SOXFORMAT_NG(7)                Sound eXchange_ng               SOXFORMAT_NG(7)

NAME
       soxformat_ng - Audio file formats supported by sox_ng and libsox_ng

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

       Format  types that 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  are  handled  by  the external library sndfile are
       marked `(with sndfile)' and format types that can only  be  read  using
       the external program ffmpeg are marked `(with ffmpeg)'

       Formats  for which SoX has internal drivers but that are also supported
       by sndfile or ffmpeg are marked (also with -t sndfile)  or  (also  with
       -t ffmpeg).   This might be useful if you have a file that doesn't work
       with SoX's built-in readers and writers.

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

       To know everything about a format - whether it reads it or writes it or
       both and what encodings it can write - enter something like --help-for-
       mat wav or, to see details for all formats, --help-format all.

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 and
              the data encoding is defined by the given suffix.  Thus f32  and
              f64  indicate files encoded as 32 and 64-bit IEEE-754 single and
              double precision floating point PCM respectively; s8,  s16,  s24
              and s32 indicate 8, 16, 24 and 32-bit signed integer PCM respec-
              tively;  u8,  u16, u24 and u32 indicate 8, 16, 24 and 32-bit un-
              signed  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 `A-law' respectively.  For all raw for-
              mats, the number of channels defaults to 1.

              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 22050Hz.

              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.

       .3gp, .3gpp (with ffmpeg)
              Third Generation Partnership Project format.

       .3g2, .3gp2, .3gpp2 (with ffmpeg)
              Third Generation Partnership Project 2 format.

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

       .aa (with ffmpeg)
              Audible Audiobook format.

              SoX can autodetect this format but, if it's coming from a  pipe,
              ffmpeg fails to decode it.

       .aac (with ffmpeg)
              Advanced Audio Coding format.

       .ac3 (with ffmpeg)
              Audio Codec 3 (Dolby Digital) format.

       .act (with ffmpeg)
              G729a  compressed speech recorded by some Chinese-made MP3 play-
              ers.

       .adts (with ffmpeg)
              Audio Data Transport Stream format.

       .aiff, .aif (also with -t sndfile or -t ffmpeg)
              AIFF files as used on old Apple Macs, Apple  IIc/IIgs  and  SGI.
              SoX's  AIFF  support  does not include multiple audio chunks nor
              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.   On
              MacOS X, AIFF has been superseded by CAF.

       .aiffc, .aifc (also with -t sndfile)
              AIFF-C  is  based on AIFF but also handles compressed audio.  It
              can also handle little-endian uncompressed linear data  that  is
              often  referred  to  as sowt encoding.  This encoding has become
              the defacto format produced by modern Macs as well as iTunes  on
              any platform.  AIFF-C files produced by other applications typi-
              cally  have  the  file extension .aif and require looking at its
              header to detect the true format.  sowt, a-law and u-law are the
              only encodings that SoX can read and write natively;  for  other
              compression types like GSM try -t ffmpeg.

              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.  Private chunks are not supported.

       alsa (optional)
              The 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 be-
              low) as well.  Examples:

                   sox_ng infile -t alsa
                   sox_ng infile -t alsa default
                   sox_ng infile -t alsa plughw:0,0
                   sox_ng -b 16 -t alsa hw:1 outfile


       .amb   Ambisonic B-Format is a specialization 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 you to get the channels together in
              the right order and at the correct amplitude.

       .amr-nb, .amr-wb (both optional, also with -t ffmpeg)
              Adaptive Multi Rate Narrow and Wide Band are lossy  formats  for
              speech  used  in  3rd generation mobile telephony and defined in
              3GPP TS 26.071 and TS 26.171

              AMR-NB audio has a fixed sampling rate of  8kHz  and  AMR-WB  of
              16kHz  and they support encoding to the following bit rates, se-
              lected by the -C option:
                             amr-nb                     amr-wb
                           -C     kbit/s              -C     kbit/s
                           0       4.75               0       6.6
                           1       5.15               1       8.85
                           2       5.9                2      12.65
                           3       6.7                3      14.25
                           4       7.4                4      15.85
                           5       7.95               5      18.25
                           6      10.2                6      19.85
                           7      12.2                7      23.05
                                                      8      23.85

       ao (optional)
              Xiph.org's Audio Output device driver only works for playing au-
              dio.  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 con-
              figuration files must be used.

