Provided by: mpg123_1.31.3-2_amd64 bug

NAME

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

SYNOPSIS

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

DESCRIPTION

       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.

OPERANDS

       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://''
               or ``https://''  is  recognized  as  URL  (see  next  section),  while  a  leading
               ``file://'' is being stripped for normal local file access, for consistency (since
               mpg123 1.30.1).

OPTIONS

       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
       process.

INPUT OPTIONS

       -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.

       --fuzzy
              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 recommended to use (subject to removal in future).

       -F, --no-frankenstein
              Disable support for Frankenstein  streams.  Normally,  mpg123  stays  true  to  the
              concept  of  MPEG audio being just a concatenation of MPEG frames. It will continue
              decoding even if the type of MPEG frames varies wildly. With this switch,  it  will
              only  decode the input as long as it does not change its character (from layer I to
              layer III, changing sampling rate, from mono to stereo), silently assuming  end  of
              stream  on  such occasion. The switch also stops decoding of compatible MPEG frames
              if there was an Info frame (Xing header, Lame tag) that contained a length  of  the
              track  in  MPEG  frames.   This comes a bit closer to the notion of a MP3 file as a
              defined collection of MPEG frames  that  belong  together,  but  gets  rid  of  the
              flexibility  that  can be fun at times but mostly is hell for the programmer of the
              parser and decoder ...

       --network  backend
              Select network  backend (helper program), choices are usually auto, wget, and curl.
              Auto means to try the first available backend.

       --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.

       -u auth, --auth auth
              HTTP authentication to use when receiving files  via  HTTP.   The  format  used  is
              user:password.  Mpg123 will clear this quickly, but it may still appear in sight of
              other users or even just in your shell history. You may seek  alternative  ways  to
              specify that to your network backend.

       --auth-file authfile
              Provide the authentication info via given file instead of command line directly.

       --ignore-mime
              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.

       --no-icy-meta
              Do not accept ICY meta data.

       --streamdump filename
              Dump a copy of the input data (as read by  libmpg123)  to  the  given  file.   This
              enables  you  to  store  a  web  stream  to  disk  while  playing, or just create a
              concatenation of the local files you play for ... why not?

       --icy-interval bytes
              This setting enables you to play a stream dump containing ICY metadata at the given
              interval  in bytes (the value of the icy-metaint HTTP response header). Without it,
              such a stream will play, but will cause regular decoding glitches with resync.

       --no-seekbuffer
              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). Furthermore, for HTTP resources,  the  MIME  type
              information  will  be  used  to  re-open  an  actual MPEG stream as such instead of
              treating it as playlist file. So you could just always use  -@  for  web  resources
              without bothering if it is a playlist or already the resolved stream address.

       -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 playing the whole list,  a negative  value
              means showing of the list of titles with their numbers...

       --continue
              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
              --random!).

       --keep-open
              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
              forever).

       -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.

       -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).

OUTPUT and PROCESSING OPTIONS

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

       --list-modules
              List the available modules.

       --list-devices
              List the available  output  devices  for  given  output  module.  If  there  is  no
              functionality  to  list  devices in the chosen module, an error will be printed and
              mpg123 will exit with a non-zero code.

       -a dev, --audiodevice dev
              Specify the audio device to use.  The default as well as the possible values depend
              on  the  active  output.  For  the  JACK output, a comma-separated list of ports to
              connect to (for each channel) can be specified.

       -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
              business.

       --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.

       --reopen
              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  option is only available when the build actually includes several
              optimized decoders.

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

       --list-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.

       --stereo
              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.

       --resample method
              Set  resampling  method  to  employ  if  forcing  an  output  rate.  Choices (case-
              insensitive) are NtoM, dirty, and fine. The  fine  resampler  is  the  default.  It
              employs  libsyn123's  low-latency  fairly  efficient  resampler  to postprocess the
              output from libmpg123 instead of the fast but very crude NtoM decoder (drop  sample
              method) that mpg123 offers since decades. If you are really low on CPU time, choose
              NtoM, as the resampler usually needs more time than the MPEG decoder  itself.   The
              mpg123  program is smart enough to combine the 2to1 or 4to1 downsampling modes with
              the postprocessing for extreme downsampling.

