Index of /www001/src/ports/audio/normalize/work/normalize-0.7.7/doc

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[DIR] fr/                     14-Sep-2005 11:27      -  
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<!DOCTYPE article PUBLIC "-//OASIS//DTD DocBook V4.2//EN">

<!--
    Copyright 2005 Chris Vaill.

    Permission is granted to copy, distribute, and/or modify this
    document under the terms of the GNU General Public License as
    published by the Free Software Foundation; either version 2 of the
    License, or (at your option) any later version.
-->

<article lang="en">

<articleinfo>
<title>Normalize</title>
<productname><application>normalize</application></productname>

<!--
<author>
<firstname>Chris</firstname>
<surname>Vaill</surname>
<affiliation>
<address>
<email>chrisvaill at gmail</email>
</address>
</affiliation>
</author>
-->

<!--
<revhistory>
<revision>
<revnumber>v1.0</revnumber>
<date>2001-09-16</date>
<authorinitials>cmv</authorinitials>
</revision>
</revhistory>
-->
<!--
<abstract>
<para>
Abstract here.
	</para>
</abstract>
-->

</articleinfo>

<para>
This is release 0.7.7 of Normalize, an audio file volume normalizer.
	</para>
<para>
Copyright &copy; 1999--2005, Chris Vaill &lt;chrisvaill at gmail&gt;
	</para>
<para>
Normalize is a tool for adjusting the volume of audio 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.
	</para>
<para>
Send bug reports, suggestions, comments to chrisvaill at gmail.
	</para>
<para>
<command>normalize</command> is free software.  See the file
<filename>COPYING</filename> for copying conditions.
	</para>

<!-- <bridgehead renderas="sect1">Installation synopsis</bridgehead> -->
<section>
<title>Installation synopsis</title>

<para>

<orderedlist>
<listitem><para><userinput>./configure</userinput>
<option><replaceable class="option">options</replaceable></option></para></listitem>
<listitem><para><userinput>make</userinput></para></listitem>
<listitem><para><userinput>make install</userinput></para></listitem>
</orderedlist>

See the file <filename>INSTALL</filename> for more extensive directions.
See the man page, <filename>normalize.1</filename>, for usage.
Run "<userinput>./configure --help</userinput>" for configure options.
	</para>
</section>

<!-- <bridgehead renderas="sect1">Dependencies</bridgehead> -->
<section>
<title>Dependencies</title>

<para>
These dependencies are <emphasis>optional</emphasis>.  Normalize
doesn't require any other packages to compile and run.
	</para>

<bridgehead renderas="other">
<ulink url="http://www.underbit.com/products/mad/">
MAD library (http://www.underbit.com/products/mad/)
</ulink>
</bridgehead>
<para>
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; earlier versions may not work.
	</para>
<para>
You can run <userinput>configure</userinput> with the
<userinput>--without-mad</userinput> option to turn off mp3 read
support.
	</para>

<bridgehead renderas="other">
<ulink url="http://www.68k.org/~michael/audiofile/">
Audiofile library (http://www.68k.org/~michael/audiofile/)
</ulink>
</bridgehead>
<para>
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.
	</para>
<para>
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
<userinput>configure</userinput> with the
<userinput>--with-audiofile</userinput> option to turn on audiofile
support.
	</para>

<bridgehead renderas="other">
<ulink url="http://www.xmms.org/">
XMMS (http://www.xmms.org/)
</ulink>
</bridgehead>
<para>
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
<userinput>--disable-xmms</userinput> prevents the plugin from being
built.
	</para>
</section>

<!-- <bridgehead renderas="sect1">Questions and Answers</bridgehead> -->
<section>
<title>Questions and Answers</title>

<QandAset>

<qandaentry id="Q1">
<question>
<para>What platforms does normalize work on?</para>
</question>
<answer>
<para>
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 Windows is concerned, you can compile
it using the <ulink url="http://www.cygwin.com/">Cygwin
toolkit</ulink>.  <link linkend="Q8">Question 8</link>, below,
contains a brief overview of this process.
	</para>
<para>
I've tried to make the code as portable as possible, so I'd appreciate
hearing whether normalize works on other platforms.
	</para>
</answer>
</qandaentry>

<qandaentry id="Q2">
<question>
<para>What is normalize useful for?</para>
</question>
<answer>
<formalpara><title>Example 1</title>
<para>
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 different 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 normalized 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:
<screen>
            normalize "Hound Dog.wav" "Blue Suede Shoes.wav" \
                      "Here Comes Santa Claus.wav" ...
</screen>
	</para>
</formalpara>
<formalpara><title>Example 2</title>
<para>
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
individual tracks should stay at the same volume relative to each
other.  For this we use <emphasis>batch mode</emphasis>.  Say the
files are named <filename>01.wav</filename> to
<filename>14.wav</filename>, and are in the current directory.  We
invoke normalize in batch mode to preserve the relative volumes, but
otherwise, everything's the default:
<screen>
            normalize -b *.wav
</screen>
You can then fire up your mp3 encoder, and the whole album will be
uniformly louder.
	</para>
</formalpara>
<formalpara><title>Example 3</title>
<para>
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:
<screen>
        > 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...
</screen>
The volume of "Ten Cents," which is track 8, is 4.58 decibels 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 standard deviation, but you can set a constant decibel
threshold with the <userinput>-t</userinput> 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.
	</para>
</formalpara>
<formalpara><title>Example 4</title>
<para>
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 <emphasis>mix mode</emphasis>
option,
<screen>
        normalize -m *.wav
</screen>
and each track will be adjusted to the average level of all the
tracks.
	</para>
</formalpara>
</answer>
</qandaentry>

