The Music Player Daemon - User's ManualIntroduction
This document is work in progress. Most of it may be incomplete
yet. Please help!
MPD (Music Player Daemon) is, as the
name suggests, a server software allowing you to remotely play
your music, handle playlists, deliver music (HTTP streams with
various sub-protocols) and organizze playlists.
It has been written with minimal resource usage and stability in
mind! Infact, it runs fine on a Pentium 75, allowing you to use
your cheap old PC to create a stereo system!
MPD supports also gapless playback,
buffered audio output, and crossfading!
The separate client and server design allows users to choose a
user interface that best suites their tastes independently of
the underlying daemon, which actually plays music!
Installation
We recommend that you use the software installation routines of
your distribution to install MPD.
Most operating systems have a MPD
package, which is very easy to install.
Installing on Debian/Ubuntu
Install the package MPD via APT:
apt-get install mpd
When installed this way, MPD by
default looks for music in
/var/lib/mpd/music/; this may not be
correct. Look at your /etc/mpd.conf
file...
Compiling from source
Download the source tarball from the
MPD home page and unpack
it:
tar xf mpd-version.tar.xz
cd mpd-version
Make sure that all the required libraries and build tools are
installed. The INSTALL file has a list.
For example, the following installs a fairly complete list of
build dependencies on Debian Wheezy:
apt-get install g++ \
libmad0-dev libmpg123-dev libid3tag0-dev \
libflac-dev libvorbis-dev libopus-dev \
libadplug-dev libaudiofile-dev libsndfile1-dev libfaad-dev \
libfluidsynth-dev libgme-dev libmikmod2-dev libmodplug-dev \
libmpcdec-dev libwavpack-dev libwildmidi-dev \
libsidplay2-dev libsidutils-dev libresid-builder-dev \
libavcodec-dev libavformat-dev \
libmp3lame-dev \
libsamplerate0-dev libsoxr-dev \
libbz2-dev libcdio-paranoia-dev libiso9660-dev libmms-dev \
libzzip-dev \
libcurl4-gnutls-dev libyajl-dev libexpat-dev \
libasound2-dev libao-dev libjack-jackd2-dev libopenal-dev \
libpulse-dev libroar-dev libshout3-dev \
libmpdclient-dev \
libnfs-dev libsmbclient-dev \
libupnp-dev \
libavahi-client-dev \
libsqlite3-dev \
libsystemd-daemon-dev libwrap0-dev \
libcppunit-dev xmlto \
libboost-dev \
libglib2.0-dev libicu-dev
Now configure the source tree:
./configure
The --help argument shows a list of
compile-time options. When everything is ready and
configured, compile:
make
And install:
make installsystemd socket activation
Using systemd, you can launch
MPD on demand when the first client
attempts to connect.
MPD comes with two
systemd unit files: a "service"
unit and a "socket" unit. These will only be installed when
MPD was configured with
--with-systemdsystemunitdir=/lib/systemd.
To enable socket activation, type:
systemctl enable mpd.socket
systemctl start mpd.socket
In this configuration, MPD will
ignore the bind_to_address and
port settings.
ConfigurationConfiguring the music directory
When you play local files, you should organize them within a
directory called the "music directory". This is configured in
MPD with the
music_directory setting.
By default, MPD follows symbolic
links in the music directory. This behavior can be switched
off: follow_outside_symlinks controls
whether MPD follows links pointing
to files outside of the music directory, and
follow_inside_symlinks lets you disable
symlinks to files inside the music directory.
Instead of using local files, you can use storage plugins to access
files on a remote file server. For example, to use music from
the SMB/CIFS server "myfileserver" on the share called
"Music", configure the music directory
"smb://myfileserver/Music". For a
recipe, read the Satellite
MPD section.
Configuring database plugins
If a music directory is configured, one database plugin is
used. To configure this plugin, add a
database block to
mpd.conf:
database {
plugin "simple"
path "/var/lib/mpd/db"
}
The following table lists the database
options valid for all plugins:
Name
Description
plugin
The name of the plugin.
More information can be found in the database plugin reference.
