Section: User Commands (1)
Updated: 29 Feb 2016
mpg123 - play audio MPEG 1.0/2.0/2.5 stream (layers 1, 2 and 3)
reads one or more
(or standard input if ``-'' is specified) or
and plays them on the audio device (default) or
outputs them to stdout.
is assumed to be an MPEG audio bit stream.
The following operands are supported:
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
(see next section).
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.
- -k num, --skip num
frames. By default the decoding starts at the first frame.
- -n num, --frames 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,
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
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
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 user:password.
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
in addition to the ones specified on the command line (if any).
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
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.
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 --random!).
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.
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).
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.
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.
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
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
, or standard output if
is used as file name. You can also use
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
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
as a CDR file. If - is used as the filename, the CDR file is written
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).
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 supports.
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
frame. This will cause the MPEG stream
to be played
times faster, which can be used for special
effects. Can also be combined with the
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
times. This will cause the MPEG stream
to be played at
speed (n times slower), which can be
used for special effects. Can also be combined with the
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
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
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
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 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.
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
- --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, ...)
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.
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
- --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,
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,
attempts to open an HTTP connection to the server in order to
retrieve that file to decode and play it.
It is often useful to retrieve files through a WWW cache or
so-called proxy. To accomplish this,
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
command line option (see the ``OPTIONS'' section). Specifying
will enforce contacting the server directly without using
any proxy, even if one of the above environment variables
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
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
and then play them from there.
If authentication is needed to access the file it can be
specified with the
When in terminal control mode, you can quit via pressing the q key,
while any time you can abort
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
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)
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
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
- 576 B
size of current frame in bytes
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
option to decode mono only, which reduces the CPU load
somewhat for layer-3 streams. See also the
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
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
switch to feed the decoded data to a program that can play it on your audio device.
Thomas Orgis <firstname.lastname@example.org>, <email@example.com>
- Original Creator:
Uses code or ideas from various people, see the AUTHORS file accompanying the source code.
is licensed under the GNU Lesser/Library General Public License, LGPL, version 2.1 .