Tribblix: manual page: soxformat.7

SoX(7) Sound eXchange SoX(7)

NAME

SoX - Sound eXchange, the Swiss Army knife of audio manipulation

DESCRIPTION

This manual describes SoX supported file formats and audio device
types; the SoX manual set starts with sox(1).

Format types that can SoX can determine by a filename extension are
listed with their names preceded by a dot. Format types that are
optionally built into SoX are marked `(optional)'.

Format types that can be handled by an external library via an
optional pseudo file type (currently sndfile) are marked e.g. `(also
with -t sndfile)'. This might be useful if you have a file that
doesn't work with SoX's default format readers and writers, and
there's an external reader or writer for that format.

To see if SoX has support for an optional format or device, enter sox
-h and look for its name under the list: `AUDIO FILE FORMATS' or
`AUDIO DEVICE DRIVERS'.

SOX FORMATS & DEVICE DRIVERS
.raw (also with -t sndfile), .f32, .f64, .s8, .s16, .s24, .s32,
.u8, .u16, .u24, .u32, .ul, .al, .lu, .la
Raw (headerless) audio files. For raw, the sample rate and
the data encoding must be given using command-line format
options; for the other listed types, the sample rate defaults
to 8kHz (but may be overridden), and the data encoding is
defined by the given suffix. Thus f32 and f64 indicate files
encoded as 32 and 64-bit (IEEE single and double precision)
floating point PCM respectively; s8, s16, s24, and s32
indicate 8, 16, 24, and 32-bit signed integer PCM
respectively; u8, u16, u24, and u32 indicate 8, 16, 24, and
32-bit unsigned integer PCM respectively; ul indicates
`<mu>-law' (8-bit), al indicates `A-law' (8-bit), and lu and
la are inverse bit order `<mu>-law' and inverse bit order `A-
law' respectively. For all raw formats, the number of
channels defaults to 1 (but may be overridden).

Headerless audio files on a SPARC computer are likely to be of
format ul; on a Mac, they're likely to be u8 but with a
sample rate of 11025 or 22050 Hz.

See .ima and .vox for raw ADPCM formats, and .cdda for raw CD
digital audio.

.f4, .f8, .s1, .s2, .s3, .s4,
.u1, .u2, .u3, .u4, .sb, .sw, .sl, .ub, .uw
Deprecated aliases for f32, f64, s8, s16, s24, s32,
u8, u16, u24, u32, s8, s16, s32, u8, and u16 respectively.

.8svx (also with -t sndfile)
Amiga 8SVX musical instrument description format.

.aiff, .aif (also with -t sndfile)
AIFF files as used on old Apple Macs, Apple IIc/IIgs and SGI.
SoX's AIFF support does not include multiple audio chunks, or
the 8SVX musical instrument description format. AIFF files
are multimedia archives and can have multiple audio and
picture chunks - you may need a separate archiver to work with
them. With Mac OS X, AIFF has been superseded by CAF.

.aiffc, .aifc (also with -t sndfile)
AIFF-C is a format based on AIFF that was created to allow
handling compressed audio. It can also handle little endian
uncompressed linear data that is often referred to as sowt
encoding. This encoding has also become the defacto format
produced by modern Macs as well as iTunes on any platform.
AIFF-C files produced by other applications typically have the
file extension .aif and require looking at its header to
detect the true format. The sowt encoding is the only
encoding that SoX can handle with this format.

AIFF-C is defined in DAVIC 1.4 Part 9 Annex B. This format is
referred from ARIB STD-B24, which is specified for Japanese
data broadcasting. Any private chunks are not supported.

alsa (optional)
Advanced Linux Sound Architecture device driver; supports both
playing and recording audio. ALSA is only used in Linux-based
operating systems, though these often support OSS (see below)
as well. Examples:
sox infile -t alsa
sox infile -t alsa default
sox infile -t alsa plughw:0,0
sox -b 16 -t alsa hw:1 outfile
See also play(1), rec(1), and sox(1) -d.

.amb Ambisonic B-Format: a specialisation of .wav with between 3
and 16 channels of audio for use with an Ambisonic decoder.
See http://www.ambisonia.com/Members/mleese/file-format-for-b-
format for details. It is up to the user to get the channels
together in the right order and at the correct amplitude.

.amr-nb (optional)
Adaptive Multi Rate - Narrow Band speech codec; a lossy format
used in 3rd generation mobile telephony and defined in 3GPP TS
26.071 et al.