Configuring neighbor plugins
All neighbor plugins are disabled by default to avoid unwanted
overhead. To enable (and configure) a plugin, add a
neighbor block to
mpd.conf:
neighbors {
plugin "smbclient"
}
The following table lists the neighbor
options valid for all plugins:
Name
Description
plugin
The name of the plugin.
More information can be found in the neighbor plugin reference.
Configuring input plugins
To configure an input plugin, add a input
block to mpd.conf:
input {
plugin "despotify"
user "foo"
password "bar"
}
The following table lists the input options
valid for all plugins:
Name
Description
plugin
The name of the plugin.
enabledyes|no
Allows you to disable a input plugin without
recompiling. By default, all plugins are enabled.
More information can be found in the input plugin reference.
Configuring decoder plugins
Most decoder plugins do not need any special configuration.
To configure a decoder, add a decoder block
to mpd.conf:
decoder {
plugin "wildmidi"
config_file "/etc/timidity/timidity.cfg"
}
The following table lists the decoder
options valid for all plugins:
Name
Description
plugin
The name of the plugin.
enabledyes|no
Allows you to disable a decoder plugin without
recompiling. By default, all plugins are enabled.
More information can be found in the decoder plugin reference.
Configuring encoder plugins
Encoders are used by some of the output plugins (such as shout). The
encoder settings are included in the
audio_output section. More information can
be found in the encoder plugin
reference.
Configuring audio outputs
Audio outputs are devices which actually play the audio chunks
produced by MPD. You can configure
any number of audio output devices, but there must be at least
one. If none is configured, MPD
attempts to auto-detect. Usually, this works quite well with
ALSA, OSS and on Mac OS X.
To configure an audio output manually, add an
audio_output block to
mpd.conf:
audio_output {
type "alsa"
name "my ALSA device"
device "hw:0"
}
The following table lists the audio_output
options valid for all plugins:
Name
Description
type
The name of the plugin.
name
The name of the audio output. It is visible to the
client. Some plugins also use it internally, e.g. as
a name registered in the PULSE server.
format
Always open the audio output with the specified audio
format (samplerate:bits:channels), regardless of the
format of the input file. This is optional for most
plugins.
Any of the three attributes may be an asterisk to
specify that this attribute should not be enforced,
example: 48000:16:*.
*:*:* is equal to not having
a format specification.
The following values are valid for
bits: 8
(signed 8 bit integer samples),
16, 24 (signed
24 bit integer samples padded to 32 bit),
32 (signed 32 bit integer
samples), f (32 bit floating
point, -1.0 to 1.0).
enabledyes|no
Specifies whether this audio output is enabled when
MPD is started. By
default, all audio outputs are enabled.
tagsyes|no
If set to no, then
MPD will not send tags to
this output. This is only useful for output plugins
that can receive tags, for example the httpd
output plugin.
always_onyes|no
If set to yes, then
MPD attempts to keep this
audio output always open. This may be useful for
streaming servers, when you don't want to disconnect
all listeners even when playback is accidentally
stopped.
mixer_typehardware|software|null|none
Specifies which mixer should be used for this audio
output: the hardware mixer (available for ALSA, OSS and PulseAudio), the
software mixer, the "null" mixer
(null; allows setting the
volume, but with no effect) or no mixer
(none). By default, the
hardware mixer is used for devices which support it,
and none for the others.
replay_gain_handlersoftware|mixer|none
Specifies how replay gain is applied. The default is
software, which uses an
internal software volume control.
mixer uses the configured
(hardware) mixer control. none
disables replay gain on this audio output.
Configuring filters
Filters are plugins which modify an audio stream.
To configure a filter, add a filter block
to mpd.conf:
filter {
plugin "volume"
name "software volume"
}
The following table lists the filter
options valid for all plugins:
Name
Description
plugin
The name of the plugin.
name
The name of the filter.
Configuring playlist plugins
Playlist plugins are used to load remote playlists. This is
not related to MPD's playlist
directory.
To configure a playlist plugin, add a
playlist_plugin block to
mpd.conf:
playlist_plugin {
name "m3u"
enabled "true"
}
The following table lists the
playlist_plugin options valid for all
plugins:
Name
Description
name
The name of the plugin.
enabledyes|no
Allows you to disable a input plugin without
recompiling. By default, all plugins are enabled.
