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Default Louder IS Better (With Lossy)

Lord Hasenpfeffer wrote:
wrote:


I would be somewhat surprised that any current CD was not NORMALISED
to somewhere around (just below) 0dbfs.


Yes, as that has apparently been "all the rage" for several years
running now. Actually, *abuse* of the 0dBFS threshold is really "all
the rage" these days. I personally am only interested in "normalizing"
older, quieter CDs mastered during the stone and middle ages before
encoding them as MP3s so that my final files sound somewhat "modern" to
my ears and not all tinny and weak when juxtaposed with MP3s encoded
from more recently mastered CDs.


I would be somewhat surprised to learn that any mainstream commercial
release has ever gone out without being normalised (using the strict
definition of the term) to somewhere within 1db of FS. Given this, it
sounds like you are actually compressing or limiting to raise the RMS
level.

In no way am I interested in brutally forcing any RMS levels to go
"through the roof". Anywhere from strictly normalized to "slightly hot"
levels (depending on genre) are all that I seek.


Well, if you are going for 'slightly hot' then you must be limiting or
clipping somewhere to keep within the actual hard limit of 0dbFS, so
irrespective of what the tool calls itself this cannot be said to be
normalising. I would also observe that a track which is quieter will
tend to sound 'tinny & weak' when compared to a louder track even without
being coded to MP3 first, it is just how humans work. I am sure that lots
of people here have stories of finding the 'extra something' at the end of
a long mix session by tweaking the control room monitor volume up a db or
two for the benefit of a client.

Many times I've
encountered WAVs from CDs that are, according to my standards, mastered
too loudly. When this is the case, I _do_ simply "rip-and-encode" the
unmodified, original WAVs (contrary to what dip**** said of me earlier
when he said I seek to indiscriminately normalize every CD I own). If
that were the case, I'd end up making half of what's in my library
*quieter* than it is on the original CDs, not louder!


Making them quieter does not undo the compression used to make them
boring to start with!

If normalization is _needed_ for a "mix-CD" sporting tracks by various
artists from various sources, I will "normalize" all the tracks
individually to bring them to a common loudness, across-the-board.


Well, this will get all the tracks to the same peak level, but they are
unlikely to all sound the same volume, as to a first approx, we hear RMS
not peak. What you probably want to do is (being careful that no
intermediate file clips (32 bit signed int may be a good intermediate
format for this)), is to equalise the average RMS levels (which will make
the peak level end up all over the place), then normalize the entire
collection to put the peak level into range. Your choice of averaging
function matters if you want good results.

However, depsite its name, the "normalize" application I use is capable
of doing more than just "textbook normalization" if I tell it to do so.
This has been a major stumbling block for me when I've previously
attempted to discuss its behaviour amongst others who've not used nor
even heard of it before.


Do you have a URL for a tarball?

"Normalize" can be made to limit the peaks
when instructed by the user to boost an RMS level beyond the textbook
normalization level. One of the guys over in the other newsgroup
suggested that we call this "limitizing" because there is no other,
readily available, predefined textbook term to describe it.


Sounds like fast attack hard knee limiting (Possibly with compression) to
me? Have you seen that compressor Dyson wrote when he was at free BSD?
It is excellent for this sort of thing.

Further the phycoacoustic model IIRC derives 'too quiet' threshold
from the RMS level of the signal, not an absolute threshold (IE at
some frequency it may be 30db below curent RMS level, NOT 40db below
0dbFS).


Mmm-HMMM.... Now *that's* something of which I was not previously aware.
If that's true then my hypothesis for boosting amps to save freqs may
indeed be fatally flawed.


Are you sure you're not thinking of frequency masking when you say this?


Yes, to a first approx, frequency masking weights towards energy in
nearby bands when calculating thresholds, what I am thinking of looks
at overall energy.

It is worth noting that whatever you do to the input data, it is only
possible to fit a fixed amount of information into the output
stream,


Now, I do understand that, however, I haven't contemplated it very much.


It is worth working thru the implications of this as it really brings home
the absence of a free lunch.

if you force the encoder to include more frequency data (higher
resolution or more bands active, then time or ampletude resolution
MUST necassarily suffer).


Excellent information. By time resolution you mean in that the file
would have to be made to play slower in order to accommodate the
increased amount of data?


No, not slower (which would imply a higher effective bit rate), but
that the information about when something happens may have to be stored
less precisely.

Also, would not such effects be greater at lower bitrates than at higher
ones?


All the compromises apply more at lower bitrates!

And if so, would this not mean that my hypothesis would actually become
more appropriately applied towards higher bitrate MP3s than at lower ones?


Because I'm guessing here, by reversing your logic, that a large enough
bitrate could eventually be employed which would cause the encoder to
either "pad the file with zeroes" or store the additional data depending
on the normalized status of the WAV being encoded.


Depends on if your hypothesis holds at all, I am yet to be convinced
that anything beyond psycological effects are at play here (Louder is
usaually perceved as better).

Consider that a 'perfect' data compression tool would simply store the
gain used in the normalisation once (after all it does not change during
playback), thus normalising has little effect on the amount of
*information* in the .wav file.

And if that's true, what bitrate may I be talking about?


There are a few lossless wav file compression tools around, some of them
are even reasonably good, find one then see how much it can reduce the
size of a typical wave file. This will give you some idea of how much
data is actually redundant in a wav file and of how much *information*
is required to represent that file.
It will be program dependent, a file containing a single 1Khz tone can
be losslessly represented in very few bits, a thrash metal gig will take
rather more (but why anyone would bother....).

