Geoff Wood wrote:

Have yo not yet read and understood the intsructions for yor favorite Linux
'Normalise' application ?

Yeah, here's one thing that piqued my curiosity.

--peak Adjust using peak levels instead of RMS levels. Each file will
be adjusted so that its maximum sample is at full scale. This
just gives a file the maximum volume possible without clipping;
no normalization is done.

What was that last line again?

"no normalization is done."

That's what I thought it said.

OK,

Myke

--

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

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

Have yo not yet read and understood the intsructions for yor favorite
Linux
'Normalise' application ?

Yeah, here's one thing that piqued my curiosity.

--peak Adjust using peak levels instead of RMS levels. Each file will
be adjusted so that its maximum sample is at full scale. This
just gives a file the maximum volume possible without clipping;
no normalization is done.

What was that last line again?

"no normalization is done."

That's what I thought it said.


What it is doubtlessly intended to say is "no compression is done". Hope
the coding is more concise that the manual.

'Peak' is the most common method of normalisation. As stated before, if
normalising to average power values (especially to the very high RMS level
that you prefer all your music at), compression is necessary if clipping is
to be avoided.


geoff

Lord Hasenpfeffer wrote:

Geoff Wood wrote:

Have yo not yet read and understood the intsructions for yor favorite Linux
'Normalise' application ?

Yeah, here's one thing that piqued my curiosity.

--peak Adjust using peak levels instead of RMS levels. Each file will
be adjusted so that its maximum sample is at full scale. This
just gives a file the maximum volume possible without clipping;
no normalization is done.

What was that last line again?

"no normalization is done."

That decription is totally false then compared to any other audio software
package... Normalising is exactly that, making the loudest sample some % of FS.
Usually 100%, but you can also normalize to, for example, 30%.

It could be argued that these applications narrow down the word "normalize" too
much, because I can imagine you could also "normalize" the dynamic range, the EQ
setting, whatever adjustable parameter there is in fact. However, the creator of
this package is creating quite a bit of confusion with its users by learning them
jargon that doesn't correspond to the standard which everyone else uses.

Erwin Timmerman

Geoff Wood wrote:

--peak Adjust using peak levels instead of RMS levels. Each file will
be adjusted so that its maximum sample is at full scale. This
just gives a file the maximum volume possible without clipping;
no normalization is done.

What was that last line again?

What it is doubtlessly intended to say is "no compression is done". Hope
the coding is more concise that the manual.

Mmm-hmmm... See, I remember reading this a time or two in the past as
well - although that's not to imply that I understood everything I was
reading either. I stated near the very beginning of that stupid
previous thread in the other newsgroup that I'd read the docs but that
they weren't very helpful because terms with which I was not familiar at
the time were being used (e.g. "normalization", "clipping", "RMS", and
"dBFS" primarily (as opposed to just "dB" which I *did* understand)).
Well, obviously the phrase "no normalization is done" did a *lot* to
help me out there - because right off the bat as soon as that other
stupid thread began there was confontation and name-calling simply
because different people were using the same term to describe different
things and nobody knew that the other person wasn't interpreting the
words in the same manner. Ditto for the term "clipping" which I at the
time thought was the "clip" that you hear whenever a sample's binary
value "wraps" - which obviously doesn't happen to anybody but me anymore
so nobody knew what I was trying to describe and I didn't know I had the
term definition incorrect. I had *seen* clipping in various WAVs in my
WAV editor before but didn't know that what I was seeing was called
"clipping" because in my book at the time "clipping" was already being
used to describe "wrapping".

So with all the honest errors occurring in miscommunication going on, a
lot of bull**** ensued and nothing was accomplished except I received a
lot of misapplied lessons in peak normalization while preserving the
original dynamic range of uncompressed CD quality audio which never did
apply to my questions and concerns about MP3 encoding techniques.

That one critically wrong line of text in the docs, however, which I
read and have naively trusted for nearly a year now did a *lot* to send
that whole entire thread over the edge into complete chaos without
resolution simply because we didn't understand each other's language.

The only good thing to come from all that **** is that I do know the
correct term definitions better than I did before and the docs make more
sense to me now than they did before - but even so, none of that still
applies directly to my hypothesis.

'Peak' is the most common method of normalisation.

From what I've been told recently, 'Peak' is the *only* form of
normalisation and this so called 'RMS normalisation' (aka 'limitizing')
is a ******* child from audio hell. g???

As stated before, if normalising to average power values (especially
to the very high RMS level that you prefer all your music at),
compression is necessary if clipping is to be avoided.

Yes, I have just created an amimated GIF of U2, "Sunday Bloody Sunday",
which clearly reveals this occurring. I have never disputed this *in
general*. What I *have* disputed it is in the *specific case* of the
MFSL "DSotM" CD which is mastered with *peaks so low* that I was able to
"RMS normalize" it to my preferred level and *still* not cause limiting,
compressing or clipping of any kind to occur. Somehow that was
perverted into the false assumption that I believed that limiting never
occurred at all. And because you believed that crap, you found
justification to dig your knives in and tear that thread even further
apart with your baseless claims.

Myke

--

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

Lord Hasenpfeffer wrote:

That one critically wrong line of text in the docs, however, which I
read and have naively trusted for nearly a year now did a *lot* to send
that whole entire thread over the edge into complete chaos without
resolution simply because we didn't understand each other's language.

Maybe it's good to mail the author of the manual about what he caused by
incorrectly using terms, thereby causing heaps of confusion with its users. If
someone acts like he knows what he is talking about, and yet is spilling
complete nonsense (just because he unknowingly doesn't apply the right words in
the right way, like you did) then it may very well look like he's not WILLING
to learn, *or* that he is very stupid. Both not being a very desireable image,
only caused by misunderstanding.

I think that is what happened to you in R.A.T.

Misunderstanding btw is the primary reason for most of the conflicts, divorces
etc. etc...

Erwin Timmerman

Erwin Timmerman wrote:

the creator of this package is creating quite a bit of confusion with
its users by learning them jargon that doesn't correspond to the
standard which everyone else uses.

Tell me about it. I learned that when I first started trying to discuss
this in another newsgroup last week. The immediate confusion which
resulted from (1) my not knowing all the terms and (2) my not knowing
that I had wrong information about the terms I thought I *did* know
caused a lot of people to immediately assume that I was a dolt and a troll.

From there on it seemed that nearly nothing I said made any sense to
anybody and all kinds of people were responding to older posts after I'd
already realized and corrected myself for prior mistakes, etc. It was a
real mess - and I am now convinced that the seed of all of that was
planted at the time I first read my "normalize" documentation which
contains that wrong information.

