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 organize 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 In any case, you need: a C++14 compiler (e.g. gcc 4.9 or clang 3.9) Boost 1.46 pkg-config Each plugin usually needs a codec library, which you also need to install. Check the plugin reference for details about required libraries. For example, the following installs a fairly complete list of build dependencies on Debian Jessie: 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 libtwolame-dev libshine-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 \ libsndio-dev \ libmpdclient-dev \ libnfs-dev libsmbclient-dev \ libupnp-dev \ libavahi-client-dev \ libsqlite3-dev \ libsystemd-dev libwrap0-dev \ libcppunit-dev xmlto \ libboost-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 install

Even though it does not "feel" like a Windows application, MPD works well under Windows. Its build process follows the "Linux style", and may seem awkward for Windows people (who are not used to compiling their software, anyway). Basically, there are three ways to compile MPD for Windows: Build on Windows for Windows. All you need to do is described above already: configure and make. For Windows users, this is kind of unusual, because few Windows users have a GNU toolchain and a UNIX shell installed. Build on Linux for Windows. This is described above already: configure and make. You need the mingw-w64 cross compiler. Pass --host=i686-w64-mingw32 (32 bit) or --host=x86_64-w64-mingw32 (64 bit) to configure. This is somewhat natural for Linux users. Many distributions have mingw-w64 packages. The remaining difficulty here is installing all the external libraries. And MPD usually needs many, making this method cumbersome for the casual user. Build on Linux for Windows using the MPD's library build script. This section is about the latter. Just like with the native build, unpack the MPD source tarball and change into the directory. Then, instead of ./configure, type: ./win32/build.py --64 This downloads various library sources, and then configures and builds MPD (for x64; to build a 32 bit binary, pass --32). The resulting EXE files is linked statically, i.e. it contains all the libraries already, and you do not need carry DLLs around. It is large, but easy to use. If you wish to have a small mpd.exe with DLLs, you need to compile manually, without the build.py script.

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/system. 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.

You can launch MPD as a systemd user unit. The service file will only be installed when MPD was configured with --with-systemduserunitdir=/usr/lib/systemd/user or --with-systemduserunitdir=$HOME/.local/share/systemd/user. Once the user unit is installed, you can start and stop MPD like any other service: systemctl --user start mpd To auto-start MPD upon login, type: systemctl --user enable mpd

MPD reads its configuration from a text file. Usually, that is /etc/mpd.conf, unless a different path is specified on the command line. If you run MPD as a user daemon (and not as a system daemon), the configuration is read from $XDG_CONFIG_HOME/mpd/mpd.conf (usually ~/.config/mpd/mpd.conf). Each line in the configuration file contains a setting name and its value, e.g.: connection_timeout "5" For settings which specify a filesystem path, the tilde is expanded: music_directory "~/Music" Some of the settings are grouped in blocks with curly braces, e.g. per-plugin settings: audio_output { type "alsa" name "My ALSA output" device "iec958:CARD=Intel,DEV=0" mixer_control "PCM" }

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.

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.

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.

To configure an input plugin, add a input block to mpd.conf: input { plugin "curl" proxy "proxy.local" } 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 one or more audio_output blocks 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), "dsd" means DSD (Direct Stream Digital). For DSD, there are special cases such as "dsd64", which allows you to omit the sample rate (e.g. dsd512:2 for stereo DSD512, i.e. 22.5792 MHz). The sample rate is special for DSD: MPD counts the number of bytes, not bits. Thus, a DSD "bit" rate of 22.5792 MHz (DSD512) is 2822400 from MPD's point of view (44100*512/8). enabled yes|no Specifies whether this audio output is enabled when MPD is started. By default, all audio outputs are enabled. This is just the default setting when there is no state file; with a state file, the previous state is restored. 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|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; this can be used as a trick to implement an external 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 (protocol commands load, listplaylist and listplaylistinfo). 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. Check the resampler plugin reference for a list of resamplers and how to configure them.

Setting Description metadata_to_use TAG1,TAG2,... Use only the specified tags, and ignore the others. This setting can reduce the database size and MPD's memory usage by omitting unused tags. By default, all tags but comment are enabled. The special value "none" disables all tags.