Regards, Dan.
--
** The email address *IS* valid, do NOT remove the spamblock
And on the evening of the first day the lord said...........
..... LX 1, GO!; and there was light.
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Erwin Timmerman
 
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WBRW wrote:

The MP3 ENCODING is fine -- it's
just the DECODER that adds its own extra clipping upon playback.
I've seen downloaded MP3s where the file had to be knocked down by 6
dB, just to get it below the level of clipping during playback!


Do you know if winamp works the same way? Would setting the volume slider
to 80% or 50% help in this case?

Erwin Timmerman

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WBRW
 
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Do you know if winamp works the same way? Would setting the volume slider
to 80% or 50% help in this case?


No, that doesn't change the actual content of the MP3 file. I
recommend you use the "MP3GAIN" utility, which will let you losslessly
normalize your MP3 files to a "safe", non-clipping volume level. If
you try to increase the loudness value too much, it will warn you that
the MP3 file will start to clip. And in automatic mode with the "/r"
switch, it will normalize all of your MP3 files to a nearly constant
RMS volume level (again, always below the level of clipping) so that
you don't have to turn the volume up for a dynamic '70s or '80s song
and then have your ears blasted away by a modern "hyper-compressed"
song. Or, you have the option of applying a constant volume change to
an entire collection of songs, so that the relative loudness
differences between the tracks of an album will be maintained, for
example.

MP3GAIN is available for download he

http://www.geocities.com/mp3gain/

Windows GUI, Windows command line, and Linux versions are available
(sorry, Mac users!).
  #4   Report Post  
Ben Bradley
 
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In rec.audio.pro, Erwin Timmerman wrote:

WBRW wrote:

The MP3 ENCODING is fine -- it's
just the DECODER that adds its own extra clipping upon playback.
I've seen downloaded MP3s where the file had to be knocked down by 6
dB, just to get it below the level of clipping during playback!


Do you know if winamp works the same way? Would setting the volume slider
to 80% or 50% help in this case?


Winamp's volume slider just controls the 'wave' volume on the
Windows 'volume control' mixer, and so does nothing to affect the
digital playback. On my system with a Delta 66 card (which has it's
own mixer/control panel), the Winamp volume control does nothing.
So no, it wouldn't help.

Erwin Timmerman



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David Morgan \(MAMS\)
 
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"Lord Hasenpfeffer" wrote in message ...
Whoa! Do you mind discussing this with me a bit via private email?
I think you've got, if not *the* answer I've been looking for, the
overall *better idea* which I've been looking for and have been trying
and probably failing to "correctfully" solve on my own.

Some of the things you've said in this post could start a flame war in
other newsgroups. I know this. I've seen it happen. It even happened
to me once. But that's beside the point. I've got some questions for
you if you don't mind answering.



Someone has finally popped up with information reagrding actions that can
be taken on the MP3s themselves. How does this relate?

Odd that you should be so taken by this, as it is a totally subsequent matter
to the encoding process. He is in reference to actions taken on the MP3
file itself... *after* the encoding process in which the frequencies that you
sold blazenly desire to KEEP, are already gone !!!

Taking it off-group is perfectly fine, but terribly unfair to those of us that
want to learn something.

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


And with regard to MP3 compression, an MP3 file can be LOSSLESSLY
NORMALIZED on a frame-by-frame basis in 1.5 dB increments. The actual
compressed data is NOT changed -- only an ancillary "loudness
scale-factor". You can even LOSSLESSLY add fade-ins and fade-outs to
MP3 files, by changing the scale-factor on a frame-by-frame basis.
(The "MP3Trim" utility can do that.) So, if you're concerned about
this whole issue, simply pre-normalize the incoming WAV file by a
multiple of 1.5 dB, and then use a utility like "MP3GAIN" to
losslessly normalize it back down to the originally intended level.

In fact, you should be doing this anyway with any MP3 files you might
download, as today's over-compressed pop music often drives MP3
decoders into extreme amounts of clipping unless the level of the MP3
file is reduced to a "safe" value. (The MP3 ENCODING is fine -- it's
just the DECODER that adds its own extra clipping upon playback.)
I've seen downloaded MP3s where the file had to be knocked down by 6
dB, just to get it below the level of clipping during playback!





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Kurt Albershardt
 
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WBRW wrote:

I recommend you use the "MP3GAIN" utility,
which will let you losslessly
normalize your MP3 files to a "safe", non-clipping volume level. If
you try to increase the loudness value too much, it will warn you that
the MP3 file will start to clip. And in automatic mode with the "/r"
switch, it will normalize all of your MP3 files to a nearly constant
RMS volume level (again, always below the level of clipping) so that
you don't have to turn the volume up for a dynamic '70s or '80s song
and then have your ears blasted away by a modern "hyper-compressed"
song. Or, you have the option of applying a constant volume change to
an entire collection of songs, so that the relative loudness
differences between the tracks of an album will be maintained, for
example.

MP3GAIN is available for download he

http://www.geocities.com/mp3gain/



Anyone compared this with replaygain? They seem to be doing similar
things, but replaygain is doing it with tags rather than modification of
the data itself http://replaygain.hydrogenaudio.org/


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WBRW wrote:
I would be somewhat surprised to learn that any mainstream commercial
release has ever gone out without being normalised (using the strict
definition of the term) to somewhere within 1db of FS.


How about this:


Snip list of tracks

Well, 89.1% (depending on if that is power or voltage) is about 1 percent
or so below FS.

