mpg123(1) User Commands mpg123(1)

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. 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)