More information can be found in the playlist plugin reference.
Audio Format SettingsGlobal Audio Format
The setting audio_output_format forces
MPD to use one audio format for
all outputs. Doing that is usually not a good idea. The
values are the same as in format in the audio_output
section.
Resampler
Sometimes, music needs to be resampled before it can be
played; for example, CDs use a sample rate of 44,100 Hz
while many cheap audio chips can only handle 48,000 Hz.
Resampling reduces the quality and consumes a lot of CPU.
There are different options, some of them optimized for high
quality and others for low CPU usage, but you can't have
both at the same time. Often, the resampler is the
component that is responsible for most of
MPD's CPU usage. Since
MPD comes with high quality
defaults, it may appear that MPD
consumes more CPU than other software.
The following resamplers are available (if enabled at
compile time):
libsamplerate
a.k.a. Secret Rabbit Code (SRC).
libsoxr,
the SoX Resampler library
internal: low CPU usage, but very poor quality. This is
the fallback if MPD was
compiled without an external resampler.
The setting samplerate_converter controls
how MPD shall resample music.
Possible values:
Value
Description
"internal"
The internal resampler. Low CPU usage, but very
poor quality.
"soxr very high"
Use libsoxr with "Very
High Quality" setting.
"soxr high" or
"soxr"
Use libsoxr with "High
Quality" setting.
"soxr medium"
Use libsoxr with "Medium
Quality" setting.
"soxr low"
Use libsoxr with "Low
Quality" setting.
"soxr quick"
Use libsoxr with "Quick"
setting.
"Best Sinc Interpolator" or
"0"
libsamplerate: Band
limited sinc interpolation, best quality, 97dB SNR,
96% BW.
"Medium Sinc Interpolator" or
"1"
libsamplerate: Band
limited sinc interpolation, medium quality, 97dB
SNR, 90% BW.
"Fastest Sinc Interpolator" or
"2"
libsamplerate: Band
limited sinc interpolation, fastest, 97dB SNR, 80%
BW.
"ZOH Sinc Interpolator" or
"3"
libsamplerate: Zero order
hold interpolator, very fast, very poor quality with
audible distortions.
"Linear Interpolator" or
"4"
libsamplerate: Linear
interpolator, very fast, poor quality.
Other SettingsThe State File
The state file is a file where
MPD saves and restores its state
(play queue, playback position etc.) to keep it persistent
across restarts and reboots. It is an optional setting.
MPD will attempt to load the
state file during startup, and will save it when shutting
down the daemon. Additionally, the state file is refreshed
every two minutes (after each state change).
SettingDescriptionstate_filePATH
Specify the state file location. The parent
directory must be writable by the
MPD user
(+wx).
state_file_intervalSECONDS
Auto-save the state file this number of seconds
after each state change. Defaults to
120 (2 minutes).
Resource Limitations
These settings are various limitations to prevent
MPD from using too many
resources (denial of service).
SettingDescriptionconnection_timeoutSECONDS
If a client does not send any new data in this time
period, the connection is closed. Clients waiting
in "idle" mode are excluded from this. Default is
60.
max_connectionsNUMBER
This specifies the maximum number of clients that
can be connected to MPD
at the same time. Default is
5.
max_playlist_lengthNUMBER
The maximum number of songs that can be in the
playlist. Default is 16384.
max_command_list_sizeKBYTES
The maximum size a command list. Default is
2048 (2 MiB).
max_output_buffer_sizeKBYTES
The maximum size of the output buffer to a client
(maximum response size). Default is
8192 (8 MiB).
Buffer Settings
Do not change these unless you know what you are doing.
SettingDescriptionaudio_buffer_sizeKBYTES
Adjust the size of the internal audio buffer.
Default is 4096 (4 MiB).
buffer_before_playPERCENT
Control the percentage of the buffer which is filled
before beginning to play. Increasing this reduces
the chance of audio file skipping, at the cost of
increased time prior to audio playback. Default is
10%.
Advanced configurationSatellite setupMPD runs well on weak machines such
as the Raspberry
Pi. However, such hardware tends to not have storage
big enough to hold a music collection. Mounting music from a
file server can be very slow, especially when updating the
database.