"The Sign" was Ace Of Base's biggest U.S. hit ever, reaching #1 on the
Billboard chart in 1994 -- yet, on the album, it actually has the
LOWEST peak level of all the tracks, only hitting 56.2% of full scale,
or -5.0 dB!


Sounds like they normalled the whole album to retain relative levels
across the tracks. This is a perfectly reasonable thing to do.

And with regard to MP3 compression, an MP3 file can be LOSSLESSLY
NORMALIZED on a frame-by-frame basis in 1.5 dB increments. The actual
compressed data is NOT changed -- only an ancillary "loudness
scale-factor". You can even LOSSLESSLY add fade-ins and fade-outs to
MP3 files, by changing the scale-factor on a frame-by-frame basis.


Indeed, I had forgotten what the resolution was, but was aware of this
functionality, not that it makes any difference if the hypothesis that
spawned this discussion holds, the damage is done by then. However the
very presence of this gain factor in each frame hints that any competent
codec should not be sensitive to input level except in so far as it
modifies this value (Otherwise you have more less entropy in the output
file then you could have, and that is bad in a compressed stream).

In fact, you should be doing this anyway with any MP3 files you might
download, as today's over-compressed pop music often drives MP3
decoders into extreme amounts of clipping unless the level of the MP3
file is reduced to a "safe" value. (The MP3 ENCODING is fine -- it's
just the DECODER that adds its own extra clipping upon playback.)
I've seen downloaded MP3s where the file had to be knocked down by 6
dB, just to get it below the level of clipping during playback!


Ouch, but I suppose some overshoot in the reconstructed audio is
going to be present....

Regards, Dan.
--
** The email address *IS* valid, do NOT remove the spamblock
And on the evening of the first day the lord said...........
..... LX 1, GO!; and there was light.
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Jonas Eckerman
 
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I recommend you use the "MP3GAIN" utility, which will let you

Anyone compared this with replaygain? They seem to be doing similar


The following text at the MP3Gain website makes me suspect that MP3Gain
implements Replay Gain for MP3 files. :-)

--8--
For a further explanation of Replay Gain (the statistical analysis done by
MP3Gain), see replaygain.hydrogenaudio.org
--8--

/Jonas
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Lord Hasenpfeffer
 
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WBRW wrote:

if you _must_ use MP3, you might as well make the best of it


May I add that to my tagline?
You really can make something so easy sound *so easy*!!

I'm reminded of the conclusion of "Breaker Morant" where the executee
says, "Don't make a mess of it, you *******s!"

-- and utilities like MP3Trim and MP3GAIN will definintely help.


Kewl and noted. Thank you, thank you.

People also apparently fail to realise that this "quiet sounds will be
eliminated" feature is RELATIVE, not ABSOLUTE.


To what does the "Absolute" in "Absolute Threshold of Hearing" refer?

It doesn't mean that "sounds at xx dB below full scale will be cut out".
Instead, it means something along the lines of "sounds at xx dB below
the peak level OF THAT PARTICULAR FRAME will be cut out".


What is your source for this information, if you don't mind me asking?
I'm not challenging you. I'd simply like to read more about it.

Any good "lossy" audio compression scheme is based upon the principles
of NMT and TMN -- "Noise Masks Tone" and "Tone Masks Noise", which are
pretty much self-explanatory for anyone with a little insight. If
these two parameters were based on FIXED levels, then the end result
would be sheer GARBAGE. Instead, they are always RELATIVE, within the
frequency ranges and instantaneous peak and RMS loudness levels of any
particular split-second instant (i.e. "frame") in the song.


OK, we seem to be dealing with the "masking" side of lossy compression
methods, not the Absolute Threshold of Hearing side of them. Correct me
if I'm wrong but ATH and masking are not the same. Masking is indeed
relative but, as I understand it, the Absolute Threshold of Hearing is
absolute - hence the inclusion of the word "Absolute" in its name. And
I am interpreting that to mean "absolute in reference to digital Full
Scale".

Also, this scheme actually works BACKWARDS compared to what people
seem to think. Take a piano solo with tape hiss in the background,
for example. When the piano is silent or very quiet, you can hear the
background hiss, so the MP3 encoder KEEPS the hiss. But when the
piano becomes loud enough to effectively "mask" the background hiss,
the encoder starts CUTTING OUT the "inaudible" hiss, since it is more
important to expend the available data bandwidth upon encoding the
piano's sound.


I don't see how people could naturally think the opposite of this to be
true - unless they're confusing the effects of masked audio removal with
standard noise reduction practices.

The beginning of "Steppin' Out" by Joe Jackson is a good example --


I have this and will check it out ASAP.

some MP2/MP3 encoders will cause particularly nasty artifacts
in the piano track as it fades in, in order to capture the intense
high-frequency percussion that's going on.


What do these nasty artifacts sound like? It will help me if I know for
what to listen.

Myke

--

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Bob Cain
 
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Lord Hasenpfeffer wrote:

To what does the "Absolute" in "Absolute Threshold of Hearing" refer?


Not much any more. It is continuously adapted to the
current "loudness" so has become rather an oxymoron. For a
sketch of the thrashing that has occured in Lame development
relative to this rather crucial issue see:

http://ee1.bradley.edu/~jodaman/dev/mp3/lame.html

I suggest again that you subscribe to the mp3encoder mailing
list where you can pose your questions to those who know
exactly what goes on and how your modifications will affect
the encoding process. You can subscribe at:

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


It doesn't mean that "sounds at xx dB below full scale will be cut out".
Instead, it means something along the lines of "sounds at xx dB below
the peak level OF THAT PARTICULAR FRAME will be cut out".


