ref: e30f496ce4e358faf9598b94f8c6ea1b6c79a306
parent: 9bddfb82cc8aa239fe499dcfa3ed32edc72f2ed6
author: cbagwell <cbagwell>
date: Sun Dec 12 16:12:56 EST 1999
Updating documentation.
--- a/CHEAT
+++ /dev/null
@@ -1,110 +1,0 @@
-CHEAT SHEET
------------
-
-This is a cheat sheet of examples using SOX to do various common
-sound file conversions. The file format examples are starting
-to become dated. Any offers to update this document to explain
-the ends and outs of each format would be appreciated.
-
-In general, sox will attempt to take an input sound file format and
-convert it to a new file format using a similar data types and sample
-rates. For instance "sox monkey.au monkey.wav" would try and convert
-the mono 8000Hz u-law .au file to a 8000Hz u-law .wav file.
-
-If an output format doesn't support the same data types as the input
-file then Sox will generally select a default format to save it in.
-You can select a data type of your choice using command line options.
-You can also override data type values to have a output file of
-higher or lower percision data (and thus higher or lower file size).
-
-Most file formats that contain complete headers will automatically
-convert to a similar format. This means .wav, .aiff, and .voc files
-will readily convert to each other without the need of complex
-command lines.
-
-If you create a sound file and you can not play it, check to make
-sure your sound card to play a file using this data type.
-
-SOX is great to use along with other command line programs. The
-currently most used example is to use mpg123 to convert mp3 files
-in to wav files. The following command line will do this:
-
-mpg123 -b 10000 -s filename.mp3 | sox -t raw -r 44100 -s -w -c 2 - filename.wav
-
-The SUN examples below all assume you have the old SUN voice-quality 8khz
-u-law hardware. If you do then you will want to have all your .au
-files in this format so that you cat do thinks like
-"cat soundfile.au > /dev/audio" and you will hear a good file.
-If the .AU file doesn't have a proper header, then you'll need the second
-command line to let sox know the values. If the .AU has a proper
-header then you can remove the "-r 8000 -U -b" in front of
-"file.au".
-
-SUN .au to Mac .snd:
-
- sox file.au -r 11025 -t ub file.snd
-or:
- sox -t ul -r 8000 file.au -r 11025 -t ub file.snd
-
-When you copy the file to the Mac, you'll have to set
-the sample rate by hand.
-
-Mac .snd to SUN .au
-
- sox -r 11025 -t ub file.snd -r 8000 -U -b file.au
-
-The Mac file might also be at sample rates 5012, 22050, or 44100.
-
-PC .voc to SUN .au
-
- sox file.voc -r 8000 -U -b file.au
-
-SUN .au to PC .voc
-
- sox file.au file.voc
-or:
- sox -r 8000 -t ul file.au file.voc
-
-SUN .au to WAV - without clipping
-
- sox file.au -s -w file.wav
-or:
- sox -t ul -r 8000 file.au -s -w file.wav
-
-WAV to SUN .au
-
- sox file.wav -r 8000 -U -b file.au
-
-WAV to VOC
- sox file.wav -u -b file.voc
-
-VOC to WAV
- sox file.voc file.wav
-
-Any file to SUN .au
-
-sox -t auto file.X -c 1 -t aiff - | sox -t aiff - -r 8000 -U -b -t au file.au
-
-Just convert file format without making a disk file.
-Example: convert input stream in AIFF format to output stream in WAV format:
-
-sox -t aiff - -t wav -
-
-Some people try to put this kind of command in scripts.
-
-It is important to understand how the internals of Sox work when
-working with compressed audio, including u-law, a-law, ADPCM, or GSM.
-Sox takes ALL input data types and converts them to uncompressed
-32-bit signed data. It will then convert this internal version into
-the requested output format. This means unneeded noise can be introduced
-from decompressing data and then recompressing, such as would happen
-when reading u-law data and writing back out u-law data. If possible,
-specify the output data to be uncompressed PCM.
-
-Under DOS, you can convert several using something similar to the
-following command line:
-
-FOR %X IN (*.RAW) DO sox -r 11025 -w -s -t raw $X $X.wav
-
-Good luck!
-
--- a/CHEAT.eft
+++ /dev/null
@@ -1,280 +1,0 @@
-This is a cheat sheet of examples using SOX to
-add effects on sound files.
-
-Introduction:
-
-The core problem is that you need some experience in using effects
-in order to say "that any old sound file sounds with effects
-absolutely hip". There isn't any rule-based system which tell you
-the correct setting of all the parameters for every effect.
-But after some time you will become an expert in using effects.
-
-Here are some examples which can be used with any music sample.
-(For a sample where only a single instrument is playing, extreme
-parameter setting may make well-known "typically" or "classical"
-sounds. Likewise, for drums, vocals or guitars.)
-Single effects will be explained and some given parameter settings
-that can be used to understand the theorie by listening to the sound file
-with the added effect.
-
-Using multiple effects in parallel or in sequel can result either
-in very perfect sound or ( mostly ) in a dramatic overloading in
-variations of sounds such that your ear may follow the sound but
-you will feel unsatisfied. Hence, for the first time using effects
-try to compose them as less as possible. We don't regard the
-composition of effects in the examples because to many combinations
-are possible and you really need a very fast maschine and a lot of
-memory to play them in real-time.
-
-And real-time playing of sounds will speed up learning the parameter
-setting.
-
-Basically, we will use the "play" front-end of SOX since it is easier
-to listen sounds coming out of the speaker or earphone instead
-of looking at cryptical data in sound files.
-
-For easy listening of file.xxx ( "xxx" is any sound format ):
-
- play file.xxx effect-name effect-parameters
-
-Or more SOX-like ( for "dsp" output ):
-
- sox file.xxx -t ossdsp -w -s /dev/dsp effect-name effect-parameters
-
-or ( for "au" output ):
-
- sox file.xxx -t sunau -w -s /dev/audio effect-name effect-parameters
-
-And for date freaks:
-
- sox file.xxx file.yyy effect-name effect-parameters
-
-Additional options can be used. However, in this case, for real-time
-playing you'll need a very fast machine.
-
-Notes:
-
-I played all examples in real-time on a Pentium 100 with 32 Mb and
-Linux 2.0.30 using a self-recorded sample ( 3:15 min long in "wav"
-format with 44.1 kHz sample rate and stereo 16 bit ).
-The sample should not contain any of the effects. However,
-if you take any recording of a sound track from radio or tape or cd,
-and it sounds like a live concert or ten people are playing the same
-rhythm with their drums or funky-groves, then take any other sample.
-(Typically, less then four different intruments and no synthesizer
-in the sample is suitable. Likewise, the combination vocal, drums, bass
-and guitar.)
-
-Effects:
-
-Echo
-----
-
-An echo effect can be naturally found in the mountains, standing somewhere
-on a moutain and shouting a single word will result in one or more repetitions
-of the word ( if not, turn a bit around ant try next, or climb to the next
-mountain ).
-
-However, the time difference between shouting and repeating is the delay
-(time), its loudness is the decay. Multiple echos can have different delays and
-decays.
-
-Very popular is using echos to play an instrument with itself together, like
-some guitar players ( Brain May from Queen ) or vocalists are doing.
-For music samples of more than one instrument, echo can be used to add a
-second sample shortly after the original one.
-This will sound as doubling the number of instruments playing the same sample:
-
- play file.xxx echo 0.8 0.88 60.0 0.4
-
-If the delay is very short then it sound like a (metallic) roboter playing
-music:
-
- play file.xxx echo 0.8 0.88 6.0 0.4
-
-Longer delay will sound like a open air concert in the mountains:
-
- play file.xxx echo 0.8 0.9 1000.0 0.3
-
-One mountain more, and:
-
- play file.xxx echo 0.8 0.9 1000.0 0.3 1800.0 0.25
-
-Echos
------
-
-Like the echo effect, echos stand for "ECHO in Sequel", that is the first echos
-takes the input, the second the input and the first echos, the third the input
-and the first and the second echos, ... and so on.
-Care should be taken using many echos ( see introduction ); a single echos
-has the same effect as a single echo.
-The sample will be bounced twice in symmetric echos:
-
- play file.xxx echos 0.8 0.7 700.0 0.25 700.0 0.3
-
-The sample will be bounced twice in asymmetric echos:
-
- play file.xxx echos 0.8 0.7 700.0 0.25 900.0 0.3
-
-The sample will sound as played in a garage:
-
- play file.xxx echos 0.8 0.7 40.0 0.25 63.0 0.3
-
-Chorus
-------
-
-The chorus effect has its name because it will often be used to make a single
-vocal sound like a chorus. But it can be applied to other instrument samples
-too.
-
-It works like the echo effect with a short delay, but the delay isn't constant.
-The delay is varied using a sinodial or triangular modulation. The modulation
-depth defines the range the modulated delay is played before or after the
-delay. Hence the delayed sound will sound slower or faster, that is the delayed
-sound tuned around the original one, like in a chorus where some vocal are
-a bit out of tune.
-
-The typical delay is around 40ms to 60ms, the speed of the modualtion is best
-near 0.25Hz and the modulation depth around 2ms.
-
-A single delay will make the sample more overloaded:
-
- play file.xxx chorus 0.7 0.9 55.0 0.4 0.25 2.0 -t
-
-Two delays of the original samples sound like this:
-
- play file.xxx chorus 0.6 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4 1.3 -s
-
-A big chorus of the sample is ( three additional samples ):
-
- play file.xxx chorus 0.5 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4 2.3 -t \
- 40.0 0.3 0.3 1.3 -s
-
-Flanger
--------
-
-The flanger effect is like the chorus effect, but the delay varies between
-0ms and maximal 5ms. It sound like wind blowing, sometimes faster or slower
-including changes of the speed.
-
-The flanger effect is widely used in funk and soul music, where the guitar
-sound varies frequently slow or a bit faster.
-
-The typical delay is around 3ms to 5ms, the speed of the modulation is best
-near 0.5Hz.
-
-Now, let's groove the sample:
-
- play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -s
-
-listen carefully between the difference of sinodial and triangular modulation:
-
- play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -t
-
-If the decay is a bit lower, than the effect sounds more popular:
-
- play file.xxx flanger 0.8 0.88 3.0 0.4 0.5 -t
-
-The drunken loundspeaker system:
-
- play file.xxx flanger 0.9 0.9 4.0 0.23 1.3 -s
-
-Reverb
-------
-
-The reverb effect is often used in audience hall which are to small or to many
-visitors disturb the reflection of sound at the walls to make the sound played
-more monumental. You can try the reverb effect in your bathroom or garage or
-sport halls by shouting loud some words. You'll hear the words reflected from
-the walls.
-
-The biggest problem in using the reverb effect is the correct setting of the
-(wall) delays such that the sound is relistic an doesn't sound like music
-playing in a tin or overloaded feedback distroys any illusion of any big hall.
-To help you for much realisitc reverb effects, you should decide first, how
-long the reverb should take place until it is not loud enough to be registered
-by your ears. This is be done by the reverb time "t", in small halls 200ms in
-bigger one 1000ms, if you like. Clearly, the walls of such a hall aren't far
-away, so you should define its setting be given every wall its delay time.
-However, if the wall is to far eway for the reverb time, you won't hear the
-reverb, so the nearest wall will be best "t/4" delay and the farest "t/2".
-You can try other distances as well, but it won't sound very realistic.
-The walls shouldn't stand to close to each other and not in a multiple interger
-distance to each other ( so avoid wall like: 200.0 and 202.0, or something
-like 100.0 and 200.0 ).
-
-Since audience halls do have a lot of walls, we will start designing one
-beginning with one wall:
-
- play file.xxx reverb 1.0 600.0 180.0
-
-One wall more:
-
- play file.xxx reverb 1.0 600.0 180.0 200.0
-
-Next two walls:
-
- play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0
-
-Now, why not a futuristic hall with six walls:
-
- play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0 280.0 300.0
-
-If you run out of machine power or memory, then stop as much applications
-as possible ( every interupt will consume a lot of cpu time which for
-bigger halls is absolutely neccessary ).
-
-Phaser
-------
-
-The phaser effect is like the flanger effect, but it uses a reverb instead of
-an echo and does phase shifting. You'll hear the difference in the examples
-comparing both effects ( simply change the effect name ).
-The delay modulation can be done sinodial or triangular, preferable is the
-later one for multiple instruments playing. For single instrument sounds
-the sinodial phaser effect will give a sharper phasing effect.
-The decay shouln't be to close to 1.0 which will cause dramatic feedback.
-A good range is about 0.5 to 0.1 for the decay.
-
-We will take a parameter setting as for the flanger before ( gain-out is
-lower since feedback can raise the output dramatically ):
-
- play file.xxx phaser 0.8 0.74 3.0 0.4 0.5 -t
-
-The drunken loundspeaker system ( now less alkohol ):
-
- play file.xxx phaser 0.9 0.85 4.0 0.23 1.3 -s
-
-A popular sound of the sample is as follows:
-
- play file.xxx phaser 0.89 0.85 1.0 0.24 2.0 -t
-
-The sample sounds if ten springs are in your ears:
-
- play file.xxx phaser 0.6 0.66 3.0 0.6 2.0 -t
-
-Other effects ( copy, rate, avg, stat, vibro, lowp, highp, band, reverb )
--------------
-
-The other effects are simply to use. However, an "easy to use manual" should
-be given here.
-
-More effects ( to do ! )
-------------
-
-There are a lot of effects around like noise gates, compressors, waw-waw,
-stereo effects and so on. They should be implemented making SOX to be more
-useful in sound mixing technics coming together with a great varity of
-different sound effects.
-
-Combining effects be using then in parallel or sequel on different channels
-needs some easy mechanism which is real-time stable.
-
-Really missing, is the changing of the parameters, starting and stoping of
-effects while playing samples in real-time!
-
-
-Good luck and have fun with all the effects!
-
- Juergen Mueller (jmueller@uia.ua.ac.be)
-
--- a/Changelog
+++ b/Changelog
@@ -60,6 +60,9 @@
o Fixed clip24() range as pointed out by Ted Powell.
o Fixed possible segfault in echos effect, as pointed out by Zinx
Verituse.
+ o Moved most documentation to new "soxexam.1" manual page so that
+ all users on a unix system will have access to important information.
+ This means no more TIPS, CHEATS, or CHEATS.eft files.
sox-12.16
---------
--- a/INSTALL
+++ b/INSTALL
@@ -57,12 +57,14 @@
can be obtained from http://www.cs.tu-berlin.de/~jutta/toast.html
After installing the GSM library you must point to this file by
commenting and modifying the appropriate section of the Makefile.
+The "configure" script should be able to detect the installation of
+the external GSM library and compile in support on its own.
Optional Compile Options
------------------------
If you're processing lots of u-law or a-law files, you should
-define FAST_ULAW_COMPRESSION and/or FAST_ALAW_COMPRESSION in your
+define FAST_ULAW_CONVERSION and/or FAST_ALAW_CONVERSION in your
Makefile. These substitute a table-based method for the standard method.
The tables are 32K, so if you don't want them, you don't have to
use them. It is automatically added to your Makefile if you run
--- a/Makefile.gcc
+++ b/Makefile.gcc
@@ -208,6 +208,7 @@
man: sox.1 libst.3
$(RM) sox.txt libst.txt
nroff -man sox.1 | col -b > sox.txt
+ nroff -man soxexam.1 | col -b > sox.txt
nroff -man libst.3 | col -b > libst.txt
install: sox
@@ -214,7 +215,7 @@
if [ -f $(BINDIR)/rec ] ; then $(RM) $(BINDIR)/rec ; fi
if [ -f $(MANDIR)/man1/rec.1 ] ; then $(RM) $(MANDIR)/man1/rec.1 ; fi
install -c -m 755 sox play $(BINDIR)
- install -c -m 644 sox.1 play.1 $(MANDIR)/man1
+ install -c -m 644 sox.1 soxexam.1 play.1 $(MANDIR)/man1
ln -s $(BINDIR)/play $(BINDIR)/rec
ln -s $(MANDIR)/man1/play.1 $(MANDIR)/man1/rec.1
--- a/README
+++ b/README
@@ -2,7 +2,7 @@
SoX (also known as Sound eXchange) translates sound samples between different
-file formats, and optionally performs various sound effects.
+file formats, and optionally performs various sound effects.
