Provided by: mpg123_1.25.10-2_amd64 bug


       mpg123 - play audio MPEG 1.0/2.0/2.5 stream (layers 1, 2 and 3)


       mpg123 [ options ] file-or-URL...


       mpg123 reads one or more files (or standard input if ``-'' is specified) or URLs and plays
       them on the audio device (default) or outputs them to stdout.  file/URL is assumed  to  be
       an MPEG audio bit stream.


       The following operands are supported:

       file(s) The  path name(s) of one or more input files.  They must be valid MPEG-1.0/2.0/2.5
               audio layer 1, 2 or 3 bit streams.  If a dash ``-'' is specified, MPEG  data  will
               be  read from the standard input.  Furthermore, any name starting with ``http://''
               is recognized as URL (see next section).


       mpg123 options may be either the traditional POSIX one letter options, or  the  GNU  style
       long options.  POSIX style options start with a single ``-'', while GNU long options start
       with ``--''.  Option arguments (if needed) follow separated  by  whitespace  (not  ``='').
       Note  that  some  options  can be absent from your installation when disabled in the build


       -k num, --skip num
              Skip first num frames.  By default the decoding starts at the first frame.

       -n num, --frames num
              Decode only num frames.  By default the complete stream is decoded.

              Enable fuzzy seeks (guessing byte offsets or using  approximate  seek  points  from
              Xing  TOC).  Without that, seeks need a first scan through the file before they can
              jump at positions.  You can decide here:  sample-accurate  operation  with  gapless
              features or faster (fuzzy) seeking.

       -y, --no-resync
              Do  NOT  try  to resync and continue decoding if an error occurs in the input file.
              Normally, mpg123 tries to keep the playback alive at all costs, including  skipping
              invalid  material  and  searching  new header when something goes wrong.  With this
              switch you can make it bail out on data errors (and perhaps spare your ears  a  bad
              time).  Note  that  this switch has been renamed from --resync.  The old name still
              works, but is not advertised or recommened to use (subject to removal in future).

       --resync-limit bytes
              Set number of bytes to search for valid MPEG data once lost  in  stream;  <0  means
              search  whole  stream.   If  you know there are huge chunks of invalid data in your
              files... here is your hammer.  Note: Only since version 1.14  this  also  increases
              the amount of junk skipped on beginning.

       -p URL | none, --proxy URL | none
              The specified proxy will be used for HTTP requests.  It should be specified as full
              URL (``http://host.domain:port/''), but the ``http://'' prefix, the port number and
              the  trailing  slash  are optional (the default port is 80).  Specifying none means
              not to use any proxy, and to retrieve files directly from the  respective  servers.
              See also the ``HTTP SUPPORT'' section.

       -u auth, --auth auth
              HTTP  authentication  to  use  when  recieving  files via HTTP.  The format used is

              Ignore MIME types given by HTTP server. If you  know  better  and  want  mpg123  to
              decode something the server thinks is image/png, then just do it.

              Disable the default micro-buffering of non-seekable streams that gives the parser a
              safer footing.

       -@ file, --list file
              Read filenames and/or URLs of  MPEG  audio  streams  from  the  specified  file  in
              addition to the ones specified on the command line (if any).  Note that file can be
              either an ordinary file, a dash ``-'' to indicate that a list of filenames/URLs  is
              to  be  read  from  the standard input, or an URL pointing to a an appropriate list
              file.  Note: only one -@ option can be used (if more than one  is  specified,  only
              the last one will be recognized).

       -l n, --listentry n
              Of  the  playlist, play specified entry only.  n is the number of entry starting at
              1. A value of 0 is the default and means playling the whole list,  a negative value
              means showing of the list of titles with their numbers...

              Enable playlist continuation mode. This changes frame skipping to apply only to the
              first track and also continues to play  following  tracks  in  playlist  after  the
              selected one. Also, the option to play a number of frames only applies to the whole
              playlist. Basically, this tries to treat the playlist  more  like  one  big  stream
              (like, an audio book).  The current track number in list (1-based) and frame number
              (0-based) are printed at exit (useful if  you  interrupted  playback  and  want  to
              continue  later).   Note  that  the continuation info is printed to standard output
              unless the switch for piping audio data to standard out is used.  Also,  it  really
              makes  sense  to  work with actual playlist files instead of lists of file names as
              arguments, to keep track positions consistent.