<qandaentry id="Q3">
<question>
<para>How does normalize work?</para>
</question>
<answer>
<para>
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:
	</para>
<para>
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.
	</para>
<para>
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 <emphasis>sustained</emphasis> 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.
	</para>
<para>
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 different target amplitude, depending on your music
collection (just make sure you normalize everything to the same
amplitude if you want it to all be the same volume!).
	</para>
<para>
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 limiter 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).
	</para>
<para>
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.
	</para>
</answer>
</qandaentry>

<qandaentry id="Q4">
<question>
<para>
Why don't you normalize using peak levels instead of RMS amplitude?
	</para>
</question>
<answer>
<para>
Well, in early (unreleased) versions, this is how it worked. I found
that this just didn't work well.  The volume that your ear hears
corresponds more closely with average RMS amplitude level than with
peak level.  Therefore, making the RMS amplitude of two files equal
makes their perceived volume equal.  (Approximately equal, anyway:
certain frequencies sound louder at the same amplitude because the ear
is just more sensitive to those frequencies.  I may try to take this
into account in a future version, but that opens up a whole new can of
worms.)
	</para>
<para>
"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 normalization being done, it's more of a
histogram expansion.  That said, since version 0.5, you can use the
<userinput>--peak</userinput> option to do this in normalize if you're
sure it's what you really want to do.
	</para>
</answer>
</qandaentry>

<qandaentry id="Q5">
<question>
<para>Can normalize operate directly on mp3 files?</para>
</question>
<answer>
<para>
Version 0.7 and up can operate directly on MPEG audio files.  An mp3
file is decoded (using Robert Leslie's <ulink
url="http://www.underbit.com/products/mad/">MAD library</ulink>) and
analyzed on the fly, without the need for large temporary WAV files.
The mp3 file is then "adjusted" by setting its <emphasis>relative
volume adjustment</emphasis> 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.
	</para>
<!--
<para>
MPEG layer III audio files store audio data as Huffman-coded frequency
components.  Theoretically, frequency components can be analyzed and
adjusted just as easily as time-domain samples (such as the PCM
samples in a WAV file).  However, because the samples are
Huffman-coded, it is likely that decoding the samples, adjusting them,
and re-encoding them will cause them to take either more or fewer bits
than before.  This is a problem, since MPEG audio files have a set
bitrate.  Layer III files have a bit reservoir mechanism that could
absorb some of the changes, but essentially there's no guarantee that
adjusted audio data will still fit in the mp3 file.  The process of
getting the data to fit in the given bitrate is basically the same as
the encoding process.  That is, it's hard.
	</para>
-->
<para>
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, <userinput>normalize-mp3</userinput>, is included in the
normalize distribution to do this for you.
	</para>
</answer>
</qandaentry>


<qandaentry id="Q6">
<question>
<para>Can normalize operate on ogg vorbis files?</para>
</question>
<answer>
<para>
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.
	</para>
<para>
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.
	</para>
<para>
The current situation is that you have to decode to WAV, normalize the
WAV, and re-encode.  The <userinput>normalize-ogg</userinput> script
is included in the normalize distribution to do this for you.
	</para>
</answer>
</qandaentry>

<qandaentry id="Q7">
<question>
<para>How do I normalize a whole tree of files recursively?</para>
</question>
<answer>
<para>
The "unix way" to do this is to use <command>find</command>:
<screen>
find . -type d -exec sh -c "normalize -b \"{}\"/*.mp3" \;
</screen>
will go directory by directory, running normalize -b on all mp3 files
in each.  If you don't want batch mode, just:
<screen>
find . -name \*.mp3 -exec normalize {} \;
</screen>
will run normalize on each mp3 file separately.  If you want to run
normalize in batch or mix mode on all files in the directory tree, use:
<screen>
find . -name \*.mp3 -print0 | xargs -0 normalize -b
</screen>
	</para>
<para>
A built-in recurse option has been a very popular request, so I'm
adding support for it in version 0.8.
	</para>
</answer>
</qandaentry>

<qandaentry id="Q8">
<question>
<para>How do I use normalize in Windows?</para>
</question>
<answer>
<para>
"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 normalize 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 systems 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.
	</para>
<para>
A discussion of compiling unix software for Windows is way beyond the
scope of this FAQ, but here's a quick rundown:
  <orderedlist>
    <listitem>
    <para>
    You first need the <ulink url="http://www.cygwin.com">Cygwin
    toolkit</ulink>.  After installing, start up a cygwin bash shell.
	</para>
    </listitem>
    <listitem>
    <para>
    Go to the directory where you unzipped the normalize archive -- it
    would be named something like
    <filename>normalize-x.y.z</filename>.
	</para>
    </listitem>
    <listitem>
    <para>
    Type "<userinput>./configure</userinput>", then
    "<userinput>make</userinput>", then
    "<userinput>make&nbsp;install</userinput>"
	</para>
    </listitem>
    <listitem>
    <para>
    If there were no errors, you can run normalize by typing
    "<userinput>normalize</userinput>" at the prompt.  Normalize is a
    command-line utility, so you have to pass it command line options.
    Run "<userinput>normalize&nbsp;--help</userinput>" for a synopsis.
	</para>
    </listitem>

  </orderedlist>
	</para>
</answer>
</qandaentry>

</QandAset>

</section>

<section>
<title></title>
<para>
Copyright &copy; 1999--2005, Chris Vaill &lt;chrisvaill at gmail&gt;
	</para>
<para>
Permission is granted to copy, distribute, and/or modify this document
under the terms of the GNU General Public License as published by the
Free Software Foundation; either version 2 of the License, or (at your
option) any later version.
	</para>
</section>

</article>