This release understands:
@@ -10,8 +10,9 @@
o Raw textual data
o Microsoft .WAV files
o PCM, u-law, a-law
- o MS ADPCM (Read only)
- o IMA ADPCM (Read only)
+ o MS ADPCM
+ o IMA ADPCM
+ o GSM
o MAUD files
o Sound Blaster .VOC files
o IRCAM SoundFile files
@@ -31,19 +32,22 @@
The sound effects include:
o Channel Averaging
- o Band-pass filter
+ o Band-pass filters
+ o Band-reject filter
o Chorus effect
o Cut out loop samples
o Add an echo
o Add a sequence of echos
o Apply a flanger effect
- o Apply a high-pass filter
- o Apply a low-pass filter
+ o Apply a high-pass filters
+ o Apply a low-pass filters
o Display a list of loops in a file
o Add masking noise to a signal
o Apply a phaser effect
o Convert from stereo to mono
- o Change sampling rates using several different algorithms.
+ o Change sampling rates using several different algorithms.A
+ 'resample' and 'poyphase' effect use high-grade signal rate
+ changes using real signal theory!
o Apply a reverb effect
o Reverse the sound samples (to search for Satanic messages ;-)
o Convert from mono to stereo
@@ -51,12 +55,6 @@
o Display general stats on a sound sample
o Add the world-famous Fender Vibro-Champ effect
-Big news! Lots of new effects have been added. This includes most the
-popular "Guitar Effects" talked about in the same named FAQ available.
-
-The 'resample' and 'polyphase' effect does high-grade signal rate
-changes using real signal theory. Yes, it's very slow.
-
History:
This is the 12th release, Patchlevel 17 of the Sound Tools.
@@ -67,6 +65,9 @@
Caveats:
+Technically, SoX is made up of a sound file processing library called libst
+and SoX is a program implemented using this library.
+
SoX is intended as the Swiss Army knife of sound processing tools. It
doesn't do anything very well, but sooner or later it comes in very handy.
SoX is really only usable day-to-day if you hide the wacky options with
@@ -78,16 +79,10 @@
SoX as described in the INSTALL file. Please read that file for further
instructions.
-Now, read TIPS, CHEAT.eft and CHEAT. These give a background on how
+Now, read sox.txt and soxexam.txt. This gives a background on how
SoX deals with sound files and how to convert this format
to that format, and apply various effects.
-SoX uses file suffices to determine the nature of a sound sample file.
-If it finds the suffix in its list, it uses the appropriate read
-or write handler to deal with that file. You may override the suffix
-by giving a different type via the '-t type' argument. See the manual
-page for more information.
-
SoX has an auto-detect feature that attempts to figure out
the nature of an unmarked sound sample. It works very well.
This feature is used if you specify '-t auto' for the file type.
@@ -95,11 +90,6 @@
I hope to inspire the creation of a common base of sound processing
tools for computer multimedia work, similar to the PBM toolkit for
image manipulation.
-
-Sound Tools may be used for any purpose. Source distributions must
-must include the copyright notices, and (lack of) warranty information.
-Binary distributions must include acknowledgements to the creators.
-Files are copyright by their respective authors.
If you have bug fixes/enhancements, please send it to me as I would like
to coordinate the releases. Please document your changes. I don't
--- a/TIPS
+++ /dev/null
@@ -1,145 +1,0 @@
-
-SOX usage:
- sox [options] from-file-args to-file-args [ effect [effect-args]]
-
-First off: the -V option makes SOX print out its idea of
-what it is doing. -V is your friend.
-
- sox -V from-file-args to-file-args
-
-From-file-args and to-file-args are the same.
-They are a series of options followed by a file name.
-The suffix on the file name usually is the file format type.
-The '-t xx' option overrides this and tells sox
-the the file format is 'xx'. The '-u/-s/-U' arguments
-say that the file is in unsigned, signed, or u-law format.
-The '-b/-w' arguments say that the file is in byte- or
-word-size (2 byte) samples. The '-r number' argument
-says that the sample rate of the file is 'number'.
-
-The extensions ub, uw, sb, sw, and ul correspond
-to raw data files of formats unsigned byte, unsigned
-word, signed byte, signed word, and u-law byte.
-Thus, '-t ul' is shorthand for '-t raw -U -b'.
-
-These conversions clip data and thus reduce sound quality,
-so be careful:
-
- Word to u-law.
- Word to byte.
- U-law to byte.
- Reduction in sample rate.
-
-Any reduction in the sample data rate loses information
-and adds noise. An increase in the data rate doesn't
-lose much information, but does add noise. See the
-note below on low-pass filtering.
-
-To convert U-law to something else without clipping,
-you'll have to convert it to (signed or unsigned) words,
-which will double the size of the file.
-
-AUTO files:
-The 'AUTO' file type reads an unknown file and
-attempts to discern its binary format.
-
-AIFF files:
-AIFF files come with complete headers and other
-info. They can in fact have multiple sound
-chunks and picture chunks. SOX only reads
-the first sound chunk.
-
-WAV files:
-WAVs use the RIFF format, which is Microsoft's
-needless imitation of AIFF. See above comments.
-
-AIFF and RIFF files need their own librarian
-programs; SOX can only do a small fraction of
-what they need.
-
-It's best if you can copy or store files in
-AIFF or WAV format. The sample rate and
-binary format are marked; also comments may
-be added to the file.
-
-SUN AU files:
-Most AU files you find are in 8khz 8-bit u-law format.
-This format was the first sound hardware SUN made available.
-Some of the files have correct headers; some do not.
-If the file has the header, this should convert it to
-another format:
-
- sox file.au to-file-args
-
-If not, this reads a raw u-law 8khz file:
-
- sox -t ul -r 8000 file.au to-file-args
-
-To convert a file to an old-style SUN .au file:
-
- sox from-file-args -r 8000 -U -b file.au
-
-AU format can have any speed and several data sizes;
-you need to specify '-r 8000 -U -b' to force SOX to
-use the old SUN format.
-
-Mac files:
-Mac files come in .snd, .aiff, and .hcom formats,
-among others; these are the most common.
-
-SND files are in unsigned byte format with no
-header. They are either 11025, 22050, or 44100 hz.
-The speed seems to be a "resource" and doesn't
-get transported to Unix when the files are.
-Thus, you just have to know.
-
- sox -r 11025 -t ub file.snd to-file-args
- sox from-file-args -r 11025 -t ub file.snd
-
-PC files:
-There are several PC sound file formats. VOC is
-common; it has headers. SND and SNDR are for
-some DOS sound package; I don't know much about them.
-WAV is the official Microsoft Windows format.
-WAV has format options for compressed sound;
-SOX doesn't implement this yet.
-
-
-Effects:
-A sound effect may be applied to the sound sample
-while it is being copied from one file to another.
-Copy is the default effect; i.e. do nothing.
-Changing the sample rate requires the 'rate'
-effect. This applies a simple linear interpolation
-to the sample. This is a poor-quality sample
-changer. After doing a rate conversion,
-you should try doing a low-pass filter to throw
-away some of the induced noise. Pick a 'center'
-frequency about 85% of the lower of the two
-frequencies, or 42.5% of the lower of the
-two sample rates. (The maximum frequency
-in a sample is 1/2 of the sample rate).
-
- sox -r 8000 file.xx -r 22050 tmp.yy
- sox tmp.yy file.yy lowp 3400
-or:
- sox -r 44100 file.xx -r 22050 tmp.yy
- sox tmp.yy file.yy lowp 9592
-
-Listen to both tmp.yy and file.yy and see if
-the low-pass filter helps. Be sure to do the
-low-pass filter before clipping the data to
-a smaller binary word size. Say you have a 16-bit
-CD-quality (44100 hz) AIFF file that you want
-to convert to a Mac sound resource:
-
- sox -r 44100 file.aiff -r 11025 tmp.sw
- sox tmp.sw -t ub file.mac lowp 9371
-
-not:
-
- sox -r 44100 file.aiff -r 11025 tmp.ub
- sox tmp.ub -t ub file.mac lowp 9371
-
-because you want to do the low-pass filter while
-you still have sixteen-bit data.
--- a/configure
+++ b/configure
@@ -1882,11 +1882,11 @@
fi
if test "$enable_fast_ulaw" = yes
then
- CFLAGS="$CFLAGS -DFAST_ULAW_COMPRESSION"
+ CFLAGS="$CFLAGS -DFAST_ULAW_CONVERSION"
fi
if test "$enable_fast_alaw" = yes
then
- CFLAGS="$CFLAGS -DFAST_ALAW_COMPRESSION"
+ CFLAGS="$CFLAGS -DFAST_ALAW_CONVERSION"
fi
LIBS="-L. -lst $LIBS"
--- a/libst.txt
+++ b/libst.txt
@@ -1,212 +1,213 @@
-C Library Functions ST(3)
+ST(3) ST(3)
-
NAME
- libst - Sound Tools : sound sample file and effects
- libraries.
+ libst - Sound Tools : sound sample file and effects
+ libraries.
SYNOPSIS
- cc file.c -o file libst.a
+ cc file.c -o file libst.a
DESCRIPTION
- Sound Tools is a library of sound sample file format
- readers/writers and sound effects processors.
+ Sound Tools is a library of sound sample file format read-
+ ers/writers and sound effects processors.
- Sound Tools includes skeleton C files to assist you in writ-
- ing new formats and effects. The full skeleton driver,
- skel.c, helps you write drivers for a new format which has
- data structures. The simple skeleton drivers help you write
- a new driver for raw (headerless) formats, or for formats
- which just have a simple header followed by raw data.
+ Sound Tools includes skeleton C files to assist you in
+ writing new formats and effects. The full skeleton
+ driver, skel.c, helps you write drivers for a new format
+ which has data structures. The simple skeleton drivers
+ help you write a new driver for raw (headerless) formats,
+ or for formats which just have a simple header followed by
+ raw data.
- Most sound sample formats are fairly simple: they are just a
- string of bytes or words and are presumed to be sampled at a
- known data rate. Most of them have a short data structure
- at the beginning of the file.
+ Most sound sample formats are fairly simple: they are just
+ a string of bytes or words and are presumed to be sampled
+ at a known data rate. Most of them have a short data
+ structure at the beginning of the file.
INTERNALS
- The Sound Tools formats and effects operate on an internal
- buffer format of signed 32-bit longs. The data processing
- routines are called with buffers of these samples, and
- buffer sizes which refer to the number of samples processed,
- not the number of bytes. File readers translate the input
- samples to signed longs and return the number of longs read.
- For example, data in linear signed byte format is left-
- shifted 24 bits.
+ The Sound Tools formats and effects operate on an internal
+ buffer format of signed 32-bit longs. The data processing
+ routines are called with buffers of these samples, and
+ buffer sizes which refer to the number of samples pro-
+ cessed, not the number of bytes. File readers translate
+ the input samples to signed longs and return the number of
+ longs read. For example, data in linear signed byte for-
+ mat is left-shifted 24 bits.
- This does cause problems in processing the data. For exam-
- ple:
- *obuf++ = (*ibuf++ + *ibuf++)/2;
- would not mix down left and right channels into one mono-
- phonic channel, because the resulting samples would overflow
- 32 bits. Instead, the ``avg'' effects must use:
- *obuf++ = *ibuf++/2 + *ibuf++/2;
+ This does cause problems in processing the data. For
+ example:
+ *obuf++ = (*ibuf++ + *ibuf++)/2;
+ would not mix down left and right channels into one mono-
+ phonic channel, because the resulting samples would over-
+ flow 32 bits. Instead, the ``avg'' effects must use:
+ *obuf++ = *ibuf++/2 + *ibuf++/2;
- Stereo data is stored with the left and right speaker data
- in successive samples. Quadraphonic data is stored in this
- order: left front, right front, left rear, right rear.
+ Stereo data is stored with the left and right speaker data
+ in successive samples. Quadraphonic data is stored in
+ this order: left front, right front, left rear, right
+ rear.
FORMATS
- A format is responsible for translating between sound sample
- files and an internal buffer. The internal buffer is store
- in signed longs with a fixed sampling rate. The format
- operates from two data structures: a format structure, and
- a private structure.
+ A format is responsible for translating between sound sam-
+ ple files and an internal buffer. The internal buffer is
+ store in signed longs with a fixed sampling rate. The
+ format operates from two data structures: a format struc-
+ ture, and a private structure.
-SunOS 5.6 Last change: October 15 1996 1
+ October 15 1996 1
+ST(3) ST(3)
-C Library Functions ST(3)
+ The format structure contains a list of control parameters
+ for the sample: sampling rate, data size (bytes, words,
+ floats, etc.), style (unsigned, signed, logarithmic), num-
+ ber of sound channels. It also contains other state
+ information: whether the sample file needs to be byte-
+ swapped, whether fseek() will work, its suffix, its file
+ stream pointer, its format pointer, and the private struc-
+ ture for the format .
+ The private area is just a preallocated data array for the
+ format to use however it wishes. It should have a defined
+ data structure and cast the array to that structure. See
+ voc.c for the use of a private data area. Voc.c has to
+ track the number of samples it writes and when finishing,
+ seek back to the beginning of the file and write it out.
+ The private area is not very large. The ``echo'' effect
+ has to malloc() a much larger area for its delay line
+ buffers.
- The format structure contains a list of control parameters
- for the sample: sampling rate, data size (bytes, words,
- floats, etc.), style (unsigned, signed, logarithmic), number
- of sound channels. It also contains other state informa-
- tion: whether the sample file needs to be byte-swapped,
- whether fseek() will work, its suffix, its file stream
- pointer, its format pointer, and the private structure for
- the format .
+ A format has 6 routines:
- The private area is just a preallocated data array for the
- format to use however it wishes. It should have a defined
- data structure and cast the array to that structure. See
- voc.c for the use of a private data area. Voc.c has to track
- the number of samples it writes and when finishing, seek
- back to the beginning of the file and write it out. The
- private area is not very large. The ``echo'' effect has to
- malloc() a much larger area for its delay line buffers.
+ startread Set up the format parameters, or read
+ in a data header, or do what needs to
+ be done.
- A format has 6 routines:
+ read Given a buffer and a length: read up
+ to that many samples, transform them
+ into signed long integers, and copy
+ them into the buffer. Return the num-
+ ber of samples actually read.
- startread Set up the format parameters, or read in
- a data header, or do what needs to be
- done.
+ stopread Do what needs to be done.
- read Given a buffer and a length: read up to
- that many samples, transform them into
- signed long integers, and copy them into
- the buffer. Return the number of sam-
- ples actually read.
+ startwrite Set up the format parameters, or write
+ out a data header, or do what needs to
+ be done.
- stopread Do what needs to be done.
+ write Given a buffer and a length: copy that
+ many samples out of the buffer, con-
+ vert them from signed longs to the
+ appropriate data, and write them to
+ the file. If it can't write out all
+ the samples, fail.
- startwrite Set up the format parameters, or write
- out a data header, or do what needs to
- be done.
+ stopwrite Fix up any file header, or do what
+ needs to be done.
- write Given a buffer and a length: copy that
- many samples out of the buffer, convert
- them from signed longs to the appropri-
- ate data, and write them to the file.
- If it can't write out all the samples,
- fail.
-
- stopwrite Fix up any file header, or do what needs
- to be done.
-
EFFECTS
- An effects loop has one input and one output stream. It has
- 5 routines.
+ An effects loop has one input and one output stream. It
+ has 5 routines.
- getopts is called with a character string argu-
- ment list for the effect.
+ getopts is called with a character string
+ argument list for the effect.
-SunOS 5.6 Last change: October 15 1996 2
+ October 15 1996 2
-C Library Functions ST(3)
+ST(3) ST(3)
+ start is called with the signal parameters
+ for the input and output streams.
- start is called with the signal parameters for
- the input and output streams.
+ flow is called with input and output data
+ buffers, and (by reference) the input
+ and output data sizes. It processes
+ the input buffer into the output
+ buffer, and sets the size variables to
+ the numbers of samples actually pro-
+ cessed. It is under no obligation to
+ fill the output buffer.
- flow is called with input and output data
- buffers, and (by reference) the input
- and output data sizes. It processes the
- input buffer into the output buffer, and
- sets the size variables to the numbers
- of samples actually processed. It is
- under no obligation to fill the output
- buffer.
+ drain is called after there are no more
+ input data samples. If the effect
+ wishes to generate more data samples
+ it copies the generated data into a
+ given buffer and returns the number of
+ samples generated. If it fills the
+ buffer, it will be called again, etc.
+ The echo effect uses this to fade
+ away.
- drain is called after there are no more input
- data samples. If the effect wishes to
- generate more data samples it copies the
- generated data into a given buffer and
- returns the number of samples generated.
- If it fills the buffer, it will be
- called again, etc. The echo effect uses
- this to fade away.