What is your source for this information, if you don't mind me asking?
I'm not challenging you. I'd simply like to read more about it.


You can ask the developers yourself. They are pretty good
with questions. I'm not sure whether they pay too much
attention to statements. :-)


Bob
--

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

A. Einstein


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Lord Hasenpfeffer
 
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Bob Cain wrote:

To what does the "Absolute" in "Absolute Threshold of Hearing" refer?


Not much any more. It is continuously adapted to the
current "loudness" so has become rather an oxymoron.


If the ATH does not use Full Scale as its point of reference what else
does it use?

For a sketch of the thrashing that has occured in Lame development
relative to this rather crucial issue see:

http://ee1.bradley.edu/~jodaman/dev/mp3/lame.html


OK, lemme click Send first.

I suggest again that you subscribe to the mp3encoder mailing
list where you can pose your questions to those who know
exactly what goes on and how your modifications will affect
the encoding process. You can subscribe at:

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


Your phrase "I suggest again" will undoubtedly be interpreted by others
in this group as "I suggested before and you have apparently ignored my
suggestion". However, this is your first post to this thread in this
newsgroup and I have not been checking in with the other thread in the
other newsgroup very much lately so this is the first I've heard of it
from you.

This would seem to be a very fine place to go for further information.
I can take my list of conclusions posted earlier here to over there and
see what they have to say about it.

Myke

--

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

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Erwin Timmerman
 
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Bob Cain wrote:

Lord Hasenpfeffer wrote:

To what does the "Absolute" in "Absolute Threshold of Hearing" refer?


Not much any more. It is continuously adapted to the
current "loudness" so has become rather an oxymoron. For a
sketch of the thrashing that has occured in Lame development
relative to this rather crucial issue see:

http://ee1.bradley.edu/~jodaman/dev/mp3/lame.html


OK, so there's the answer, at least for the LAME encoder. The ATH level is
dynamically adjusted to content. I must say that some other websites are not too
clear about it, and just talk about the FM-curve and that they apply it. To
what? To FS?

That seemed strange to me all along with all the varying playback levels that
can occur. And indeed now it shows that the effect is dynamically applied. Thus
changing the volume of a wave file does nothing but change the ATH level with
it. No difference whatsoever. No frequencies kicked out.

End of story.

Erwin Timmerman

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

You can ask the developers yourself. They are pretty good
with questions. I'm not sure whether they pay too much
attention to statements. :-)

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


This would seem to be a very fine place to go for further information.
I can take my list of conclusions posted earlier here to over there and
see what they have to say about it.


Myke, with all due respect... so far (to me) it would appear that your
communication problem is that you had rather push your conclusions
onto others, rather than to seek their validity through questioning.

Please heed Bob's last statement as repeated above.

Good luck to you.

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


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WBRW
 
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To what does the "Absolute" in "Absolute Threshold of Hearing" refer?

It is the standard "hearing sensitivity curve" that I believe Dolby
used to help develop their Noise Reduction systems. It is a curve
showing how the ear is most sensitive to mid-range frequencies (around
1-2 kHz) and how the sensitivity decreases quite a bit above 10 kHz
(in order words, high frequency sounds have to be LOUDER in order to
he heard). This curve attempts to show the QUIETEST sound levels that
the average human ear can detect across its frequency range (20 -
20,000 Hz).

Also, this spectral graph might help explain how MP3 encoding behaves:

http://rvcc2.raritanval.edu/ktek9053/noisetest.gif

The top half is the spectrum analysis of an uncompressed WAV file
containing white noise that fades out gradually, with a constant 2000
Hz triangle wave applied on top at a constant level (the "bars" are
the harmonics that this triangle wave generates).

The lower half is this same audio, as run through the popular
Fraunhofer "Fast" MP3 encoder at 96 kbps. You can see how it starts
to increasingly "cut out" the white noise as the tone begins to "mask"
more of it.

What do these nasty artifacts sound like? It will help me if I know for
what to listen.


It almost sounds like dead spots on magnetic tape -- the piano will
simply drop out momentarily, creating a "chattering" effect which is
actually quite common in low-quality lossy audio compression.
  #15   Report Post  
Lord Hasenpfeffer
 
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Erwin Timmerman wrote:

http://ee1.bradley.edu/~jodaman/dev/mp3/lame.html


OK, so there's the answer, at least for the LAME encoder. The ATH level is
dynamically adjusted to content. I must say that some other websites are not too
clear about it, and just talk about the FM-curve and that they apply it. To
what? To FS?

That seemed strange to me all along with all the varying playback levels that
can occur. And indeed now it shows that the effect is dynamically applied. Thus
changing the volume of a wave file does nothing but change the ATH level with
it. No difference whatsoever. No frequencies kicked out.

End of story.


This is definitely a good lead. This page does seem to indicate that at
least the _point of reference_ for the scale changes on a frame-by-frame
basis based upon the loudness of the maximum peak within each frame
rather than remaining fixed at digital Full Scale. The implications of
this, however, I do not yet fully comprehend therefore as far as I'm
concerned it is still a bit premature to declare "End of story" as you
so eagerly have. Nevertheless, I hope you are correct.