Perhaps now you can why I have been so adamant about trying to keep
things more on-topic this time over in *this* newsgroup. Things are
definitely going much better for me now in here than they were for me
over there last week.

Myke




--

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

Erwin Timmerman wrote:

Maybe it's good to mail the author of the manual about what he caused by
incorrectly using terms, thereby causing heaps of confusion with its users. If
someone acts like he knows what he is talking about, and yet is spilling
complete nonsense (just because he unknowingly doesn't apply the right words in
the right way, like you did) then it may very well look like he's not WILLING
to learn, *or* that he is very stupid. Both not being a very desireable image,
only caused by misunderstanding.

I think that is what happened to you in R.A.T.

Thank you. So do I and I tried to explain that to them on more than one
occasion but more and more people kept jumping in and replying to the
older posts that were sent before the problem was understood, etc. etc.
It was a real mess that will exist now primarily my expense forever in
the archives at Google!

And the first reply here which was about horse**** vs. dog**** really
didn't help matters - even though John LeBlanc was probably completely
unaware of what I'd just come from experiencing in the other NG.

Myke

--

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

"Lord Hasenpfeffer" wrote in message ...

And the first reply here which was about horse**** vs. dog**** really
didn't help matters - even though John LeBlanc was probably completely
unaware of what I'd just come from experiencing in the other NG.

But as you must be discovering, he was certainly not unaware of
the typical results that occur from the scenario you put forth.

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

--peak Adjust using peak levels instead of RMS levels. Each file will
be adjusted so that its maximum sample is at full scale. This
just gives a file the maximum volume possible without clipping;
no normalization is done.

What was that last line again?

What it is doubtlessly intended to say is "no compression is done". Hope
the coding is more concise that the manual.

I had *seen* clipping in various WAVs in my
WAV editor before but didn't know that what I was seeing was called
"clipping" because in my book at the time "clipping" was already being
used to describe "wrapping".

Wrapping ?? ... Around a bridge abutment, perhaps, as in a "train wreck".

preserving the
original dynamic range of uncompressed CD quality audio which never did
apply to my questions and concerns about MP3 encoding techniques.

It's time to re-post your so-called "mission statement".

Here you are again, saying that processing which takes place before encoding
is totally irrevelant, but I have just responded to a number of your posts where
you are making a big deal of it.

'Peak' is the most common method of normalisation.

From what I've been told recently, 'Peak' is the *only* form of
normalisation ....

That is NOT what you've been told, at least not by me.

and this so called 'RMS normalisation' (aka 'limitizing')
is a ******* child from audio hell. g???

It can be, but both are 'forms' of normalization.

As stated before, if normalising to average power values (especially
to the very high RMS level that you prefer all your music at),
compression is necessary if clipping is to be avoided.

Yes, I have just created an amimated GIF of U2, "Sunday Bloody Sunday",
which clearly reveals this occurring. I have never disputed this *in
general*. What I *have* disputed it is in the *specific case* of the
MFSL "DSotM" CD which is mastered with *peaks so low* that I was able to
"RMS normalize" it to my preferred level and *still* not cause limiting,
compressing or clipping of any kind to occur.

But this is supposedly irrelevant to your purpose.

you found
justification to dig your knives in and tear that thread even further
apart with your baseless claims.

What baseless claims? We are essentially talking about encoding
but we are *forced* by you to discuss what goes on beforehand, as it
directly effects your "hypothesis" that normalizing files changes the
frequency content that is preserved on MP3. It does not, it merely
changes the amplitude of the file that is acted on in the same manner
by the codec.

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

"David Morgan (MAMS)" wrote in message
...

"Lord Hasenpfeffer" wrote in message
...

And the first reply here which was about horse**** vs. dog**** really
didn't help matters - even though John LeBlanc was probably completely
unaware of what I'd just come from experiencing in the other NG.

But as you must be discovering, he was certainly not unaware of
the typical results that occur from the scenario you put forth.


But actually it was a very apt description of the 'problem'. You are
worrying about very minor changes to something you have alerady degraded
hugely.


geoff

Geoff Wood wrote:

Turning it into a Linux v. Windows battle didn't help.

That I turned anything into a Linux v. Windows battle is a figment of
your imagination.

I use Linux and Windows and Macintosh as needed. Not just one and not
the other(s).

Myke

--

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

Geoff Wood wrote:

From what I've been told recently, 'Peak' is the *only* form of
normalisation and this so called 'RMS normalisation' (aka 'limitizing')
is a ******* child from audio hell. g???

No, normalising to an RMS or average value exists, but without compression
may result in clipping. Your "normalise" application may (or may not) do
this.

"normalize" employs a limiter by default to prevent clipping.
The limiter can be deliberately switched out by the user so that
clipping results where limiting would otherwise occur.

Either way, it is severely dicking with the music.

I ripped U2, "War", track 1, "Sunday Bloody Sunday" last night and
copied the WAV twice. I ran "normalize" across copy #1 using the
default -12dBFS value. I then ran "normalize" across copy #2 using my
preferred -10dBFS value. Then I produced an animated screenshot which
compared the differences between all three. I can clearly see about 5
peaks in the song which are being obviously limited in the -10dBFS WAV.
But how you can label this as "severely dicking with the music" I do
not understand. If I were producing a master CD for commercial use, the
situation would be different, but since I am not, it's not that great of
a concern to me.

http://www.mykec.com/mykec/images/Su...Sunday_012.gif

But on the other hand, so is MP3 encoding .....

Duh, however, lossy encoding is (1) an assumed and (2) required element
for the purpose of this discussion - so let's stop debating compressed
vs. uncompressed and start focusing on what common sense techniques can
be employed in order to minimize the damage which MP3 encoding
inherently causes to occur, OK?

Myke

--

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

"Lord Hasenpfeffer" wrote in message

I ripped U2, "War", track 1, "Sunday Bloody Sunday" last night and
copied the WAV twice. I ran "normalize" across copy #1 using the
default -12dBFS value. I then ran "normalize" across copy #2 using my
preferred -10dBFS value. Then I produced an animated screenshot which
compared the differences between all three. I can clearly see about 5
peaks in the song which are being obviously limited in the -10dBFS WAV.
But how you can label this as "severely dicking with the music" I do
not understand. If I were producing a master CD for commercial use, the
situation would be different, but since I am not, it's not that great of
a concern to me.

You are satisfied with MP3-quality results @128kbps (or 192 or whatever) -
I am not. So it's horses-for-courses.


geoff

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

From what I've been told recently, 'Peak' is the *only* form of
normalisation and this so called 'RMS normalisation' (aka 'limitizing')
is a ******* child from audio hell. g???

