalize?

More specifically?

Myke

--

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David Morgan (MAMS) wrote:

It is still *very* unclear as to what kind of Linux 'command line' options
there are in this software called "Normalize".

Ask and ye shall receive. (I can post the entire man page which came
with my version of Normalize if you'd care to see it.)

Unfortunately with these screenshots, we haven't been seeing the same
song(s) used as a continuous reference.

Nobody's made any particular suggestions yet in that regard.

Myke

--

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"Lord Hasenpfeffer" wrote in message ...
David Morgan (MAMS) wrote:

It is still *very* unclear as to what kind of Linux 'command line' options
there are in this software called "Normalize".

Ask and ye shall receive. (I can post the entire man page which came
with my version of Normalize if you'd care to see it.)

Sure.

Unfortunately with these screenshots, we haven't been seeing the same
song(s) used as a continuous reference.

Nobody's made any particular suggestions yet in that regard.


Destroy something on purpose. RMS normalize an older song to say
-6dBFS. Find the highest average RMS value area of the normalized
song and start some screen shots from about 30 seconds in duration,
expanding the .wav in each subsequent shot until you reach only a
second or so in duration. Provide several incremental screen shots
of this alonf with a before and after of the entire song. ....Just an idea
to start with so we can try to see what the software is doing and then
ask for other, supportive screenshots later.


--
David Morgan (MAMS)
http://www.m-a-m-s.com
http://www.artisan-recordingstudio.com

Jonas Eckerman wrote:

Below are the values CoolEdit shows for the file
http://mmm.truls.org/temp/AAahAAA-She(3).wav

--8--
0dB = FS Sine Wave
Using RMS Window of 50 ms
Left Right
Minimum RMS Power: -56.77 dB -56.24 dB
Maximum RMS Power: -3.92 dB -3.68 dB
Average RMS Power: -11.81 dB -11.95 dB
Total RMS Power: -10.64 dB -10.81 dB
--8--
0dB = FS Square Wave
Using RMS Window of 50 ms
Left Right
Minimum RMS Power: -59.78 dB -59.25 dB
Maximum RMS Power: -6.93 dB -6.69 dB
Average RMS Power: -14.82 dB -14.96 dB
Total RMS Power: -13.65 dB -13.82 dB
--8--


[mykec@sillygoose mykec]$ normalize -n AAahAAA-She(3).wav
Computing levels...
level peak gain
-8.1064dBFS -0.2637dBFS -3.8936dB AAahAAA-She(3).wav


Myke

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On Sun, 06 Jul 2003 16:51:49 -0500, Lord Hasenpfeffer wrote:

Martin Tillman wrote:

Hmm... If that screenshot is distorted so much due to the scaling that
the points I'm talking about AREN'T so hugely different, then, well,
words fail me.

Many times with Audacity I've seen peaks which appear higher than they
really are because pixel resolution is only so good. Zooming in on the
waveform solves this.

It's the opposite in CEP, but we're hardly talking individual samples
here.

Even the mp3 would do - it's not going to look so different from the wav.

Yeah, but what ****wit would send you an MP3 in a discussion like this?

The peaks won't be significantly different from the wav.

I'd hate to think that a recreation of your 'remaster' would be any
different from the original remaster - can you be absolutely certain
of your original settings in Normalize?

More specifically?

How can I be more specific than 'can you be absolutely certain of your
original settings in Normalize?' ?

Left Right
0dB = FS Sine Wave
Maximum RMS Power: -3.92 dB -3.68 dB
Average RMS Power: -11.81 dB -11.95 dB
[...]
0dB = FS Square Wave
Maximum RMS Power: -6.93 dB -6.69 dB
Average RMS Power: -14.82 dB -14.96 dB

level peak gain
-8.1064dBFS -0.2637dBFS -3.8936dB AAahAAA-She(3).wav

Well... Now we really don't know what Normalize bases it's calculations on,
wich is what I suspected a test on a real music file would result in. :-)

Does the above numbers mean that Normalize will lower the amplitude of this
specific file with 3.9dB?

Regards
/Jonas

On Mon, 07 Jul 2003 16:43:15 -0500, Lord Hasenpfeffer wrote:

How can I be more specific than 'can you be absolutely
certain of your original settings in Normalize?' ?

Are you meaning in terms of recreating my steps?

If so, absolutely certain. There's only one setting that I specified:
-10dBFS as a target "level".

That's OK then.

Jonas Eckerman wrote:

P.S. Have you sent your song there to Dr. Demento?

Dr. Demento? Is that some guy I'd recognize if I'd been to the US?

I thought surely Dr. Demento was an internationally recognized
phenomenon by now... Weekly radio show... Lots of oddball various
artists compilations... He has a penchant for playing a lot of old
novelty 78s and a lot of new recordings submitted by his fans and
listeners; stuff like that to which virtually no other
commercially-minded program director / air personality would ever
consider giving airtime.

He played a key role in helping Weird Al Yankovic get his start back in
1979.

Why I'd even betcha some of the regulars in this newsgroup know him
personally!

http://www.drdemento.com/

Myke

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Lord Hasenpfeffer wrote:

Why I'd even betcha some of the regulars in this newsgroup know him
personally!

http://www.drdemento.com/

Cool site! I had forgotten that the Dr. had played "Lee Harvey Was a
Friend of Mine" by my old buddy Homer Henderson back in '89.

Artie

Myke

Martin Tillman wrote:

We'll start with the screenshots.

(Of DSOTM)

And are we going to see them anytime soon?

I'm sorry, my hard drive's full of a whole bunch of other WAVs right at
the moment and I'm back to work now. Things take longer than they do
when I'm on vacation.

Myke

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Windows...It's rebootylicious!!!
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