AMR-NB audio has a fixed sampling rate of 8 kHz and supports
encoding to the following bit-rates (as selected by the -C
option): 0 = 4.75 kbit/s, 1 = 5.15 kbit/s, 2 = 5.9 kbit/s, 3 =
6.7 kbit/s, 4 = 7.4 kbit/s 5 = 7.95 kbit/s, 6 = 10.2 kbit/s, 7
= 12.2 kbit/s.

.amr-wb (optional)
Adaptive Multi Rate - Wide Band speech codec; a lossy format
used in 3rd generation mobile telephony and defined in 3GPP TS
26.171 et al.

AMR-WB audio has a fixed sampling rate of 16 kHz and supports
encoding to the following bit-rates (as selected by the -C
option): 0 = 6.6 kbit/s, 1 = 8.85 kbit/s, 2 = 12.65 kbit/s, 3
= 14.25 kbit/s, 4 = 15.85 kbit/s 5 = 18.25 kbit/s, 6 = 19.85
kbit/s, 7 = 23.05 kbit/s, 8 = 23.85 kbit/s.

ao (optional)
Xiph.org's Audio Output device driver; works only for playing
audio. It supports a wide range of devices and sound systems
- see its documentation for the full range. For the most
part, SoX's use of libao cannot be configured directly;
instead, libao configuration files must be used.

The filename specified is used to determine which libao plugin
to use. Normally, you should specify `default' as the
filename. If that doesn't give the desired behavior then you
can specify the short name for a given plugin (such as pulse
for pulse audio plugin). Examples:
sox infile -t ao
sox infile -t ao default
sox infile -t ao pulse
See also play(1) and sox(1) -d.

.au, .snd (also with -t sndfile)
Sun Microsystems AU files. There are many types of AU file;
DEC has invented its own with a different magic number and
byte order. To write a DEC file, use the -L option with the
output file options.

Some .au files are known to have invalid AU headers; these are
probably original Sun <mu>-law 8000 Hz files and can be dealt
with using the .ul format (see below).

It is possible to override AU file header information with the
-r and -c options, in which case SoX will issue a warning to
that effect.

.avr Audio Visual Research format; used by a number of commercial
packages on the Mac.

.caf (optional)
Apple's Core Audio File format.

.cdda, .cdr
`Red Book' Compact Disc Digital Audio (raw audio). CDDA has
two audio channels formatted as 16-bit signed integers (big
endian)at a sample rate of 44.1 kHz. The number of (stereo)
samples in each CDDA track is always a multiple of 588.

coreaudio (optional)
Mac OSX CoreAudio device driver: supports both playing and
recording audio. If a filename is not specific or if the name
is "default" then the default audio device is selected. Any
other name will be used to select a specific device. The
valid names can be seen in the System Preferences->Sound menu
and then under the Output and Input tabs.

Examples:
sox infile -t coreaudio
sox infile -t coreaudio default
sox infile -t coreaudio "Internal Speakers"
See also play(1), rec(1), and sox(1) -d.

.cvsd, .cvs
Continuously Variable Slope Delta modulation. A headerless
format used to compress speech audio for applications such as
voice mail. This format is sometimes used with bit-reversed
samples - the -X format option can be used to set the bit-
order.

.cvu Continuously Variable Slope Delta modulation (unfiltered).
This is an alternative handler for CVSD that is unfiltered but
can be used with any bit-rate. E.g.
sox infile outfile.cvu rate 28k
play -r 28k outfile.cvu sinc -3.4k

.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, and one line that contains the
number of channels. Subsequent lines contain two or more
numeric data intems: the time since the beginning of the first
sample and the sample value for each channel.

Values are normalized so that the maximum and minimum are 1
and -1. This file format can be used to create data files for
external programs such as FFT analysers or graph routines.
SoX can also convert a file in this format back into one of
the other file formats.

Example containing only 2 stereo samples of silence:

; Sample Rate 8012
; Channels 2
0 0 0
0.00012481278 0 0

.dvms, .vms
Used in Germany to compress speech audio for voice mail. A
self-describing variant of cvsd.

.fap (optional)
See .paf.

.flac (optional; also with -t sndfile)
Xiph.org's Free Lossless Audio CODEC compressed audio. FLAC
is an open, patent-free CODEC designed for compressing music.
It is similar to MP3 and Ogg Vorbis, but lossless, meaning
that audio is compressed in FLAC without any loss in quality.

SoX can read native FLAC files (.flac) but not Ogg FLAC files
(.ogg). [But see .ogg below for information relating to
support for Ogg Vorbis files.]