No, normalising to an RMS or average value exists, but without compression
may result in clipping. Your "normalise" application may (or may not) do
this.

"normalize" employs a limiter by default to prevent clipping.
The limiter can be deliberately switched out by the user so that
clipping results where limiting would otherwise occur.

Again, when you get inside the electronics of limiting, essentially
it *is* clipping. That's precisely what the circuit design in most limiters
is called... a "clipper circuit".

Duh, however, lossy encoding is (1) an assumed and (2) required element
for the purpose of this discussion - so let's stop debating compressed
vs. uncompressed and start focusing on what common sense techniques can
be employed in order to minimize the damage which MP3 encoding
inherently causes to occur, OK?

Already been there... you musta' missed it. Either design a new Codec,
or visit the resources given to you (like Stephen Paul's site), or start doing
some various codec listening tests. Of course, we could go back to the topic
that you waffle back and forth on as to it's importance.... processing the
original raw .wav files with some various other tools before encoding,
testing various gain levels for effectiveness along the way, etc..

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

"Lord Hasenpfeffer" wrote in message ...


That I turned anything into a Linux v. Windows battle is a figment of
your imagination.



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

Geoff Wood wrote:

You are satisfied with MP3-quality results @128kbps (or 192 or whatever) -
I am not. So it's horses-for-courses.

Which is *exactly why* I've repeatedly asked you why you insist on being
a butt to me in these discussions when obviously you don't give a damn
about MP3s in your CD-only world! Nearly everything you've ever said to
me in both the previous newsgroup and this one has nothing to do with
MP3s. With all due respect, your arguments are generally irrelevant to
the topic of ATH-based lossy compression encoding schemes. What is so
difficult for you to understand about this I haven't the foggiest idea.
This is only about the 10th time I've said this.

Myke

--

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Windows...It's rebootylicious!!!
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David Morgan (MAMS) wrote:
"Lord Hasenpfeffer" wrote in message ...

That I turned anything into a Linux v. Windows battle is a figment of
your imagination.

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

Oh......

Where is Linux mentioned in my tagline?

--

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Windows...It's rebootylicious!!!
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David Morgan (MAMS) wrote:

That I turned anything into a Linux v. Windows battle is a figment of
your imagination.

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

Oh......

Windows *IS* rebootylicious, David.

--

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

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

You are satisfied with MP3-quality results @128kbps (or 192 or whatever) -
I am not. So it's horses-for-courses.

Which is *exactly why* I've repeatedly asked you why you insist on being
a butt to me in these discussions when obviously you don't give a damn
about MP3s in your CD-only world! Nearly everything you've ever said to
me in both the previous newsgroup and this one has nothing to do with
MP3s.

But Lord.... you are inquiring (or boasting, not sure which) about preparing
raw audio for conversion to MP3. Or was it about ingnoring what happens
there because you like louder MP3s? Like I said, you've confused me.

So now, your replys have been reduced to combattance with a r.a.p.
contributor who has already tried to help you understand the same
points in other groups. (Here's to the learning machine!) Since I read
much of your other posting history on this subject, I wasn't surprised
when you told Geoff to "butt out" of *YOUR* thread here early on. He
had already given you many answers with references or links to back
them up and other supporting replies on another group, but you refused
to accept them.

With all due respect, your arguments are generally irrelevant to
the topic of ATH-based lossy compression encoding schemes. What is so
difficult for you to understand about this I haven't the foggiest idea.

The problem with foggy ideas Lord, is that you continually move away
from MP3 to preprocessing the .wav files prior to encoding. You will have
to decide which it is that you choose to discuss. IOW, I will ask you for the
5th time today.... for the benefit of the entire group, please reiterate your
so-called "Mission Statement" and clarify please, EXACTLY what is your
question and EXACTLY what is it that you are trying to accomplish here?

You had a "theory". It's been sufficiently challenged and virtually disproven
here and on two other newsgroups.

This is only about the 10th time I've said this.

Are you ignoring my posts because I am now making a point of the
fact that you are continuously waffling subject matter back and forth
between preparation for the encoding and the actual encoding? I would
be happy to ask you 5 more times to reiterate your goals, would that
make things even? g

You are about as likely to rescend your "theory" as Geoff is to stop
supporting his own experiences.

The very least you could do after creating a *very* long thread of people
who have contributed their time, energy and experience, free of charge
to advise you, WHAT HAVE YOU LEARNED ??

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

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

That I turned anything into a Linux v. Windows battle is a figment of
your imagination.

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

Oh......

Windows *IS* rebootylicious, David.


I agree. It doesn't hurt to reboot every month or so just to tighten up the
registry and run the init sequence again. ;-)

"Lord Hasenpfeffer" wrote in message ...

Fine. What is the associations between limiting/clipping and ATH-based
filtration mechanisms?

There is only the association that you have created by the processes
you espouse to implement.

start focusing on what common sense techniques can
be employed in order to minimize the damage which MP3 encoding
inherently causes to occur, OK?

Already been there... you musta' missed it.

Who musta' missed what?

You musta' missed the 40 or so posts in this thread alone that have provided
information regarding the above point.

Of course, we could go back to the topic that you waffle back and forth
on as to it's importance.... processing the original raw .wav files
with some various other tools before encoding, testing various gain
levels for effectiveness along the way, etc..

Yes, let's.

Do we have a new "Mission Statement" here ?? If so, I must refer you
to the idea of re-reading this entire thread as well as those that rest on
two other newsgroups which address directly the above topic.

I fear we are chatting for the sake of chatting - because at this point, all
people can do is repeat themselves.

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

David Morgan (MAMS) wrote:

Do we have a new "Mission Statement" here ??

I'm not following your logic here.

There have been many posts to this thread so far which have helped to
move my hypothesis closer to being proven false or true. I have no
desire to change my mission statement and don't understand why you
believe that I do or have or even intend to.

What have I learned?

1) Pre-"normalizing" a WAV to a higher initial amplitude *should* enable
more frequencies from the source data to successfully "jump the hurdles"
imposed by the Absolute Threshold of Hearing filtration mechanism of
most lossy compression schemes as the data is being compressed into a
destination file or form such as MP3, AAC, or MiniDisc.

2) With regard to MP3s, VBR encoding is most likely preferable over CBR
encoding since the additional quantity of surviving frequencies
resulting from the earlier WAV normalization could result in a need for
more storage space than CBR-encoding can afford.

3) Larger bitrates provide more room for data storage than do lower
bitrates, therefore, greater potential for "better sound" with them exists.