              The filename is used to determine which libao plugin to use  and
              normally,  you  should  specify `default'.  If that doesn't give
              the desired behavior, you can specify the short name for a given
              plugin (such as pulse for the  PulseAudio  plugin  or  null  for
              testing. See http://xiph.org/ao).  Examples:

                   sox_ng infile -t ao
                   sox_ng infile -t ao default
                   sox_ng infile -t ao pulse


       .ape (with ffmpeg)
              Monkey's Audio format.

       .apm (with ffmpeg)
              Ubisoft Rayman 2 APM format.

       .aptx (with ffmpeg)
              Audio Processing Technology for Bluetooth format.

              SoX  can only autodetect this type of file from its filename ex-
              tension; if it is read from `standard input' (stdin) or  from  a
              file whose name does not end in `.aptx', you will need to prefix
              it with `-t ffmpeg'.

       .argo_asf (with ffmpeg)
              Argonaut Games ASF format.

       .asf (with ffmpeg)
              Advanced / Active Streaming Format.

       .ast (with ffmpeg)
              AST Audio Stream format.

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

              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.

              It  is  possible to override AU file header information with the
              -r (sampling rate) and -c (number of channels) options, in which
              case SoX will issue a warning about the mismatch.

       .avi (with ffmpeg)
              Audio Video Interleaved format.

       .avr (also with -t ffmpeg or -t sndfile)
              Audio Visual Research format, used by  a  number  of  commercial
              packages on the Mac.

       .caf (with sndfile, also with -t ffmpeg)
              Apple's Core Audio File format.

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

              By  default, SoX expects and produces CDDA in big-endian format.
              However, the endianess of CDDA files can vary depending  on  the
              tools  and  hardware used.  To produce and to read little-endian
              CDDA files, predece the filename with the -L option.

       coreaudio (optional)
              The MacOS X CoreAudio device driver supports  both  playing  and
              recording.   If a filename is not specified or is "default", the
              default audio device is selected.  Any other name will  be  used
              to select a specific device.

              Examples:

                  sox_ng infile -t coreaudio
                  sox_ng infile -t coreaudio "Internal Speakers"

              You  can  also  select a device by its number; to see the device
              numbers:

                  sox_ng -V -n -t coreaudio nonexistent


       .cvsd, .cvs
              Continuously Variable Slope Delta  modulation  is  a  headerless
              format  used  to  compress speech audio for applications such as
              voice mail with a fixed bit rate of 8kHz.  This format is  some-
              times  used with bit-reversed samples; the -X option can be used
              to set the bit order.

       .cvu   Unfiltered Continuously Variable Slope Delta  modulation  is  an
              alternative  handler for CVSD that is unfiltered but can be used
              with any sampling rate. As it is a headerless format,  you  have
              to  specify  the  sampling  rate with -r if it is different from
              8kHz.

                   sox_ng infile outfile.cvu rate 28k
                   play -r 28k outfile.cvu sinc -3.4k


       .dat   Text Data files contain a textual representation of sample data.
              There is one line at the beginning that contains the sample rate
              and one that contains the number of channels.  Subsequent  lines
              contain  two  or more numeric data items: the time since the be-
              ginning of the first sample and the sample value for each  chan-
              nel.

              Values  are  normalized so the maximum and minimum are 1 and -1.
              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  for-
              mats.

              Example containing only 2 stereo samples of silence:


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


       .dfpwm (with ffmpeg)
              Dynamic Filter Pulse Width Modulation format 1a.

              SoX  can only autodetect this type of file from its filename ex-
              tension; if it is read from `standard input' (stdin) or  from  a
              file  whose name does not end in `.dfpwm', you will need to pre-
              fix it with `-t ffmpeg'.

       .dts (with ffmpeg)
              Digital Theatre Systems format.

              SoX can only autodetect this type of file from its filename  ex-
              tension;  if  it is read from `standard input' (stdin) or from a
              file whose name does not end in `.dts', you will need to  prefix
              it with `-t ffmpeg'.