One approach for optimization is running
MPD on the file server, which not
only exports raw files, but also provides access to a readily
scanned database. Example configuration:
music_directory "nfs://fileserver.local/srv/mp3"
#music_directory "smb://fileserver.local/mp3"
database {
plugin "proxy"
host "fileserver.local"
}
The music_directory
setting tells MPD to read files
from the given NFS server. It does this by connecting to the
server from userspace. This does not actually mount the file
server into the kernel's virtual file system, and thus
requires no kernel cooperation and no special privileges. It
does not even require a kernel with NFS support, only the
nfs
storage plugin (using the libnfs
userspace library). The same can be done with SMB/CIFS using
the smbclient
storage plugin (using libsmbclient).
The database
setting tells MPD to pass all
database queries on to the MPD
instance running on the file server (using the proxy
plugin).
Using MPDThe client
After you have installed, configured and started
MPD, you choose a client to control
the playback.
The most basic client is mpc, which
provides a command line interface. It is useful in shell
scripts. Many people bind specific mpc
commands to hotkeys.
The MPD
Wiki contains an extensive list of clients to choose
from.
The music directory and the database
The "music directory" is where you store your music files.
MPD stores all relevant meta
information about all songs in its "database". Whenever you
add, modify or remove songs in the music directory, you have
to update the database, for example with
mpc:
mpc update
Depending on the size of your music collection and the speed
of the storage, this can take a while.
To exclude a file from the update, create a file called
.mpdignore in its parent directory. Each
line of that file may contain a list of shell wildcards.
The queue
The queue (sometimes called "current playlist") is a list of
songs to be played by MPD. To play
a song, add it to the queue and start playback. Most clients
offer an interface to edit the queue.
Advanced usageBit-perfect playback
"Bit-perfect playback" is a phrase used by audiophiles to
describe a setup that plays back digital music as-is, without
applying any modifications such as resampling, format
conversion or software volume. Naturally, this implies a
lossless codec.
By default, MPD attempts to do
bit-perfect playback, unless you tell it not to. Precondition
is a sound chip that supports the audio format of your music
files. If the audio format is not supported,
MPD attempts to fall back to the
nearest supported audio format, trying to lose as little
quality as possible.
To verify if MPD converts the audio
format, enable verbose logging, and watch for these lines:
decoder: audio_format=44100:24:2, seekable=true
output: opened plugin=alsa name="An ALSA output" audio_format=44100:16:2
output: converting from 44100:24:2
This example shows that a 24 bit file is being played, but the
sond chip cannot play 24 bit. It falls back to 16 bit,
discarding 8 bit.
However, this does not yet prove bit-perfect playback;
ALSA may be fooling
MPD that the audio format is
supported. To verify the format really being sent to the
physical sound chip, try:
cat /proc/asound/card*/pcm*p/sub*/hw_params
access: RW_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 4096
buffer_size: 16384
Obey the "format" row, which indicates that the current
playback format is 16 bit (signed 16 bit integer, little
endian).
Check list for bit-perfect playback:
Use the ALSA output
plugin.
Disable sound processing inside
ALSA by configuring a
"hardware" device (hw:0,0 or
similar).
Don't use software volume (setting mixer_type).
Don't force MPD to use a
specific audio format (settings format,
audio_output_format).
Verify that you are really doing bit-perfect playback
using MPD's verbose log and
/proc/asound/card*/pcm*p/sub*/hw_params.
Some DACs can also indicate the audio format.
Direct Stream Digital (DSD)
DSD (Direct
Stream Digital) is a digital format that stores audio
as a sequence of single-bit values at a very high sampling
rate.
MPD understands the file formats
dff
and dsf. There
are three ways to play back DSD:
Native DSD playback. Requires
ALSA 1.0.27.1 or later, a sound
driver/chip that supports DSD and of course a DAC that
supports DSD.
DoP (DSD over PCM) playback. This wraps DSD inside fake
24 bit PCM according to the DoP
standard. Requires a DAC that supports DSD. No
support from ALSA and the sound chip required (except for
bit-perfect 24 bit PCM
support).
Convert DSD to PCM on-the-fly.