4) More than likely, greater bitrates also lessen the need to be
concerned about whether one should choose VBR over CBR when encoding an MP3.

5) Software utilities such as MP3Gain can be used to achieve a nice
balance in loudness levels across wide ranges of MP3 which have been
encoded from a variety of different sound sources. However, simply
increasing or decreasing the loudness levels of these MP3s does nothing
to ensure that as many of the frequencies present in the original WAVs
are saved from being filtered by the ATH-aspects of the MP3 lossy
compression method.

6) The curve table which visually depicts the Absolute Threshold of
Hearing for frequencies ranging 20-20,000Hz is fixed (i.e. Absolute) in
relation to digital Full Scale - although this is still in dispute by
another contributor to this thread.

Myke

--

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

David Morgan (MAMS) wrote:

But Lord.... you are inquiring (or boasting, not sure which)
about preparing raw audio for conversion to MP3.

Of thee I inquire, "Do aspects of normalization exist about which a man
such as I might find reason to boast?"

Or was it about ingnoring what happens there because you like louder MP3s?

I'm primarily concerned with preventing unnecessary ATH-centric
filtration of frequencies which is inherent to (I believe) all lossy
audio data-compression methods.

I wasn't surprised when you told Geoff to "butt out" of *YOUR* thread
here early on. He had already given you many answers with references
or links to back them up and other supporting replies on another
group, but you refused to accept them.

I believe most of the information Geoff has told me regarding
uncompressed CD/WAV audio has been useful. However, Geoff has been
unnecessarily and extremely rude to me on multiple occasions and to that
I do not take kindly. To this end I cite the lies about me which he
submitted in his original post to this thread.

The problem with foggy ideas Lord, is that you continually move away
from MP3 to preprocessing the .wav files prior to encoding.

The preprocessing of .wav file in order to create "better sounding" MP3s
is central to my hypothesis as defined in the post which spawned this
thread. I am not moving from one issue to another here.

Preprocessing WAVs to produce "better" MP3s *is* *the* issue.

please reiterate your so-called "Mission Statement" and clarify
please EXACTLY what is your question and EXACTLY what is it that
you are trying to accomplish here?

No need to reiterate, just click he

http://groups.google.com/groups?q=Louder+IS+Better+(With+Lossy)&hl=en&lr=&i e=UTF-8&oe=UTF-8&selm=3EFE7206.8030709%40spamsucks.ionet.net&rnum =2

You had a "theory".

I had (and still have) a hypothesis, not a theory.

It's been sufficiently challenged and virtually disproven here and on
two other newsgroups.

It has been debated. It has yet to be proven true or false.

You are about as likely to rescend your "theory" as Geoff is to stop
supporting his own experiences.

I am perfectly willing to abandon my theory if it can indeed be proven
false.

The very least you could do after creating a *very* long thread

Unless you've forgotten, I'm the one who was duly "spanked" for
attempting to moderate this thread in an effort to prevent it from
growing as large as it has. Apparently in unmoderated newsgroups such
as this, staying on-topic is not required and (thread)size matters not.

Let freedom ring!

Myke

--

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

"Lord Hasenpfeffer" wrote in message I believe
most of the information Geoff has told me regarding
uncompressed CD/WAV audio has been useful. However, Geoff has been
unnecessarily and extremely rude to me on multiple occasions and to that
I do not take kindly. To this end I cite the lies about me which he
submitted in his original post to this thread.

The problem with foggy ideas Lord, is that you continually move away
from MP3 to preprocessing the .wav files prior to encoding.

That's not mine Lord.

The preprocessing of .wav file in order to create "better sounding" MP3s
is central to my hypothesis as defined in the post which spawned this
thread. I am not moving from one issue to another here.

Trouble is, you are thrash a 'hypothesis' but have a poor understanding of
the background. When commonly known *facts* are presented that don't fit
well with your 'hypothesis' , you vehemently blame the messenger rather
than graciously absorbing the info that has been presented to you and
re-evaluating you ideas.

I am perfectly willing to abandon my theory if it can indeed be proven
false.

As you evidently don't have the necessary technical kowledge to understand
the explanations you have been given, and appear to have an antipathy
towards gaining that knowledge, then is is impossible to prove your theory
(oops, hypothesis) false, to you.


geoff

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

Geoff Wood wrote:

Trouble is, you are thrash a 'hypothesis' but have a poor understanding of
the background. When commonly known *facts* are presented that don't fit
well with your 'hypothesis' , you vehemently blame the messenger rather
than graciously absorbing the info that has been presented to you and
re-evaluating you ideas.

What previous post(s) can you site as evidence of this?

As you evidently don't have the necessary technical kowledge to understand
the explanations you have been given,

Such blanket statements are impossible to interpret.

and appear to have an antipathy towards gaining that knowledge,
then is is impossible to prove your theory
(oops, hypothesis) false, to you.

And the thread lengthens!

Myke

--

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Windows...It's rebootylicious!!!
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"Lord Hasenpfeffer" wrote in message ...
David Morgan (MAMS) wrote:

But Lord.... you are inquiring (or boasting, not sure which)
about preparing raw audio for conversion to MP3.

Of thee I inquire, "Do aspects of normalization exist about which a man
such as I might find reason to boast?"

May I refer you to your original post, wherein you state for the record
that normalizing your raw files has created a means by which you
believe that you have overcome the limitations of encoding to MP3
thereby resulting in the conclusion that "Louder IS better (With Lossy)".

I'm primarily concerned with preventing unnecessary ATH-centric
filtration of frequencies which is inherent to (I believe) all lossy
audio data-compression methods.

Then, my new acquaintance, you must design a new encoder.

I wasn't surprised when you told Geoff to "butt out" of *YOUR* thread
here early on. He had already given you many answers with references
or links to back them up and other supporting replies on another
group, but you refused to accept them.

I believe most of the information Geoff has told me regarding
uncompressed CD/WAV audio has been useful. However, Geoff has been
unnecessarily and extremely rude to me on multiple occasions and to that
I do not take kindly. To this end I cite the lies about me which he
submitted in his original post to this thread.

That would be this :

qoute from ...

For those who don't know, My Myke here insists on normalising every piece of
his 2100-strong CD collection to -10dB average RMS, and considers any tracks
that don't meet this criteria to be flawed, and incompetently produced.
Despite serious attempts to clue him up he clings to total misconceptions
regarding levels, amplification, attenuation, normalisation, the mastering
process, etc.

He dismisses MFSLs Dark Side Of The Moon as being a peice of excrement
because the highest peak is -4dB or so, and that buyers have been ripped
off. (they didn't get all the bits they paid for ?).

end quote

This was actually what caused me to research all of your previous posts to
other groups, which validated it's content 100%.