+ stop is called when there are no more input
+ samples to process. stop may generate
+ output samples on its own. See echo.c
+ for how to do this, and see that what
+ it does is absolutely bogus.
- stop is called when there are no more input
- samples to process. stop may generate
- output samples on its own. See echo.c
- for how to do this, and see that what it
- does is absolutely bogus.
-
COMMENTS
- Theoretically, formats can be used to manipulate several
- files inside one program. Multi-sample files, for example
- the download for a sampling keyboard, can be handled cleanly
- with this feature.
+ Theoretically, formats can be used to manipulate several
+ files inside one program. Multi-sample files, for example
+ the download for a sampling keyboard, can be handled
+ cleanly with this feature.
PORTABILITY PROBLEMS
- Many computers don't supply arithmetic shifting, so do mul-
- tiplies and divides instead of << and >>. The compiler will
- do the right thing if the CPU supplies arithmetic shifting.
+ Many computers don't supply arithmetic shifting, so do
+ multiplies and divides instead of << and >>. The compiler
+ will do the right thing if the CPU supplies arithmetic
+ shifting.
- Do all arithmetic conversions one stage at a time. I've had
- too many problems with "obviously clean" combinations.
+ Do all arithmetic conversions one stage at a time. I've
+ had too many problems with "obviously clean" combinations.
- In general, don't worry about "efficiency". The sox.c base
- translator is disk-bound on any machine (other than a 8088
- PC with an SMD disk controller). Just comment your code and
- make sure it's clean and simple. You'll find that DSP code
- is extremely painful to write as it is.
+ In general, don't worry about "efficiency". The sox.c
+ base translator is disk-bound on any machine (other than a
+ 8088 PC with an SMD disk controller). Just comment your
+ code and make sure it's clean and simple. You'll find
+ that DSP code is extremely painful to write as it is.
BUGS
- The HCOM format is not re-entrant; it can only be used once
- in a program.
+ The HCOM format is not re-entrant; it can only be used
+ once in a program.
+ The program/library interface is pretty weak. There's too
+ October 15 1996 3
-SunOS 5.6 Last change: October 15 1996 3
+ST(3) ST(3)
-C Library Functions ST(3)
+ much ad-hoc information which a program is supposed to
+ gather up. Sound Tools wants to be an object-oriented
+ dataflow architecture.
- The program/library interface is pretty weak. There's too
- much ad-hoc information which a program is supposed to
- gather up. Sound Tools wants to be an object-oriented
- dataflow architecture.
@@ -258,7 +259,6 @@
-SunOS 5.6 Last change: October 15 1996 4
-
+ October 15 1996 4
--- a/sox.1
+++ b/sox.1
@@ -9,7 +9,7 @@
.if t .sp .5v
.if n .sp
..
-.TH SoX 1 "November 22, 1999"
+.TH SoX 1 "December 10, 1999"
.SH NAME
sox \- Sound eXchange : universal sound sample translator
.SH SYNOPSIS
@@ -24,9 +24,9 @@
.B sox
[\fI general options \fB ]
[ \fIformat options \fB ]
-\fIifile\fB
+\fIinfile\fB
[ \fIformat options \fB ]
-\fIofile\fB
+\fIoutfile\fB
[ \fIeffect\fR [ \fIeffect options ...\fB ] ]
.P
\fIGeneral options:\fB
@@ -105,18 +105,53 @@
.br
vibro \fIspeed \fB[ \fIdepth\fB ]
.SH DESCRIPTION
-.I Sox
-translates sound files from one format to another,
-possibly doing a sound effect.
+.I SoX
+is a command line program that can convert most popular audio files
+to most other popular audio file formats. It can optionally apply a
+sound effect to the file during this translation.
+.P
+There are two types of audio files formats that
+.I SoX
+can work with. The first are self-describing file formats. These
+contain a header that completely describe the characteristics of
+the audio data that follows.
+.P
+The second type are headerless data, or sometimes called raw data. A
+user must pass enough information to
+.I SoX
+on the command line so that it knows what type of data it contains.
+.P
+Audio data can usually be totally described by four characteristics:
+.TP 10
+rate
+The sample rate is in samples per second. For example, CD sample rates are at 44100.
+.TP 10
+data type
+What format the data is stored in. Most popular are 8-bit or 16-bit words.
+.TP 10
+data format
+What encoding the data type uses. Examples are u-law, ADPCM, or signed linear data.
+.TP 10
+channels
+How many channels are contained in the audio data. Mono and Stereo are the two most common.
+.P
+Please refer to the
+.B soxexam(1)
+manual page for a long description with examples on how to use sox with
+various types of file formats.
.SH OPTIONS
The option syntax is a little grotty, but in essence:
+.P
.br
sox file.au file.voc
+.P
.br
-translates a sound sample in SUN Sparc .AU format
+translates a sound file in SUN Sparc .AU format
into a SoundBlaster .VOC file, while
+.P
.br
sox -v 0.5 file.au -r 12000 file.voc rate
+.P
.br
does the same format translation but also
lowers the amplitude by 1/2 and changes
@@ -127,12 +162,12 @@
.PP
Format options:
.PP
-Format options effect the file that they immediately percede. If
+Format options effect the audio samples that they immediately percede. If
they are placed before the input file name then they effect the input
data. If they are placed before the output file name then they will
-effect the output data. It is also possible to read a given file in
-and output it in any supported data format by specifying output format
-options.
+effect the output data. By taking advantage of this, you can override
+a input file's currupted header or produce an output file that is totally
+different style then the input file.
.TP 10
\fB-t\fI filetype
gives the type of the sound sample file.
@@ -147,7 +182,7 @@
used with its default parameters.
.TP 10
\fB-s/-u/-U/-A/-a/-i/-g\fR
-The sample data is signed linear (2's complement),
+The sample data format is signed linear (2's complement),
unsigned linear, U-law (logarithmic), A-law (logarithmic),
ADPCM, IMA_ADPCM, or GSM.
U-law and A-law are the U.S. and international
@@ -163,7 +198,7 @@
quality.
.TP 10
\fB-b/-w/-l/-f/-d/-D\fR
-The sample data is in bytes, 16-bit words, 32-bit longwords,
+The sample data type is in bytes, 16-bit words, 32-bit longwords,
32-bit floats, 64-bit double floats, or 80-bit IEEE floats.
Floats and double floats are in native machine format.
.TP 10
@@ -189,9 +224,10 @@
General options:
.TP 10
\fB-e\fR
-after the input file allows you to avoid giving
-an output file and just name an effect.
-This is mainly useful with the
+When used after the input file (so that it applies to the output file)
+it allows you to avoid giving an output filename and will not
+produce an output file. It will apply any specified effects
+to the input file. This is mainly useful with the
.B stat
effect but can be used with others.
.TP 10
@@ -217,48 +253,20 @@
Useful for figuring out exactly how
.I sox
is mangling your sound samples.
-.PP
-The input and output files may be standard input and output.
-This is specified by '-'.
-The
-.B -t\ \fItype
-option must be given in this case,
-else
-.I sox
-will not know the format of the given file.
-The
-.B -t,
-.B -r,
-.B -s/-u/-U/-A,
-.B -b/-w/-l/-f/-d/-D
-and
-.B -x
-options refer to the input data when given before the
-input file name. After, they refer to the output data.
-.PP
-If you don't give an output file name,
-.I sox
-will just read the input file.
-This is useful for validating structured file formats;
-the
-.B stat
-effect may also be used
-via the
-.B -e
-option.
.SH FILE TYPES
-.I Sox
-needs to know the formats of the input and output files.
+.I SoX
+uses the file extension of the input and output file to determine what
+type of file format to use. This can be overriden by specifying the
+"-t" option on the command line.
+.P
+The input and output files may be read from standard in and out. This
+is done by specifing '-' as the filename.
+.P
File formats which have headers are checked,
if that header doesn't seem right,
the program exits with an appropriate message.
-Currently, raw (no header) binary and textual data,
-Amiga 8SVX, Apple/SGI AIFF, SPARC .AU (w/header), AVR, NeXT .SND,
-CD-R, CVSD, GSM 06.10, Mac HCOM, Sound Tools MAUD, OSS device drivers,
-Turtle Beach .SMP, Sound Blaster, Sndtool, and Sounder,
-Sun Audio device driver,
-Yamaha TX-16W Sampler, IRCAM Sound Files, Creative Labs VOC,
-Psion .WVE, and Microsoft RIFF/WAV are supported.
+.P
+The following file formats are supported:
.PP
.TP 10
.B .8svx
@@ -440,10 +448,7 @@
override the file header, and you will be warned to this effect.
You had better know what you are doing! Output format
options will cause a format conversion, and the \fB.wav\fR
-will written appropriately. Note that it is possible to
-write data of a type that cannot be specified by
-the \fB.wav\fR header, and you will be warned that
-you are writing a bad file !
+will written appropriately.
Sox currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
It can write all of these formats including
.B (NEW!)
@@ -530,6 +535,12 @@
.I center
frequency and settling around it.
See \fBfilter\fR for a bandpass effect with steeper shoulders.
+.TP 10
+bandpass
+Butterworth bandpass filter. Description coming soon!
+.TP 10
+bandreject
+Butterworth bandreject filter. Description coming soon!
.TP
chorus \fIgain-in gain-out delay decay speed depth
.TP 10
@@ -628,6 +639,9 @@
The slope of the filter is quite gentle.
See \fBfilter\fR for a highpass effect with sharper cutoff.
.TP 10
+highpass
+Butterworth highpass filter. Description comming soon!
+.TP 10
lowp \fIcenter
Apply a low-pass filter.
The frequency response drops logarithmically with
@@ -636,6 +650,9 @@
The slope of the filter is quite gentle.
See \fBfilter\fR for a lowpass effect with sharper cutoff.
.TP 10
+lowpass
+Butterworth lowpass filter. Description coming soon!
+.TP 10
map
Display a list of loops in a sample,
and miscellaneous loop info.
@@ -711,7 +728,7 @@
or
.B polyphase.
If you are wondering which of
-.B Sox's
+.B SoX's
rate changing effects to use, you will want to read a
detailed analysis of all of them at http://eakaw2.et.tu-dresden.de/~andreas/resample/resample.html
[Nov,1999: These tests need to be updated for sox-12.17, which has bugfixes to the
@@ -864,31 +881,16 @@
." or other channel mixing
effect must be requested.
.SH BUGS
-The syntax is horrific.
-It's very tempting to include a default system that allows
-an effect name as the program name
-and just pipes a sound sample from standard input
-to standard output, but the problem of inputting the
-sample rates makes this unworkable.
+The syntax is horrific. Thats the breaks when trying to handle all things from the command line.
.P
Please report any bugs found in this version of sox to Chris Bagwell (cbagwell@sprynet.com)
.SH FILES
.SH SEE ALSO
-.BR play (1) ,
-.BR rec (1)
+.BR play (1),
+.BR rec (1),
+.BR soxexam(1)
.SH NOTICES
-The echoplex effect is:
-Copyright (C) 1989 by Jef Poskanzer.
-
-Permission to use, copy, modify, and distribute this software and its
-documentation for any purpose and without fee is hereby granted, provided
-that the above copyright notice appear in all copies and that both that
-copyright notice and this permission notice appear in supporting
-documentation. This software is provided "as is" without express or
-implied warranty.
-
The version of Sox that accompanies this manual page is support by
Chris Bagwell (cbagwell@sprynet.com). Please refer any questions
regarding it to this address. You may obtain the latest version at the
the web site http://home.sprynet.com/~cbagwell/sox.html
-
--- a/sox.txt
+++ b/sox.txt
@@ -1,912 +1,971 @@
-User Commands SoX(1)
+SoX(1) SoX(1)
-
NAME
- sox - Sound eXchange : universal sound sample translator
+ sox - Sound eXchange : universal sound sample translator
SYNOPSIS
- sox infile outfile
- sox infile outfile [ effect [ effect options ... ] ]
- sox infile -e effect [ effect options ... ]
- sox [ general options ] [ format options ] ifile [ format
- options ] ofile [ effect [ effect options ... ] ]
+ sox infile outfile
+ sox infile outfile [ effect [ effect options ... ] ]
+ sox infile -e effect [ effect options ... ]
+ sox [ general options ] [ format options ] infile [ for-
+ mat options ] outfile [ effect [ effect options ... ] ]
- General options: [ -e ] [ -h ] [ -p ] [ -v volume ] [ -V ]
+ General options: [ -e ] [ -h ] [ -p ] [ -v volume ] [ -V ]
- Format options: [ -t filetype ] [ -r rate ] [ -s/-u/-U/-
- A/-a/-g ] [ -b/-w/-l/-f/-d/-D ] [ -c channels ] [ -x ]
+ Format options: [ -t filetype ] [ -r rate ] [
+ -s/-u/-U/-A/-a/-i/-g ] [ -b/-w/-l/-f/-d/-D ] [ -c channels
+ ] [ -x ]
- Effects:
- avg [ -l | -r ]
- band [ -n ] center [ width ]
- check
- chorus gain-in gain out delay decay speed depth
- -s | -t [ delay decay speed depth -s | -fI-t ]
- compand attack1,decay1[,attack2,decay2...]
- in-dB1,out-dB1[,in-dB2,out-dB2...]
- [gain] [initial-volume]
- copy
- cut
- deemph
- echo gain-in gain-out delay decay [ delay decay ...]
- echos gain-in gain-out delay decay [ delay decay ...]
- filter [ low ]-[ high ] [ window-len [ beta ]]
- flanger gain-in gain-out delay decay speed -s | -fI-t
- highp center
- lowp center
- map
- mask
- phaser gain-in gain-out delay decay speed -s | -t
- pick
- polyphase [ -w < num / ham > ]
- [ -width < long / short / # > ]
- [ -cutoff # ]
- rate
- resample
- reverb gain-out reverb-time delay [ delay ... ]
- reverse
- split
- stat [ debug | -v ]
- swap [ 1 2 3 4 ]
- vibro speed [ depth ]
+ Effects:
+ avg [ -l | -r ]
+ band [ -n ] center [ width ]
+ check
+ chorus gain-in gain out delay decay speed depth
+ -s | -t [ delay decay speed depth -s | -fI-t ]
+ compand attack1,decay1[,attack2,decay2...]
+ in-dB1,out-dB1[,in-dB2,out-dB2...]
+ [gain] [initial-volume]
+ copy
+ cut
+ deemph
+ echo gain-in gain-out delay decay [ delay decay ...]
+ echos gain-in gain-out delay decay [ delay decay ...]
+ filter [ low ]-[ high ] [ window-len [ beta ]]
+ flanger gain-in gain-out delay decay speed -s | -fI-t
+ highp center
+ lowp center
+ map
+ mask
+ phaser gain-in gain-out delay decay speed -s | -t
+ pick
+ polyphase [ -w < nut / ham > ]
+ [ -width < long / short / # > ]
+ [ -cutoff # ]
+ rate
+ resample
+ reverb gain-out reverb-time delay [ delay ... ]
+ reverse
+ split
+ stat [ debug | -v ]
+ swap [ 1 2 3 4 ]
+ vibro speed [ depth ]
DESCRIPTION
- Sox translates sound files from one format to another, pos-
- sibly doing a sound effect.
+ SoX is a command line program that can convert most popu-
+ lar audio files to most other popular audio file formats.
+ It can optionally apply a sound effect to the file during
-SunOS 5.6 Last change: November 8, 1999 1
+ December 10, 1999 1
+SoX(1) SoX(1)
-User Commands SoX(1)
+ this translation.
+ There are two types of audio files formats that SoX can
+ work with. The first are self-describing file formats.
+ These contain a header that completely describe the char-
+ acteristics of the audio data that follows.
-OPTIONS
- The option syntax is a little grotty, but in essence:
- sox file.au file.voc
- translates a sound sample in SUN Sparc .AU format into a
- SoundBlaster .VOC file, while
- sox -v 0.5 file.au -r 12000 file.voc rate
- does the same format translation but also lowers the ampli-
- tude by 1/2 and changes the sampling rate from 8000 hertz to
- 12000 hertz via the rate sound effect loop.
+ The second type are headerless data, or sometimes called
+ raw data. A user must pass enough information to SoX on
+ the command line so that it knows what type of data it
+ contains.
- Format options:
+ Audio data can usually be totally described by four char-
+ acteristics:
- Format options effect the file that they immediately per-
- cede. If they are placed before the input file name then
- they effect the input data. If they are placed before the
- output file name then they will effect the output data. It
- is also possible to read a given file in and output it in
- any supported data format by specifying output format
- options.
+ rate The sample rate is in samples per second. For
+ example, CD sample rates are at 44100.