Myke

--

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Bob Cain
 
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Lord Hasenpfeffer wrote:

Your phrase "I suggest again" will undoubtedly be interpreted by others
in this group as "I suggested before and you have apparently ignored my
suggestion". However, this is your first post to this thread in this
newsgroup and I have not been checking in with the other thread in the
other newsgroup very much lately so this is the first I've heard of it
from you.


I suggested it to you very early in the thread that you
originally began in alt.audio.minidisc nearly a week ago.
Maybe it wasn't that long ago but it's begun to feel like
it. :-)


Bob
--

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

A. Einstein
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Bob Cain
 
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Erwin Timmerman wrote:


That seemed strange to me all along with all the varying playback levels that
can occur. And indeed now it shows that the effect is dynamically applied. Thus
changing the volume of a wave file does nothing but change the ATH level with
it. No difference whatsoever. No frequencies kicked out.

End of story.


To be fair to Myke, it shows that his hypothesis does in
fact make sense and has long been a major consideration
among the codec developers. Perhaps he is using an older
version of Lame in which it was in fact a fixed function and
he can perceive the subtleties involved.

A little research, however, would have shown that he was
reinventing the wheel.


Bob
--

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

A. Einstein
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Lord Hasenpfeffer
 
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Bob Cain wrote:

It shows that you were on the right track. It also says
that if you use a modern version of Lame you will get a more
effective application of the principle than you can effect
by global level changes of the file since they reconsider
and try to optimize the masking thresholds on a frame by
frame basis.


I absolutely agree with you and I appreciate your honest and fair
assessment of my situation.

All of this talk about newer versions of LAME has indeed made me excited
about upgrading for the first time in 2 years. If it can indeed - with
the combined use of MP3Gain - eliminate my perceived need to always
normalize my older, quieter WAVs then *great* - as it reduces both the
time and effort required to do what I have to do.

Myke

--

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WBRW
 
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Well, 89.1% (depending on if that is power or voltage) is about 1 percent
or so below FS.


In this context, 100% is referenced to a digital sample value of +/-
32767, which is the maximum level that 16-bit audio can handle, and is
commonly referenced as "full scale" or "0 dB".

Working backwards from this benchmark, 89.1% represents 1.00 dB below
full scale, or "-1.00 dB". That is a good "safety margin" to ensure
that no clipping occurs, however, even as far back as 1986, CDs were
commonly mastered with peak levels reaching exactly full-scale, or
"100%". However, that is only "acceptable" if a SINGLE digital sample
is at the 100% mark. If two or more samples in a row reach full
scale, that is defined to be "clipping", and on some equipment, it
will actually cause a "clip" light to illuminate when that occurs
during recording or playback.

In order to prevent this while still allowing for MASSIVE increases in
loudness, modern "hyper-compressed" CDs typical limit their maximum
level to something like 98% or 99%, or -0.1 or -0.2 dB. That way, you
can have 10 or 20 samples in a row all "slammed" against this
arbitrary limit, resulting in a "hacked-off" waveform that is
essentially a square wave -- but since it's not TECHNICALLY "clipping"
since it doesn't reach the 100% mark, the proper term for this is
"hard-knee limiting" -- which has become so common on CDs within the
past decade that it is the NORM... very few popular music CDs are NOT
subjected to large amounts of "hard-knee limiting" these days.

For more information on this, please see the following
well-illustrated web site:

http://rvcc2.raritanval.edu/ktek9053/cdpage

Note that on this web site, only the term "clipping" is used, because
that's merely what "hard-knee limiting" is an euphemism for (sort of
like putting extra chrome on a Toyota and calling it a "Lexus").


  #21   Report Post  
Lord Hasenpfeffer
 
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Bob Cain wrote:

So *this* is what a train-wreck looks like! ;-) ;-)


Not necessasarily. I'm still considering the part that says
that mild compression, that I would otherwise avoid, may
have a benefit relative to lossy encoding.


I was merely kidding with that comment - hence the wink, wink...

As for the mild compression... It was not my point to say that
compression itself is beneficial to the encoding but rather the higher
levels of the entire remainder of the song at the expense of perhaps 5
short peak bursts that seem to me to be more due to happenstance than
deliberation.

Then again, I am conducting some tests which do seem to confirm Geoff's
position on how much of a level boost is required before an aurally
detectable difference is produced in the sound of a resulting MP3. At
least with these tests I am getting some "hard numbers and examples"
which may be useful for supporting his position.

At this point I've been using "normalize" to shove the gain down on
copies of that "Sunday Bloody Sunday" WAV in -5dB increments to see how
far down I have to go before it flatlines.

At -40dB it's just barely identifiable.
At -50dB it's no longer there.

Encoding 128kb/s MP3s of it at -5dB, -10dB, -15dB, and -25dB down from
it's original level and then importing the MP3s and amplifying them back
to just under Full Scale with Audacity reveals no appreciable difference
at least to my ears - even though the animated close-ups I've been able
to create from various screenshots of these "loudness-restored" WAVs do
reveal visual discrepancies.

Then again, (1) I'm not listening to these WAVs with expert ears, (2)
I'm probably not using good source test material, and (3) I still don't
know if my version of notlame (which uses the LAME v3.70 engine) uses
fixed or relative ATH positioning.

I downloaded and installed a Linux sine tone generator and have been
able to get it to play some tones, but it's idea of "writing to a file"
on the hard drive involves dumping values to a text file instead of
creating a WAV file with which I can conduct more appropriate tests.

And, yes, I do recall the R.A.P. 5 suggestion...