Preprocessing WAVs to produce "better" MP3s *is* *the* issue.

Gosh... I can only ask why then, you lashed out at all of those who showed
you the err in your ways with regard to your method... followed by changing
the subject to the encoding process.

I even made it a point in one of your subsequent posts to that effect, to
offer you all of the help or advice I could with regard to this after an
admission that I could not help you woth encoding issues that are
out of my control.

Now, everything that I could likely offer has not only been covered by me,
but by the vast majority of replies to your thread, here and elsewhere.

EXACTLY what is your question and EXACTLY what is it that
you are trying to accomplish here?

No need to reiterate, just click he


http://groups.google.com/groups?q=Louder+IS+Better+(With+Lossy)&hl=en&lr=&i e=UTF-8&oe=UTF-8&selm=3EFE7206.8030709%40spamsucks.ionet.net&rnum =2

There is No question here. There is No statement of intent. There
is nothing more than what I have already repeadly noted. This is
your original post - a duplicate of that made on another group - which
is merely a synopsis of your personal methodology, experiences,
and listening preferences, culminating with a blanket statement of
louder is better.

A mission statement lays out a set of goals to accomplish and
steps that must be taken to accomplish those goals; it usually
defines current knowledge and knowledge needed to complete
the mission as well as the tools and supplies needed, etc..

You posted an opinion, not a goal to accomplish or a question for
the group.

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

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

Do we have a new "Mission Statement" here ??

I'm not following your logic here.

I am chiding you Myke, because you have never HAD a mission statement.

There have been many posts to this thread so far which have helped to
move my hypothesis closer to being proven false or true. I have no
desire to change my mission statement and don't understand why you
believe that I do or have or even intend to.

What have I learned?

1) Pre-"normalizing" a WAV to a higher initial amplitude *should* enable
more frequencies from the source data to successfully "jump the hurdles"
imposed by the Absolute Threshold of Hearing filtration mechanism of
most lossy compression schemes as the data is being compressed into a
destination file or form such as MP3, AAC, or MiniDisc.

Then I must assume that you have chosen to deny the high number of replies
that state otherwise.

2) With regard to MP3s, VBR encoding is most likely preferable over CBR
encoding

Very Good !!! I picked up on that as well. :-)

3) Larger bitrates provide more room for data storage than do lower
bitrates, therefore, greater potential for "better sound" with them exists.

Since you're stuck on a dial-up connection, I suppose that I'll understand
why you may not have know this from the first day you even thought
about the word MP3 - or had not seen and listened to the various bitrate
results using your encoding software.

The varying quality of the radio samples on your website would have led
me to believe that you had already used more than one bit rate.

4) More than likely, greater bitrates also lessen the need to be
concerned about whether one should choose VBR over CBR
when encoding an MP3.

Perhaps. This is over my head and I didn't garner this impression.

5) Software utilities such as MP3Gain can be used to achieve a nice
balance in loudness levels across wide ranges of MP3 which have been
encoded from a variety of different sound sources. However, simply
increasing or decreasing the loudness levels of these MP3s does nothing
to ensure that as many of the frequencies present in the original WAVs
are saved from being filtered by the ATH-aspects of the MP3 lossy
compression method.

Yes. What's done is done. From there, only manipulating playback
options will have any effect.

6) The curve table which visually depicts the Absolute Threshold of
Hearing for frequencies ranging 20-20,000Hz is fixed (i.e. Absolute) in
relation to digital Full Scale - although this is still in dispute by
another contributor to this thread.

Myke

I learned a great deal more than you then, and I appreciate the motivation
you provided me to research MP3 things with more intensity than before.


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

Jonas Eckerman wrote:

No, you have not learned this. Nowhere in this thread has there been any
conclusive evidence that most lossy compression schemes has a fixed
threshold below wich they filter everything.

I believe my statement assumed this as true since it had not yet been
proven false and all ATH charts which I had seen at the time seemed to
indicate a fixed position. I may have only thought I learned this. At
any rate, David asked me a question. I answered it. Your reply to my
answer is valid and, therefore, proof that this entire thread has not
been a complete waste of time for all involved.

I just spent 10 minutes googling about this

What were your search criteria? If I were a little more keen in this
regard I could have probably done the same on my own. I Google for most
everything I ever need to learn via both web and groups provided that I
know what I'm searching for in the first place.

My initial decision to create a thread in this newsgroup on this topic,
however, was in direct response to a dare by someone in another
newsgroup who obviously assumes that the members of this group are much
more akin to ravenous wolves than I. Something about one's "tolerance
for gore not being up the spectacle" comes to mind at this point. So
for those who would choose to believe that I am unjustly convicting
innocents by merely expecting to encounter rudeness at every turn in
Usenet, my actual presence here and willingness to accept the
rec.audio.pro challenge stands as evidence of my actual real faith in
the process regardless of what my *defensively-positioned* expectations
of it may actually be.

and I found that some encoders does have a fixed threshold below
wich everything is filtered out, while other encoders does not.

My "notlame" encoder is approximately 2 years old at this time.

What I learned from that is simjply that in this regard some encoders are
better than others, and that if I want as good a quality as possible in the
compromise that lossy encoding allways is I should use the best encoder I
can find. Of course we all knew this allready.

Yeah, but I can also understand from a community such as this why so
many would be so more concerned initially about attempting to straighten
my crooked ways when dealing with the merits of WAV normalization as
opposed to the actual issue at hand. I asked very early of if this
perhaps may be an off-topic thread in a newsgroup such as this and was
not told "yes". And Bob Cain wouldn't suggest that I subscribe to the
mp3encoders mailing list either until late last night or early this morning.

2) With regard to MP3s, VBR encoding is most likely preferable over
CBR encoding

Depends on your priorities.

See, responses like this convince me that posting my answer to Dave's
question was indeed another "good thing" for me to do. It seems to have
caused a bit of focus to return to this unmoderated thread - which
pleases me to no end.

A constant bitrate at the highest value the encoder supports
(320 is a common max for encoders) will result in better
quality than VBR. VBR is simply a better way to keep the file size down
than a constant and not very high bitrate.

The rationale to support the absolute nature of your claim here is lost
on me at this time. I have always used CBR because when I last read up
on the merits of VBR in early 2002, it was still receiving quite a bit
of bad press. Apparently it's been improved since then, however, if I
can still safely use CBR in most instances, I think I'd like to - simply
because my old biases in its favour still remain... even though I
believe I understand the potential downside for it as well as a result
of participating in this most lengthy discussion.