Native DSD playback is used automatically if available. DoP
is only used if enabled explicitly using the dop option,
because there is no way for MPD to
find out whether the DAC supports it. DSD to PCM conversion
is the fallback if DSD cannot be used directly.
Plugin referenceDatabase pluginssimple
The default plugin. Stores a copy of the database in
memory. A file is used for permanent storage.
SettingDescriptionpath
The path of the database file.
cache_directory
The path of the cache directory for additional
storages mounted at runtime. This setting is
necessary for the mount protocol
command.
compressyes|no
Compress the database file using
gzip? Enabled by default (if
built with zlib).
proxy
Provides access to the database of another
MPD instance using
libmpdclient. This is useful when you
run mount the music directory via NFS/SMB, and the file
server already runs a MPD
instance. Only the file server needs to update the
database.
SettingDescriptionhost
The host name of the "master"
MPD instance.
port
The port number of the "master"
MPD instance.
upnp
Provides access to UPnP media servers.
Storage pluginslocal
The default plugin which gives
MPD access to local files. It is
used when music_directory refers to a
local directory.
smbclient
Load music files from a SMB/CIFS server. It used used when
music_directory contains a
smb:// URI, for example
"smb://myfileserver/Music".
nfs
Load music files from a NFS server. It used used when
music_directory contains a
nfs:// URI according to RFC2224,
for example "nfs://servername/path".
This plugin uses libnfs,
which supports only NFS version 3. Since
MPD is not allowed to bind to
"privileged ports", the NFS server needs to enable the
"insecure" setting; example
/etc/exports:
/srv/mp3 192.168.1.55(ro,insecure)
Don't fear: "insecure" does not mean that your NFS server is
insecure. A few decades ago, people thought the concept of
"privileged ports" would make network services "secure",
which was a fallacy. The absence of this obsolete
"security" measure means little.
Neighbor pluginssmbclient
Provides a list of SMB/CIFS servers on the local network.
upnp
Provides a list of UPnP servers on the local network.
Input pluginsalsa
Allows MPD on Linux to play audio
directly from a soundcard using the scheme
alsa://. Audio is formatted as 44.1 kHz
16-bit stereo (CD format). Examples:
mpc add alsa:// plays audio from device hw:0,0
mpc add alsa://hw:1,0 plays audio from device
hw:1,0
cdio_paranoia
Plays audio CDs. The URI has the form:
"cdda://[DEVICE][/TRACK]". The
simplest form cdda:// plays the whole
disc in the default drive.
SettingDescriptiondefault_byte_orderlittle_endian|big_endian
If the CD drive does not specify a byte order,
MPD assumes it is the
CPU's native byte order. This setting allows
overriding this.
curl
Opens remote files or streams over HTTP.
SettingDescriptionproxy
Sets the address of the HTTP proxy server.
proxy_user,
proxy_password
Configures proxy authentication.
verify_peeryes|no
Verify the peer's SSL certificate? More
information.
verify_hostyes|no
Verify the certificate's name against host? More
information.
despotify
Plays Spotify tracks using the despotify
library. The despotify plugin uses a spt:// URI and a Spotify
URL. So for example, you can add a song with:
mpc add spt://spotify:track:5qENVY0YEdZ7fiuOax70x1
You need a Spotify premium account to use this plugin, and you need
to setup username and password in the configuration file. The
configuration settings are global since the despotify playlist plugin
use the same settings.
SettingDescriptiondespotify_user
Sets up the Spotify username (required)
despotify_password
Sets up the Spotify password (required)
despotify_high_bitrate
Set up if high bitrate should be used for Spotify tunes.
High bitrate sounds better but slow systems can have problems
with playback (default yes).
file
Opens local files.
mms
Plays streams with the MMS protocol.
nfs
Allows MPD to access files on
NFSv3 servers without actually mounting them (i.e. in
userspace, without help from the kernel's VFS layer). All
URIs with the nfs:// scheme are used
according to RFC2224.
Example:
mpc add nfs://servername/path/filename.ogg
Note that this usually requires enabling the "insecure" flag
in the server's /etc/exports file,
because MPD cannot bind to
so-called "privileged" ports. Don't fear: this will not
make your file server insecure; the flag was named in a time
long ago when privileged ports were thought to be meaningful
for security. By today's standards, NFSv3 is not secure at
all, and if you believe it is, you're already doomed.
smbclient
Allows MPD to access files on
SMB/CIFS servers (e.g. Samba or Microsoft Windows). All
URIs with the smb:// scheme are used.