       .dff   Direct Stream Digital Interchange File Format (DSDIFF) is a for-
              mat  defined by Philips for storing 1-bit DSD data, used in SACD
              mastering and occasionally for  online  distribution.   SoX  can
              write  single-, double- and quad-sample rates of 64, 128 and 256
              times 44100Hz (2822400, 5644800 and 11289600Hz), known as DSD64,
              DSD128 and DSD256, but not the octuple- and sexdecuple- variants
              (DSD512 and DSD1024).

       .dsf, .wsd
              DSD Stream File and Wideband Single-bit Data are formats defined
              by Sony and by Sharp and Pioneer for  storing  1-bit  DSD  data,
              commonly  used for online distribution of audiophile recordings,
              supporting the same sample rates as dff.  The wsd variant can be
              read but not written.

       .dvms, .vms
              The Digital Voice Messaging System format is used in Germany  to
              compress  speech  audio for voice mail.  It is a self-describing
              variant of cvsd.

       .ea (with ffmpeg)
              Electronic Arts Multimedia format.

       .eac3 (with ffmpeg)
              Enhanced AC-3 Audio.

       .f4v (with ffmpeg)
              Another name for .mov.

       .fap (with sndfile)
              See .paf.

       ffmpeg (optional)
              This is a pseudo-type that uses the external program  ffmpeg  if
              it  is  installed.  It  can only read files, not write them, and
              will extract the sound track from many video file formats.  ffm-
              peg deduces the actual file type from the file's contents with a
              far more advanced algorithm than that used by  SoX,  which  only
              recognizes up to two fixed byte sequences at fixed offsets.

       .flac (optional; also with -t sndfile or -t ffmpeg)
              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, so the audio is
              compressed without any loss in quality.

              SoX can read native FLAC files (.flac) but  can  only  read  Ogg
              FLAC files (.oga) if ffmpeg is installed.

              See  .ogg below for information relating to support for Ogg Vor-
              bis 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_ng(1)) with a whole number from 0 to 8.

       .flv (with ffmpeg)
              Macromedia Flash Video format.

       .fssd  Flexible Sound Studio Data format, a raw format that defaults to
              .u8 at 8kHz.

       .gsrt  Grandstream  ring-tone  files.  Whilst this file format can con-
              tain 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_ng music.wav -t gsrt ring.bin
                 play ring.bin


       .gsm (optional; also with -t sndfile or -t ffmpeg)
              GSM 06.10 Lossy Speech Compression.  A  lossy  format  for  com-
              pressing speech, used in the Global Standard for Mobile telecom-
              munications  (GSM)  and  in some voice mail applications, with a
              fixed sample rate of 8kHz.

              Although SoX can create gsm files with up to  sixteen  channels,
              it  is a headerless format so input is assumed to be mono unless
              it is told otherwise.

       .gxf (with ffmpeg)
              General eXchange Format.

       .hcom (also with -t ffmpeg)
              Macintosh HCOM files.  These are Mac  FSSD  files  with  Huffman
              compression.

       .hip   The  -t hip  format  forces decoding of MP3 to use mp3lame's de-
              coder based on mpglib by Michael Hipp.

       .htk (also with -t sndfile)
              Single channel 16-bit PCM format used  by  HTK,  a  toolkit  for
              building Hidden Markov Model speech processing tools.

       .ircam (also with -t sndfile or -t ffmpeg)
              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.

       .ism (with ffmpeg)
              ISM streaming video format.

       .kvag (with ffmpeg)
              Simon & Schuster Interactive VAG format.

       .lpc, .lpc10
              LPC-10  is  a  compression  scheme  for  speech developed by the
              United     States     Department      of      Defense.       See
              https://github.com/jafingerhut/lpc10  for  details.  There is no
              associated file format, so SoX's implementation is headerless.

       .m4a (with ffmpeg)
              MPEG-4 Audio format.

       .m4b (with ffmpeg)
              Another name for .mov.

       .m4v, .mp4 (with ffmpeg)
              MPEG-4 Video format.

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

       .m3u   A  playlist  format,  containing 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 and published along with the `Toccata' sound  card
              on  the  Amiga  allows  8bit  linear,  16bit  linear,  A-Law and
              <mu>-law in mono and stereo.

       .mj2 (with ffmpeg)
              Another name for .mov.

       .mkv, .webm (with ffmpeg)
              Matroska video format.