6) The curve table which visually depicts the Absolute Threshold of
Hearing for frequencies ranging 20-20,000Hz is fixed (i.e. Absolute)
in relation to digital Full Scale

How you could have learned this from this thread is a mystery too me. I've
only managed to learn that it might be so.

Until I had reason to believe otherwise, this was my assumption.

Myke

--

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

I just spent 10 minutes googling about this

What were your search criteria?

I don't remember exactly. Something similar to:
+mpeg +layer +frequencies +ath

My "notlame" encoder is approximately 2 years old at this time.

The you should be able to check the state of the Lame encoder at that time
by reading it's history document if there is also a reference about what
version of Lame came at what time. I don't know if there is one.

I would however suggest that you use a much more recent version of Lame
instead. The Lame encoder is continously being improved, and they have made
important improvements in the last two years.

A constant bitrate at the highest value the encoder supports
(320 is a common max for encoders) will result in better
quality than VBR. VBR is simply a better way to keep the file size
down than a constant and not very high bitrate.

The rationale to support the absolute nature of your claim here is
lost on me at this time.

It's quite simple.

When you use VBR encoding, different pieces of audio will be encoded at
different bitrates. When the encoder decides that a high bitrate is
necessary to get accepted quality a high bitrate will be used, when it
decides that a low bitrate is enough to get accepted quality a low bitrate
will be used.

This has proven to be a good way to get a balance between file size and
perceived quality. Wether it works or not depends on wether the encoder can
make good decisions about wich bitrate to use when.

The current version of Lame is pretty good in this regard, but of course
the first attempts at this technique weren't as good.

When using Lame to encode VBR MP3s you also have a bunch of options. You
can specify the maximum and minimum bitrates to be used, and you can
specify a "quality level". The whole idea is to give the user the freedom
to decide what (s)he sees as a good balance between file size and quality.

Using CBR at the encoders highest bitrate will give you higher quality than
VBR. The reason is simply that all frames will then be encoded using the
highest bitrate.

Personally I allways use VBR, for a very simply reason. File size does
matter, and I want a balance between file size and quality. If file size
does not matter, I don't encode to MP3.

I have always used CBR because when I last
read up on the merits of VBR in early 2002, it was still receiving
quite a bit of bad press.

There has been both problems and misunderstanding about VBR. A couple of
things that gave it a bad reputation:

* There has been encoders that simply didn't do a good job of it. I suppose
there still are such encoders.

* Not all MP3 decoders could handle VBR files, so there were compatibility
issues. I haven't seen such a decoder for a long time now.

* Some people simply didn't understand what VBR was. There were people
saying that VBR allways and automatically sounded better than CBR, and
there was people who believed them, tested it, and was dissapointed when
they found out that it wasn't true.

How you could have learned this from this thread is a mystery too me.
I've only managed to learn that it might be so.

Until I had reason to believe otherwise, this was my assumption.

Aahh.. Now I understand you better, or maybe worse. I'm confused. :-)

Did you assume, for the sake of testing and argument, that your hypothesis
was true, or did you actually believe it was true? There's a great big
difference there.

When you say you have learned something, I take that to mean that you have
actually gained that knowledge.

Making an assumption about a hypothesis in order to test it is common
scientific method, but taking that assumption and without proof promote it
from an assumption to knowledge is not scientific. You must allways
remember that an assumption is just an assumption and therefore doesn't
have anything to with actual knowledge.

Regards
/Jonas Eckerman

David Morgan (MAMS) wrote:

May I refer you to your original post, wherein you state for the record
that normalizing your raw files has created a means by which you
believe that you have overcome the limitations of encoding to MP3
thereby resulting in the conclusion that "Louder IS better (With Lossy)".

The limitations of encoding to MP3 are *assumed*, therefore, they cannot
be overcome. Meanwhile, I do believe it is possible to deliberately do
things which can minimize the adverse effects of said limitations.
Basically this all boils down to "Should we try harder to minimize the
adverse effects?" v. "It doesn't pay to try harder so let come what may."

I'm primarily concerned with preventing unnecessary ATH-centric
filtration of frequencies which is inherent to (I believe) all lossy
audio data-compression methods.

Then, my new acquaintance, you must design a new encoder.

Addendum: ...given the state of existing encoders with which I must contend.

My Myke here insists on normalising every piece of his 2100-strong
CD collection to -10dB average RMS,

That's lie #1. I only insist on normalizing those which fall below
-10dBFS. Most newer CDs in my collection go on to be encoded exactly as
they are ripped with no modification whatsoever.

and considers any tracks that don't meet this criteria to be flawed

That's lie #2. The tracks in and of themselves are fine. But *may*
need to be "better prepared" prior to being encoded.

and incompetently produced.

That's lie #3. MP3 encoding was not a factor in the decision-making
process when my CDs were mastered... and the rules which govern success
in CD and MP3 creation are not the same. Those who produce CDs take
only matters related to uncompressed audio into consideration. They do
not base decisions upon what works best for *both* CD and MP3! This
does not imply incompetence on their behalf at all.

Despite serious attempts to clue him up he clings to total misconceptions
regarding levels, amplification, attenuation, normalisation, the mastering
process, etc.

Nothing I can say will affect anyone's opinion about this.

He dismisses MFSLs Dark Side Of The Moon as being a peice of excrement
because the highest peak is -4dB or so,

That's lie #4. I do *not* consider it exrement because of its low
level. For CD audio alone, it's perfectly fine, given the intent of its
makers. For MP3 encoding? I don't believe it's appropriate in its
default, low-amp state. This, however, I do not hold against MFSL. MP3
didn't exist when that disc was mastered and even if it *did*, it would
not have affected their decision-making process at all.

and that buyers have been ripped off.

That's lie #5. Geoff implies here that I still believe something which
I no longer do. I did believe this at one time with regard to that
particular CD, yes, but have long since rescinded my position on that
after it was explained to me why that particular CD's levels are so low.

(they didn't get all the bits they paid for ?).

And that was *never* the reason for my prior belief in the "rip-off"
even when I *did* think that way, therefore, he's clearly
misrepresenting my position yet again.

This was actually what caused me to research all of your previous posts to
other groups, which validated it's content 100%.

These same tired accusations are bogus from the start and do not deserve
to be further discussed since they are nothing but baseless, filler
arguments obviously being injected into this thread for the purpose of
*starting* the so-called "train wreck" in this newsgroup which I have
sought to avoid.

Preprocessing WAVs to produce "better" MP3s *is* *the* issue.

Gosh... I can only ask why then, you lashed out at all of those who showed
you the err in your ways with regard to your method... followed by changing
the subject to the encoding process.