Example:
mpc add smb://servername/sharename/filename.oggDecoder pluginsdsdiff
Decodes DFF files containing DSDIFF data (e.g. SACD rips).
SettingDescriptionlsbitfirstyes|no
Decode the least significant bit first. Default is
no.
dsf
Decodes DSF files containing DSDIFF data (e.g. SACD rips).
fluidsynth
MIDI decoder based on FluidSynth.
SettingDescriptionsample_rate
The sample rate that shall be synthesized by the
plugin. Defaults to 48000.
soundfont
The absolute path of the soundfont file. Defaults
to
/usr/share/sounds/sf2/FluidR3_GM.sf2.
mikmod
Module player based on MikMod.
SettingDescriptionloopyes|no
Allow backward loops in modules. Default is
no.
sample_rate
Sets the sample rate generated by
libmikmod. Default is 44100.
modplug
Module player based on MODPlug.
SettingDescriptionloop_count
Number of times to loop the module if it uses backward loops.
Default is 0 which prevents looping.
-1 loops forever.
sidplay
C64 SID decoder based on
libsidplay.
SettingDescriptionsonglength_databasePATH
Location of your songlengths file, as distributed
with the HVSC. The sidplay
plugin checks this for matching MD5 fingerprints.
See http://www.c64.org/HVSC/DOCUMENTS/Songlengths.faq.
default_songlengthSECONDS
This is the default playing time in seconds for
songs not in the songlength database, or in case
you're not using a database. A value of 0 means
play indefinitely.
filteryes|no
Turns the SID filter emulation on or off.
wildmidi
MIDI decoder based on libwildmidi.
SettingDescriptionconfig_file
The absolute path of the timidity config file. Defaults
to
/etc/timidity/timidity.cfg.
Encoder pluginsflac
Encodes into FLAC (lossless).
SettingDescriptioncompression
Sets the libFLAC compression
level. The levels range from 0 (fastest, least
compression) to 8 (slowest, most compression).
lame
Encodes into MP3 using the LAME
library.
SettingDescriptionquality
Sets the quality for VBR. 0 is the highest quality,
9 is the lowest quality. Cannot be used with
bitrate.
bitrate
Sets the bit rate in kilobit per second. Cannot be
used with quality.
null
Does not encode anything, passes the input PCM data as-is.
shine
Encodes into MP3 using the Shine
library.
SettingDescriptionbitrate
Sets the bit rate in kilobit per second.
twolame
Encodes into MP2 using the TwoLAME
library.
SettingDescriptionquality
Sets the quality for VBR. 0 is the highest quality,
9 is the lowest quality. Cannot be used with
bitrate.
bitrate
Sets the bit rate in kilobit per second. Cannot be
used with quality.
vorbis
Encodes into Ogg
Vorbis.
SettingDescriptionquality
Sets the quality for VBR. -1 is the lowest quality,
10 is the highest quality. Cannot be used with
bitrate.
bitrate
Sets the bit rate in kilobit per second. Cannot be
used with quality.
wave
Encodes into WAV (lossless).
Output pluginsalsa
The Advanced
Linux Sound Architecture
(ALSA) plugin uses
libasound. It is recommended if you
are using Linux.
SettingDescriptiondeviceNAME
Sets the device which should be used. This can be
any valid ALSA device name. The default value is
"default", which makes
libasound choose a device. It
is recommended to use a "hw" or "plughw" device,
because otherwise, libasound
automatically enables "dmix", which has major
disadvantages (fixed sample rate, poor resampler,
...).
use_mmapyes|no
If set to yes, then
libasound will try to use
memory mapped I/O.
buffer_timeUS
Sets the device's buffer time in microseconds.
Don't change unless you know what you're doing.
period_timeUS
Sets the device's period time in microseconds.