       .mlp (with ffmpeg)
              Meridian Lossless Packing format.

       .mod   Amiga Module format, typically created with Protracker, decoding
              only.

       .mov (with ffmpeg)
              MPEG-1 Systems / MPEG program stream format.

       .mp1, .m1a, .mp2, .mp3 (optional, also with -t sndfile, -t ffmpeg or
       -t hip)
              MP1, MP2 and MP3 compressed audio (MPEG 1 Layers 1, 2 and 3) are
              part of the MPEG standards for audio and video compression whose
              patents have expired.  They are lossy compression  formats  that
              achieve good compression rates with little quality loss.

              Decoding

              libmad,  which  SoX  uses to decode MPEG files, does not work on
              files with a bit-rate higher than 192K but you  can  read  those
              using:

              -t sndfile
                     which  uses libmpg123, generally gives better quality re-
                     sults, is better at decoding damaged or corrupt files and
                     is twice as fast as libmad if you have an FPU;

              -t ffmpeg
                     which uses yet another decoder, the one in libavcodec;

              -t hip which uses LAME's internal libmpg but the LAME program is
                     able to decode more MPEG files than -t hip as it  decodes
                     the  file  header  itself before calling the hip decoder,
                     whereas SoX makes the decoder do all the work.

              Yet more ways to decode MPEG files are input filenames like:

                 '|madplay -q --no-tty-control -o wav:- in.mp3'
                 '|mpg123 -q --no-control --wav - in.mp3'
                 '|lame --quiet --decode in.mp3 -'
                 '|ffmpeg -loglevel quiet -nostdin -i in.mp3 -'

              As these piped commands cannot rewind their output pipe  to  fix
              the  file  length in the header, you may prefer to add --ignore-
              length before them, which makes SoX warn "premature EOF on input
              file" but keeps reading to the end.

              The decoding speeds of the default libmad, -t hip and -t sndfile
              are in proportions 1:2:2.5 with an  FPU.   Without  one,  libmad
              will be the fastest instead of the slowest.

              With  mad or sndfile, SoX decodes MPEG files with a precision of
              28 bits but it declares a precision of 16 bits so that  decoding
              MP3s  to  WAVs gives the CD quality that most people expect. You
              can specify a higher precision before  the  output  filename  to
              keep this extra information.

              Encoding

              SoX can only read MP1 files, not write them.

              SoX  uses twolame to write MP2 files, and the target bit rate is
              set using the -C option in kbps: 32, 48, 56, 64,  80,  96,  112,
              128, 160 or 192 for a constant bit rate (CBR) for mono files and
              double  those for stereo ones.  Variable bit rate encoding (VBR)
              is selected by values from -50 to 50 (of which only  -10  to  10
              are  said  to be useful); if you really want VBR at qualities 32
              or 48, use something like 32.01 or 48.01.

              Unfortunately, twolame outputs 0.005 seconds of quiet garbage at
              the start and truncates the end by 0.002 seconds.

              It can also encode MP2s with -t sndfile, which also uses twolame
              and has the same defect.

              SoX uses libmp3lame to write MP3 files and can use up to 25 bits
              of precision but, at present, SoX's use of libmp3lame adds  0.01
              seconds  of  quiet garbage at the start and at the end; encoding
              with -t sndfile gets the length right and no garbage.

              MP3 compression parameters can be selected using SoX's -C option
              as follows:

              The primary parameter to the LAME MP3 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 "quality" 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 or lowest quality (128.01 or 128.99).

              LAME uses bitrate to specify a constant bitrate but higher qual-
              ity 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
              we use negative numbers to select VBR. -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 high-
              est quality VBR (size) and high quality (speed).

              See Ogg Vorbis and opus for similar formats that achieve  higher
              signal quality with less bandwidth.

       .mp4 (with ffmpeg)
              MPEG-4 video format.

       .mpeg, .mpg (with ffmpeg)
              MPEG-1 Systems / MPEG program stream format.

       .mpegts (with ffmpeg)
              MPEG-TS (MPEG-2 Transport Stream) format.

       .mul   Tomb Raider Multiplex Stream (write-only support).

       .mxf, .mxf_opatom (with ffmpeg)
              Material eXchange Format Operational Pattern OP1A "OP-Atom" for-
              mat (SMPTE 390M).

