Provided by: sox_14.3.2-3_i386 bug


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


       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 or ffmpeg) 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

       .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 `μ-law' (8-bit), al indicates `A-law'
              (8-bit), and lu and la are inverse bit order `μ-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 -2 -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
              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)
    '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.

                   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.  Subsequent lines contain two  numeric
              data items: the time since the beginning of the first sample and
              the sample value.  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.

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

       .fap (optional)
              See .paf.

       ffmpeg (optional)
              This  is a pseudo-type that forces ffmpeg to be used. The actual
              file type is deduced from the file name (it cannot  be  used  on
              stdio).   It  can  read  a wide range of audio files, not all of
              which are documented here, and also  the  audio  track  of  many
              video  files  (including AVI, WMV and MPEG). At present only the
              first audio track of a file can be read.

       .flac (optional; also with -t sndfile)
    '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

       .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  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.

       .mp4, .m4a (optional)
              MP4  compressed  audio.   MP3  (MPEG  4)  is  part  of  the MPEG
              standards for audio and video compression.   See  mp3  for  more

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

       .ogg, .vorbis (optional)
    '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

              See also .mp3 for a similar format.

       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 -2 -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 -i 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.
                   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.

       .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

       .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
                   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
                   sox infile -t sunau -U -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.

              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, μ-law, A-law, MS  ADPCM,  and
              IMA (or DVI) ADPCM.  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.

              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

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


       sox(1), soxi(1), libsox(3), octave(1), wget(1)

       The SoX web page at
       SoX scripting examples at

       [1]    Wikipedia, M3U,

       [2]    Wikipedia, PLS,


       Copyright 1998-2011 Chris Bagwell and SoX Contributors.
       Copyright 1991 Lance Norskog and Sundry Contributors.


       Chris  Bagwell  (   Other  authors  and
       contributors are listed in the ChangeLog file that is distributed  with
       the source code.