       --loop times
              for looping track(s) a certain number of times, < 0 means infinite loop  (not  with

              For remote control mode: Keep loaded file open after reaching end.

       --timeout seconds
              Timeout  in  (integer)  seconds  before  declaring  a  stream  dead  (if <= 0, wait

       -z, --shuffle
              Shuffle play.  Randomly shuffles the order of files specified on the command  line,
              or in the list file.

       -Z, --random
              Continuous  random  play.  Keeps picking a random file from the command line or the
              play list.  Unlike shuffle play above, random play never ends, and plays individual
              songs more than once.

              Do not accept ICY meta data.

       -i, --index
              Index  /  scan  through  the track before playback.  This fills the index table for
              seeking (if enabled in libmpg123) and may make the operating system cache the  file
              contents for smoother operating on playback.

       --index-size size
              Set the number of entries in the seek frame index table.

       --preframes num
              Set  the  number of frames to be read as lead-in before a seeked-to position.  This
              serves to fill the layer 3 bit reservoir, which is needed to faithfully reproduce a
              certain  sample  at  a  certain position.  Note that for layer 3, a minimum of 1 is
              enforced (because of frame overlap), and for layer 1 and 2, this is  limited  to  2
              (no bit reservoir in that case, but engine spin-up anyway).


       -o module, --output module
              Select audio output module. You can provide a comma-separated list to use the first
              one that works.

              List the available modules.

       -a dev, --audiodevice dev
              Specify the  audio  device  to  use.   The  default  is  system-dependent  (usually
              /dev/audio  or  /dev/dsp).   Use this option if you have multiple audio devices and
              the default is not what you want.

       -s, --stdout
              The decoded audio samples are written to standard output, instead of  playing  them
              through  the  audio device.  This option must be used if your audio hardware is not
              supported by mpg123.  The output format per default is raw (headerless) linear  PCM
              audio data, 16 bit, stereo, host byte order (you can force mono or 8bit).

       -O file, --outfile
              Write  raw  output  into a file (instead of simply redirecting standard output to a
              file with the shell).

       -w file, --wav
              Write output as WAV file. This will cause the MPEG stream to be decoded  and  saved
              as  file file , or standard output if - is used as file name. You can also use --au
              and --cdr for AU and CDR format, respectively. Note that  WAV/AU  writing  to  non-
              seekable  files,  or redirected stdout, needs some thought. Since 1.16.0, the logic
              changed to writing the header with the first actual data. This avoids spurious  WAV
              headers  in a pipe, for example. The result of decoding nothing to WAV/AU is a file
              consisting just of the header when it is seekable and really nothing when not  (not
              even  a header). Correctly writing data with prophetic headers to stdout is no easy

       --au file
              Does not play the MPEG file but writes it to file in SUN audio  format.   If  -  is
              used  as  the  filename,  the AU file is written to stdout. See paragraph about WAV
              writing for header fun with non-seekable streams.

       --cdr file
              Does not play the MPEG file but writes it to file as a CDR file.  If - is  used  as
              the filename, the CDR file is written to stdout.

              Forces reopen of the audiodevice after ever song

       --cpu decoder-type
              Selects  a  certain  decoder (optimized for specific CPU), for example i586 or MMX.
              The list of available decoders can vary; depending on the build and what  your  CPU
              supports.   This  options is only availabe when the build actually includes several
              optimized decoders.

              Tests your CPU and prints a list of possible choices for --cpu.

              Lists all available decoder choices, regardless of support by your CPU.

       -g gain, --gain gain
              [DEPRECATED] Set audio hardware output gain (default: don't change).  The  unit  of
              the  gain  value  is hardware and output module dependent.  (This parameter is only
              provided for backwards compatibility and may be removed in the future without prior
              notice. Use the audio player for playing and a mixer app for mixing, UNIX style!)