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

       .nsp (also with -t ffmpeg)
              SoX  can read Computerized Speech Lab NSP files that may contain
              both audio and bioelectric data.  Typically, the  first  channel
              is  sound pressure (audio) and additional channels are data such
              as laryngeal kinematic  or  aerodynamic  (air  pressure  or  air
              flow).

              The  NSP  file  format  was  also used for the Phonetic Database
              (PDB) from Speech Technology Research who had a free NSP Player,
              SpeakNSP.  CSL NSP file reading and writing is supported by  the
              WaveSurfer package.

       .nut (with ffmpeg)
              NUT  is  a low overhead generic container format that stores au-
              dio, video, subtitle and user-defined streams in  a  simple  yet
              efficient way.

       .oga (with ffmpeg)
              Various Xiph.org audio formats in an Ogg container.

       .ogg, .vorbis (optional, also with -t sndfile or -t ffmpeg))
              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 and AAC) that achieves good
              compression rates with a minimal 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 with a number from -1 to 10; frac-
              tional numbers (e.g.  3.6) are also allowed.  Decoding is  some-
              what CPU intensive and encoding is very CPU intensive.

              See .mp3 for a similar format.

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

              SoX can only read Opus files, not write them.

       oss (optional)
              The Open Sound System /dev/dsp device driver supports both play-
              ing  and recording audio.  OSS support is available in Unix-like
              operating systems, sometimes  together  with  alternative  sound
              systems (such as ALSA).  Examples:

                   sox_ng infile -t oss
                   sox_ng infile -t oss /dev/dsp
                   sox_ng -b 16 -t oss /dev/dsp outfile


       .paf, .fap (with sndfile, also with -t ffmpeg)
              Ensoniq PARIS file format (big and little-endian respectively).

       .pls   A  playlist  format  containing  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.

       .pcm   Audio track format for the MSU-1 (Media Streaming Unit, revision
              1), an enhancement chip specification for the Super Nintendo En-
              tertainment System (SNES) / Super Famicom (SFC), designed by the
              late Byuu (aka Near).  Audio files are 44.1 kHz,  16-bit  stereo
              uncompressed  signed  PCM  in little-endian order ("Red Book" CD
              audio is big-endian), left channel first.  They  begin  with  an
              8-byte  header: the first four bytes are "MSU1" in ASCII and the
              next four are a 32-bit unsigned integer  that  define  the  loop
              point in samples.

       .prc   Psion  Record  are  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,  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 is a  cross-platform  networked  sound  server.   The
              PulseAudio driver supports both playing and recording of audio.

              If  a  file  name  is given to this driver, it selects the pulse
              server, the audio source or sink device and/or  the  application
              name  under which SoX registers itself with PulseAudio.  app:foo
              changes the application name from the default of  SoX,  anything
              else  with a colon on it is a server name and anything without a
              colon is the device name.  To specify more than one of these, or
              multiple servers to try in sequence, use a single filename  with
              the  entries  separated by spaces; as application names can con-
              tain spaces, an "app:" field must come last.  For example:

              sox_ng in.au -t pulseaudio alsa_output.pci-0000_01_00.1.hdmi-stereo


              Where pactl list sinks short gives a list of playing devices and
              pactl list sources short a list of recording devices.

              You can also set the PulseAudio device with SoX's  AUDIODEV  and
              PulseAudio's  PULSE_SOURCE  and PULSE_SINK environment variables
              (AUDIODEV overrides PULSE_* and a filename overrides  both)  and
              the  server  with PULSE_SERVER. For details of the server name's
              syntax, search for "pulseaudio format of server strings".

       .pvf (with sndfile)
              Portable Voice Format.

       .ra (with ffmpeg)
              RealAudio format.

       raw    Headerless audio data. See the first entry in this list for  de-
              tails.

       .rm (with ffmpeg)
              RealMedia format.

       .rso (with ffmpeg)
              Lego Mindstorms RSO format.

              SoX  can only autodetect this type of file from its filename ex-
              tension; if it is read from `standard input' (stdin) or  from  a
              file  whose name does not end in `.rso', you will need to prefix
              it with `-t ffmpeg'.

       .sbc (with ffmpeg)
              Bluetooth SIG low-complexity subband audio format.