My arguments in favor of preprocessing the WAVs have *always* been made
with the subject of the encoding process in mind. Geoff is the one who
has gross dislike of the encoding process, not I, therefore his
arguments which centered around uncompressed audio only were completely
beside the point, IMHO.

There is No question here. There is No statement of intent. There
is nothing more than what I have already repeadly noted. This is
your original post - a duplicate of that made on another group

No, it was posted *here* *first*. It was re-posted elsewhere later.
That post was specifically intended for this newsgroup when I wrote it.

- which is merely a synopsis of your personal methodology, experiences,
and listening preferences,

This is so that the terms which were *not* well-defined at the beginning
of the previous thread in the other newsgroup could already be defined
from the beginning of this thread in this newsgroup; the purpose of this
being to avoid the miscommunications here which resulted in so much
disaster before.

culminating with a blanket statement of louder is better.

No, louder is better *with* *lossy*. It's right there (and quite
deliberately placed) in the subject line. I wanted to make it quite
clear here that everyone understood right from the start that I was not
here to discuss the merits of "louder is better" with regard to
uncompressed CD/WAV audio - as Geoff insisted on believing.

A mission statement lays out a set of goals to accomplish
You posted an opinion, not a goal to accomplish or a question for
the group.

The mission is to confirm or deny the sense in preprocessing WAV files
with "normalize" with regard to minimizing the adverse effects which are
caused by ATH-based frequency filtration techniques.

The mission never was to discuss the effects of "normalize" on WAVs with
regard to what it does to the WAV *only*.

Myke

--

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

"Lord Hasenpfeffer" wrote
What have I learned?

1) Pre-"normalizing" a WAV to a higher initial amplitude *should* enable
more frequencies from the source data to successfully "jump the hurdles"
imposed by the Absolute Threshold of Hearing filtration mechanism

So, what are you waiting for? Do it both ways, listen to the results
and chose what you like best. You could have done that 10 times over
considering how much time you've spent arguing here. It doesn't matter if
theory says door number 2 should sound better, if door number 1 actually
sounds better then choose door number 1.

- Tommy

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

May I refer you to your original post, wherein you state for the record
that normalizing your raw files has created a means by which you
believe that you have overcome the limitations of encoding to MP3


A few days was spent puting you straight on what normalisation, peak, RMS,
avarage, clipping, compression, etc actual means, and well basis decibels
theory. Now you are delving into the science and fundamentals of perceptual
coding..

And then ascertaining that your use of the word "normalise" related purely
to the Linux command-line program which *apparently* can apply compression
to attain normalisation on an RMS basis without clipping. You continue to
use that term with quotes, which is not conducive to clear discussion.
Please call it "RMS-normalise-compress" or quote specify the application
each time ou refer to it, rather than just using quotes, for the benefit of
those who 'missed out' on the earlier earlier threads in r.a.t .

1 - No, CDs are not mastered with the intention of providing material
best-prepared for MP3 encoders who prefer high RMS average level.
2 - Yes, "louder is better", for you.
3- No, peak normalisation of a few dB has inaudible effect on resultant MP3
'frequencies'.
4 - Yes, RMS normalisation may have profound effects on resultant MP3s
5 - No, it has zilch to do with threshold levels of frequency bands.
6 - It has to do with the hyper-compression that you demand.

Hypercompression is the subject of a whole bunch of scorn because of the
blanding of music it has created over the last few years, but that's a
different thread.

geoff

Geoff Wood wrote:

A few days was spent puting you straight on what normalisation, peak, RMS,
avarage, clipping, compression, etc actual means, and well basis decibels

Yeah, and next I suppose you'll say that I barged in all hell-bent to
teach everyone the differences between up and down - when in reality I
openly admitted from the start that I was not exactly clear on all the
terms - but that I did know how to successfully achieve my desired
results despite that fact. "I know how to walk the walk; just not talk
the talk", remember? And with regard to everything I believe(d) I
needed to do with "normalize" at the time, that still holds true - even
though my practices may not be the best thing for *you* to do in *your*
own life which are pursuant to *your* own goals.

And then ascertaining that your use of the word "normalise" related purely
to the Linux command-line program which *apparently* can apply compression
to attain normalisation on an RMS basis without clipping. You continue to
use that term with quotes, which is not conducive to clear discussion.
Please call it "RMS-normalise-compress" or quote specify the application
each time ou refer to it, rather than just using quotes, for the benefit of
those who 'missed out' on the earlier earlier threads in r.a.t .

I actually prefer the word "limitize" as was put forth as a proposed
solution in the other thread - because what I routinely use that
application to do does not involve compression; only limiting.

Hypercompression is the subject of a whole bunch of scorn because of the
blanding of music it has created over the last few years, but that's a
different thread.

Yes, yes, yes... and I've both told you and have shown you that I do
*not* *hyper-compress* anything by way of 'limitizing' to an average RMS
of -10dBFS.

Q: Where is the evidence of any (hyper-)compression in this screenshot?
A: There isn't any!

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

Myke

--

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

"Lord Hasenpfeffer" wrote in message ...


My arguments in favor of preprocessing the WAVs have *always* been made
with the subject of the encoding process in mind. Geoff is the one who
has gross dislike of the encoding process, not I, therefore his
arguments which centered around uncompressed audio only were
completely beside the point, IMHO.

You're entitled to that. But I don't know anyone who frequents this group
who does not dislike the MP3 encoding processes to some degree - so I'd
be inclined to let him off the hook. It really changes the audio a bit too much
to be respected a great deal by people who do their best to make it as
good as possible for a living. Once up in the seriously high bit rates, it
can be really good.

There is No question here. There is No statement of intent. There
is nothing more than what I have already repeatedly noted. This is
your original post - a duplicate of that made on another group

No, it was posted *here* *first*. It was re-posted elsewhere later.
That post was specifically intended for this newsgroup when I wrote it.

Don't forget Google... (which is really having some problems right now).
The message went to these groups... alt.audio.minidisc, rec.audio.tech,
and rec.audio.misc - on this date... (Saturday) 2003-06-28 21:57:53 PST

It appeared here on (Sunday) June 29, 2003 1:07 AM CST

By the way... thank you for not crossposting - and if Google is reporting
these times incorrectly, I apologize.

No, louder is better *with* *lossy*. It's right there (and quite
deliberately placed) in the subject line.

Now you're picking on me with semantics. g I have quoted your
whole phrase enough times that you understand what I mean.

A mission statement lays out a set of goals to accomplish
You posted an opinion, not a goal to accomplish or a question for
the group.