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).

"Stickers" are pieces of information attached to songs. Some clients use them to store ratings and other volatile data. This feature requires SQLite, compile-time configure option --enable-sqlite. Setting Description sticker_file PATH The location of the sticker database.

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%.

If Zeroconf support (Avahi or Apple's Bonjour) was enabled at compile time with --with-zeroconf=..., MPD can announce its presence on the network. The following settings control this feature: Setting Description zeroconf_enabled yes|no Enables or disables this feature. Default is yes. zeroconf_name NAME The service name to publish via Zeroconf. The default is "Music Player".

MPD 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).

On Linux, MPD attempts to configure real-time scheduling for some threads that benefit from it. This is only possible you allow MPD to do it. This privilege is controlled by RLIMIT_RTPRIO RLIMIT_RTTIME. You can configure this privilege with ulimit before launching MPD: ulimit -HS -r 50; mpd Or you can use the prlimit program from the util-linux package: prlimit --rtprio=50 --rttime=unlimited mpd The systemd service file shipped with MPD comes with this setting. This works only if the Linux kernel was compiled with CONFIG_RT_GROUP_SCHED disabled. Use the following command to check this option for your current kernel: zgrep ^CONFIG_RT_GROUP_SCHED /proc/config.gz There is a rumor that real-time scheduling improves audio quality. That is not true. All it does is reduce the probability of skipping (audio buffer xruns) when the computer is under heavy load.

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. Matching files in the current directory and all subdirectories are excluded.

When scanning or playing a song, MPD parses its metadata. The following tags are supported:

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.

Stored playlists are some kind of secondary playlists which can be created, saved, edited and deleted by the client. They are addressed by their names. Its contents can be loaded into the queue, to be played back. The playlist_directory setting specifies where those playlists are stored.

"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 setting 'mixer_type "null"' asks MPD to pretend that there is a mixer, but not actually do something. This allows you to implement a MPD client which listens for mixer events, queries the current (fake) volume, and uses it to program an external mixer. For example, your client can forward this setting to your amplifier.

Make sure you have the latest MPD version (via mpd --version, not mpc version). All the time, bugs are found and fixed, and your problem might be a bug that is fixed already. Do not ask for help unless you have the latest MPD version. The most common excuse is when your distribution ships an old MPD version - in that case, please ask your distribution for help, and not the MPD project. Check the log file. Configure 'log_level "verbose"' or pass --verbose to mpd. Sometimes, it is helpful to run MPD in a terminal and follow what happens. This is how to do it: mpd --stdout --no-daemon --verbose

The MPD project runs a forum and an IRC channel (#mpd on Freenode) for requesting help. Visit the MPD help page for details on how to get help.

I can't see my music in the MPD database! Check your music_directory setting. Does the MPD user have read permission on all music files, and read+execute permission on all music directories (and all of their parent directories)? Did you update the database? (mpc update) Did you enable all relevant decoder plugins at compile time? mpd --version will tell you. MPD doesn't read ID3 tags! You probably compiled MPD without libid3tag. mpd --version will tell you. I can't hear music on my client! That problem usually follows a misunderstanding of the nature of MPD. MPD is a remote-controlled music player, not a music distribution system. Usually, the speakers are connected to the box where MPD runs, and the MPD client only sends control commands, but the client does not actually play your music. MPD has output plugins which allow hearing music on a remote host (such as httpd), but that is not MPD's primary design goal. "Device or resource busy" This ALSA error means that another program uses your sound hardware exclusively. You can stop that program to allow MPD to use it. Sometimes, this other program is PulseAudio, which can multiplex sound from several applications, to allow them to share your sound chip. In this case, it might be a good idea for MPD to use PulseAudio as well, instead of using ALSA directly.