- -t filetype
- gives the type of the sound sample file.
+ data type What format the data is stored in. Most popular
+ are 8-bit or 16-bit words.
- -r rate Give sample rate in Hertz of file. To cause the
- output file to have a different sample rate then
- the input file, include this option with the
- appropriate rate value along with the output
- options. If the input and output files have dif-
- ferent rates then a sample rate change effect must
- be ran. If a sample rate changing effect is not
- specified then a default one will be used with its
- default parameters.
+ data format
+ What encoding the data type uses. Examples are
+ u-law, ADPCM, or signed linear data.
- -s/-u/-U/-A/-a/-g
- The sample data is signed linear (2's complement),
- unsigned linear, U-law (logarithmic), A-law (loga-
- rithmic), ADPCM, or GSM. U-law and A-law are the
- U.S. and international standards for logarithmic
- telephone sound compression. ADPCM is form of
- sound compression that has a good compromise
- between good sound quality and fast
- encoding/decoding time. GSM is a standard used
- for telephone sound compression in European coun-
- tries and its gaining popularity because of its
- quality.
+ channels How many channels are contained in the audio
+ data. Mono and Stereo are the two most common.
- -b/-w/-l/-f/-d/-D
- The sample data is in bytes, 16-bit words, 32-bit
- longwords, 32-bit floats, 64-bit double floats, or
- 80-bit IEEE floats. Floats and double floats are
- in native machine format.
+ Please refer to the soxexam(1) manual page for a long
+ description with examples on how to use sox with various
+ types of file formats.
+OPTIONS
+ The option syntax is a little grotty, but in essence:
+ sox file.au file.voc
+ translates a sound file in SUN Sparc .AU format into a
+ SoundBlaster .VOC file, while
-SunOS 5.6 Last change: November 8, 1999 2
+ sox -v 0.5 file.au -r 12000 file.voc rate
+ does the same format translation but also lowers the
+ amplitude by 1/2 and changes the sampling rate from 8000
+ hertz to 12000 hertz via the rate sound effect loop.
+ Format options:
+ Format options effect the audio samples that they immedi-
+ ately percede. If they are placed before the input file
+ name then they effect the input data. If they are placed
+ before the output file name then they will effect the out-
+ put data. By taking advantage of this, you can override a
+ input file's currupted header or produce an output file
-User Commands SoX(1)
+ December 10, 1999 2
- -x The sample data is in XINU format; that is, it
- comes from a machine with the opposite word order
- than yours and must be swapped according to the
- word-size given above. Only 16-bit and 32-bit
- integer data may be swapped. Machine-format
- floating-point data is not portable. IEEE floats
- are a fixed, portable format.
- -c channels
- The number of sound channels in the data file.
- This may be 1, 2, or 4; for mono, stereo, or quad
- sound data. To cause the output file to have a
- different number of channels then the input file,
- include this option with the approraite value with
- the output file options. If the input and output
- file have a different number of channels then the
- avg effect must be used. If the avg effect is not
- specified on the command line it will be invoked
- with default parameters.
- General options:
+SoX(1) SoX(1)
- -e after the input file allows you to avoid giving an
- output file and just name an effect. This is
- mainly useful with the stat effect but can be used
- with others.
- -h Print version number and usage information.
+ that is totally different style then the input file.
- -p Run in preview mode and run fast. This will some-
- what speed up sox when the output format has a
- different number of channels and a different rate
- then the input file. The order that the effects
- are run in will be arranged for maximum speed and
- not quality.
+ -t filetype
+ gives the type of the sound sample file.
- -v volume Change amplitude (floating point); less than 1.0
- decreases, greater than 1.0 increases. Note: we
- perceive volume logarithmically, not linearly.
- Note: see the stat effect.
+ -r rate Give sample rate in Hertz of file. To cause the
+ output file to have a different sample rate than
+ the input file, include this option with the
+ appropriate rate value along with the output
+ options. If the input and output files have
+ different rates then a sample rate change effect
+ must be ran. If a sample rate changing effect
+ is not specified then a default one will be used
+ with its default parameters.
- -V Print a description of processing phases. Useful
- for figuring out exactly how sox is mangling your
- sound samples.
+ -s/-u/-U/-A/-a/-i/-g
+ The sample data format is signed linear (2's
+ complement), unsigned linear, U-law (logarith-
+ mic), A-law (logarithmic), ADPCM, IMA_ADPCM, or
+ GSM. U-law and A-law are the U.S. and interna-
+ tional standards for logarithmic telephone sound
+ compression. ADPCM is form of sound compression
+ that has a good compromise between good sound
+ quality and fast encoding/decoding time.
+ IMA_ADPCM is also a form of adpcm compression,
+ slightly simpler and slightly lower fidelity
+ than Microsoft's flavor of ADPCM. IMA_ADPCM is
+ also called DVI_ADPCM. GSM is a standard used
+ for telephone sound compression in European
+ countries and its gaining popularity because of
+ its quality.
- The input and output files may be standard input and output.
- This is specified by '-'. The -t type option must be given
- in this case, else sox will not know the format of the given
- file. The -t, -r, -s/-u/-U/-A, -b/-w/-l/-f/-d/-D and -x
- options refer to the input data when given before the input
- file name. After, they refer to the output data.
+ -b/-w/-l/-f/-d/-D
+ The sample data type is in bytes, 16-bit words,
+ 32-bit longwords, 32-bit floats, 64-bit double
+ floats, or 80-bit IEEE floats. Floats and dou-
+ ble floats are in native machine format.
+ -x The sample data is in XINU format; that is, it
+ comes from a machine with the opposite word
+ order than yours and must be swapped according
+ to the word-size given above. Only 16-bit and
+ 32-bit integer data may be swapped. Machine-
+ format floating-point data is not portable.
+ IEEE floats are a fixed, portable format.
+ -c channels
+ The number of sound channels in the data file.
+ This may be 1, 2, or 4; for mono, stereo, or
+ quad sound data. To cause the output file to
+ have a different number of channels than the
+ input file, include this option with the appro-
+ raite value with the output file options. If
+ the input and output file have a different
-SunOS 5.6 Last change: November 8, 1999 3
+ December 10, 1999 3
-User Commands SoX(1)
+SoX(1) SoX(1)
+ number of channels then the avg effect must be
+ used. If the avg effect is not specified on the
+ command line it will be invoked with default
+ parameters.
- If you don't give an output file name, sox will just read
- the input file. This is useful for validating structured
- file formats; the stat effect may also be used via the -e
- option.
+ General options:
-FILE TYPES
- Sox needs to know the formats of the input and output files.
- File formats which have headers are checked, if that header
- doesn't seem right, the program exits with an appropriate
- message. Currently, raw (no header) binary and textual
- data, Amiga 8SVX, Apple/SGI AIFF, SPARC .AU (w/header), AVR,
- NeXT .SND, CD-R, CVSD, GSM 06.10, Mac HCOM, Sound Tools
- MAUD, OSS device drivers, Turtle Beach .SMP, Sound Blaster,
- Sndtool, and Sounder, Sun Audio device driver, Yamaha TX-16W
- Sampler, IRCAM Sound Files, Creative Labs VOC, Psion .WVE,
- and Microsoft RIFF/WAV are supported.
+ -e When used after the input file (so that it
+ applies to the output file) it allows you to
+ avoid giving an output filename and will not
+ produce an output file. It will apply any spec-
+ ified effects to the input file. This is mainly
+ useful with the stat effect but can be used with
+ others.
- .8svx Amiga 8SVX musical instrument description format.
+ -h Print version number and usage information.
- .aiff AIFF files used on Apple IIc/IIgs and SGI. Note:
- the AIFF format supports only one SSND chunk. It
- does not support multiple sound chunks, or the
- 8SVX musical instrument description format. AIFF
- files are multimedia archives and and can have
- multiple audio and picture chunks. You may need a
- separate archiver to work with them.
+ -p Run in preview mode and run fast. This will
+ somewhat speed up sox when the output format has
+ a different number of channels and a different
+ rate than the input file. The order that the
+ effects are run in will be arranged for maximum
+ speed and not quality.
- .au SUN Microsystems AU files. There are apparently
- many types of .au files; DEC has invented its own
- with a different magic number and word order. The
- .au handler can read these files but will not
- write them. Some .au files have valid AU headers
- and some do not. The latter are probably original
- SUN u-law 8000 hz samples. These can be dealt
- with using the .ul format (see below).
+ -v volume Change amplitude (floating point); less than 1.0
+ decreases, greater than 1.0 increases. Note: we
+ perceive volume logarithmically, not linearly.
+ Note: see the stat effect.
- .avr Audio Visual Research
- The AVR format is produced by a number of commer-
- cial packages on the Mac.
+ -V Print a description of processing phases. Use-
+ ful for figuring out exactly how sox is mangling
+ your sound samples.
- .cdr CD-R
- CD-R files are used in mastering music Compact
- Disks. The file format is, as you might expect,
- raw stereo raw unsigned samples at 44khz. But,
- there's some blocking/padding oddity in the for-
- mat, so it needs its own handler.
+FILE TYPES
+ SoX uses the file extension of the input and output file
+ to determine what type of file format to use. This can be
+ overriden by specifying the "-t" option on the command
+ line.
- .cvs Continuously Variable Slope Delta modulation
- Used to compress speech audio for applications
- such as voice mail.
+ The input and output files may be read from standard in
+ and out. This is done by specifing '-' as the filename.
- .dat Text Data files
+ File formats which have headers are checked, if that
+ header doesn't seem right, the program exits with an
+ appropriate message.
+ The following file formats are supported:
-SunOS 5.6 Last change: November 8, 1999 4
+ .8svx Amiga 8SVX musical instrument description for-
+ mat.
+ .aiff AIFF files used on Apple IIc/IIgs and SGI.
+ Note: the AIFF format supports only one SSND
+ December 10, 1999 4
-User Commands SoX(1)
- These files contain a textual representation of
- the sample data. There is one line at the begin-
- ning that contains the sample rate. Subsequent
- lines contain two numeric data items: the time
- since the beginning of the sample and the sample
- value. Values are normalized so that the maximum
- and minimum are 1.00 and -1.00. This file format
- can be used to create data files for external pro-
- grams such as FFT analyzers or graph routines.
- SoX can also convert a file in this format back
- into one of the other file formats.
+SoX(1) SoX(1)
- .gsm GSM 06.10 Lossy Speech Compression
- A standard for compressing speech which is used in
- the Global Standard for Mobil telecommunications
- (GSM). Its good for its purpose, shrinking audio
- data size, but it will introduce lots of noise
- when a given sound sample is encoded and decoded
- multiple times. This format is used by some voice
- mail applications. It is rather CPU intensive.
- GSM in sox is optional and requires access to an
- external GSM library. To see if there is support
- for gsm run sox -h and look for it under the list
- of supported file formats.
- .hcom Macintosh HCOM files. These are (apparently) Mac
- FSSD files with some variant of Huffman compres-
- sion. The Macintosh has wacky file formats and
- this format handler apparently doesn't handle all
- the ones it should. Mac users will need your
- usual arsenal of file converters to deal with an
- HCOM file under Unix or DOS.
+ chunk. It does not support multiple sound
+ chunks, or the 8SVX musical instrument descrip-
+ tion format. AIFF files are multimedia archives
+ and and can have multiple audio and picture
+ chunks. You may need a separate archiver to
+ work with them.
- .maud An Amiga format
- An IFF-conform sound file type, registered by MS
- MacroSystem Computer GmbH, published along with
- the "Toccata" sound-card on the Amiga. Allows
- 8bit linear, 16bit linear, A-Law, u-law in mono
- and stereo.
+ .au SUN Microsystems AU files. There are apparently
+ many types of .au files; DEC has invented its
+ own with a different magic number and word
+ order. The .au handler can read these files but
+ will not write them. Some .au files have valid
+ AU headers and some do not. The latter are
+ probably original SUN u-law 8000 hz samples.
+ These can be dealt with using the .ul format
+ (see below).
- ossdsp OSS /dev/dsp device driver
- This is a psuedo-file type and can be optionally
- compiled into Sox. Run sox -h to see if you have
- support for this file type. When this driver is
- used it allows you to open up the OSS /dev/dsp
- file and configure it to use the same data type as
- passed in to Sox. It works for both playing and
- recording sound samples. When playing sound files
- it attempts to set up the OSS driver to use the
- same format as the input file. It is suggested to
- always override the output values to use the
- highest quality samples your sound card can
+ .avr Audio Visual Research
+ The AVR format is produced by a number of com-
+ mercial packages on the Mac.
+ .cdr CD-R
+ CD-R files are used in mastering music Compact
+ Disks. The file format is, as you might expect,
+ raw stereo raw unsigned samples at 44khz. But,
+ there's some blocking/padding oddity in the for-
+ mat, so it needs its own handler.
+ .cvs Continuously Variable Slope Delta modulation
+ Used to compress speech audio for applications
+ such as voice mail.
-SunOS 5.6 Last change: November 8, 1999 5
+ .dat Text Data files
+ These files contain a textual representation of
+ the sample data. There is one line at the
+ beginning that contains the sample rate. Subse-
+ quent lines contain two numeric data items: the
+ time since the beginning of the sample and the
+ sample value. Values are normalized so that the
+ maximum and minimum are 1.00 and -1.00. This
+ file format can be used to create data files for
+ external programs such as FFT analyzers or graph
+ routines. SoX can also convert a file in this
+ format back into one of the other file formats.
+ .gsm GSM 06.10 Lossy Speech Compression
+ A standard for compressing speech which is used
+ in the Global Standard for Mobil telecommunica-
+ tions (GSM). Its good for its purpose, shrink-
+ ing audio data size, but it will introduce lots
+ of noise when a given sound sample is encoded
+ and decoded multiple times. This format is used
+ by some voice mail applications. It is rather
+ CPU intensive. GSM in sox is optional and
+ December 10, 1999 5
-User Commands SoX(1)
- handle. Example: -t ossdsp -w -s /dev/dsp
+SoX(1) SoX(1)
- .sf IRCAM Sound Files.
- SoundFiles are used by academic music software
- such as the CSound package, and the MixView sound
- sample editor.
- .smp Turtle Beach SampleVision files.
- SMP files are for use with the PC-DOS package Sam-
- pleVision by Turtle Beach Softworks. This package
- is for communication to several MIDI samplers. All
- sample rates are supported by the package,
- although not all are supported by the samplers
- themselves. Currently loop points are ignored.
+ requires access to an external GSM library. To
+ see if there is support for gsm run sox -h and
+ look for it under the list of supported file
+ formats.
+
+ .hcom Macintosh HCOM files. These are (apparently)
+ Mac FSSD files with some variant of Huffman com-
+ pression. The Macintosh has wacky file formats
+ and this format handler apparently doesn't han-
+ dle all the ones it should. Mac users will need
+ your usual arsenal of file converters to deal
+ with an HCOM file under Unix or DOS.
+
+ .maud An Amiga format
+ An IFF-conform sound file type, registered by MS
+ MacroSystem Computer GmbH, published along with
+ the "Toccata" sound-card on the Amiga. Allows
+ 8bit linear, 16bit linear, A-Law, u-law in mono
+ and stereo.
+
+ ossdsp OSS /dev/dsp device driver
+ This is a pseudo-file type and can be optionally
+ compiled into Sox. Run sox -h to see if you
+ have support for this file type. When this
+ driver is used it allows you to open up the OSS
+ /dev/dsp file and configure it to use the same
+ data type as passed in to Sox. It works for
+ both playing and recording sound samples. When
+ playing sound files it attempts to set up the
+ OSS driver to use the same format as the input
+ file. It is suggested to always override the
+ output values to use the highest quality samples
+ your sound card can handle. Example: -t ossdsp
+ -w -s /dev/dsp
- sunau Sun /dev/audio device driver
- This is a psuedo-file type and can be optionally
- compiled into Sox. Run sox -h to see if you have
- support for this file type. When this driver is
- used it allows you to open up a Sun /dev/audio
- file and configure it to use the same data type as
- passed in to Sox. It works for both playing and
- recording sound samples. When playing sound files
- it attempts to set up the audio driver to use the
- same format as the input file. It is suggested to
- always override the output values to use the
- highest quality samples your hardware can handle.
- Example: -t sunau -w -s /dev/audio or -t sunau -U
- -c 1 /dev/audio for older sun equipment.
+ .sf IRCAM Sound Files.
+ SoundFiles are used by academic music software
+ such as the CSound package, and the MixView
+ sound sample editor.
- .txw Yamaha TX-16W sampler.