SoX can write native FLAC files according to a given or
default compression level. 8 is the default compression level
and gives the best (but slowest) compression; 0 gives the
least (but fastest) compression. The compression level is
selected using the -C option [see sox(1)] with a whole number
from 0 to 8.

.fssd An alias for the .u8 format.

.gsrt Grandstream ring-tone files. Whilst this file format can
contain A-Law, <mu>-law, GSM, G.722, G.723, G.726, G.728, or
iLBC encoded audio, SoX supports reading and writing only A-
Law and <mu>-law. E.g.
sox music.wav -t gsrt ring.bin
play ring.bin

.gsm (optional; also with -t sndfile)
GSM 06.10 Lossy Speech Compression. A lossy format for
compressing speech which is used in the Global Standard for
Mobile telecommunications (GSM). It's good for its purpose,
shrinking audio data size, but it will introduce lots of noise
when a given audio signal is encoded and decoded multiple
times. This format is used by some voice mail applications.
It is rather CPU intensive.

.hcom Macintosh HCOM files. These are Mac FSSD files with Huffman
compression.

.htk Single channel 16-bit PCM format used by HTK, a toolkit for
building Hidden Markov Model speech processing tools.

.ircam (also with -t sndfile)
Another name for .sf.

.ima (also with -t sndfile)
A headerless file of IMA ADPCM audio data. IMA ADPCM claims
16-bit precision packed into only 4 bits, but in fact sounds
no better than .vox.

.lpc, .lpc10
LPC-10 is a compression scheme for speech developed in the
United States. See http://www.arl.wustl.edu/~jaf/lpc/ for
details. There is no associated file format, so SoX's
implementation is headerless.

.mat, .mat4, .mat5 (optional)
Matlab 4.2/5.0 (respectively GNU Octave 2.0/2.1) format (.mat
is the same as .mat4).

.m3u A playlist format; contains a list of audio files. SoX can
read, but not write this file format. See [1] for details of
this format.

.maud An IFF-conforming audio 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, <mu>-law in mono and stereo.

.mp3, .mp2 (optional read, optional write)
MP3 compressed audio; MP3 (MPEG Layer 3) is a part of the
patent-encumbered MPEG standards for audio and video
compression. It is a lossy compression format that achieves
good compression rates with little quality loss.

Because MP3 is patented, SoX cannot be distributed with MP3
support without incurring the patent holder's fees. Users who
require SoX with MP3 support must currently compile and build
SoX with the MP3 libraries (LAME & MAD) from source code, or,
in some cases, obtain pre-built dynamically loadable
libraries.

When reading MP3 files, up to 28 bits of precision is stored
although only 16 bits is reported to user. This is to allow
default behavior of writing 16 bit output files. A user can
specify a higher precision for the output file to prevent
lossing this extra information. MP3 output files will use up
to 24 bits of precision while encoding.

MP3 compression parameters can be selected using SoX's -C
option as follows (note that the current syntax is subject to
change):

The primary parameter to the LAME encoder is the bit rate. If
the value of the -C value is a positive integer, it's taken as
the bitrate in kbps (e.g. if you specify 128, it uses 128
kbps).

The second most important parameter is probably "quality"
(really performance), which allows balancing encoding speed
vs. quality. In LAME, 0 specifies highest quality but is very
slow, while 9 selects poor quality, but is fast. (5 is the
default and 2 is recommended as a good trade-off for high
quality encodes.)

Because the -C value is a float, the fractional part is used
to select quality. 128.2 selects 128 kbps encoding with a
quality of 2. There is one problem with this approach. We need
128 to specify 128 kbps encoding with default quality, so 0
means use default. Instead of 0 you have to use .01 (or .99)
to specify the highest quality (128.01 or 128.99).

LAME uses bitrate to specify a constant bitrate, but higher
quality can be achieved using Variable Bit Rate (VBR). VBR
quality (really size) is selected using a number from 0 to 9.
Use a value of 0 for high quality, larger files, and 9 for
smaller files of lower quality. 4 is the default.

In order to squeeze the selection of VBR into the the -C value
float we use negative numbers to select VRR. -4.2 would select
default VBR encoding (size) with high quality (speed). One
special case is 0, which is a valid VBR encoding parameter but
not a valid bitrate. Compression value of 0 is always treated
as a high quality vbr, as a result both -0.2 and 0.2 are
treated as highest quality VBR (size) and high quality
(speed).

See also Ogg Vorbis for a similar format.