              SoX can only autodetect this type of file from its filename  ex-
              tension;  if  it is read from `standard input' (stdin) or from a
              file whose name does not end in `.sbc', you will need to  prefix
              it with `-t ffmpeg'.

       .sd2 (with sndfile)
              Sound Designer 2 format.

       .sds (with sndfile, also with -t ffmpeg)
              MIDI Sample Dump Standard.

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

       .sln (also with -t ffmpeg)
              Asterisk PBX `signed linear' 8khz, 16-bit signed  integer,  lit-
              tle-endian raw format.

       .smjpeg (with ffmpeg)
              Loki SDL MJPEG.

       .smp   SMP  files  are  for use with the PC-DOS package SampleVision by
              Turtle Beach Softworks, which  communicates  with  several  MIDI
              samplers.   All  sample  rates  are supported by the package al-
              though not all are supported by the samplers  themselves.   Loop
              points are currently ignored.

       .snd   Several file formats use the .snd extension.

              The  main one was by NeXT, essentially the same as Sun Microsys-
              tems' .au format. See .au

              Apple made another .snd format in which the first two bytes  are
              a  16-bit  integer representing the numbers 1 or 2 but which can
              often be read as a raw format.

              Akai had an audio file format for its MPC range of  samplers  of
              which  the  first  byte contains the number 1 and the second the
              number 4. See .mpc2k

              There are also Sounder and SoundTool files  from  MS-DOS/Windows
              in the early '90s.  See .sndr and .sndt.

              Lastly,  the  HOM-BOT  Robot Vacuum Cleaner and the V.Flash Home
              Entertainment System use .snd audio files which are raw  single-
              channel  16-bit  16kHz PCM and the Unity Game Engine uses a com-
              pressed format called .snd.

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

       sndio (optional)
              The OpenBSD  audio  device  driver  supports  both  playing  and
              recording audio.

       .sndr  Sounder  files  are an MS-DOS/Windows format from the early '90s
              that usually have the extension `.snd'.

       .sndt  SoundTool files are another MS-DOS/Windows format from the early
              '90s that usually have the extension `.snd'.

       .sou   An alias for the .u8 raw format.

       .sox (also with -t ffmpeg)
              SoX's native uncompressed PCM format is intended for storing  or
              piping audio at intermediate processing points between SoX invo-
              cations.   It  has  much  in common with WAV, AIFF and AU uncom-
              pressed PCM formats but has the following specific  characteris-
              tics:  the PCM samples are stored as 32 bit signed integers, the
              samples are stored (by default) as `native endian' and the  num-
              ber of samples in the file is recorded as a 64-bit integer. Com-
              ments are also supported.

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

       .spdif (with ffmpeg)
              IEC 61937 S/PDIF format.

              SoX  can only autodetect this type of file from its filename ex-
              tension; if it is read from `standard input' (stdin) or  from  a
              file  whose name does not end in `.spdif', you will need to pre-
              fix it with `-t ffmpeg'.

       .sph, .nist (also with -t sndfile or -t ffmpeg)
              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 con-
              tain <mu>-law and PCM data.  It will ignore  header  information
              that  says  the data is compressed using shorten compression and
              will treat the data as either <mu>-law or PCM.  SoX and the com-
              mand line shorten program can be run together using pipes to en-
              compasses the data and then pass the result to SoX for  process-
              ing.

       .spx, .speex (with ffmpeg)
              Ogg  Speex format is for high compression of speech that, in VBR
              mode, achieves higher quality than AMR or GSM, but is  now  con-
              sidered superseded by their more recent Opus codec.

       sunau (optional)
              The  Sun  /dev/audio  device  driver  supports  both playing and
              recording audio.  For example:

                   sox_ng infile -t sunau /dev/audio

              or

                   sox_ng infile -t sunau -e mu-law -c 1 /dev/audio

              for older Sun equipment.


       .svcd (with ffmpeg)
              Another name for .mov.

       .tta (with ffmpeg)
              True Audio format.

       .vag (with ffmpeg)
              Sony PS2 VAG format.

       .txw   TXW is a file format from the Yamaha  TX-16W  sampling  keyboard
              which  wrote  samples onto IBM/PC-format 3.5" floppies at sample
              rates of 16kHz, 33kHz and 50kHz, all exact divisors of 100kHz.

