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Amiga Plus 2002 #11
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Amiga Plus CD - 2002 - No. 11.iso
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Normalize
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README
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2002-10-27
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Note for Amiga users:
Please download and install the ixemul environment, available in my website.
It makes easier to install and increase compatibility with programs that use
ixemul.library.
Email me if you have problems: fr3dy@ya.com
------------------------------------------------------------------------
This is release 0.7.6 of Normalize, a wave file volume normalizer.
Copyright (C) 1999--2002, Chris Vaill <cvaill@cs.columbia.edu>
Normalize is an overly complicated tool for adjusting the volume of
wave files to a standard level. This is useful for things like
creating mixed CD's and mp3 collections, where different recording
levels on different albums can cause the volume to vary greatly from
song to song.
Send bug reports, suggestions, comments to cvaill@cs.columbia.edu.
normalize is free software. See the file COPYING for copying conditions.
-------------------------------------------------------------------------
INSTALLATION SYNOPSIS
./configure
make
make install
See the file INSTALL for more extensive directions.
See the man page, normalize.1, for usage.
-------------------------------------------------------------------------
DEPENDENCIES
These dependencies are optional. Normalize doesn't require any other
packages to compile and run.
MAD library (http://www.mars.org/home/rob/proj/mpeg/).
Normalize will use the MAD MPEG Audio Decoder library if you have it
(highly recommended). This gives normalize the ability to read mp3
files. MAD support in normalize was developed using MAD version
0.14.2b; other versions may not work.
You can run configure with the --without-mad option to turn off mp3
read support.
XMMS (http://www.xmms.org).
If you have xmms installed, the configure system will build the
xmms-rva plugin, which honors the relative volume adjustment frames
that normalize adds to ID3 tags. The option --disable-xmms prevents
the plugin from being built.
Audiofile library (http://oss.sgi.com/projects/audiofile/).
Normalize can use the audiofile library if version 0.2.2 or later is
available on your system. This gives normalize the ability to read
and write AIFF, AIFF-C, WAV, NeXT/Sun .snd/.au, Berkeley/IRCAM/CARL,
and whatever else the audiofile library people decide to implement
in the future.
Audiofile support is not turned on by default, because the built-in
WAV support is faster (only because it's specifically tailored for
PCM WAVs), and because I'm guessing most people only ever need to
normalize standard PCM WAV and mp3 files. If you only want to use
normalize on standard PCM WAV and mp3 files, you don't need
audiofile. If, however, you would like to be able to normalize all
the different audio file formats that audiofile handles, run
configure with the --with-audiofile option to turn on audiofile
support.
-------------------------------------------------------------------------
QUESTIONS AND ANSWERS
1 What platforms does normalize work on?
I've tested normalize on GNU/Linux and FreeBSD on x86,
Solaris on Sparc, and Irix on MIPS. I've heard that it
works on GNU/Linux on Alpha and on BeOS R5. As far as Win
dows is concerned, you can compile it using the Cygwin
toolkit. Question 7, below, contains a brief overview of
this process.
I've tried to make the code as portable as possible, so
I'd appreciate hearing whether normalize works on other
platforms.
2 What is normalize useful for?
Let's say you've got a bunch of wav files containing what
are, in your estimation, Elvis's greatest hits, collected
from various albums. You want to encode them as mp3's and
add them to an established collection, but since they're
all from different albums, they're all recorded at differ
ent volumes from each other and from the rest of your mp3
collection. If you've been using normalize on all your wav
files before you encode them, your collection is normal
ized to the default volume level, and you want these new
additions to be at the same level. Just run normalize with
no options on the files, and each will be adjusted to the
proper volume level:
normalize "Hound Dog.wav" "Blue Suede Shoes.wav" \
"Here Comes Santa Claus.wav" ...
Example 2. Suppose now you've just extracted all the wav
files from the Gorilla Biscuits album "Start Today,"
which, you may know, is recorded at a particularly low
volume. We want to make the whole album louder, but indi
vidual tracks should stay at the same volume relative to
each other. For this we use batch mode. Say the files are
named 01.wav to 14.wav, and are in the current directory.
We invoke normalize in batch mode to preserve the relative
volumes, but otherwise, everything's the default:
normalize -b *.wav
You can then fire up your mp3 encoder, and the whole album
will be uniformly louder.
Example 3. Now suppose we want to encode the Converge
album "When Forever Comes Crashing." This album has one
song, "Ten Cents," that is really quiet while the rest of
the songs have about the same (loud) volume. We'll turn
up the verbosity so we can see what's going on:
> normalize -bv *.wav
Computing levels...
Level for track01.cdda.wav: -9.3980dBFS (0.0000dBFS peak)
Level for track02.cdda.wav: -9.2464dBFS (-0.1538dBFS peak)
Level for track03.cdda.wav: -8.6308dBFS (-0.2520dBFS peak)
Level for track04.cdda.wav: -8.7390dBFS (0.0000dBFS peak)
Level for track05.cdda.wav: -8.1000dBFS (-0.0003dBFS peak)
Level for track06.cdda.wav: -8.2215dBFS (-0.1754dBFS peak)
Level for track07.cdda.wav: -8.9346dBFS (-0.1765dBFS peak)
Level for track08.cdda.wav: -13.6175dBFS (-0.4552dBFS peak)
Level for track09.cdda.wav: -9.0107dBFS (-0.1778dBFS peak)
Level for track10.cdda.wav: -8.1824dBFS (-0.4519dBFS peak)
Level for track11.cdda.wav: -8.5700dBFS (-0.1778dBFS peak)
Standard deviation is 1.47 dB
Throwing out level of -13.6175dBFS (different by 4.58dB)
Average level: -8.6929dBFS
Applying adjustment of -3.35dB...