The mission is to confirm or deny the sense in preprocessing WAV files
with "normalize" with regard to minimizing the adverse effects which are
caused by ATH-based frequency filtration techniques.

Do you have ANY other audio processing tools besides "normalize" in your kit ?

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

"Lord Hasenpfeffer" wrote in message ...

Yes, yes, yes... and I've both told you and have shown you that I do
*not* *hyper-compress* anything by way of 'limitizing' to an average RMS
of -10dBFS.

You haven't even told us if you have a compression tool or if you know how
to use it.

Q: Where is the evidence of any (hyper-)compression in this screenshot?
A: There isn't any!

It doesn't look like it, but compression is harder to detect visually. Limiting,
otherwise known as 'clipping' of the waveform, is readily apparent and is
often much more destructive to the audio.

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

In article ,
Lord Hasenpfeffer wrote:

Of thee I inquire, "Do aspects of normalization exist about which a man
such as I might find reason to boast?"

Okay, that does it- I apologize for giving Myke the letters 'ATH' to
play with shoulda kept my big mouth shut.



Chris Johnson, uniquely guilty of setting this guy up with more
information to get confused by

David Morgan (MAMS) wrote:

My arguments in favor of preprocessing the WAVs have *always* been made
with the subject of the encoding process in mind. Geoff is the one who
has gross dislike of the encoding process, not I, therefore his
arguments which centered around uncompressed audio only were
completely beside the point, IMHO.

You're entitled to that. But I don't know anyone who frequents this group
who does not dislike the MP3 encoding processes to some degree - so I'd
be inclined to let him off the hook.

The only time he's "on the hook" as far as I'm concerned is (1) when he
refers to my by names other than my own, (2) when he blatantly
misrepresents my positions for the sake of causing me more trouble than
I'm due to receive and (3) when he continues to speak about things which
I perceive to pertain only to uncompressed audio in a discussion which
assumes the presence of lossy compression.

Other than that, I think Geoff's a really great guy who I would not
hesitate to consult for technical advice pertaining to his particular
field(s) of endeavor. There is definitely some overlap between my life
and his in that regard. I really don't enjoy being "at odds" with him.

It really changes the audio a bit too much to be respected a great deal by people who do their best to make it as good as possible for a
living.

And that's perfectly understandable. The RIAA considers it be even
*more* abhorrent ... due to it's fear of being rendered obsolete.

Once up in the seriously high bit rates, it
can be really good.

At what point does it become better than common, high-bias audiotape?
My ignorant ears "say" 128kb/s.

By the way... thank you for not crossposting.

Thank you for noticing... and you're welcome.
That other conversation was already 3 NGs deep once I joined in.

and if Google is reporting these times incorrectly, I apologize.

No apology needed. I wasn't offended by your initial comment.

No, louder is better *with* *lossy*. It's right there (and quite
deliberately placed) in the subject line.

Now you're picking on me with semantics. g I have quoted your
whole phrase enough times that you understand what I mean.

I can see why you'd consider it as such, however, the subject of this
thread is very deliberately and specifically worded so as to make it
clear in no uncertain terms that *this* discussion in *this* newsgroup
assumes and requires the presence of lossy compression in order for it
to make any sense. There was too much "Just turn it up!" being said in
the other NG - which *even* *if* *true* completely fails to address the
ATH effects of lossy compression which lies at the core of that which
I'd been trying to discuss from the beginning.

Do you have ANY other audio processing tools besides "normalize" in your kit ?

Yes. But none that perform as well as "normalize" for its intended and
stated purpose.

Myke

--

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

David Morgan (MAMS) wrote:

You haven't even told us if you have a compression tool or if you know how
to use it.

If I have a "compression tool" I'm not aware of it.

Q: Where is the evidence of any (hyper-)compression in this screenshot?
A: There isn't any!

It doesn't look like it, but compression is harder to detect visually.

Hmmm... Geoff has accused me many times of compressing and/or
hyper-compressing the WAV in that screenshot and "severely dicking with
the sound" but where any of that is evident in that image, he's yet to
reveal.

Limiting, otherwise known as 'clipping' of the waveform, is readily
apparent and is often much more destructive to the audio.

Are we still talking about the "Dark Side Of The Moon" screenshot?
Or the "Sunday Bloody Sunday" animation?

In the case of DSotM, I see no clipping either. There is one peak that
reaches full scale at the bottom near the 33 minute mark and I have
posted at my site addition, medium and extreme closeups of that peak as
well which clearly reveal no limiting or clipping has taken place.

....unless I just still don't have my terms as well-defined as think I
do! :-)

http://www.mykec.com/mykec/images/20...FSL_Zoom_1.png
http://www.mykec.com/mykec/images/20...FSL_Zoom_2.png

Myke

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Windows...It's rebootylicious!!!
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Chris Johnson wrote:

I apologize for giving Myke the letters 'ATH' to
play with

Ha! You didn't. Stereo Review introduced me to both ATH and masking in
1991 (or was it 1992?) in the first article about MiniDisc which I'd
ever seen in my life. I was transfixed by the explanation of ATH as I
stood reading about it at the grocery store. It was one of the kewlest
ideas regarding audio processing of which I'd ever heard. The concept
of masking was kewl too but far less impressive to me.

Myke

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Windows...It's rebootylicious!!!
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Geoff Wood wrote:

Q: Where is the evidence of any (hyper-)compression in this screenshot?
A: There isn't any!

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

Maybe not in this track,

Good. The reason this screenshot even exists is so that I could be able
to defend myself against all the outrageous claims that by limitizing
"Dark Side" to -10dBFS I was not inadvertently spray painting the Mona Lisa.

And as far as my whooping the ass of MFSL is concerned... According to
the the idea(s) as described in my hypothesis, I found a way to
"improve" their original WAV for the purpose of creating a better MP3.
I did not improve upon their *CD*. A lot of the "anger and disgust" I
had for MFSL early on had already existed literally for *years* as a
direct result of my not understanding the differences in industry
standard CD mastering practices during the 80s v. the 90s. It didn't
take much explaining from a considerably more knowledgeable source to
set me straight on that issue. I feel much better today knowing why the
disc is mastered as quietly as it is because my old confusion is gone, a
new understanding has been born, and my faith and willingness to listen
to that particular CD again has returned after some 6-7 years.

but it's hard to tell at that magnification (both
axiis)

Which is why I also posted these additional close-ups of the one extreme
peak which touched Full Scale on the bottom in the left channel at about
the 33 minute mark.

http://www.mykec.com/mykec/images/20...FSL_Zoom_1.png
http://www.mykec.com/mykec/images/20...FSL_Zoom_2.png

Myke

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Windows...It's rebootylicious!!!
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