.nist (also with -t sndfile)
See .sph.

.ogg, .vorbis (optional)
Xiph.org's Ogg Vorbis compressed audio; an open, patent-free
CODEC designed for music and streaming audio. It is a lossy
compression format (similar to MP3, VQF & AAC) that achieves
good compression rates with a minimum amount of quality loss.

SoX can decode all types of Ogg Vorbis files, and can encode
at different compression levels/qualities given as a number
from -1 (highest compression/lowest quality) to 10 (lowest
compression, highest quality). By default the encoding
quality level is 3 (which gives an encoded rate of approx.
112kbps), but this can be changed using the -C option (see
above) with a number from -1 to 10; fractional numbers (e.g.
3.6) are also allowed. Decoding is somewhat CPU intensive and
encoding is very CPU intensive.

See also .mp3 for a similar format.

.opus (optional)
Xiph.org's Opus compressed audio; an open, lossy, low-latency
codec offering a wide range of compression rates. It uses the
Ogg container.

SoX can only read Opus files, not write them.

oss (optional)
Open Sound System /dev/dsp device driver; supports both
playing and recording audio. OSS support is available in
Unix-like operating systems, sometimes together with
alternative sound systems (such as ALSA). Examples:
sox infile -t oss
sox infile -t oss /dev/dsp
sox -b 16 -t oss /dev/dsp outfile
See also play(1), rec(1), and sox(1) -d.

.paf, .fap (optional)
Ensoniq PARIS file format (big and little-endian
respectively).

.pls A playlist format; contains a list of audio files. SoX can
read, but not write this file format. See [2] for details of
this format.

Note: SoX support for SHOUTcast PLS relies on wget(1) and is
only partially supported: it's necessary to specify the audio
type manually, e.g.
play -t mp3 "http://a.server/pls?rn=265&file=filename.pls"
and SoX does not know about alternative servers - hit Ctrl-C
twice in quick succession to quit.

.prc Psion Record. Used in Psion EPOC PDAs (Series 5, Revo and
similar) for System alarms and recordings made by the built-in
Record application. When writing, SoX defaults to A-law,
which is recommended; if you must use ADPCM, then use the -e
ima-adpcm switch. The sound quality is poor because Psion
Record seems to insist on frames of 800 samples or fewer, so
that the ADPCM CODEC has to be reset at every 800 frames,
which causes the sound to glitch every tenth of a second.

pulseaudio (optional)
PulseAudio driver; supports both playing and recording of
audio. PulseAudio is a cross platform networked sound server.
If a file name is specified with this driver, it is ignored.
Examples:
sox infile -t pulseaudio
sox infile -t pulseaudio default
See also play(1), rec(1), and sox(1) -d.

.pvf (optional)
Portable Voice Format.

.sd2 (optional)
Sound Designer 2 format.

.sds (optional)
MIDI Sample Dump Standard.

.sf (also with -t sndfile)
IRCAM SDIF (Institut de Recherche et Coordination
Acoustique/Musique Sound Description Interchange Format). Used
by academic music software such as the CSound package, and the
MixView sound sample editor.

.sln Asterisk PBX `signed linear' 8khz, 16-bit signed integer,
little-endian raw format.

.sph, .nist (also with -t sndfile)
SPHERE (SPeech HEader Resources) is a file format defined by
NIST (National Institute of Standards and Technology) and is
used with speech audio. SoX can read these files when they
contain <mu>-law and PCM data. It will ignore any header
information that says the data is compressed using shorten
compression and will treat the data as either <mu>-law or PCM.
This will allow SoX and the command line shorten program to be
run together using pipes to encompasses the data and then pass
the result to SoX for processing.

.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.

.snd See .au, .sndr and .sndt.

sndfile (optional)
This is a pseudo-type that forces libsndfile to be used. For
writing files, the actual file type is then taken from the
output file name; for reading them, it is deduced from the
file.

sndio (optional)
OpenBSD audio device driver; supports both playing and
recording audio.
sox infile -t sndio
See also play(1), rec(1), and sox(1) -d.

.sndr Sounder files. An MS-DOS/Windows format from the early '90s.
Sounder files usually have the extension `.SND'.

.sndt SoundTool files. An MS-DOS/Windows format from the early
'90s. SoundTool files usually have the extension `.SND'.

.sou An alias for the .u8 raw format.