       -2, --2to1; -4, --4to1
              Performs a downsampling of ratio 2:1 (22 kHz from 44.1 kHz) or 4:1 (11 kHz) on  the
              output  stream,  respectively. Saves some CPU cycles, but of course throws away the
              high frequencies, as the decoder does not bother producing them.

       --pitch value
              Set a pitch change (speedup/down, 0 is neutral; 0.05  is  5%  speedup).   When  not
              enforcing  an  output rate, this changes the output sampling rate, so it only works
              in the range your audio system/hardware supports. When  you  combine  this  with  a
              fixed output rate, it modifies a software resampling ratio instead.

       --8bit Forces 8bit output

       --float
              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.   Also
              note  that  these are the 32 bands of the MPEG codec, not spaced like you would see
              for a usual graphic equalizer. The upside is that there is zero computational  cost
              in  addition  to  decoding.  The  downside is that you roughly have bass in band 0,
              (upper) mids in band 1, treble in all others.

       --gapless
              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 .

       --no-gapless
              Disable the gapless code. That gives you MP3 decodings that include  encoder  delay
              and padding plus mpg123's decoder delay.

       --no-infoframe
              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.

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

MISC OPTIONS

       -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
              occur.

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

       -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.

       --no-control
              Disable terminal control even if terminal is detected.

       --title
              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.

       --pauseloop seconds
              Set the length of the loop interval in terminal control paused mode, away from  the
              default of 0.5 seconds, as a floating point number.

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

       --long-tag
              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.

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

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

       --aggressive
              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.

       --longhelp
              Shows long usage instructions.

       --version
              Print the version string.

HTTP SUPPORT

       In addition to reading MPEG audio streams from ordinary files and from the standard input,
       mpg123 supports retrieval of MPEG audio streams 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://  or  https://.   When a file with that prefix is encountered,
       mpg123 since 1.30.0 will by default call an external helper  program  (either  wget(1)  or
       curl(1),  see the --network option) to retrieve the resource. You can configure access via
       a proxy server using the  standard  environment  variables  those  programs  support.  The
       --proxy option that mpg123 before 1.30.0 used for its internal network code is gone in the
       default build now and will probably disappear for good with 1.31.1.

       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.

       Streams with embedded ICY metadata are supported, the interval being communicated via HTTP
       headers or --icy-interval.

INTERRUPT

       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.

PLAYBACK STATUS LINE

       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
       p+0.014

       The information consists of, in order:

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

       0291+0955
              current frame offset and number of remaining frames after the plus sign

       00:01.68+00:28.22
              current position from and remaining time in human terms (hours, minutes, seconds)

       [00:05.30]
              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))

       100=085
              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)

       p+0.014
              pitch change (increased/decreased playback sampling rate on user request)

NOTES

       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).

EXIT CODE

       Up  to  version  1.25.x,  mpg123 always returned exit code 0 also for complete junk on the
       input side. Fatal errors were only  considered  for  output.  With  version  1.26.0,  this
       changed to the behaviour described below.

       When not using the remote control interface (which returns input errors as text messages),
       the process exit code is zero (success) only if all tracks in a playlist had at least  one
       frame  parsed,  even  if  it did not decode cleanly, or are empty, MPEG-wise (perhaps only
       metadata, or really an empty file).  When you decode nothing, nothing is  the  result  and
       that  is  fine.  When  a  track  later aborts because of parser errors or breakdown of the
       network communication, this is treated as end of a track, but does not make the process as
       such  fail.  One really bad (or non-existing) stream in the playlist results in a non-zero
       error code, consistent with other UNIX tools.

       An error in audio output  results  in  the  process  ending  with  a  non-zero  exit  code
       immediately,  regardless  of  how  much  data  has  been  successfully  played before. The
       forgiveness is only on the input side.

BUGS

       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. But the decoder is built and tested to behave nicely
       with damaged streams. Mostly, damaged frames will just be silent.

       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.

AUTHORS

       Maintainer:
              Thomas Orgis <maintainer@mpg123.org>, <thomas@orgis.org>

       Original Creator:
              Michael Hipp

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

LICENSE

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

WEBSITE

       http://www.mpg123.org
       http://sourceforge.net/projects/mpg123

                                           11 Jul 2022                                  mpg123(1)