Don't change unless you really know what you're
doing.
auto_resampleyes|no
If set to no, then
libasound will not attempt to
resample, handing the responsibility over to
MPD. It is recommended
to let MPD resample (with
libsamplerate), because
ALSA is quite poor at doing so.
auto_channelsyes|no
If set to no, then
libasound will not attempt to
convert between different channel numbers.
auto_formatyes|no
If set to no, then
libasound will not attempt to
convert between different sample formats (16 bit, 24
bit, floating point, ...).
dopyes|no
If set to yes, then DSD over
PCM according to the DoP
standard is enabled. This wraps DSD
samples in fake 24 bit PCM, and is understood by
some DSD capable products, but may be harmful to
other hardware. Therefore, the default is
no and you can enable the
option at your own risk.
The according hardware mixer plugin understands the
following settings:
SettingDescriptionmixer_deviceDEVICE
Sets the ALSA mixer device name, defaulting to
default which lets ALSA
pick a value.
mixer_controlNAME
Choose a mixer control, defaulting to
PCM. Type amixer
scontrols to get a list of available
mixer controls.
mixer_indexNUMBER
Choose a mixer control index. This is necessary if
there is more than one control with the same name.
Defaults to 0 (the first
one).
ao
The ao plugin uses the portable libao
library. Use only if there is no native plugin for your
operating system.
SettingDescriptiondriverD
The libao driver to use for
audio output. Possible values depend on what libao
drivers are available. See http://www.xiph.org/ao/doc/drivers.html
for information on some commonly used drivers.
Typical values for Linux include "oss" and "alsa09".
The default is "default", which causes libao to
select an appropriate plugin.
optionsO
Options to pass to the selected
libao driver.
write_sizeO
This specifies how many bytes to write to the audio
device at once. This parameter is to work around a
bug in older versions of libao on sound cards with
very small buffers. The default is 1024.
fifo
The fifo plugin writes raw PCM data to a
FIFO (First In, First Out) file. The data can be read by
another program.
SettingDescriptionpathP
This specifies the path of the FIFO to write to.
Must be an absolute path. If the path does not
exist, it will be created when
MPD is started, and
removed when MPD is
stopped. The FIFO will be created with the same
user and group as MPD is
running as. Default permissions can be modified by
using the builtin shell command
umask. If a FIFO already
exists at the specified path it will be reused, and
will not be removed when
MPD is stopped. You can
use the "mkfifo" command to create this, and then
you may modify the permissions to your liking.
jack
The jack plugin connects to a JACK
server.
SettingDescriptionclient_nameNAME
The name of the JACK
client. Defaults to "Music Player Daemon".
server_nameNAME
Optional name of the JACK
server.
autostartyes|no
If set to yes, then
libjack will automatically
launch the JACK daemon.
Disabled by default.
source_portsA,B
The names of the JACK
source ports to be created. By default, the ports
"left" and "right" are created. To use more ports,
you have to tweak this option.
destination_portsA,B
The names of the JACK
destination ports to connect to.
ringbuffer_sizeNBYTES
Sets the size of the ring buffer for each channel.
Do not configure this value unless you know what
you're doing.
httpd
The httpd plugin creates a HTTP server,
similar to ShoutCast
/ IceCast.
HTTP streaming clients like
mplayer can connect to it.
It is highly recommended to configure a fixed
format, because a stream cannot switch
its audio format on-the-fly when the song changes.
SettingDescriptionportP
Binds the HTTP server to the specified port.
bind_to_addressADDR
Binds the HTTP server to the specified address (IPv4 or
IPv6). Multiple addresses in parallel are not supported.
encoderNAME
Chooses an encoder plugin. A list of encoder
plugins can be found in the encoder plugin
reference.
max_clientsMC
Sets a limit, number of concurrent clients. When set
to 0 no limit will apply.
null
The null plugin does nothing. It
discards everything sent to it.
SettingDescriptionsyncyes|no
If set to no, then the timer
is disabled - the device will accept PCM chunks at
arbitrary rate (useful for benchmarking). The
default behaviour is to play in real time.
oss
The "Open Sound System" plugin is supported on most Unix
platforms.
On Linux, OSS has been superseded
by ALSA. Use the ALSA output
plugin instead of this one on Linux.
SettingDescriptiondevicePATH
Sets the path of the PCM device. If not specified,
then MPD will attempt to
open /dev/sound/dsp and
/dev/dsp.
