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.


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INSTALLATION SYNOPSIS

./configure
make
make install

See the file INSTALL for more extensive directions.
See the man page, normalize.1, for usage.


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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.


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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.
