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! 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.

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...

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++ automake autoconf \ 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 \ libbz2-dev libcdio-paranoia-dev libiso9660-dev libmms-dev \ libzzip-dev \ libcurl4-gnutls-dev libyajl-dev \ libasound2-dev libao-dev libjack-jackd2-dev libopenal-dev \ libpulse-dev libroar-dev libshout3-dev \ libmpdclient-dev \ libavahi-client-dev \ libsqlite3-dev \ libsystemd-daemon-dev libwrap0-dev \ libcppunit-dev xmlto \ libboost-dev \ libglib2.0-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 install

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.

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".

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.

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. enabled yes|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.

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. enabled yes|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.

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.

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). enabled yes|no Specifies whether this audio output is enabled when MPD is started. By default, all audio outputs are enabled. tags yes|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_on yes|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_type hardware|software|none Specifies which mixer should be used for this audio output: the hardware mixer (available for ALSA, OSS and PulseAudio), the software mixer or no mixer (none). By default, the hardware mixer is used for devices which support it, and none for the others. replay_gain_handler software|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.

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.

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. enabled yes|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.

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.

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.

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). Setting Description state_file PATH Specify the state file location. The parent directory must be writable by the MPD user (+wx). state_file_interval SECONDS Auto-save the state file this number of seconds after each state change. Defaults to 120 (2 minutes).

These settings are various limitations to prevent MPD from using too many resources (denial of service). Setting Description connection_timeout SECONDS 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_connections NUMBER This specifies the maximum number of clients that can be connected to MPD at the same time. Default is 5. max_playlist_length NUMBER The maximum number of songs that can be in the playlist. Default is 16384. max_command_list_size KBYTES The maximum size a command list. Default is 2048 (2 MiB). max_output_buffer_size KBYTES The maximum size of the output buffer to a client (maximum response size). Default is 8192 (8 MiB).

Do not change these unless you know what you are doing. Setting Description audio_buffer_size KBYTES Adjust the size of the internal audio buffer. Default is 4096 (4 MiB). buffer_before_play PERCENT 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%.

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" 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 (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.

"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.

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.

The default plugin. Stores a copy of the database in memory. A file is used for permanent storage. Setting Description path The path of the database file. compress yes|no Compress the database file using gzip? Enabled by default (if built with zlib).

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. Setting Description host The host name of the "master" MPD instance. port The port number of the "master" MPD instance.

Provides access to UPnP media servers.

The default plugin which gives MPD access to local files. It is used when music_directory refers to a local directory.

Load music files from a SMB/CIFS server. It used used when music_directory contains a smb:// URI, for example "smb://myfileserver/Music".

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".

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

Plays audio CDs. The URI has the form: "cdda://[DEVICE][/TRACK]". The simplest form cdda:// plays the whole disc in the default drive. Setting Description default_byte_order little_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.

Opens remote files or streams over HTTP. Setting Description proxy Sets the address of the HTTP proxy server. proxy_user, proxy_password Configures proxy authentication. verify_peer yes|no Verify the peer's SSL certificate? More information. verify_host yes|no Verify the certificate's name against host? More information.

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. Setting Description despotify_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).

Opens local files.

Plays streams with the MMS protocol.

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.

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.ogg

Decodes DFF files containing DSDIFF data (e.g. SACD rips). Setting Description lsbitfirst yes|no Decode the least significant bit first. Default is no.

Decodes DSF files containing DSDIFF data (e.g. SACD rips).

MIDI decoder based on FluidSynth. Setting Description sample_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.

Module player based on MikMod. Setting Description loop yes|no Allow backward loops in modules. Default is no. sample_rate Sets the sample rate generated by libmikmod. Default is 44100.

Module player based on MODPlug. Setting Description loop_count Number of times to loop the module if it uses backward loops. Default is 0 which prevents looping. -1 loops forever.

MIDI decoder based on libwildmidi. Setting Description config_file The absolute path of the timidity config file. Defaults to /etc/timidity/timidity.cfg.

Encodes into FLAC (lossless). Setting Description compression Sets the libFLAC compression level. The levels range from 0 (fastest, least compression) to 8 (slowest, most compression).

Encodes into MP3 using the LAME library. Setting Description quality 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.

Does not encode anything, passes the input PCM data as-is.

Encodes into MP3 using the Shine library. Setting Description bitrate Sets the bit rate in kilobit per second.

Encodes into MP2 using the TwoLAME library. Setting Description quality 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.

Encodes into Ogg Vorbis. Setting Description quality 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.

Encodes into WAV (lossless).