The according hardware mixer plugin understands the
following settings:
SettingDescriptionmixer_deviceDEVICE
Sets the OSS mixer device path, defaulting to
/dev/mixer.
mixer_controlNAME
Choose a mixer control, defaulting to
PCM.
openal
The "OpenAL" plugin uses libopenal.
It is supported on many platforms. Use only if there is no
native plugin for your operating system.
SettingDescriptiondeviceNAME
Sets the device which should be used. This can be
any valid OpenAL device name. If not specified, then
libopenal will choose a default device.
osx
The "Mac OS X" plugin uses Apple's CoreAudio API.
pipe
The pipe plugin starts a program and
writes raw PCM data into its standard input.
SettingDescriptioncommandCMD
This command is invoked with the shell.
pulse
The pulse plugin connects to a PulseAudio
server.
SettingDescriptionserverHOSTNAME
Sets the host name of the
PulseAudio server. By
default, MPD connects to
the local PulseAudio
server.
sinkNAME
Specifies the name of the
PulseAudio sink
MPD should play on.
roar
The roar plugin connects to a RoarAudio
server.
SettingDescriptionserverHOSTNAME
The host name of the RoarAudio server. If not
specified, then MPD will
connect to the default locations.
roleROLE
The "role" that MPD
registers itself as in the RoarAudio server. The
default is "music".
recorder
The recorder plugin writes the audio
played by MPD to a file. This
may be useful for recording radio streams.
SettingDescriptionpathP
Write to this file.
encoderNAME
Chooses an encoder plugin. A list of encoder
plugins can be found in the encoder plugin
reference.
shout
The shout plugin connects to a ShoutCast
or IceCast
server. It forwards tags to this server.
You must set a format.
SettingDescriptionhostHOSTNAME
Sets the host name of the ShoutCast
/ IceCast
server.
portPORTNUMBER
Connect to this port number on the specified host.
timeoutSECONDS
Set the timeout for the shout connection in seconds.
Defaults to 2 seconds.
protocolicecast2|icecast1|shoutcast
Specifies the protocol that wil be used to connect
to the server. The default is
"icecast2".
mountURI
Mounts the MPD stream in
the specified URI.
userUSERNAME
Sets the user name for submitting the stream to the
server. Default is "source".
passwordPWD
Sets the password for submitting the stream to the
server.
nameNAME
Sets the name of the stream.
genreGENRE
Sets the genre of the stream (optional).
descriptionDESCRIPTION
Sets a short description of the stream (optional).
urlURL
Sets a URL associated with the stream (optional).
publicyes|no
Specifies whether the stream should be "public".
Default is no.
encoderPLUGIN
Chooses an encoder plugin. Default is vorbis.
A list of encoder plugins can be found in the encoder plugin
reference.
solaris
The "Solaris" plugin runs only on SUN Solaris, and plays via
/dev/audio.
SettingDescriptiondevicePATH
Sets the path of the audio device, defaults to
/dev/audio.
Playlist pluginsembcue
Reads CUE sheets from the "CUESHEET" tag of song files.
m3u
Reads .m3u playlist files.
extm3u
Reads extended .m3u playlist files.
pls
Reads .pls playlist files.
xspf
Reads XSPF
playlist files.
despotify
Adds Spotify
playlists. Spotify playlists use the spt:// URI,
and a Spotify playlist URL. So for example, you can load a playlist
with
mpc load spt://spotify:user:simon.kagstrom:playlist:3SUwkOe5VbVHysZcidEZtH
See the despotify input plugin for configuration options (username
and password needs to be setup)
soundcloud
Adds Soundcloud
playlists. SoundCloud playlists use the soundcloud:// URI,
and with a number of arguments, you may load different playlists with
mpc load soundcloud://track/TRACK_ID
mpc load soundcloud://playlist/PLAYLIST_ID
mpc load soundcloud://user/USERNAME
mpc load soundcloud://search/SEARCH_QUERY
mpc load soundcloud://url/https://soundcloud.com/ARTIST/TRACK-NAME
SettingDescriptionapikeyclient_id
User apikey/client_id can override the
MPD token provided by
SoundCloud.