SoX(1)

NAME
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 ... ] ]

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 ]

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 ]
    copy
    cut
    deemph
    echo gain-in gain-out delay decay [ delay decay ...]
    echos gain-in gain-out delay decay [ delay decay ...]
    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 ]

DESCRIPTION
Sox translates sound files from one format to another, possibly doing a sound effect.

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 amplitude by 1/2 and changes the sampling rate from 8000 hertz to 12000 hertz via the rate sound effect loop.

File type options:

-t filetype
    gives the type of the sound sample file.

-r rate
    Give sample rate in Hertz of file.

-s/-u/-U/-A/-a/-g
    The sample data is signed linear (2's complement), unsigned linear, U-law (logarithmic), A-law (logarithmic), 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 countries and its gaining popularity because of its quality.

-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.

-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.

General options:

-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.

-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 then the input file. The order that the effects are run in will be arranged for maximum speed and not quality.

-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.

-V  Print a description of processing phases. Useful for figuring out exactly how sox is mangling your sound samples.

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.

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.

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), 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.

.8svx
    Amiga 8SVX musical instrument description format.

.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.

.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).

.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 format, so it needs its own handler.

.cvs
    Continuously Variable Slope Delta modulation
    Used to compress speech audio for applications such as voice mail.

.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. 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 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 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 compression. 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.

.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 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 handle. Example: -t ossdsp -w -s /dev/dsp

.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 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.

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.

.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.

.vms
    More info to come.
    Used to compress speech audio for applications such as voice mail.

.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 appropriately. On output, silence is not detected, nor are impossible sample rates.

.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 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.

.wve
    Psion 8-bit alaw
    These are 8-bit a-law 8khz sound files used on the Psion palmtop portable computer.

.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
    These are several suffices which serve as a shorthand 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.) 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 message. 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.

avg [ -l | -r ]
    Reduce the number of channels by averaging the samples, or duplicate channels to increase the number of channels. Valid combinations are 1 - 2, 1 - 4, 2 - 4, 4 - 2, 4 - 1, 2 - 1. The -l or -r option is not really averaging but either duplicates or leaves just the left or right channel, depending on if your increasing or decreasing the number of output 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 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 instrumental music. The -n (for noise) option uses the alternate mode for un-pitched signals. Band introduces noise in the shape of the filter, i.e. peaking at the center frequency and settling around it.

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 milliseconds 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.

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 rectified. 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.

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.

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 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 frequency in the middle of the drop. The slope of the filter is quite gentle.

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.

map Display a list of loops in a sample, and miscellaneous 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 milliseconds and the decay (relative to gain-in) with a modulation speed in Hz. The modulation is either sinodial (-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 sample, or one of four channels in a quadrophonic sample.

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 samples. 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 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 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 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 ues, you will want to read a detailed analysis of all of them at http://eakaw2.et.tu-dresden.de/~andreas/resample/resample.html

resample [ 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. 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 therefor should be something between 0. and 1., in practice 0.8-0.95. beta trades stop band rejection against transition width from passband to stop band. Larger beta means a slower transition and greater stopband rejection. beta should be at least greater than 2. The default is rollof 0.8, beta 17.5, which is rather conservative with respect to aliasing. Lower beta and higher rolloff values preserve more high frequency signal energy, but introduce measurable artifacts. This is the default effect if the two files have different sampling rates.

reverb gain-out delay [ delay ... ]
    Add reverbation 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 reverbation. Gain-out is the volume of the output.

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.

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 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.

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 different 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.

Please report any bugs found in this version of sox to Chris Bagwell (cbagwell@sprynet.com)

FILES

SEE ALSO
play(1), rec(1)

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