The volume of "Ten Cents," which is track 8, is 4.58 deci
bels off the average, which, given a standard deviation of
1.47 decibels, makes it a statistical aberration (which
I've defined as anything off by more that twice the stan
dard deviation, but you can set a constant decibel thresh
old with the -t option). Therefore, it isn't counted in
the average, and the adjustment applied to the album isn't
thrown off because of one song. Although the aberrant
song's volume is not counted in the average, it is
adjusted along with the rest of the files.
Example 4. Finally, say you want to make a mixed CD of
80's songs for your mom or something. You won't allow any
80's songs to taint your hallowed mp3 collection, so the
absolute volumes of these tracks don't matter, as long as
they're all about the same, so mom doesn't have to keep
adjusting the volume. For this, use the mix mode option,
normalize -m *.wav
and each track will be adjusted to the average level of
all the tracks.
3 How does normalize work?
A little background on how normalize computes the volume
of a wav file, in case you want to know just how your
files are being munged:
The volumes calculated are RMS amplitudes, which corre
spond (roughly) to perceived volume. Taking the RMS ampli
tude 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 max
imum 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 maxi
mum sustained RMS amplitude.
As for the default target amplitude of 0.25 (-12dBFS),
I've found that it's pretty close to the level of most of
my albums already, but not so high as to cause a lot of
limiting on quieter albums. You may want to choose a dif
ferent target amplitude, depending on your music collec
tion (just make sure you normalize everything to the same
amplitude if you want it to all be the same volume!).
Regarding clipping: since version 0.6, a limiter is
employed to eliminate clipping. The limiter is on by
default; you don't have to do anything to use it. The 0.5
series had a -c option to turn on limiting, but that lim
iter caused problems with inexact volume adjustment. The
new limiter doesn't have this problem, and the -c option
is considered deprecated (it will be removed in version
1.0).
Please note that I'm not a recording engineer or an elec
trical engineer, so my signal processing theory may be
off. I'd be glad to hear from any signal processing wiz
ards if I've made faulty assumptions regarding signal
power, perceived volume, or any of that fun signal theory
stuff.
4 Why don't you normalize using peak levels instead of RMS amplitude?
Well, in early (unreleased) versions, this is how it
worked. I found that this just didn't work well. The vol
ume that your ear hears corresponds more closely with
average RMS amplitude level than with peak level. There
fore, making the RMS amplitude of two files equal makes
their perceived volume equal. (Approximately equal, any
way: certain frequencies sound louder at the same ampli
tude because the ear is just more sensitive to those fre
quencies. I may try to take this into account in a future
version, but that opens up a whole new can of worms.)
"Normalizing" by peak level generally makes files with
small dynamic range very loud and does nothing to files
with large dynamic ranges. There's not really any normal
ization being done, it's more of a histogram expansion.
That said, since version 0.5, you can use the --peak
option to do this in normalize.
5 Can you make normalize operate directly on mp3 files?
Version 0.7 and up can operate directly on MPEG audio
files. An mp3 file is decoded (using Robert Leslie's MAD
library) and analyzed on the fly, without the need for
large temporary WAV files. The mp3 file is then "adjusted"
by setting its relative volume adjustment information
(technically, an "RVA2" frame is set in its ID3v2 tag).
The advantage of this method is that the audio data
doesn't need to be touched, and you don't incur the cost
of re-encoding. The disadvantage is that your mp3 player
needs to read and use relative volume adjustment ID3
frames. The normalize distribution now includes a plugin
for xmms that honors volume adjustment frames. If you use
an mp3 player other than xmms, you'll have to bug the
author to support RVA2 frames in ID3 tags.
If you'd rather change the volume of the mp3 audio data
itself, you still have to decode to WAV, normalize the
WAV, and re-encode. A script, normalize-mp3, is included
in the normalize distribution to do this for you.
6 Can normalize operate on ogg vorbis files?
Version 0.8 will at least be able to read vorbis audio
files. Adjusting is harder, though: the problem is that,
unlike with ID3, as far as I know there's no standardized
volume adjustment tag for ogg. I could just use, say,
"VOLUME_ADJUST=X.XXdB" as an ogg comment, but there would
be no reason for players to support it.
It may be possible to twiddle the vorbis data itself to
alter the volume in a lossless way. I'm looking into
this, but it would be a big undertaking, not something
that would be finished anytime soon.
The current situation is that you have to decode to WAV,
normalize the WAV, and re-encode. The normalize-ogg
script is included in the normalize distribution to do
this for you.
7 How do I use normalize in Windows?
"I click on INSTALL but nothing happens. What's wrong?"
Okay, here's the deal: normalize is free software, written
for free operating systems such as Linux and FreeBSD.
These happen to be unix-style operating systems, so nor
malize generally works on other non-free flavors of unix
as well. Unlike Windows software, unix software such as
normalize is meant to run on many different operating sys
tems on many different architectures, so usually it comes
in source code form and you have to compile it for your
particular setup. If you are running some form of unix,
normalize should compile right out of the box (let me know
if it doesn't!). For other operating systems, such as
Amiga, BeOS, OS/2, or Windows, you may have to jump
through some hoops to get it to compile.
A discussion of compiling unix software for Windows is way
beyond the scope of this FAQ, but here's a quick rundown:
1. You first need the Cygwin toolkit
(http://www.cygwin.com). After installing, start up a cygwin
bash shell.
2. Go to the directory where you unzipped the normalize archive
-- it would be named something like normalize-x.y.z.
3. Type "./configure", then "make", then "make install"
4. If there were no errors, you can run normalize by typ ing
"normalize" at the prompt. Normalize is a command-line
utility, so you have to pass it command line options. Run
"normalize --help" for a synopsis.