       -f factor, --scale factor
              Change scale factor (default: 32768).

       --rva-mix, --rva-radio
              Enable  RVA  (relative  volume  adjustment)  using the values stored for ReplayGain
              radio mode / mix mode with all tracks roughly  equal  loudness.   The  first  valid
              information found in ID3V2 Tags (Comment named RVA or the RVA2 frame) or ReplayGain
              header in Lame/Info Tag is used.

       --rva-album, --rva-audiophile
              Enable RVA (relative volume adjustment) using  the  values  stored  for  ReplayGain
              audiophile  mode  /  album mode with usually the effect of adjusting album loudness
              but keeping relative loudness inside album.  The first valid information  found  in
              ID3V2  Tags  (Comment  named  RVA_ALBUM  or the RVA2 frame) or ReplayGain header in
              Lame/Info Tag is used.

       -0, --single0; -1, --single1
              Decode only channel 0 (left) or channel 1 (right), respectively.  These options are
              available for stereo MPEG streams only.

       -m, --mono, --mix, --singlemix
              Mix both channels / decode mono. It takes less CPU time than full stereo decoding.

              Force stereo output

       -r rate, --rate rate
              Set  sample  rate  (default: automatic).  You may want to change this if you need a
              constant bitrate independent of the mpeg stream rate. mpg123 automagically converts
              the rate. You should then combine this with --stereo or --mono.

       -2, --2to1; -4, --4to1
              Performs a downsampling of ratio 2:1 (22 kHz) or 4:1 (11 kHz) on the output stream,
              respectively. Saves some CPU cycles, but at least the 4:1 ratio sounds ugly.

       --pitch value
              Set hardware pitch (speedup/down, 0 is neutral;  0.05  is  5%).  This  changes  the
              output  sampling  rate,  so  it  only works in the range your audio system/hardware

       --8bit Forces 8bit output

              Forces f32 encoding

       -e enc, --encoding enc
              Choose output sample encoding. Possible  values  look  like  f32  (32-bit  floating
              point), s32 (32-bit signed integer), u32 (32-bit unsigned integer) and the variants
              with different numbers of bits (s24, u24,  s16,  u16,  s8,  u8)  and  also  special
              variants  like  ulaw  and  alaw  8-bit.   See  the  output of mpg123's longhelp for
              actually available encodings.

       -d n, --doublespeed n
              Only play every n'th frame.  This will cause the MPEG stream to be played  n  times
              faster,  which  can  be  used  for  special effects.  Can also be combined with the
              --halfspeed option to play 3 out of 4 frames etc.  Don't expect great sound quality
              when using this option.

       -h n, --halfspeed n
              Play  each  frame  n times.  This will cause the MPEG stream to be played at 1/n'th
              speed (n times slower), which can be used for special effects. Can also be combined
              with  the  --doublespeed  option  to  double every third frame or things like that.
              Don't expect great sound quality when using this option.

       -E file, --equalizer
              Enables equalization, taken from file.  The file needs to contain 32 lines of data,
              additional  comment  lines  may be prefixed with #.  Each data line consists of two
              floating-point entries, separated by whitespace.  They specify the multipliers  for
              left  and  right channel of a certain frequency band, respectively.  The first line
              corresponds to the lowest, the 32nd to the highest frequency band.  Note  that  you
              can control the equalizer interactively with the generic control interface.

              Enable  code  that  cuts  (junk)  samples  at beginning and end of tracks, enabling
              gapless transitions between MPEG files when encoder padding and codec delays  would
              prevent it.  This is enabled per default beginning with mpg123 version 1.0.0 .

              Disable  the  gapless code. That gives you MP3 decodings that include encoder delay
              and padding plus mpg123's decoder delay.

              Do not parse the Xing/Lame/VBR/Info frame, decode it instead just like a stupid old
              MP3  hardware  player.  This implies disabling of gapless playback as the necessary
              information is in said metadata frame.

       -D n, --delay n
              Insert a delay of n seconds before each track.

       -o h, --headphones
              Direct audio output to the headphone connector (some hardware only; AIX, HP, SUN).

