-
- 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 With Mac OS X, AIFF has been superseded by CAF.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 μ-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,
μ-law, GSM, G.722, G.723, G.726, G.728, or iLBC encoded audio, SoX
supports reading and writing only A-Law and μ-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, μ-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 μ-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 μ-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 μ-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,
μ-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.