- A file format from a Yamaha sampling keyboard
- which wrote IBM-PC format 3.5" floppies. Handles
- reading of files which do not have the sample rate
- field set to one of the expected by looking at
- some other bytes in the attack/loop length fields,
- and defaulting to 33kHz if the sample rate is
- still unknown.
+ .smp Turtle Beach SampleVision files.
+ SMP files are for use with the PC-DOS package
+ SampleVision by Turtle Beach Softworks. This
+ package is for communication to several MIDI
+ samplers. All sample rates are supported by the
+ package, although not all are supported by the
+ samplers themselves. Currently loop points are
+ ignored.
- .vms More info to come.
- Used to compress speech audio for applications
- such as voice mail.
+ sunau Sun /dev/audio device driver
+ This is a pseudo-file type and can be optionally
+ compiled into Sox. Run sox -h to see if you
+ have support for this file type. When this
+ driver is used it allows you to open up a Sun
- .voc Sound Blaster VOC files.
- VOC files are multi-part and contain silence
- parts, looping, and different sample rates for
- different chunks. On input, the silence parts are
- filled out, loops are rejected, and sample data
- with a new sample rate is rejected. Silence with
- a different sample rate is generated appropri-
- ately. On output, silence is not detected, nor
- are impossible sample rates.
+ December 10, 1999 6
-SunOS 5.6 Last change: November 8, 1999 6
+SoX(1) SoX(1)
-User Commands SoX(1)
+ /dev/audio file and configure it to use the same
+ data type as passed in to Sox. It works for
+ both playing and recording sound samples. When
+ playing sound files it attempts to set up the
+ audio driver to use the same format as the input
+ file. It is suggested to always override the
+ output values to use the highest quality samples
+ your hardware can handle. Example: -t sunau -w
+ -s /dev/audio or -t sunau -U -c 1 /dev/audio for
+ older sun equipment.
+ .txw Yamaha TX-16W sampler.
+ A file format from a Yamaha sampling keyboard
+ which wrote IBM-PC format 3.5" floppies. Han-
+ dles reading of files which do not have the sam-
+ ple rate field set to one of the expected by
+ looking at some other bytes in the attack/loop
+ length fields, and defaulting to 33kHz if the
+ sample rate is still unknown.
+ .vms More info to come.
+ Used to compress speech audio for applications
+ such as voice mail.
- .wav Microsoft .WAV RIFF files.
- These appear to be very similar to IFF files, but
- not the same. They are the native sound file for-
- mat of Windows. (Obviously, Windows was of such
- incredible importance to the computer industry
- that it just had to have its own sound file for-
- mat.) Normally .wav files have all formatting
- information in their headers, and so do not need
- any format options specified for an input file. If
- any are, they will override the file header, and
- you will be warned to this effect. You had better
- know what you are doing! Output format options
- will cause a format conversion, and the .wav will
- written appropriately. Note that it is possible
- to write data of a type that cannot be specified
- by the .wav header, and you will be warned that
- you a writing a bad file ! Sox currently can read
- PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
- It can output all of these formats except the
- ADPCM styles.
+ .voc Sound Blaster VOC files.
+ VOC files are multi-part and contain silence
+ parts, looping, and different sample rates for
+ different chunks. On input, the silence parts
+ are filled out, loops are rejected, and sample
+ data with a new sample rate is rejected.
+ Silence with a different sample rate is gener-
+ ated appropriately. On output, silence is not
+ detected, nor are impossible sample rates.
- .wve Psion 8-bit alaw
- These are 8-bit a-law 8khz sound files used on the
- Psion palmtop portable computer.
+ .wav Microsoft .WAV RIFF files.
+ These appear to be very similar to IFF files,
+ but not the same. They are the native sound
+ file format of Windows. (Obviously, Windows was
+ of such incredible importance to the computer
+ industry that it just had to have its own sound
+ file format.) Normally .wav files have all for-
+ matting information in their headers, and so do
+ not need any format options specified for an
+ input file. If any are, they will override the
+ file header, and you will be warned to this
+ effect. You had better know what you are doing!
+ Output format options will cause a format con-
+ version, and the .wav will written appropri-
+ ately. Sox currently can read PCM, ULAW, ALAW,
+ MS ADPCM, and IMA (or DVI) ADPCM. It can write
+ all of these formats including (NEW!) the ADPCM
+ styles.
- .raw Raw files (no header).
- The sample rate, size (byte, word, etc), and style
- (signed, unsigned, etc.) of the sample file must
- be given. The number of channels defaults to 1.
+ .wve Psion 8-bit alaw
- .ub, .sb, .uw, .sw, .ul
- These are several suffices which serve as a short-
- hand for raw files with a given size and style.
- Thus, ub, sb, uw, sw, and ul correspond to
- "unsigned byte", "signed byte", "unsigned word",
- "signed word", and "ulaw" (byte). The sample rate
- defaults to 8000 hz if not explicitly set, and the
- number of channels (as always) defaults to 1.
- There are lots of Sparc samples floating around in
- u-law format with no header and fixed at a sample
- rate of 8000 hz. (Certain sound management
- software cheerfully ignores the headers.) Simi-
- larly, most Mac sound files are in unsigned byte
- format with a sample rate of 11025 or 22050 hz.
- .auto This is a ``meta-type'': specifying this type for
- an input file triggers some code that tries to
- guess the real type by looking for magic words in
- the header. If the type can't be guessed, the
- program exits with an error message. The input
- must be a plain file, not a pipe. This type can't
- be used for output files.
+ December 10, 1999 7
-SunOS 5.6 Last change: November 8, 1999 7
+SoX(1) SoX(1)
+ These are 8-bit a-law 8khz sound files used on
+ the Psion palmtop portable computer.
-User Commands SoX(1)
+ .raw Raw files (no header).
+ The sample rate, size (byte, word, etc), and
+ style (signed, unsigned, etc.) of the sample
+ file must be given. The number of channels
+ defaults to 1.
+ .ub, .sb, .uw, .sw, .ul, .sl
+ These are several suffices which serve as a
+ shorthand for raw files with a given size and
+ style. Thus, ub, sb, uw, sw, ul and sl corre-
+ spond to "unsigned byte", "signed byte",
+ "unsigned word", "signed word", "ulaw" (byte),
+ and "signed long". The sample rate defaults to
+ 8000 hz if not explicitly set, and the number of
+ channels (as always) defaults to 1. There are
+ lots of Sparc samples floating around in u-law
+ format with no header and fixed at a sample rate
+ of 8000 hz. (Certain sound management software
+ cheerfully ignores the headers.) Similarly,
+ most Mac sound files are in unsigned byte format
+ with a sample rate of 11025 or 22050 hz.
+ .auto This is a ``meta-type'': specifying this type
+ for an input file triggers some code that tries
+ to guess the real type by looking for magic
+ words in the header. If the type can't be
+ guessed, the program exits with an error mes-
+ sage. The input must be a plain file, not a
+ pipe. This type can't be used for output files.
EFFECTS
- Only one effect from the palette may be applied to a sound
- sample. To do multiple effects you'll need to run sox in a
- pipeline.
+ Only one effect from the palette may be applied to a sound
+ sample. To do multiple effects you'll need to run sox in
+ a pipeline.
- avg [ -l | -r ]
- Reduce the number of channels by averaging the
- samples, or duplicate channels to increase the
- number of channels. This effect is automatically
- used when the number of input samples differ then
- the number of output channels. When reducing the
- number of channels it is possible to manually
- specify the avg effect and use the -l and -r
- options to select only the left or right channel
- for the output instead of averaging the two chan-
- nels.
+ avg [ -l | -r ]
+ Reduce the number of channels by averaging the
+ samples, or duplicate channels to increase the
+ number of channels. This effect is automati-
+ cally used when the number of input samples dif-
+ fer from the number of output channels. When
+ reducing the number of channels it is possible
+ to manually specify the avg effect and use the
+ -l and -r options to select only the left or
+ right channel for the output instead of averag-
+ ing the two channels.
- band [ -n ] center [ width ]
- Apply a band-pass filter. The frequency response
- drops logarithmically around the center frequency.
- The width gives the slope of the drop. The fre-
- quencies at center + width and center - width will
- be half of their original amplitudes. Band
- defaults to a mode oriented to pitched signals,
- i.e. voice, singing, or instrumental music. The
- -n (for noise) option uses the alternate mode for
- un-pitched signals. Warning: -n introduces a
- power-gain of about 11dB in the filter, so beware
- of output clipping. Band introduces noise in the
- shape of the filter, i.e. peaking at the center
- frequency and settling around it. See filter for
- a bandpass effect with steeper shoulders.
+ band [ -n ] center [ width ]
+ Apply a band-pass filter. The frequency
+ response drops logarithmically around the center
+ frequency. The width gives the slope of the
- chorus gain-in gain-out delay decay speed deptch
- -s | -t [ delay decay speed depth -s | -t ... ]
- Add a chorus to a sound sample. Each quadtuple
- delay/decay/speed/depth gives the delay in mil-
- liseconds and the decay (relative to gain-in) with
- a modulation speed in Hz using depth in mil-
- liseconds. The modulation is either sinodial (-s)
- or triangular (-t). Gain-out is the volume of the
- output.
- compand attack1,decay1[,attack2,decay2...]
+ December 10, 1999 8
- in-dB1,out-dB1[,in-dB2,out-dB2...]
- [gain] [initial-volume]
- Compand (compress or expand) the dynamic range of
- a sample. The attack and decay time specify the
- integration time over which the absolute value of
-SunOS 5.6 Last change: November 8, 1999 8
+SoX(1) SoX(1)
+ drop. The frequencies at center + width and
+ center - width will be half of their original
+ amplitudes. Band defaults to a mode oriented to
+ pitched signals, i.e. voice, singing, or instru-
+ mental music. The -n (for noise) option uses
+ the alternate mode for un-pitched signals.
+ Warning: -n introduces a power-gain of about
+ 11dB in the filter, so beware of output clip-
+ ping. Band introduces noise in the shape of the
+ filter, i.e. peaking at the center frequency and
+ settling around it. See filter for a bandpass
+ effect with steeper shoulders.
+ bandpass Butterworth bandpass filter. Description coming
+ soon!
+ bandreject
+ Butterworth bandreject filter. Description com-
+ ing soon!
+ chorus gain-in gain-out delay decay speed depth
-User Commands SoX(1)
+ -s | -t [ delay decay speed depth -s | -t ... ]
+ Add a chorus to a sound sample. Each quadtuple
+ delay/decay/speed/depth gives the delay in mil-
+ liseconds and the decay (relative to gain-in)
+ with a modulation speed in Hz using depth in
+ milliseconds. The modulation is either sinodial
+ (-s) or triangular (-t). Gain-out is the volume
+ of the output.
+ compand attack1,decay1[,attack2,decay2...]
+ in-dB1,out-dB1[,in-dB2,out-dB2...]
- the input signal is integrated to determine its
- volume. Where more than one pair of attack/decay
- parameters are specified, each channel is treated
- separately and the number of pairs must agree with
- the number of input channels. The second parame-
- ter is a list of points on the compander's
- transfer function specified in dB relative to the
- maximum possible signal amplitude. The input
- values must be in a strictly increasing order but
- the transfer function does not have to be monoton-
- ically rising. The special value -inf may be used
- to indicate that the input volume should be asso-
- ciated output volume. The points -inf,-inf and
- 0,0 are assumed; the latter may be overridden, but
- the former may not. The third (optional) parame-
- ter is a postprocessing gain in dB which is
- applied after the compression has taken place; the
- fourth (optional) parameter is an initial volume
- to be assumed for each channel when the effect
- starts. This permits the user to supply a nominal
- level initially, so that, for example, a very
- large gain is not applied to initial signal levels
- before the companding action has begun to operate:
- it is quite probable that in such an event, the
- output would be severely clipped while the com-
- pander gain properly adjusts itself.
+ [gain] [initial-volume]
+ Compand (compress or expand) the dynamic range
+ of a sample. The attack and decay time specify
+ the integration time over which the absolute
+ value of the input signal is integrated to
+ determine its volume. Where more than one pair
+ of attack/decay parameters are specified, each
+ channel is treated separately and the number of
+ pairs must agree with the number of input chan-
+ nels. The second parameter is a list of points
+ on the compander's transfer function specified
+ in dB relative to the maximum possible signal
+ amplitude. The input values must be in a
+ strictly increasing order but the transfer func-
+ tion does not have to be monotonically rising.
+ The special value -inf may be used to indicate
+ that the input volume should be associated out-
+ put volume. The points -inf,-inf and 0,0 are
+ assumed; the latter may be overridden, but the
- copy Copy the input file to the output file. This is
- the default effect if both files have the same
- sampling rate.
- cut loopnumber
- Extract loop #N from a sample.
- deemph Apply a treble attenuation shelving filter to sam-
- ples in audio cd format. The frequency response
- of pre-emphasized recordings is rectified. The
- filtering is defined in the standard document ISO
- 908.
+ December 10, 1999 9
- echo gain-in gain-out delay decay [ delay decay ... ]
- Add echoing to a sound sample. Each delay/decay
- part gives the delay in milliseconds and the decay
- (relative to gain-in) of that echo. Gain-out is
- the volume of the output.
- echos gain-in gain-out delay decay [ delay decay ... ]
- Add a sequence of echos to a sound sample. Each
- delay/decay part gives the delay in milliseconds
- and the decay (relative to gain-in) of that echo.
- Gain-out is the volume of the output.
+SoX(1) SoX(1)
-SunOS 5.6 Last change: November 8, 1999 9
+ former may not. The third (optional) parameter
+ is a postprocessing gain in dB which is applied
+ after the compression has taken place; the
+ fourth (optional) parameter is an initial volume
+ to be assumed for each channel when the effect
+ starts. This permits the user to supply a nomi-
+ nal level initially, so that, for example, a
+ very large gain is not applied to initial signal
+ levels before the companding action has begun to
+ operate: it is quite probable that in such an
+ event, the output would be severely clipped
+ while the compander gain properly adjusts
+ itself.
+ copy Copy the input file to the output file. This is
+ the default effect if both files have the same
+ sampling rate.
+ cut loopnumber
+ Extract loop #N from a sample.
+ deemph Apply a treble attenuation shelving filter to
+ samples in audio cd format. The frequency
+ response of pre-emphasized recordings is recti-
+ fied. The filtering is defined in the standard
+ document ISO 908.
+ echo gain-in gain-out delay decay [ delay decay ... ]
+ Add echoing to a sound sample. Each delay/decay
+ part gives the delay in milliseconds and the
+ decay (relative to gain-in) of that echo. Gain-
+ out is the volume of the output.
-User Commands SoX(1)
+ echos gain-in gain-out delay decay [ delay decay ... ]
+ Add a sequence of echos to a sound sample. Each
+ delay/decay part gives the delay in milliseconds
+ and the decay (relative to gain-in) of that
+ echo. Gain-out is the volume of the output.
+ filter [ low ]-[ high ] [ window-len [ beta ] ]
+ Apply a Sinc-windowed lowpass, highpass, or
+ bandpass filter of given window length to the
+ signal. low refers to the frequency of the
+ lower 6dB corner of the filter. high refers to
+ the frequency of the upper 6dB corner of the
+ filter.
+ A lowpass filter is obtained by leaving low
+ unspecified, or 0. A highpass filter is
+ obtained by leaving high unspecified, or 0, or
+ greater than or equal to the Nyquist frequency.
- filter [ low ]-[ high ] [ window-len [ beta ] ]
- Apply a Sinc-windowed lowpass, highpass, or
- bandpass filter of given window length to the sig-
- nal. low refers to the frequency of the lower 6dB
- corner of the filter. high refers to the fre-
- quency of the upper 6dB corner of the filter.
+ The window-len, if unspecified, defaults to 128.
+ Longer windows give a sharper cutoff, smaller
- A lowpass filter is obtained by leaving low
- unspecified, or 0. A highpass filter is obtained
- by leaving high unspecified, or 0, or greater than
- or equal to the Nyquist freq.
- The window-len, if unspecified, defaults to 128.
- Longer windows give a sharper cutoff, smaller win-
- dows a more gradual cutoff.
- The beta, if unspecified, defaults to 16. This
- selects a Kaiser window. You can select a Nuttall
- window by specifying anything <= 2.0 here. For
- more discussion of beta, look under the resample
- effect.
+ December 10, 1999 10
- flanger gain-in gain-out delay decay speed -s | -t
- Add a flanger to a sound sample. Each triple
- delay/decay/speed gives the delay in milliseconds
- and the decay (relative to gain-in) with a modula-
- tion speed in Hz. The modulation is either sino-
- dial (-s) or triangular (-t). Gain-out is the
- volume of the output.