       -o s, --speaker
              Direct audio output to the speaker  (some hardware only; AIX, HP, SUN).

       -o l, --lineout
              Direct audio output to the line-out connector (some hardware only; AIX, HP, SUN).

       -b size, --buffer size
              Use an audio output buffer of size Kbytes.  This is useful to bypass short  periods
              of  heavy  system  activity,  which  would  normally  cause  the audio output to be
              interrupted.  You should specify a buffer size of at least 1024 (i.e. 1  Mb,  which
              equals  about  6  seconds of audio data) or more; less than about 300 does not make
              much sense.  The default is 0, which turns buffering off.

       --preload fraction
              Wait for the buffer to be filled to fraction  before  starting  playback  (fraction
              between 0 and 1). You can tune this prebuffering to either get faster sound to your
              ears or safer uninterrupted web radio.  Default is 0.2 (wait for 20 % of buffer  to
              be full, changed from 1 in version 1.23).

       --devbuffer seconds
              Set  device  buffer  in seconds; <= 0 means default value. This is the small buffer
              between the application  and  the  audio  backend,  possibly  directly  related  to
              hardware buffers.

              Keep  buffer  over  track  boundaries  --  meaning, do not empty the buffer between
              tracks for possibly some added smoothness.


       -t, --test
              Test mode.  The audio stream is decoded, but no output occurs.

       -c, --check
              Check for filter range violations (clipping), and report them for each frame if any

       -v, --verbose
              Increase  the  verbosity  level.   For  example,  displays the frame numbers during

       -q, --quiet
              Quiet.  Suppress diagnostic messages.

       -C, --control
              Enable terminal control keys. This  is  enabled  automatically  if  a  terminal  is
              detected.   By  default  use  's' or the space bar to stop/restart (pause, unpause)
              playback, 'f' to jump forward to the next song, 'b' to jump back to  the  beginning
              of  the  song, ',' to rewind, '.' to fast forward, and 'q' to quit.  Type 'h' for a
              full list of available controls.

              Disable terminal control even if terminal is detected.

              In an xterm, rxvt, screen, iris-ansi  (compatible,  TERM  environment  variable  is
              examined), change the window's title to the name of song currently playing.

       --name name
              Set  the  name  of  this  instance,  possibly used in various places. This sets the
              client name for JACK output.

              Display ID3 tag info always in long format with one line per item  (artist,  title,

       --utf8 Regardless of environment, print metadata in UTF-8 (otherwise, when not using UTF-8
              locale, you'll get ASCII stripdown).

       -R, --remote
              Activate generic control interface.  mpg123 will then  read  and  execute  commands
              from  stdin.  Basic usage is ``load <filename> '' to play some file and the obvious
              ``pause'', ``command.  ``jump <frame>'' will jump/seek to a given point (MPEG frame
              number).  Issue ``help'' to get a full list of commands and syntax.

              Print responses for generic control mode to standard error, not standard out.  This
              is automatically triggered when using -s N.

       --fifo path
              Create a fifo / named pipe on the given path and  use  that  for  reading  commands
              instead of standard input.

              Tries to get higher priority

       -T, --realtime
              Tries  to  gain realtime priority.  This option usually requires root privileges to
              have any effect.

       -?, --help
              Shows short usage instructions.

              Shows long usage instructions.

              Print the version string.


       In addition to reading MPEG audio streams from ordinary files and from the standard input,
       mpg123 supports retrieval of MPEG audio files or playlists via the HTTP protocol, which is
       used in the World Wide Web (WWW).  Such files are specified using a so-called  URL,  which
       starts  with ``http://''.  When a file with that prefix is encountered, mpg123 attempts to
       open an HTTP connection to the server in order to retrieve that file to  decode  and  play

       It  is  often  useful  to  retrieve  files  through  a  WWW  cache or so-called proxy.  To
       accomplish this, mpg123 examines  the  environment  for  variables  named  MP3_HTTP_PROXY,
       http_proxy  and HTTP_PROXY, in this order.  The value of the first one that is set will be
       used as proxy specification.  To override this, you can use the  -p  command  line  option
       (see  the  ``OPTIONS''  section).   Specifying  -p none will enforce contacting the server
       directly without using any proxy, even if one of the above environment variables is set.

