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