- highp center
- Apply a high-pass filter. The frequency response
- drops logarithmically with center frequency in the
- middle of the drop. The slope of the filter is
- quite gentle. See filter for a highpass effect
- with sharper cutoff.
- lowp center
- Apply a low-pass filter. The frequency response
- drops logarithmically with center frequency in the
- middle of the drop. The slope of the filter is
- quite gentle. See filter for a lowpass effect
- with sharper cutoff.
- map Display a list of loops in a sample, and miscel-
- laneous loop info.
+SoX(1) SoX(1)
- mask Add "masking noise" to signal. This effect deli-
- berately adds white noise to a sound in order to
- mask quantization effects, created by the process
- of playing a sound digitally. It tends to mask
+ windows a more gradual cutoff.
+ The beta, if unspecified, defaults to 16. This
+ selects a Kaiser window. You can select a Nut-
+ tall window by specifying anything <= 2.0 here.
+ For more discussion of beta, look under the
+ resample effect.
-SunOS 5.6 Last change: November 8, 1999 10
+ flanger gain-in gain-out delay decay speed -s | -t
+ Add a flanger to a sound sample. Each triple
+ delay/decay/speed gives the delay in millisec-
+ onds and the decay (relative to gain-in) with a
+ modulation speed in Hz. The modulation is
+ either sinodial (-s) or triangular (-t). Gain-
+ out is the volume of the output.
+ highp center
+ Apply a high-pass filter. The frequency
+ response drops logarithmically with center fre-
+ quency in the middle of the drop. The slope of
+ the filter is quite gentle. See filter for a
+ highpass effect with sharper cutoff.
+ highpass Butterworth highpass filter. Description com-
+ ming soon!
+ lowp center
+ Apply a low-pass filter. The frequency response
+ drops logarithmically with center frequency in
+ the middle of the drop. The slope of the filter
+ is quite gentle. See filter for a lowpass
+ effect with sharper cutoff.
+ lowpass Butterworth lowpass filter. Description coming
+ soon!
-User Commands SoX(1)
+ map Display a list of loops in a sample, and miscel-
+ laneous loop info.
+ mask Add "masking noise" to signal. This effect
+ deliberately adds white noise to a sound in
+ order to mask quantization effects, created by
+ the process of playing a sound digitally. It
+ tends to mask buzzing voices, for example. It
+ adds 1/2 bit of noise to the sound file at the
+ output bit depth.
+ phaser gain-in gain-out delay decay speed -s | -t
+ Add a phaser to a sound sample. Each triple
+ delay/decay/speed gives the delay in millisec-
+ onds and the decay (relative to gain-in) with a
+ modulation speed in Hz. The modulation is
+ either sinodial (-s) or triangular (-t). The
- buzzing voices, for example. It adds 1/2 bit of
- noise to the sound file at the output bit depth.
- phaser gain-in gain-out delay decay speed -s | -t
- Add a phaser to a sound sample. Each triple
- delay/decay/speed gives the delay in milliseconds
- and the decay (relative to gain-in) with a modula-
- tion speed in Hz. The modulation is either sino-
- dial (-s) or triangular (-t). The decay should be
- less than 0.5 to avoid feedback. Gain-out is the
- volume of the output.
- pick Select the left or right channel of a stereo sam-
- ple, or one of four channels in a quadrophonic
- sample.
+ December 10, 1999 11
- polyphase [ -w < num / ham > ]
- [ -width < long / short / # > ]
- [ -cutoff # ]
- Translate input sampling rate to output sampling
- rate via polyphase interpolation, a DSP algorithm.
- This method is slow and uses lots of RAM, but
- gives much better results then rate.
- -w < nut / ham > : select either a Nuttal (~90 dB
- stopband) or Hamming (~43 dB stopband) window.
- Warning: Nuttall windows require 2x length than
- Hamming windows. Default is nut.
- -width long / short / # : specify the width of the
- filter. long is 1024 samples; short is 128 sam-
- ples. Alternatively, an exact number can be used.
- Default is long.
- -cutoff # : specify the filter cutoff frequency in
- terms of fraction of bandwidth. If upsampling,
- then this is the fraction of the orignal signal
- that should go through. If downsampling, this is
- the fraction of the signal left after downsam-
- pling. Default is 0.95. Remember that this is a
- float.
- rate Translate input sampling rate to output sampling
- rate via linear interpolation to the Least Common
- Multiple of the two sampling rates. This is the
- default effect if the two files have different
- sampling rates and the preview options was speci-
- fied. This is fast but noisy: the spectrum of
- the original sound will be shifted upwards and
- duplicated faintly when up-translating by a multi-
- ple. Lerp-ing is acceptable for cheap 8-bit sound
- hardware, but for CD-quality sound you should
+SoX(1) SoX(1)
+ decay should be less than 0.5 to avoid feedback.
+ Gain-out is the volume of the output.
-SunOS 5.6 Last change: November 8, 1999 11
+ pick Select the left or right channel of a stereo
+ sample, or one of four channels in a quadro-
+ phonic sample.
+ polyphase [ -w < nut / ham > ]
+ [ -width < long / short / # > ]
+ [ -cutoff # ]
+ Translate input sampling rate to output sampling
+ rate via polyphase interpolation, a DSP algo-
+ rithm. This method is slow and uses lots of
+ RAM, but gives much better results than rate.
+ -w < nut / ham > : select either a Nuttal (~90
+ dB stopband) or Hamming (~43 dB stopband) win-
+ dow. Default is nut.
+ -width long / short / # : specify the (approxi-
+ mate) width of the filter. long is 1024 sam-
+ ples; short is 128 samples. Alternatively, an
+ exact number can be used. Default is long. The
+ short option is not recommended, as it produces
+ poor quality results.
+ -cutoff # : specify the filter cutoff frequency
+ in terms of fraction of bandwidth. If upsam-
+ pling, then this is the fraction of the original
+ signal that should go through. If downsampling,
+ this is the fraction of the signal left after
+ downsampling. Default is 0.95. Remember that
+ this is a float.
+ rate Translate input sampling rate to output sampling
+ rate via linear interpolation to the Least Com-
+ mon Multiple of the two sampling rates. This is
+ the default effect if the two files have differ-
+ ent sampling rates and the preview options was
+ specified. This is fast but noisy: the spectrum
+ of the original sound will be shifted upwards
+ and duplicated faintly when up-translating by a
+ multiple. Lerp-ing is acceptable for cheap
+ 8-bit sound hardware, but for CD-quality sound
+ you should instead use either resample or
+ polyphase. If you are wondering which of SoX's
+ rate changing effects to use, you will want to
+ read a detailed analysis of all of them at
+ http://eakaw2.et.tu-dresden.de/~andreas/resam-
+ ple/resample.html [Nov,1999: These tests need to
+ be updated for sox-12.17, which has bugfixes to
+ the resample and polyphase code.]
-User Commands SoX(1)
- instead use either resample or polyphase. If you
- are wondering which of Sox's rate changing effects
- to ues, you will want to read a detailed analysis
- of all of them at http://eakaw2.et.tu-
- dresden.de/~andreas/resample/resample.html
- [Nov,1999: These tests need to be updated for
- sox-12.18, which has bugfixes to the resample and
- polyphase code.]
- resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
- Translate input sampling rate to output sampling
- rate via simulated analog filtration. This method
- is slower than rate, but gives much better
- results.
+ December 10, 1999 12
- The -qs, -q, or -ql options specify increased
- accuracy at the cost of lower execution speed. By
- default, linear interpolation is used, with a win-
- dow width about 37 samples at the lower rate.
- This gives an accuracy of about 16 bits, but
- insufficient stopband rejection in the case that
- you want to have rolloff greater than about 0.85
- of the Nyquist frequency. The -q* options use
- quadratic interpolation of filter coefficients,
- resulting in about 22 bits precision. -qs, -q, or
- -ql use window lengths of 37, 75, or 150 samples,
- respectively, at the lower sample-rate of the two
- files. This means progressively sharper stop-band
- rejection, at proportionally slower execution
- times.
- rolloff refers to the cut-off frequency of the low
- pass filter and is given in terms of the Nyquist
- frequency for the lower sample rate. rolloff
- therefore should be something between 0. and 1.,
- in practice 0.8-0.95. The default is 0.8.
- The beta parameter determines the type of filter
- window used. Any value greater than 2.0 is the
- beta for a Kaiser window. Beta <= 2.0 selects a
- Nuttall window. If unspecified, the default is a
- Kaiser window with beta 16.
- In the case of Kaiser window beta > 2.0, lower
- betas produce a somewhat faster transition from
- passband to stopband, at the cost of noticeable
- artifacts. A beta of 16 is the default, beta less
- than 10 is not recommended. If you want a sharper
- cutoff, don't use low beta's, use a longer sample
- window. A Nuttall window is selected by specify-
- ing any 'beta' <= 2, and the Nuttall window has
- somewhat steeper cutoff than the default Kaiser
+SoX(1) SoX(1)
-SunOS 5.6 Last change: November 8, 1999 12
+ resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
+ Translate input sampling rate to output sampling
+ rate via simulated analog filtration. This
+ method is slower than rate, but gives much bet-
+ ter results.
+ The -qs, -q, or -ql options specify increased
+ accuracy at the cost of lower execution speed.
+ By default, linear interpolation is used, with a
+ window width about 45 samples at the lower rate.
+ This gives an accuracy of about 16 bits, but
+ insufficient stopband rejection in the case that
+ you want to have rolloff greater than about 0.80
+ of the Nyquist frequency. The -q* options use
+ quadratic interpolation of filter coefficients,
+ resulting in about 24 bits precision.
+ Following is a table of the reasonable defaults
+ which are built-in to sox:
+ Option Window rolloff beta interpolation
+ ------ ------ ------- ---- -------------
+ (none) 45 0.80 16 linear
+ -qs 45 0.80 16 quadratic
+ -q 75 0.875 16 quadratic
+ -ql 149 0.94 16 quadratic
+ ------ ------ ------- ---- -------------
+ -qs, -q, or -ql use window lengths of 45, 75, or
+ 149 samples, respectively, at the lower sample-
+ rate of the two files. This means progressively
+ sharper stop-band rejection, at proportionally
+ slower execution times.
+ rolloff refers to the cut-off frequency of the
+ low pass filter and is given in terms of the
+ Nyquist frequency for the lower sample rate.
+ rolloff therefore should be something between 0.
+ and 1., in practice 0.8-0.95. The defaults are
+ indicated above.
+ The beta parameter determines the type of filter
+ window used. Any value greater than 2.0 is the
+ beta for a Kaiser window. Beta <= 2.0 selects a
+ Nuttall window. If unspecified, the default is
+ a Kaiser window with beta 16.
+ In the case of Kaiser window (beta > 2.0), lower
+ betas produce a somewhat faster transition from
+ passband to stopband, at the cost of noticeable
+ artifacts. A beta of 16 is the default, beta
+ less than 10 is not recommended. If you want a
+ sharper cutoff, don't use low beta's, use a
+ longer sample window. A Nuttall window is
+ selected by specifying any 'beta' <= 2, and the
+ Nuttall window has somewhat steeper cutoff than
+ the default Kaiser window. You will probably
-User Commands SoX(1)
+ December 10, 1999 13
- window. You will probably not need to use the
- beta parameter at all, unless you are just curious
- about comparing the effects of Nuttall vs. Kaiser
- windows.
- This is the default effect if the two files have
- different sampling rates. Default parameters are
- Kaiser window of length 37, rolloff 0.80, beta 16,
- linear interpolation. -qs is only slightly
- slower, but more accurate for 16-bit or higher
- precision.
- reverb gain-out delay [ delay ... ]
- Add reverbation to a sound sample. Each delay is
- given in milliseconds and its feedback is depend-
- ing on the reverb-time in milliseconds. Each
- delay should be in the range of half to quarter of
- reverb-time to get a realistic reverbation.
- Gain-out is the volume of the output.
+SoX(1) SoX(1)
- reverse Reverse the sound sample completely. Included for
- finding Satanic subliminals.
- split Turn a mono sample into a stereo sample by copying
- the input channel to the left and right channels.
+ not need to use the beta parameter at all,
+ unless you are just curious about comparing the
+ effects of Nuttall vs. Kaiser windows.
- stat [ debug | -v ]
- Do a statistical check on the input file, and
- print results on the standard error file. stat
- may copy the file untouched from input to output,
- if you select an output file. The "Volume Adjust-
- ment:" field in the statistics gives you the argu-
- ment to the -v number which will make the sample
- as loud as possible without clipping. There is an
- optional parameter -v that will print out the
- "Volume Adjustment:" field's value and return.
- This could be of use in scripts to auto convert
- the volume. There is an also an optional parame-
- ter debug that will place sox into debug mode and
- print out a hex dump of the sound file from the
- internal buffer that is in 32-bit signed PCM data.
- This is mainly only of use in tracking down endian
- problems that creep in to sox on cross-platform
- versions.
+ This is the default effect if the two files have
+ different sampling rates. Default parameters
+ are, as indicated above, Kaiser window of length
+ 45, rolloff 0.80, beta 16, linear interpolation.
- swap [ 1 2 3 4 ]
- Swap channels in multi-channel sound files. In
- files with more than 2 channels you may specify
- the order that the channels should be rearranged
- in.
+ NOTE: -qs is only slightly slower, but more
+ accurate for 16-bit or higher precision.
+ NOTE: In many cases of up-sampling, no interpo-
+ lation is needed, as exact filter coefficients
+ can be computed in a reasonable amount of space.
+ To be precise, this is done when
+ input_rate < output_rate
+ &&
+ output_rate/gcd(input_rate,output_rate) <= 511
+ reverb gain-out delay [ delay ... ]
+ Add reverberation to a sound sample. Each delay
+ is given in milliseconds and its feedback is
+ depending on the reverb-time in milliseconds.
+ Each delay should be in the range of half to
+ quarter of reverb-time to get a realistic rever-
+ beration. Gain-out is the volume of the output.
-SunOS 5.6 Last change: November 8, 1999 13
+ reverse Reverse the sound sample completely. Included
+ for finding Satanic subliminals.
+ split Turn a mono sample into a stereo sample by copy-
+ ing the input channel to the left and right
+ channels.
+ stat [ debug | -v ]
+ Do a statistical check on the input file, and
+ print results on the standard error file. stat
+ may copy the file untouched from input to out-
+ put, if you select an output file. The "Volume
+ Adjustment:" field in the statistics gives you
+ the argument to the -v number which will make
+ the sample as loud as possible without clipping.
+ There is an optional parameter -v that will
+ print out the "Volume Adjustment:" field's value
+ and return. This could be of use in scripts to
+ auto convert the volume. There is an also an
+ optional parameter debug that will place sox
+ into debug mode and print out a hex dump of the
+ sound file from the internal buffer that is in
+ 32-bit signed PCM data. This is mainly only of
+ use in tracking down endian problems that creep
+ in to sox on cross-platform versions.
+ December 10, 1999 14
-User Commands SoX(1)
- vibro speed [ depth ]
- Add the world-famous Fender Vibro-Champ sound
- effect to a sound sample by using a sine wave as
- the volume knob. Speed gives the Hertz value of
- the wave. This must be under 30. Depth gives the
- amount the volume is cut into by the sine wave,
- ranging 0.0 to 1.0 and defaulting to 0.5.
- Sox enforces certain effects. If the two files have dif-
- ferent sampling rates, the requested effect must be one of
- copy, or rate, If the two files have different numbers of
- channels, the avg effect must be requested.
+SoX(1) SoX(1)
+
+ swap [ 1 2 3 4 ]
+ Swap channels in multi-channel sound files. In
+ files with more than 2 channels you may specify
+ the order that the channels should be rearranged
+ in.
+
+ vibro speed [ depth ]
+ Add the world-famous Fender Vibro-Champ sound
+ effect to a sound sample by using a sine wave as
+ the volume knob. Speed gives the Hertz value of
+ the wave. This must be under 30. Depth gives
+ the amount the volume is cut into by the sine
+ wave, ranging 0.0 to 1.0 and defaulting to 0.5.
+
+ Sox enforces certain effects. If the two files have dif-
+ ferent sampling rates, the requested effect must be one of
+ copy, or rate, If the two files have different numbers of
+ channels, the avg effect must be requested.
+
BUGS
- The syntax is horrific. It's very tempting to include a
- default system that allows an effect name as the program
- name and just pipes a sound sample from standard input to
- standard output, but the problem of inputting the sample
- rates makes this unworkable.
