Provided by: mpg123_1.33.3-1_amd64 
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. This interacts fine with --random since version 1.33, but not --shuffle, as that
destroys the notion of a stable position in the playlist.
--loop times
for playing track(s) a certain number of times in a loop, < 0 means infinite loop.
--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 every track.
--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. The A-B loop feature with key 'o'
changes the preset loop interval to the interval between two presses of 'o', the third press (or
'p') ending the looped playback. The key 'p' will use the updated loop interval after that.
--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 fixed looping mode, away from the default
of 0.5 seconds, as a floating point number. This value can be overwritten at runtime using the A-B
loop feature.
--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.
--libversion
Print version information on the mpg123 libraries being used (libmpg123, libout123, libsyn123).
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)