Myke

--

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  #22   Report Post  
Geoff Wood
 
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"Lord Hasenpfeffer" wrote in message

At this point I've been using "normalize" to shove the gain down on
copies of that "Sunday Bloody Sunday" WAV in -5dB increments to see how
far down I have to go before it flatlines.


Mayber it would help to give "normalise" a capital "N", as it is the name of
an appplication, and thus aproper noun. It would lead to less confusion, as
would them giving it a sensible name in the first place.

geoff


  #23   Report Post  
Bob Cain
 
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Lord Hasenpfeffer wrote:

Encoding 128kb/s MP3s of it at -5dB, -10dB, -15dB, and -25dB down from
it's original level and then importing the MP3s and amplifying them back
to just under Full Scale with Audacity reveals no appreciable difference
at least to my ears - even though the animated close-ups I've been able
to create from various screenshots of these "loudness-restored" WAVs do
reveal visual discrepancies.


You should at least hear signifigant increases in the noise
level when you get as far as -25 dB. Don't you? Or are you
working at greater than 16 bits for these trials?


Bob
--

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

A. Einstein
  #24   Report Post  
Lord Hasenpfeffer
 
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Bob Cain wrote:

Lord Hasenpfeffer wrote:

Encoding 128kb/s MP3s of it at -5dB, -10dB, -15dB, and -25dB down from
it's original level and then importing the MP3s and amplifying them back
to just under Full Scale with Audacity reveals no appreciable difference
at least to my ears - even though the animated close-ups I've been able
to create from various screenshots of these "loudness-restored" WAVs do
reveal visual discrepancies.


You should at least hear signifigant increases in the noise
level when you get as far as -25 dB. Don't you? Or are you
working at greater than 16 bits for these trials?


That was *my* assumption to, however, I cannot. Then again, I'm
listening to a rather loud rock song. And never in my life have I
ever been compelled to stick my ear this close the floor either. So I
really don't know what one sounds like. My best guess would be
something kinda like the natural electronics noise my sound card makes
whenever I turn the volume on my speakers all the way up - only a lot
smoother (aka pink noise?).

I could not tell an aural difference between any of the attenuated and
subsequently amplified MP3s although the visual differences of the
original attenuated WAVs is readily apparent - so I know I'm working
with the correct set of files. At -25dB, the WAV was as thin as a
toothpick in Audacity, yet after reamplification, no difference!

That's why I jumped down to -50dB next and discovered it to be
"flatlining" or whatever that would be called. "Silent" might be more
appropriate - except for the noise of my soundcard at full vol.

I went back up to -40dB and could just barely make out the music at full
volume. So I reason that somewhere between -25dB and -40dB there
*should* be some kind of obvious drop-off point in terms of the
resultant MP3 quality but I haven't made it that far in my testing yet.

I got so bored with that I went off trying to find a sine wave tone
generator that could write a WAV to my hard drive. I found the
generator. But it doesn't write a WAV. I kept looking but haven't
found anything else yet.

Have you a suggestion for a better music source? What besides rock?

I could use A.L.W.'s "Phantom Of The Opera"...

I could use the obligatory Mozart, "Eine Kleine Nachtmusik" by way of
Sir Neville Marriner & Academy of St. Martin-In-The-Fields'...

Vivaldi's a favourite.

Or how about a little Shoenberg or Berg on Deutsche Grammophon?

Or Cage, maybe? I have some Cage. He's usually pretty quiet.

Oh, there's the required "Beethoven: Complete Symphonies" thing put out
via Musical Heritage Society.

Or how about Elvis? (Presley or Costello, take your pick.)

The Sex Pistols? Naaaah. That's getting too far back to where I am
right now...

Hmmm...

Myke

--

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  #25   Report Post  
Jonas Eckerman
 
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P.S. And does anyone here object to my use of the non-word "limitize"
when describing the standard behaviour of Normalize in terms of both


To me the word "limitize" looks like an unusual way of saying "limiting".
It does not look like a shorter way of saying "RMS normalization with
limiting".

So, yes, I object. If you use "limitize" in this discussion I know what you
mean, but if I see the same word used by someone else in another discussion
I would probably interpret it as "limiting".

I think calling the process "RMS normalizing" is fine.

Regards
/Jonas


  #26   Report Post  
Mark T. Wieczorek
 
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take your favorite piece of music, something fairly rich and compressed
already - i.e. without so much dynamic range.

Then take 10 second slices and attenuate each by 5 or 10 db or so,
intersperse them with silence.

This way you only have to run that 1 file and encode it and you'll have a
wide range of volumes right there.

Re-boost them back and you're done.

I.e.

10 seconds: 0db
5 seconds: silence
10 seconds: -10db
5 seconds: silence
10 seconds: -20db

etc. Should be easy to create in any audio program.

Regards,
Mark

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

http://www.prosoundreview.com/
User reviews of pro audio gear
  #27   Report Post  
Lord Hasenpfeffer
 
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Bob Cain wrote:

You should at least hear signifigant increases in the noise
level when you get as far as -25 dB. Don't you? Or are you
working at greater than 16 bits for these trials?


16-bit's all I'm working with here.