              SoX handles reading of files which do not have the  sample  rate
              field  set to one of the expected rates by looking at some other
              bytes in the attack/loop length fields and defaulting  to  33kHz
              if the sample rate is still unknown.

       .vcd (with ffmpeg)
              Another name for .mov.

       .vms   See .dvms.

       .vob (with ffmpeg)
              Another name for .mov.

       .voc (also with -t sndfile or -t ffmpeg)
              Sound  Blaster  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  reads  but  cannot  write  VOC files with multiple
              blocks and files containing <mu>-law, A-law and 2/3/4-bit  ADPCM
              samples.

       .vorbis
              See .ogg.

       .vox   Headerless  files of Dialogic/OKI ADPCM audio data commonly come
              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 AD-
              PCM  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.  While 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_ng input.vox output.wav highpass 10


       .w64 (with sndfile, also with -t ffmpeg)
              Sonic Foundry's 64-bit RIFF/WAV format.

              SoX  can only autodetect this type of file from its filename ex-
              tension; if it is read from `standard input' (stdin) or  from  a
              file  whose name does not end in `.w64', you will need to prefix
              it with `-t w64'.

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

              Normally  .wav  files  have  all formatting information in their
              headers, so format options do not usually need to  be  specified
              for  input files.  If any are, they override the file header and
              you will be warned to this effect.  Output format  options  will
              cause a format conversion and the .wav is 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 other ways not currently  sup-
              ported  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


              Natively, SoX can only read WAV files with a bit-depth of 8, 16,
              24 or 32; files with other bit-depths can be read  by  preceding
              them with -t sndfile.

              Big  endian  versions  of RIFF files, called RIFX, are also sup-
              ported.  To write a RIFX file, use the -B output file option.

              See also .wavpcm.

       waveaudio (optional)
              The MS-Windows native audio device driver.  Examples:

                   sox_ng infile -t waveaudio
                   sox_ng infile -t waveaudio default
                   sox_ng infile -t waveaudio 1
                   sox_ng infile -t waveaudio "High Definition Audio Device"

              If the device name is omitted, -1, or default, you get the  `Mi-
              crosoft  Wave Mapper' device.  Wave Mapper means `use the system
              default audio devices' and you can control what `default'  means
              via the OS Control Panel.

              If  the given device name is some other number, you get that au-
              dio device by its 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 device name 0
              will  get  the  first  output device (usually the only audio de-
              vice).

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


       .wavpcm
              A non-standard but widely used variant of .wav.   Some  applica-
              tions  cannot  read  a  standard WAV file header for PCM-encoded
              data with a 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 sup-
              port 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.  SoX will automatically de-
              tect and read WAV files with a non-standard header.

              The most common use of this file type is likely to be along  the
              following lines:

                   sox_ng infile.any -t wavpcm -e signed-integer outfile.wav


       .webm (with ffmpeg)
              See .mkv.

       .wma (with ffmpeg)
              Windows Media Audio format.

       .wsaud (with ffmpeg)
              Westwood Studios audio format.

       .wsd   Wideband  Single-bit Data is the same as .dsf but with a differ-
              ent header.

       .wtv (with ffmpeg)
              Windows Television format.

       .wv (also with -t sndfile or -t ffmpeg)
              WavPack lossless audio compression.  Note that, when  converting
              .wav  to this format and back again, the RIFF header is not nec-
              essarily preserved losslessly, though the audio is.

       .wve (also with -t sndfile)
              Psion 8-bit A-law is used on Psion SIBO PDAs (Series 3 and simi-
              lar).

       .xa (also with -t ffmpeg)
              Maxis XA files 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 (with sndfile)
              Fasttracker 2 Extended Instrument format.

REFERENCES
       [1]    Wikipedia, M3U, http://en.wikipedia.org/wiki/M3U

       [2]    Wikipedia, PLS, http://en.wikipedia.org/wiki/PLS_(file_format)

SEE ALSO
       sox_ng(1), https://codeberg.org/sox_ng

AUTHORS
       Lance  Norskog,  Chris  Bagwell and many other authors and contributors
       listed in the README file that is distributed with the source code.

SoX                            November 28, 2024               SOXFORMAT_NG(7)
