mpg123(1) mpg123(1)
26 Apr 2020
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://'' is recognized as URL (see next section).
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
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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).
-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 ...
--resync-limit bytes
Set number of bytes to search for valid MPEG data once lost in
stream; <0 means search whole stream. If you know there are huge
chunks of invalid data in your files... here is your hammer.
Note: Only since version 1.14 this also increases the amount of
junk skipped on beginning.
-p URL | none, --proxy URL | none
The specified proxy will be used for HTTP requests. It should be
specified as full URL (``http://host.domain:port/''), but the
``http://'' prefix, the port number and the trailing slash are
optional (the default port is 80). Specifying none means not to
use any proxy, and to retrieve files directly from the respective
servers. See also the ``HTTP SUPPORT'' section.
-u auth, --auth auth
HTTP authentication to use when recieving files via HTTP. The
format used is user:password.
--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
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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).
-l n, --listentry n
Of the playlist, play specified entry only. n is the number of
entry starting at 1. A value of 0 is the default and means
playling the whole list, a negative value means showing of the
list of titles with their numbers...
--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.
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--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.
-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
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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 options
is only availabe 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
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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,
Decode only channel 0 (left) or channel 1 (right), respectively.
These options are available for stereo MPEG streams only.
-m, --mono, --mix,
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.
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-2, --2to1; -4,
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.
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--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.
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--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.
--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> ''
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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 N.
--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 files or
playlists via the HTTP protocol, which is used in the World Wide Web
(WWW). Such files are specified using a so-called URL, which starts
with ``http://''. When a file with that prefix is encountered, mpg123
attempts to open an HTTP connection to the server in order to retrieve
that file to decode and play it.
It is often useful to retrieve files through a WWW cache or so-called
proxy. To accomplish this, mpg123 examines the environment for
variables named MP3_HTTP_PROXY, http_proxy and HTTP_PROXY, in this
order. The value of the first one that is set will be used as proxy
specification. To override this, you can use the -p command line
option (see the ``OPTIONS'' section). Specifying -p none will enforce
contacting the server directly without using any proxy, even if one of
the above environment variables is set.
Note that, in order to play MPEG audio files from a WWW server, it is
necessary that the connection to that server is fast enough. For
example, a 128 kbit/s MPEG file requires the network connection to be
at least 128 kbit/s (16 kbyte/s) plus protocol overhead. If you
suffer from short network outages, you should try the -b option
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(buffer) to bypass such outages. If your network connection is
generally not fast enough to retrieve MPEG audio files in realtime,
you can first download the files to your local harddisk (e.g. using
wget(1)) and then play them from there.
If authentication is needed to access the file it can be specified
with the -u user:pass.
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))
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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
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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.
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
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