+ The syntax is horrific. Thats the breaks when trying to
+ handle all things from the command line.
- Please report any bugs found in this version of sox to Chris
- Bagwell (cbagwell@sprynet.com)
+ Please report any bugs found in this version of sox to
+ Chris Bagwell (cbagwell@sprynet.com)
FILES
SEE ALSO
- play(1), rec(1)
+ play(1), rec(1), soxexam(1)
NOTICES
- The echoplex effect is: Copyright (C) 1989 by Jef
- Poskanzer.
+ The version of Sox that accompanies this manual page is
+ support by Chris Bagwell (cbagwell@sprynet.com). Please
+ refer any questions regarding it to this address. You may
+ obtain the latest version at the the web site
+ http://home.sprynet.com/~cbagwell/sox.html
- Permission to use, copy, modify, and distribute this
- software and its documentation for any purpose and without
- fee is hereby granted, provided that the above copyright
- notice appear in all copies and that both that copyright
- notice and this permission notice appear in supporting docu-
- mentation. This software is provided "as is" without
- express or implied warranty.
- The version of Sox that accompanies this manual page is sup-
- port by Chris Bagwell (cbagwell@sprynet.com). Please refer
- any questions regarding it to this address. You may obtain
- the latest version at the the web site
- http://home.sprynet.com/~cbagwell/sox.html
@@ -918,7 +977,14 @@
-SunOS 5.6 Last change: November 8, 1999 14
+
+
+
+
+
+
+
+ December 10, 1999 15
--- /dev/null
+++ b/soxexam.1
@@ -1,0 +1,441 @@
+.de Sh
+.br
+.ne 5
+.PP
+\fB\\$1\fR
+.PP
+..
+.de Sp
+.if t .sp .5v
+.if n .sp
+..
+.TH SoX 1 "December 10, 1999"
+.SH NAME
+soxexam - SoX Examples (CHEAT SHEET)
+.SH CONVERSIONS
+.B Introduction
+.P
+In general, sox will attempt to take an input sound file format and
+convert it to a new file format using a similar data type and sample
+rate. For instance, "sox monkey.au monkey.wav" would try and convert
+the mono 8000Hz u-law sample .au file that comes with sox to a 8000Hz
+u-law .wav file.
+.P
+If an output format doesn't support the same data type as the input file
+then sox will generally select a default data type to save it in.
+You can override the default data type selection by using command line
+options. This is also useful for producing a output file with higher
+or lower percision data and/or sample rate.
+.P
+Most file formats that contain headers can automatically be read in.
+When working with headerless file formats then a user must manually
+tell sox the data type and sample rate using command line options.
+.P
+When working with headerless files (raw files), you may take advantage of
+they pseudo-file types of .ub, .uw, .sb, .sw, .ul, and .sl. By using these
+extensions on your filenames you will not have to specify the corrisponding
+options on the command line.
+.P
+.B Percision
+.P
+The following data types and formats can be represented by their total
+uncompressed bit percision. When converting from one data type to another
+care must be taken to insure it has an equal or greater percision. If not
+then the audio quality will be degraded. This is not always a bad thing
+when your working with things such as voice audio and are concerned about
+disk space or bandwidth of the audio data.
+.P
+.br
+ Data Format Percision
+.br
+ ___________ _________
+.br
+ unsigned byte 8-bit
+.br
+ signed byte 8-bit
+.br
+ u-law 12-bit
+.br
+ a-law 12-bit
+.br
+ unsigned word 16-bit
+.br
+ signed word 16-bit
+.br
+ ADPCM 16-bit
+.br
+ GSM 16-bit
+.br
+ unsigned long 32-bit
+.br
+ signed long 32-bit
+.br
+ ___________ _________
+.P
+.B Examples
+.P
+Use the '-V' option on all your command lines. It makes SoX print out its
+idea of what is going on. '-V' is your friend.
+.P
+To convert from unsigned bytes at 8000 Hz to signed words at 8000 Hz:
+.P
+.br
+ sox -r 8000 -c 1 filename.ub newfile.sw
+.P
+To convert from Apple's AIFF format to Microsoft's WAV format:
+.P
+.br
+ sox filename.aiff filename.wav
+.P
+To convert from mono raw 8000 Hz 8-bit unsigned PCM data to a WAV file:
+.P
+.br
+ sox -r 8000 -u -b -c 1 filename.raw filename.wav
+.P
+.I SoX
+is great to use along with other command line programs by passing data
+between the programs using pipelines. The most common example is to use
+mpg123 to convert mp3 files in to wav files. The following command line will
+do this:
+.P
+.br
+ mpg123 -b 10000 -s filename.mp3 | sox -t raw -r 44100 -s -w -c 2 - filename.wav
+.P
+When working with totally unknown audio data then the "auto" file format may
+be of use. It attempts to guess what the file type is and then you may
+save it in to a known audio format.
+.P
+.br
+ sox -V -t auto filename.snd filename.wav
+.P
+It is important to understand how the internals of
+.I SoX
+work with
+compressed audio including u-law, a-law, ADPCM, or GSM.
+.I SoX
+takes ALL input data types and converts them to uncompressed 32-bit
+signed data. It will then convert this internal version into the
+requested output format. This means unneeded noise can be introduced
+from decompressing data and then recompressing. If applying multiple
+effects to audio data it is best to save the intermediate data as PCM
+data. After the final effect is performed then you can specify it as
+a compressed output format. This will keep noise introduction to a minimum.
+.P
+The following example is to apply various effects to an 8000 Hz ADPCM input
+file and then end up with the final file as 44100 Hz ADPCM.
+.P
+.br
+ sox firstfile.wav -r 44100 -s -w secondfile.wav
+.br
+ sox secondfile.wav thirdfile.wav swap
+.br
+ sox thirdfile.wav -a -b finalfile.wav mask
+.P
+Under a DOS shell, you can convert several audio files to an new output
+format using something similar to the following command line:
+.P
+.br
+ FOR %X IN (*.RAW) DO sox -r 11025 -w -s -t raw $X $X.wav
+.SH EFFECTS
+Special thanks goes to Juergen Mueller (jmeuller@uia.au.ac.be) for this
+write up on effects.
+.P
+.B Introduction:
+.P
+The core problem is that you need some experience in using effects
+in order to say "that any old sound file sounds with effects
+absolutely hip". There isn't any rule-based system which tell you
+the correct setting of all the parameters for every effect.
+But after some time you will become an expert in using effects.
+.P
+Here are some examples which can be used with any music sample.
+(For a sample where only a single instrument is playing, extreme
+parameter setting may make well-known "typically" or "classical"
+sounds. Likewise, for drums, vocals or guitars.)
+.P
+Single effects will be explained and some given parameter settings
+that can be used to understand the theorie by listening to the sound file
+with the added effect.
+.P
+Using multiple effects in parallel or in sequel can result either
+in very perfect sound or ( mostly ) in a dramatic overloading in
+variations of sounds such that your ear may follow the sound but
+you will feel unsatisfied. Hence, for the first time using effects
+try to compose them as less as possible. We don't regard the
+composition of effects in the examples because to many combinations
+are possible and you really need a very fast maschine and a lot of
+memory to play them in real-time.
+.P
+And real-time playing of sounds will speed up learning the parameter
+setting.
+.P
+Basically, we will use the "play" front-end of SOX since it is easier
+to listen sounds coming out of the speaker or earphone instead
+of looking at cryptical data in sound files.
+.P
+For easy listening of file.xxx ( "xxx" is any sound format ):
+.P
+.BR
+ play file.xxx effect-name effect-parameters
+.P
+Or more SOX-like ( for "dsp" output ):
+.P
+.BR
+ sox file.xxx -t ossdsp -w -s /dev/dsp effect-name effect-parameters
+.P
+or ( for "au" output ):
+.P
+.BR
+ sox file.xxx -t sunau -w -s /dev/audio effect-name effect-parameters
+.P
+And for date freaks:
+.P
+.BR
+ sox file.xxx file.yyy effect-name effect-parameters
+.P
+Additional options can be used. However, in this case, for real-time
+playing you'll need a very fast machine.
+.P
+Notes:
+.P
+I played all examples in real-time on a Pentium 100 with 32 Mb and
+Linux 2.0.30 using a self-recorded sample ( 3:15 min long in "wav"
+format with 44.1 kHz sample rate and stereo 16 bit ).
+The sample should not contain any of the effects. However,
+if you take any recording of a sound track from radio or tape or cd,
+and it sounds like a live concert or ten people are playing the same
+rhythm with their drums or funky-groves, then take any other sample.
+(Typically, less then four different intruments and no synthesizer
+in the sample is suitable. Likewise, the combination vocal, drums, bass
+and guitar.)
+.P
+Effects:
+.P
+.B Echo
+.P
+An echo effect can be naturally found in the mountains, standing somewhere
+on a moutain and shouting a single word will result in one or more repetitions
+of the word ( if not, turn a bit around ant try next, or climb to the next
+mountain ).
+.P
+However, the time difference between shouting and repeating is the delay
+(time), its loudness is the decay. Multiple echos can have different delays and
+decays.
+.P
+Very popular is using echos to play an instrument with itself together, like
+some guitar players ( Brain May from Queen ) or vocalists are doing.
+For music samples of more than one instrument, echo can be used to add a
+second sample shortly after the original one.
+.P
+This will sound as doubling the number of instruments playing the same sample:
+.P
+.BR
+ play file.xxx echo 0.8 0.88 60.0 0.4
+.P
+If the delay is very short then it sound like a (metallic) roboter playing
+music:
+.P
+.BR
+ play file.xxx echo 0.8 0.88 6.0 0.4
+.P
+Longer delay will sound like a open air concert in the mountains:
+.P
+.BR
+ play file.xxx echo 0.8 0.9 1000.0 0.3
+.P
+One mountain more, and:
+.P
+.BR
+ play file.xxx echo 0.8 0.9 1000.0 0.3 1800.0 0.25
+.P
+.B Echos
+.P
+Like the echo effect, echos stand for "ECHO in Sequel", that is the first echos
+takes the input, the second the input and the first echos, the third the input
+and the first and the second echos, ... and so on.
+Care should be taken using many echos ( see introduction ); a single echos
+has the same effect as a single echo.
+.P
+The sample will be bounced twice in symmetric echos:
+.P
+.BR
+ play file.xxx echos 0.8 0.7 700.0 0.25 700.0 0.3
+.P
+The sample will be bounced twice in asymmetric echos:
+.P
+.BR
+ play file.xxx echos 0.8 0.7 700.0 0.25 900.0 0.3
+.P
+The sample will sound as played in a garage:
+.P
+.BR
+ play file.xxx echos 0.8 0.7 40.0 0.25 63.0 0.3
+.P
+.B Chorus
+.P
+The chorus effect has its name because it will often be used to make a single
+vocal sound like a chorus. But it can be applied to other instrument samples
+too.
+.P
+It works like the echo effect with a short delay, but the delay isn't constant.
+The delay is varied using a sinodial or triangular modulation. The modulation
+depth defines the range the modulated delay is played before or after the
+delay. Hence the delayed sound will sound slower or faster, that is the delayed
+sound tuned around the original one, like in a chorus where some vocal are
+a bit out of tune.
+.P
+The typical delay is around 40ms to 60ms, the speed of the modualtion is best
+near 0.25Hz and the modulation depth around 2ms.
+.P
+A single delay will make the sample more overloaded:
+.P
+.BR
+ play file.xxx chorus 0.7 0.9 55.0 0.4 0.25 2.0 -t
+.P
+Two delays of the original samples sound like this:
+.P
+.BR
+ play file.xxx chorus 0.6 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4 1.3 -s
+.P
+A big chorus of the sample is ( three additional samples ):
+.P
+.BR
+ play file.xxx chorus 0.5 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4 2.3 -t \
+ 40.0 0.3 0.3 1.3 -s
+.P
+.B Flanger
+.P
+The flanger effect is like the chorus effect, but the delay varies between
+0ms and maximal 5ms. It sound like wind blowing, sometimes faster or slower
+including changes of the speed.
+.P
+The flanger effect is widely used in funk and soul music, where the guitar
+sound varies frequently slow or a bit faster.
+.P
+The typical delay is around 3ms to 5ms, the speed of the modulation is best
+near 0.5Hz.
+.P
+Now, let's groove the sample:
+.P
+.BR
+ play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -s
+.P
+listen carefully between the difference of sinodial and triangular modulation:
+.P
+.BR
+ play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -t
+.P
+If the decay is a bit lower, than the effect sounds more popular:
+.P
+.BR
+ play file.xxx flanger 0.8 0.88 3.0 0.4 0.5 -t
+.P
+The drunken loundspeaker system:
+.P
+.BR
+ play file.xxx flanger 0.9 0.9 4.0 0.23 1.3 -s
+.P
+.B Reverb
+.P
+The reverb effect is often used in audience hall which are to small or to many
+visitors disturb the reflection of sound at the walls to make the sound played
+more monumental. You can try the reverb effect in your bathroom or garage or
+sport halls by shouting loud some words. You'll hear the words reflected from
+the walls.
+.P
+The biggest problem in using the reverb effect is the correct setting of the
+(wall) delays such that the sound is relistic an doesn't sound like music
+playing in a tin or overloaded feedback distroys any illusion of any big hall.
+To help you for much realisitc reverb effects, you should decide first, how
+long the reverb should take place until it is not loud enough to be registered
+by your ears. This is be done by the reverb time "t", in small halls 200ms in
+bigger one 1000ms, if you like. Clearly, the walls of such a hall aren't far
+away, so you should define its setting be given every wall its delay time.
+However, if the wall is to far eway for the reverb time, you won't hear the
+reverb, so the nearest wall will be best "t/4" delay and the farest "t/2".
+You can try other distances as well, but it won't sound very realistic.
+The walls shouldn't stand to close to each other and not in a multiple interger
+distance to each other ( so avoid wall like: 200.0 and 202.0, or something
+like 100.0 and 200.0 ).
+.P
+Since audience halls do have a lot of walls, we will start designing one
+beginning with one wall:
+.P
+.BR
+ play file.xxx reverb 1.0 600.0 180.0
+.P
+One wall more:
+.P
+.BR
+ play file.xxx reverb 1.0 600.0 180.0 200.0
+.P
+Next two walls:
+.P
+.BR
+ play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0
+.P
+Now, why not a futuristic hall with six walls:
+.P
+.BR
+ play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0 280.0 300.0
+.P
+If you run out of machine power or memory, then stop as much applications
+as possible ( every interupt will consume a lot of cpu time which for
+bigger halls is absolutely neccessary ).
+.P
+.B Phaser
+.P
+The phaser effect is like the flanger effect, but it uses a reverb instead of
+an echo and does phase shifting. You'll hear the difference in the examples
+comparing both effects ( simply change the effect name ).
+The delay modulation can be done sinodial or triangular, preferable is the
+later one for multiple instruments playing. For single instrument sounds
+the sinodial phaser effect will give a sharper phasing effect.
+The decay shouln't be to close to 1.0 which will cause dramatic feedback.
+A good range is about 0.5 to 0.1 for the decay.
+.P
+We will take a parameter setting as for the flanger before ( gain-out is
+lower since feedback can raise the output dramatically ):
+.P
+.BR
+ play file.xxx phaser 0.8 0.74 3.0 0.4 0.5 -t
+.P
+The drunken loundspeaker system ( now less alkohol ):
+.P
+.BR
+ play file.xxx phaser 0.9 0.85 4.0 0.23 1.3 -s
+.P
+A popular sound of the sample is as follows:
+.P
+.BR
+ play file.xxx phaser 0.89 0.85 1.0 0.24 2.0 -t
+.P
+The sample sounds if ten springs are in your ears:
+.P
+.BR
+ play file.xxx phaser 0.6 0.66 3.0 0.6 2.0 -t
+.P
+.B Other effects ( copy, rate, avg, stat, vibro, lowp, highp, band, reverb )
+.P
+The other effects are simply to use. However, an "easy to use manual" should
+be given here.
+.P
+.B More effects ( to do ! )
+.P
+There are a lot of effects around like noise gates, compressors, waw-waw,
+stereo effects and so on. They should be implemented making SOX to be more
+useful in sound mixing technics coming together with a great varity of
+different sound effects.
+.P
+Combining effects be using then in parallel or sequel on different channels
+needs some easy mechanism which is real-time stable.
+.P
+Really missing, is the changing of the parameters, starting and stoping of
+effects while playing samples in real-time!
+.P
+Good luck and have fun with all the effects!