Here are the readings for the 8 source WAVs I've used so far...

level peak
-18.7053dBFS -5.9999dBFS N01-05.wav
-23.7053dBFS -11.0000dBFS N02-10.wav
-28.7053dBFS -16.0004dBFS N03-15.wav
-33.7048dBFS -21.0011dBFS N04-20.wav
-38.7053dBFS -26.0013dBFS N05-25.wav
-43.7052dBFS -30.9956dBFS N06-30.wav
-48.7052dBFS -36.0054dBFS N07-35.wav
-53.7048dBFS -41.0011dBFS N08-40.wav

In all 8 cases so far, I've copied an original source WAV, adjusted each
copy via "Normalize -g -5dB", "Normalize -g -10dB", etc., on down to
-40dB. Then I've encoded each attenuated WAV to MP3 at 128kb/ps. Then
I've imported each of the resulting MP3s into Audacity where I've then
amplified them up to "-0.5". Audacity shows Full Scale as +/- 1.0 with
'0' in the middle, so -0.5 is just slightly below FS. I then save the
final decompressed MP3-WAV files back as WAVs to my hard drive.
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.

level peak
-14.0943dBFS -0.9601dBFS N01-05.mp3.wav
-13.7098dBFS -0.7672dBFS N02-10.mp3.wav
-14.2900dBFS -1.0749dBFS N03-15.mp3.wav
-14.1163dBFS -0.9832dBFS N04-20.mp3.wav
-13.9006dBFS -0.9975dBFS N05-25.mp3.wav
-13.9302dBFS -1.0002dBFS N06-30.mp3.wav
-14.1153dBFS -0.9492dBFS N07-35.mp3.wav
-14.1134dBFS -1.0802dBFS N08-40.mp3.wav

Myke

--

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Windows...It's rebootylicious!!!
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  #28   Report Post  
Lord Hasenpfeffer
 
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Jonas Eckerman wrote:

I think calling the process "RMS normalizing" is fine.


I think this is a good term for it as well but I think a few days ago
the term "RMS normalizing" caused someone to believe that I was
attempting to set the RMS level to -0dBFS. (What a noise *that* would
make!) Since textbook-styled "peak normalizing" involves setting the
maximum peak level to -0dBFS. IIRC, this caused a major uproar as
everybody thought I was a *really* deaf, dumb and blind kid then.

Myke

--

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  #29   Report Post  
Lord Hasenpfeffer
 
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Mark T. Wieczorek wrote:

Re-boost them back and you're done.


Sorry... Caught it the 2nd time around (post-Send)!

Myke

--

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Windows...It's rebootylicious!!!
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  #30   Report Post  
Geoff Wood
 
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"Lord Hasenpfeffer" wrote in message
...
Jonas Eckerman wrote:

I think calling the process "RMS normalizing" is fine.


I think this is a good term for it as well but I think a few days ago
the term "RMS normalizing" caused someone to believe that I was
attempting to set the RMS level to -0dBFS. (What a noise *that* would
make!) Since textbook-styled "peak normalizing" involves setting the
maximum peak level to -0dBFS. IIRC, this caused a major uproar as
everybody thought I was a *really* deaf, dumb and blind kid then.


"RMS normalising" is not sufficient is one is implying that limiting or
compression is also being applied to prevent the otherwise likely clipping.

geoff




  #31   Report Post  
Geoff Wood
 
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"Lord Hasenpfeffer" wrote in message
...
Bob Cain wrote:

You should at least hear signifigant increases in the noise
level when you get as far as -25 dB. Don't you? Or are you
working at greater than 16 bits for these trials?


16-bit's all I'm working with here.

Here are the readings for the 8 source WAVs I've used so far...

level peak
-18.7053dBFS -5.9999dBFS N01-05.wav


Stop right there ! In the interests of not confusing people even further,
dont't label the first column 'level" . Label it correctly - "Average RMS
level".

geoff


  #32   Report Post  
Lord Hasenpfeffer
 
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Mark T. Wieczorek wrote:

etc. Should be easy to create in any audio program.


Definitely easier. Thanks.

I tried attenuating the segments with Audacity this time and got
dramtically different (i.e. noticeably worse) results with each
post-encode reamplification. There was a big difference especially
between what I believe (from memory) were the -20dB and -25dB segments.
I then listened back to just the first and last segments and could
tell a definite difference in the clarity of the vocals - but this is
not a time during which I can really pump it up for better detection of
artifacts or whatever. I'll have to continue later.

Interesting to see how Audacity's concept of a -5dB is so much more
severe than Normalize's. I guarantee you I was getting virtually
identical results everytime when I was attenuating with Normalize and
reamplifying with Audacity. But with Audacity alone, things were
noticeably different at each step along the way.

This Normalize seems to have some rather odd properties about it. I'm
starting to get the impression that what equates to -10dBFS RMS to me
using Normalize ain't nothing like what -10dBFS RMS means to you guys
using CoolEdit Pro or whatever.

I sure am wishing there was another Normalize user in here with whom I
could compare results because I'm *really* at a loss now to explain the
differences I have now witnessed.

Myke

--

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  #33   Report Post  
Lord Hasenpfeffer
 
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Geoff Wood wrote:

level peak
-18.7053dBFS -5.9999dBFS N01-05.wav



Stop right there ! In the interests of not confusing people even further,
dont't label the first column 'level" . Label it correctly - "Average RMS
level".


Sorry. That's what the program itself displays in my shell as it's
running. What you're seeing is exactly what I'm seeing. Perhaps this
is why the first time you asked me if I knew what "Average RMS level"
meant, I said "no" when really I apparently did by way of experience;
just not by name.

Myke

--

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  #34   Report Post  
Erwin Timmerman
 
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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.

Good to learn a thing or 2 though about the background process. But the end
result doesn't seem to suffer as much as you've thought. If 40 dB doesn't
matter much (although I'd say that amount should even matter with
uncompressed 16 bit audio!), think about how insignificant a change 5 dB
would make to the sound.