.sox SoX's native uncompressed PCM format, intended for storing (or
piping) audio at intermediate processing points (i.e. between
SoX invocations). It has much in common with the popular WAV,
AIFF, and AU uncompressed PCM formats, but has the following
specific characteristics: the PCM samples are always stored as
32 bit signed integers, the samples are stored (by default) as
`native endian', and the number of samples in the file is
recorded as a 64-bit integer. Comments are also supported.

See `Special Filenames' in sox(1) for examples of using the
.sox format with `pipes'.

sunau (optional)
Sun /dev/audio device driver; supports both playing and
recording audio. For example:
sox infile -t sunau /dev/audio
or
sox infile -t sunau -e mu-law -c 1 /dev/audio
for older sun equipment.

See also play(1), rec(1), and sox(1) -d.

.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 33 kHz if the
sample rate is still unknown.

.vms See .dvms.

.voc (also with -t sndfile)
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. SoX supports reading (but not
writing) VOC files with multiple blocks, and files containing
<mu>-law, A-law, and 2/3/4-bit ADPCM samples.

.vorbis
See .ogg.

.vox (also with -t sndfile)
A headerless file of Dialogic/OKI ADPCM audio data commonly
comes with the extension .vox. This ADPCM data has 12-bit
precision packed into only 4-bits.

Note: some early Dialogic hardware does not always reset the
ADPCM encoder at the start of each vox file. This can result
in clipping and/or DC offset problems when it comes to
decoding the audio. Whilst little can be done about the
clipping, a DC offset can be removed by passing the decoded
audio through a high-pass filter, e.g.:
sox input.vox output.wav highpass 10

.w64 (optional)
Sonic Foundry's 64-bit RIFF/WAV format.

.wav (also with -t sndfile)
Microsoft .WAV RIFF files. This is the native audio file
format of Windows, and widely used for uncompressed audio.

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.

SoX can read and write linear PCM, floating point, <mu>-law,
A-law, MS ADPCM, and IMA (or DVI) ADPCM encoded samples. WAV
files can also contain audio encoded in many other ways (not
currently supported with SoX) e.g. MP3; in some cases such a
file can still be read by SoX by overriding the file type,
e.g.
play -t mp3 mp3-encoded.wav
Big endian versions of RIFF files, called RIFX, are also
supported. To write a RIFX file, use the -B option with the
output file options.

waveaudio (optional)
MS-Windows native audio device driver. Examples:
sox infile -t waveaudio
sox infile -t waveaudio default
sox infile -t waveaudio 1
sox infile -t waveaudio "High Definition Audio Device ("
If the device name is omitted, -1, or default, then you get
the `Microsoft Wave Mapper' device. Wave Mapper means `use
the system default audio devices'. You can control what
`default' means via the OS Control Panel.

If the device name given is some other number, you get that
audio device by index; so recording with device name 0 would
get the first input device (perhaps the microphone), 1 would
get the second (perhaps line in), etc. Playback using 0 will
get the first output device (usually the only audio device).

If the device name given is something other than a number, SoX
tries to match it (maximum 31 characters) against the names of
the available devices.

See also play(1), rec(1), and sox(1) -d.

.wavpcm
A non-standard, but widely used, variant of .wav. Some
applications cannot read a standard WAV file header for PCM-
encoded data with sample-size greater than 16-bits or with
more than two channels, but can read a non-standard WAV
header. It is likely that such applications will eventually
be updated to support the standard header, but in the mean
time, this SoX format can be used to create files with the
non-standard header that should work with these applications.
(Note that SoX will automatically detect and read WAV files
with the non-standard header.)

The most common use of this file-type is likely to be along
the following lines:
sox infile.any -t wavpcm -e signed-integer outfile.wav

.wv (optional)
WavPack lossless audio compression. Note that, when
converting .wav to this format and back again, the RIFF header
is not necessarily preserved losslessly (though the audio is).

.wve (also with -t sndfile)
Psion 8-bit A-law. Used on Psion SIBO PDAs (Series 3 and
similar). This format is deprecated in SoX, but will continue
to be used in libsndfile.

.xa Maxis XA files. These are 16-bit ADPCM audio files used by
Maxis games. Writing .xa files is currently not supported,
although adding write support should not be very difficult.

.xi (optional)
Fasttracker 2 Extended Instrument format.

LICENSE

AUTHORS

Chris Bagwell (cbagwell@users.sourceforge.net). Other authors and
contributors are listed in the ChangeLog file that is distributed
with the source code.

soxformat December 31, 2014 SoX(7)

NAME

DESCRIPTION

SEE ALSO

References

LICENSE

AUTHORS