The Advanced Linux Sound Architecture (ALSA) plugin uses libasound. It is recommended if you are using Linux. Setting Description device NAME 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_mmap yes|no If set to yes, then libasound will try to use memory mapped I/O. buffer_time US Sets the device's buffer time in microseconds. Don't change unless you know what you're doing. period_time US Sets the device's period time in microseconds. Don't change unless you really know what you're doing. auto_resample yes|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_channels yes|no If set to no, then libasound will not attempt to convert between different channel numbers. auto_format yes|no If set to no, then libasound will not attempt to convert between different sample formats (16 bit, 24 bit, floating point, ...). dop yes|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: Setting Description mixer_device DEVICE Sets the ALSA mixer device name, defaulting to default which lets ALSA pick a value. mixer_control NAME Choose a mixer control, defaulting to PCM. Type amixer scontrols to get a list of available mixer controls. mixer_index NUMBER 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).

The ao plugin uses the portable libao library. Use only if there is no native plugin for your operating system. Setting Description driver D 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. options O Options to pass to the selected libao driver. write_size O 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.

The fifo plugin writes raw PCM data to a FIFO (First In, First Out) file. The data can be read by another program. Setting Description path P 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.

The jack plugin connects to a JACK server. Setting Description client_name NAME The name of the JACK client. Defaults to "Music Player Daemon". server_name NAME Optional name of the JACK server. autostart yes|no If set to yes, then libjack will automatically launch the JACK daemon. Disabled by default. source_ports A,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_ports A,B The names of the JACK destination ports to connect to. ringbuffer_size NBYTES Sets the size of the ring buffer for each channel. Do not configure this value unless you know what you're doing.

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. Setting Description port P Binds the HTTP server to the specified port. bind_to_address ADDR Binds the HTTP server to the specified address (IPv4 or IPv6). Multiple addresses in parallel are not supported. encoder NAME Chooses an encoder plugin. A list of encoder plugins can be found in the encoder plugin reference. max_clients MC Sets a limit, number of concurrent clients. When set to 0 no limit will apply.

The null plugin does nothing. It discards everything sent to it. Setting Description sync yes|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.

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. Setting Description device PATH 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: Setting Description mixer_device DEVICE Sets the OSS mixer device path, defaulting to /dev/mixer. mixer_control NAME Choose a mixer control, defaulting to PCM.

The "OpenAL" plugin uses libopenal. It is supported on many platforms. Use only if there is no native plugin for your operating system. Setting Description device NAME 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.

The "Mac OS X" plugin uses Apple's CoreAudio API.

The pipe plugin starts a program and writes raw PCM data into its standard input. Setting Description command CMD This command is invoked with the shell.

The pulse plugin connects to a PulseAudio server. Setting Description server HOSTNAME Sets the host name of the PulseAudio server. By default, MPD connects to the local PulseAudio server. sink NAME Specifies the name of the PulseAudio sink MPD should play on.

The roar plugin connects to a RoarAudio server. Setting Description server HOSTNAME The host name of the RoarAudio server. If not specified, then MPD will connect to the default locations. role ROLE The "role" that MPD registers itself as in the RoarAudio server. The default is "music".

The recorder plugin writes the audio played by MPD to a file. This may be useful for recording radio streams. Setting Description path P Write to this file. encoder NAME Chooses an encoder plugin. A list of encoder plugins can be found in the encoder plugin reference.

The shout plugin connects to a ShoutCast or IceCast server. It forwards tags to this server. You must set a format. Setting Description host HOSTNAME Sets the host name of the ShoutCast / IceCast server. port PORTNUMBER Connect to this port number on the specified host. timeout SECONDS Set the timeout for the shout connection in seconds. Defaults to 2 seconds. protocol icecast2|icecast1|shoutcast Specifies the protocol that wil be used to connect to the server. The default is "icecast2". mount URI Mounts the MPD stream in the specified URI. user USERNAME Sets the user name for submitting the stream to the server. Default is "source". password PWD Sets the password for submitting the stream to the server. name NAME Sets the name of the stream. genre GENRE Sets the genre of the stream (optional). description DESCRIPTION Sets a short description of the stream (optional). url URL Sets a URL associated with the stream (optional). public yes|no Specifies whether the stream should be "public". Default is no. encoder PLUGIN Chooses an encoder plugin. Default is vorbis. A list of encoder plugins can be found in the encoder plugin reference.

The "Solaris" plugin runs only on SUN Solaris, and plays via /dev/audio. Setting Description device PATH Sets the path of the audio device, defaults to /dev/audio.

Reads CUE sheets from the "CUESHEET" tag of song files.

Reads .m3u playlist files.

Reads extended .m3u playlist files.

Reads .pls playlist files.

Reads XSPF playlist files.

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)

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 Setting Description apikey client_id User apikey/client_id can override the MPD token provided by SoundCloud.