+
+ Juergen Mueller (jmueller@uia.ua.ac.be)
+
+.SH SEE ALSO
+sox(1), play(1), rec(1)
--- /dev/null
+++ b/soxexam.txt
@@ -1,0 +1,594 @@
+
+
+
+SoX(1) SoX(1)
+
+
+NAME
+ soxexam - SoX Examples (CHEAT SHEET)
+
+CONVERSIONS
+ Introduction
+
+ In general, sox will attempt to take an input sound file
+ format and convert it to a new file format using a similar
+ data type and sample rate. For instance, "sox monkey.au
+ monkey.wav" would try and convert the mono 8000Hz u-law
+ sample .au file that comes with sox to a 8000Hz u-law .wav
+ file.
+
+ If an output format doesn't support the same data type as
+ the input file then sox will generally select a default
+ data type to save it in. You can override the default
+ data type selection by using command line options. This
+ is also useful for producing a output file with higher or
+ lower percision data and/or sample rate.
+
+ Most file formats that contain headers can automatically
+ be read in. When working with headerless file formats
+ then a user must manually tell sox the data type and sam-
+ ple rate using command line options.
+
+ When working with headerless files (raw files), you may
+ take advantage of they pseudo-file types of .ub, .uw, .sb,
+ .sw, .ul, and .sl. By using these extensions on your
+ filenames you will not have to specify the corrisponding
+ options on the command line.
+
+ Percision
+
+ The following data types and formats can be represented by
+ their total uncompressed bit percision. When converting
+ from one data type to another care must be taken to insure
+ it has an equal or greater percision. If not then the
+ audio quality will be degraded. This is not always a bad
+ thing when your working with things such as voice audio
+ and are concerned about disk space or bandwidth of the
+ audio data.
+
+ Data Format Percision
+ ___________ _________
+ unsigned byte 8-bit
+ signed byte 8-bit
+ u-law 12-bit
+ a-law 12-bit
+ unsigned word 16-bit
+ signed word 16-bit
+ ADPCM 16-bit
+ GSM 16-bit
+ unsigned long 32-bit
+ signed long 32-bit
+
+
+
+ December 10, 1999 1
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+ ___________ _________
+
+ Examples
+
+ Use the '-V' option on all your command lines. It makes
+ SoX print out its idea of what is going on. '-V' is your
+ friend.
+
+ To convert from unsigned bytes at 8000 Hz to signed words
+ at 8000 Hz:
+
+ sox -r 8000 -c 1 filename.ub newfile.sw
+
+ To convert from Apple's AIFF format to Microsoft's WAV
+ format:
+
+ sox filename.aiff filename.wav
+
+ To convert from mono raw 8000 Hz 8-bit unsigned PCM data
+ to a WAV file:
+
+ sox -r 8000 -u -b -c 1 filename.raw filename.wav
+
+ SoX is great to use along with other command line programs
+ by passing data between the programs using pipelines. The
+ most common example is to use mpg123 to convert mp3 files
+ in to wav files. The following command line will do this:
+
+ mpg123 -b 10000 -s filename.mp3 | sox -t raw -r 44100 -s
+ -w -c 2 - filename.wav
+
+ When working with totally unknown audio data then the
+ "auto" file format may be of use. It attempts to guess
+ what the file type is and then you may save it in to a
+ known audio format.
+
+ sox -V -t auto filename.snd filename.wav
+
+ It is important to understand how the internals of SoX
+ work with compressed audio including u-law, a-law, ADPCM,
+ or GSM. SoX takes ALL input data types and converts them
+ to uncompressed 32-bit signed data. It will then convert
+ this internal version into the requested output format.
+ This means unneeded noise can be introduced from decom-
+ pressing data and then recompressing. If applying multi-
+ ple effects to audio data it is best to save the interme-
+ diate data as PCM data. After the final effect is per-
+ formed then you can specify it as a compressed output for-
+ mat. This will keep noise introduction to a minimum.
+
+ The following example is to apply various effects to an
+ 8000 Hz ADPCM input file and then end up with the final
+ file as 44100 Hz ADPCM.
+
+
+
+
+ December 10, 1999 2
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+ sox firstfile.wav -r 44100 -s -w secondfile.wav
+ sox secondfile.wav thirdfile.wav swap
+ sox thirdfile.wav -a -b finalfile.wav mask
+
+ Under a DOS shell, you can convert several audio files to
+ an new output format using something similar to the fol-
+ lowing command line:
+
+ FOR %X IN (*.RAW) DO sox -r 11025 -w -s -t raw $X $X.wav
+
+EFFECTS
+ Special thanks goes to Juergen Mueller
+ (jmeuller@uia.au.ac.be) for this write up on effects.
+
+ Introduction:
+
+ The core problem is that you need some experience in using
+ effects in order to say "that any old sound file sounds
+ with effects absolutely hip". There isn't any rule-based
+ system which tell you the correct setting of all the
+ parameters for every effect. But after some time you will
+ become an expert in using effects.
+
+ Here are some examples which can be used with any music
+ sample. (For a sample where only a single instrument is
+ playing, extreme parameter setting may make well-known
+ "typically" or "classical" sounds. Likewise, for drums,
+ vocals or guitars.)
+
+ Single effects will be explained and some given parameter
+ settings that can be used to understand the theorie by
+ listening to the sound file with the added effect.
+
+ Using multiple effects in parallel or in sequel can result
+ either in very perfect sound or ( mostly ) in a dramatic
+ overloading in variations of sounds such that your ear may
+ follow the sound but you will feel unsatisfied. Hence, for
+ the first time using effects try to compose them as less
+ as possible. We don't regard the composition of effects in
+ the examples because to many combinations are possible and
+ you really need a very fast maschine and a lot of memory
+ to play them in real-time.
+
+ And real-time playing of sounds will speed up learning the
+ parameter setting.
+
+ Basically, we will use the "play" front-end of SOX since
+ it is easier to listen sounds coming out of the speaker or
+ earphone instead of looking at cryptical data in sound
+ files.
+
+ For easy listening of file.xxx ( "xxx" is any sound format
+ ):
+
+
+
+
+ December 10, 1999 3
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+ play file.xxx effect-name effect-parameters
+
+ Or more SOX-like ( for "dsp" output ):
+
+ sox file.xxx -t ossdsp -w -s /dev/dsp effect-name
+ effect-parameters
+
+ or ( for "au" output ):
+
+ sox file.xxx -t sunau -w -s /dev/audio effect-name
+ effect-parameters
+
+ And for date freaks:
+
+ sox file.xxx file.yyy effect-name effect-parameters
+
+ Additional options can be used. However, in this case, for
+ real-time playing you'll need a very fast machine.
+
+ Notes:
+
+ I played all examples in real-time on a Pentium 100 with
+ 32 Mb and Linux 2.0.30 using a self-recorded sample ( 3:15
+ min long in "wav" format with 44.1 kHz sample rate and
+ stereo 16 bit ). The sample should not contain any of the
+ effects. However, if you take any recording of a sound
+ track from radio or tape or cd, and it sounds like a live
+ concert or ten people are playing the same rhythm with
+ their drums or funky-groves, then take any other sample.
+ (Typically, less then four different intruments and no
+ synthesizer in the sample is suitable. Likewise, the com-
+ bination vocal, drums, bass and guitar.)
+
+ Effects:
+
+ Echo
+
+ An echo effect can be naturally found in the mountains,
+ standing somewhere on a moutain and shouting a single word
+ will result in one or more repetitions of the word ( if
+ not, turn a bit around ant try next, or climb to the next
+ mountain ).
+
+ However, the time difference between shouting and repeat-
+ ing is the delay (time), its loudness is the decay. Multi-
+ ple echos can have different delays and decays.
+
+ Very popular is using echos to play an instrument with
+ itself together, like some guitar players ( Brain May from
+ Queen ) or vocalists are doing. For music samples of more
+ than one instrument, echo can be used to add a second sam-
+ ple shortly after the original one.
+
+ This will sound as doubling the number of instruments
+
+
+
+ December 10, 1999 4
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+ playing the same sample:
+
+ play file.xxx echo 0.8 0.88 60.0 0.4
+
+ If the delay is very short then it sound like a (metallic)
+ roboter playing music:
+
+ play file.xxx echo 0.8 0.88 6.0 0.4
+
+ Longer delay will sound like a open air concert in the
+ mountains:
+
+ play file.xxx echo 0.8 0.9 1000.0 0.3
+
+ One mountain more, and:
+
+ play file.xxx echo 0.8 0.9 1000.0 0.3 1800.0 0.25
+
+ Echos
+
+ Like the echo effect, echos stand for "ECHO in Sequel",
+ that is the first echos takes the input, the second the
+ input and the first echos, the third the input and the
+ first and the second echos, ... and so on. Care should be
+ taken using many echos ( see introduction ); a single
+ echos has the same effect as a single echo.
+
+ The sample will be bounced twice in symmetric echos:
+
+ play file.xxx echos 0.8 0.7 700.0 0.25 700.0 0.3
+
+ The sample will be bounced twice in asymmetric echos:
+
+ play file.xxx echos 0.8 0.7 700.0 0.25 900.0 0.3
+
+ The sample will sound as played in a garage:
+
+ play file.xxx echos 0.8 0.7 40.0 0.25 63.0 0.3
+
+ Chorus
+
+ The chorus effect has its name because it will often be
+ used to make a single vocal sound like a chorus. But it
+ can be applied to other instrument samples too.
+
+ It works like the echo effect with a short delay, but the
+ delay isn't constant. The delay is varied using a sin-
+ odial or triangular modulation. The modulation depth
+ defines the range the modulated delay is played before or
+ after the delay. Hence the delayed sound will sound slower
+ or faster, that is the delayed sound tuned around the
+ original one, like in a chorus where some vocal are a bit
+ out of tune.
+
+
+
+
+ December 10, 1999 5
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+ The typical delay is around 40ms to 60ms, the speed of the
+ modualtion is best near 0.25Hz and the modulation depth
+ around 2ms.
+
+ A single delay will make the sample more overloaded:
+
+ play file.xxx chorus 0.7 0.9 55.0 0.4 0.25 2.0 -t
+
+ Two delays of the original samples sound like this:
+
+ play file.xxx chorus 0.6 0.9 50.0 0.4 0.25 2.0 -t
+ 60.0 0.32 0.4 1.3 -s
+
+ A big chorus of the sample is ( three additional samples
+ ):
+
+ play file.xxx chorus 0.5 0.9 50.0 0.4 0.25 2.0 -t
+ 60.0 0.32 0.4 2.3 -t 40.0 0.3 0.3 1.3 -s
+
+ Flanger
+
+ The flanger effect is like the chorus effect, but the
+ delay varies between 0ms and maximal 5ms. It sound like
+ wind blowing, sometimes faster or slower including changes
+ of the speed.
+
+ The flanger effect is widely used in funk and soul music,
+ where the guitar sound varies frequently slow or a bit
+ faster.
+
+ The typical delay is around 3ms to 5ms, the speed of the
+ modulation is best near 0.5Hz.
+
+ Now, let's groove the sample:
+
+ play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -s
+
+ listen carefully between the difference of sinodial and
+ triangular modulation:
+
+ play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -t
+
+ If the decay is a bit lower, than the effect sounds more
+ popular:
+
+ play file.xxx flanger 0.8 0.88 3.0 0.4 0.5 -t
+
+ The drunken loundspeaker system:
+
+ play file.xxx flanger 0.9 0.9 4.0 0.23 1.3 -s
+
+ Reverb
+
+ The reverb effect is often used in audience hall which are
+
+
+
+ December 10, 1999 6
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+ to small or to many visitors disturb the reflection of
+ sound at the walls to make the sound played more monumen-
+ tal. You can try the reverb effect in your bathroom or
+ garage or sport halls by shouting loud some words. You'll
+ hear the words reflected from the walls.
+
+ The biggest problem in using the reverb effect is the cor-
+ rect setting of the (wall) delays such that the sound is
+ relistic an doesn't sound like music playing in a tin or
+ overloaded feedback distroys any illusion of any big hall.
+ To help you for much realisitc reverb effects, you should
+ decide first, how long the reverb should take place until
+ it is not loud enough to be registered by your ears. This
+ is be done by the reverb time "t", in small halls 200ms in
+ bigger one 1000ms, if you like. Clearly, the walls of such
+ a hall aren't far away, so you should define its setting
+ be given every wall its delay time. However, if the wall
+ is to far eway for the reverb time, you won't hear the
+ reverb, so the nearest wall will be best "t/4" delay and
+ the farest "t/2". You can try other distances as well,
+ but it won't sound very realistic. The walls shouldn't
+ stand to close to each other and not in a multiple
+ interger distance to each other ( so avoid wall like:
+ 200.0 and 202.0, or something like 100.0 and 200.0 ).
+
+ Since audience halls do have a lot of walls, we will start
+ designing one beginning with one wall:
+
+ play file.xxx reverb 1.0 600.0 180.0
+
+ One wall more:
+
+ play file.xxx reverb 1.0 600.0 180.0 200.0
+
+ Next two walls:
+
+ play file.xxx reverb 1.0 600.0 180.0 200.0 220.0
+ 240.0
+
+ Now, why not a futuristic hall with six walls:
+
+ play file.xxx reverb 1.0 600.0 180.0 200.0 220.0
+ 240.0 280.0 300.0
+
+ If you run out of machine power or memory, then stop as
+ much applications as possible ( every interupt will con-
+ sume a lot of cpu time which for bigger halls is abso-
+ lutely neccessary ).
+
+ Phaser
+
+ The phaser effect is like the flanger effect, but it uses
+ a reverb instead of an echo and does phase shifting.
+ You'll hear the difference in the examples comparing both
+
+
+
+ December 10, 1999 7
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+ effects ( simply change the effect name ). The delay mod-
+ ulation can be done sinodial or triangular, preferable is
+ the later one for multiple instruments playing. For single
+ instrument sounds the sinodial phaser effect will give a
+ sharper phasing effect. The decay shouln't be to close to
+ 1.0 which will cause dramatic feedback. A good range is
+ about 0.5 to 0.1 for the decay.
+
+ We will take a parameter setting as for the flanger before
+ ( gain-out is lower since feedback can raise the output
+ dramatically ):
+
+ play file.xxx phaser 0.8 0.74 3.0 0.4 0.5 -t
+
+ The drunken loundspeaker system ( now less alkohol ):
+
+ play file.xxx phaser 0.9 0.85 4.0 0.23 1.3 -s
+
+ A popular sound of the sample is as follows:
+
+ play file.xxx phaser 0.89 0.85 1.0 0.24 2.0 -t
+
+ The sample sounds if ten springs are in your ears:
+
+ play file.xxx phaser 0.6 0.66 3.0 0.6 2.0 -t
+
+ Other effects ( copy, rate, avg, stat, vibro, lowp, highp,
+ band, reverb )
+
+ The other effects are simply to use. However, an "easy to
+ use manual" should be given here.
+
+ More effects ( to do ! )
+
+ There are a lot of effects around like noise gates, com-
+ pressors, waw-waw, stereo effects and so on. They should
+ be implemented making SOX to be more useful in sound mix-
+ ing technics coming together with a great varity of dif-
+ ferent sound effects.
+
+ Combining effects be using then in parallel or sequel on
+ different channels needs some easy mechanism which is
+ real-time stable.
+
+ Really missing, is the changing of the parameters, start-
+ ing and stoping of effects while playing samples in real-
+ time!
+
+ Good luck and have fun with all the effects!
+
+ Juergen Mueller (jmueller@uia.ua.ac.be)
+
+
+
+
+
+
+ December 10, 1999 8
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+SEE ALSO
+ sox(1), play(1), rec(1)
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ December 10, 1999 9
+
+
--- a/src/Makefile.in
+++ b/src/Makefile.in
@@ -72,8 +72,9 @@
$(AR) libst.a $(LIBOBJS)
$(RANLIB) libst.a
-man: sox.1 libst.3
+man: sox.1 soxexam.1 libst.3
nroff -man sox.1 | col -b > sox.txt
+ nroff -man soxexam.1 | col -b > soxexam.txt
nroff -man libst.3 | col -b > libst.txt
PLAY_INSTALL_0 =
@@ -82,6 +83,7 @@
install: sox $(PLAY_INSTALL_$(PLAY_SUPPORT))
$(INSTALL) -c -m 755 sox $(BINDIR)
$(INSTALL) -c -m 644 sox.1 $(MANDIR)/man1
+ $(INSTALL) -c -m 644 soxexam.1 $(MANDIR)/man1
install-play:
if [ -f $(BINDIR)/rec ] ; then $(RM) $(BINDIR)/rec; fi