As a side-topic: on your own web site you write:

"MykecEdit, MykecView and all original content used to render this webpage
is copyright 2001-2003 by Myke Carter. All rights reserved. Unauthorized
duplication and/or use of any content contained herein is a violation of
applicable laws and just plain rude, so don't do that"

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

Erwin Timmerman

  #35   Report Post  
Jonas Eckerman
 
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This Normalize seems to have some rather odd properties about it.

Just looked up the docs. There are some things about your Normalize
application that makes it different from the majority of simple normalizers
used by CD rippers.

If anyone else is interested in reading about what the app does, the docs
are at:
http://www1.cs.columbia.edu/~cvaill/...ze/README.html

This text shows that the application does in fact not "normalize" based on
a file's (or batch's) RMS. It does some more stuff in an effort to not
completely destroy the sound.

Quote:
--8--
The volumes calculated are RMS amplitudes, which correspond (roughly) to
perceived volume. Taking the RMS amplitude of an entire file would not give
us quite the measure we want, though, because a quiet song punctuated by
short loud parts would average out to a quiet song, and the adjustment we
would compute would make the loud parts excessively loud.

What we want is to consider the maximum volume of the file, and normalize
according to that. We break up the signal into 100 chunks per second, and
get the signal power of each chunk, in order to get an estimation of
"instantaneous power" over time. This "instantaneous power" signal varies
too much to get a good measure of the original signal's maximum sustained
power, so we run a smoothing algorithm over the power signal (specifically,
a mean filter with a window width of 100 elements). The maximum point of
the smoothed power signal turns out to be a good measure of the maximum
sustained power of the file. We can then take the square root of the power
to get maximum sustained RMS amplitude.
--8--

I think that if you will ever again enter a discussion where the behaviour
of this application is a fundamental piece of the puzzle, you should post a
link to the applications documentation so that other participants have a
chance of knowing what it's doing with the sound.

If we had all known what the apps documentation says, a lot of
misunderstandings might could have been avoided.

Another 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--

This is probably the reason why some audio terms, as used by the author,
isn't used the way we're used to.

It is also a good hint that the reader of those docs should not take the
docs as a source for knowledge about terms used for audio/signal
processing.

Regards
/Jonas


  #36   Report Post  
David Morgan \(MAMS\)
 
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"Lord Hasenpfeffer" wrote in message ...

P.S. And does anyone here object to my use of the non-word "limitize"
when describing the standard behaviour of Normalize in terms of both its
default -12dBFS and my preferred -10dBFS RMS-normalization operation?


Let's don't pollute the waters any more than we have to. g
I think "RMS normalization" is the most standard.

DM


  #37   Report Post  
David Morgan \(MAMS\)
 
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"Lord Hasenpfeffer" wrote in message ...
Jonas Eckerman wrote:

I think calling the process "RMS normalizing" is fine.


I think this is a good term for it as well but I think a few days ago
the term "RMS normalizing" caused someone to believe that I was
attempting to set the RMS level to -0dBFS.


I don't really recall that. You would most certainly have had a mountain
of angry people knocking on *that* post. If I missed it... ah well, this is
getting long.

(What a noise *that* would make!)


And as I'm trying to say, even though not so destructive, limiting does
make a sort of 'noise', albeit low in level and not always readily audible
it still effects the perception, alters the sound and fatigues the ear.

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


  #38   Report Post  
David Morgan \(MAMS\)
 
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"Jonas Eckerman" wrote in message ...
Replying to myself again:

I think calling the process "RMS normalizing" is fine.


After reading the docs, I've changed my mind. "RMS normalizing" or "RMS
normalizing with limiting" is not enough to convey what the app does.

Quoting the documentation or providing a link to it is nessecary.

The authors explanation or what the app does:
http://www1.cs.columbia.edu/~cvaill/...DME.html#AEN88

Regards
/Jonas


I won't be specific, but I have a few quibbles with the 'presentation' of this
FAQ, mostly in use of terminology and vagueness of applying calculations.


  #39   Report Post  
David Morgan \(MAMS\)
 
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"Lord Hasenpfeffer" wrote in message ...

Interesting to see how Audacity's concept of a -5dB is so much more
severe than Normalize's. I guarantee you I was getting virtually
identical results everytime when I was attenuating with Normalize and
reamplifying with Audacity. But with Audacity alone, things were
noticeably different at each step along the way.


This is why my initial thoughts were to experiment more and with more
software... certainly more with your ears.

I sure am wishing there was another Normalize user in here with whom I
could compare results because I'm *really* at a loss now to explain the
differences I have now witnessed.


There are all sorts of 'normalize' users in here who understand corruption,
but perhaps not so many "Normalize" users.

I, for one, am simply glad that you are experimenting and listening.

DM


  #40   Report Post  
David Morgan \(MAMS\)
 
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"Jonas Eckerman" wrote in message . 1...

Another 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--

This is probably the reason why some audio terms, as used by the author,
isn't used the way we're used to.

It is also a good hint that the reader of those docs should not take the
docs as a source for knowledge about terms used for audio/signal
processing.



A-hah! I see that I'm not the only one who appears to believe that there's a
little 'voodoo' going on here. ;-)

It's kinda funny, the first paragraph that you qouted (I didn't repeat it here)
from the document, is exactly where I stopped reading it, came back and
stated that I didn't quite buy the terminology.

DM


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