       Note that, in order to play MPEG audio files from a WWW server, it is necessary  that  the
       connection  to  that  server is fast enough.  For example, a 128 kbit/s MPEG file requires
       the network connection to be at least 128 kbit/s (16 kbyte/s) plus protocol overhead.   If
       you  suffer  from  short  network outages, you should try the -b option (buffer) to bypass
       such outages.  If your network connection is generally not fast enough  to  retrieve  MPEG
       audio  files  in  realtime,  you can first download the files to your local harddisk (e.g.
       using wget(1)) and then play them from there.

       If authentication is needed to access the file it can be specified with the -u user:pass.


       When in terminal control mode, you can quit via pressing the q key, while any time you can
       abort  mpg123  by  pressing Ctrl-C. If not in terminal control mode, this will skip to the
       next file (if any). If you want to abort playing immediately in that  case,  press  Ctrl-C
       twice in short succession (within about one second).

       Note  that the result of quitting mpg123 pressing Ctrl-C might not be audible immediately,
       due to audio data buffering in the audio device.  This delay is system dependent,  but  it
       is usually not more than one or two seconds.


       In  verbose  mode,  mpg123  updates  a  line with various information centering around the
       current playback position. On any decent terminal, the line also works as a  progress  bar
       in the current file by reversing video for a fraction of the line according to the current
       position. An example for a full line is this:

            > 0291+0955  00:01.68+00:28.22 [00:05.30] mix 100=085 192 kb/s  576 B acc    18  clip

       The information consists of, in order:

       >      single-character  playback  state  (``>''  for  playing, ``='' for pausing/looping,
              ``_'' for stopped)

              current frame offset and number of remaining frames after the plus sign

              current position from and remaining time in human terms (hours, minutes, seconds)

              fill of the output buffer in terms of playback time, if the buffer is enabled

       mix    selected RVA mode (possible values: mix, alb (album), and --- (neutral, off))

              set volume and the RVA-modified effective volume after the equal sign

       192 kb/s
              current bitrate

       576 B  size of current frame in bytes

       acc    if positions are accurate, possible values are ``acc'' for  accurate  positions  or
              ``fuz'' for fuzzy (with guessed byte offsets using mean frame size)

       18 clip
              amount  of  clipped  samples,  non-zero only if decoder reports that (generic does,
              some optimized ones not)

              pitch change (increased/decreased playback sampling rate on user request)


       MPEG audio decoding requires a good deal  of  CPU  performance,  especially  layer-3.   To
       decode  it in realtime, you should have at least an i486DX4, Pentium, Alpha, SuperSparc or
       equivalent processor.  You can also use the -m option to decode mono only,  which  reduces
       the CPU load somewhat for layer-3 streams.  See also the -2 and -4 options.

       If everything else fails, have mpg123 decode to a file and then use an appropriate utility
       to play that file with less CPU load.  Most probably you can configure mpg123 to produce a
       format suitable for your audio device (see above about encodings and sampling rates).

       If  your  system  is  generally fast enough to decode in realtime, but there are sometimes
       periods of heavy system load (such as cronjobs, users logging  in  remotely,  starting  of
       ``big'' programs etc.) causing the audio output to be interrupted, then you should use the
       -b option to use a buffer of reasonable size (at least 1000 Kbytes).


       Mostly MPEG-1 layer 2 and 3 are tested in real life.  Please report any issues and provide
       test files to help fixing them.

       No CRC error checking is performed.

       Some  platforms  lack audio hardware support; you may be able to use the -s switch to feed
       the decoded data to a program that can play it on your audio device.


              Thomas Orgis <>, <>

       Original Creator:
              Michael Hipp

       Uses code or ideas from various people, see the AUTHORS file accompanying the source code.


       mpg123 is licensed under the GNU Lesser/Library General Public License, LGPL, version  2.1


                                           29 Feb 2016                                  mpg123(1)