If you believe you found a bug in MPD, report it on the bug tracker. Your bug report should contain: the output of mpd --version your configuration file (mpd.conf) relevant portions of the log file (--verbose) be clear about what you expect MPD to do, and what is actually happening

All MPD crashes are bugs which must be fixed by a developer, and you should write a bug report. (Many crash bugs are caused by codec libraries used by MPD, and then that library must be fixed; but in any case, the MPD bug tracker is a good place to report it first if you don't know.) A crash bug report needs to contain a "backtrace". First of all, your MPD executable must not be "stripped" (i.e. debug information deleted). The executables shipped with Linux distributions are usually stripped, but some have so-called "debug" packages (package mpd-dbg or mpd-dbgsym on Debian, mpd-debug on other distributions). Make sure this package is installed. You can extract the backtrace from a core dump, or by running MPD in a debugger, e.g.: gdb --args mpd --stdout --no-daemon --verbose run As soon as you have reproduced the crash, type "bt" on the gdb command prompt. Copy the output to your bug report.

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. cache_directory The path of the cache directory for additional storages mounted at runtime. This setting is necessary for the mount protocol command. 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. Note that unless overridden by the below settings (e.g. by setting them to a blank value), general curl configuration from environment variables such as http_proxy or specified in ~/.curlrc will be in effect. Setting Description host The host name of the "master" MPD instance. port The port number of the "master" MPD instance. keepalive yes|no Send TCP keepalive packets to the "master" MPD instance? This option can help avoid certain firewalls dropping inactive connections, at the expensive of a very small amount of additional network traffic. Disabled by default.

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.

A WebDAV client using libcurl. It is used when music_directory contains a http:// or https:// URI, for example "https://the.server/dav/".

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

Load music files from a NFS server. It is 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.

Provides a list of SMB/CIFS servers on the local network.

Provides a list of UPnP servers on the local network.

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 using libcdio. 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 using libcurl. Note that unless overridden by the below settings (e.g. by setting them to a blank value), general curl configuration from environment variables such as http_proxy or specified in ~/.curlrc will be in effect. 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.

Access to various network protocols implemented by the FFmpeg library: gopher://, rtp://, rtsp://, rtmp://, rtmpt://, rtmps://

Opens local files.

Plays streams with the MMS protocol using libmms.

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 AdLib files using libadplug. Setting Description sample_rate The sample rate that shall be synthesized by the plugin. Defaults to 48000.

Decodes WAV and AIFF files using libaudiofile.

Decodes AAC files using libfaad.

Decodes various codecs using FFmpeg. Setting Description analyzeduration VALUE Sets the FFmpeg muxer option analyzeduration, which specifies how many microseconds are analyzed to probe the input. The FFmpeg formats documentation has more information. probesize VALUE Sets the FFmpeg muxer option probesize, which specifies probing size in bytes, i.e. the size of the data to analyze to get stream information. The FFmpeg formats documentation has more information.

Decodes FLAC files using libFLAC.

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.

Video game music file emulator based on game-music-emu. Setting Description accuracy yes|no Enable more accurate sound emulation.

Decodes MP3 files using libmad.

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.

Decodes Musepack files using libmpcdec.

Decodes MP3 files using libmpg123.

Decodes Opus files using libopus.

Read raw PCM samples. It understands the "audio/L16" MIME type with parameters "rate" and "channels" according to RFC 2586. It also understands the MPD-specific MIME type "audio/x-mpd-float".

C64 SID decoder based on libsidplay. Setting Description songlength_database PATH Location of your songlengths file, as distributed with the HVSC. The sidplay plugin checks this for matching MD5 fingerprints. See . default_songlength SECONDS 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. filter yes|no Turns the SID filter emulation on or off.

Decodes WAV and AIFF files using libsndfile.

Decodes Ogg-Vorbis files using libvorbis.

Decodes WavPack files using libwavpack.

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 Opus. Setting Description bitrate Sets the data rate in bit per second. The special value "auto" lets libopus choose a rate (which is the default), and "max" uses the maximum possible data rate. complexity Sets the Opus complexity. signal Sets the Opus signal type. Valid values are "auto" (the default), "voice" and "music".

Encodes into Ogg Vorbis. Setting Description quality Sets the quality for VBR. -1 is the lowest quality, 10 is the highest quality. Defaults to 3. Cannot be used with bitrate. bitrate Sets the bit rate in kilobit per second. Cannot be used with quality.

Encodes into WAV (lossless).

The resampler can be configured in a block named resampler, for example: resampler { plugin "soxr" quality "very high" } The following table lists the resampler options valid for all plugins: Name Description plugin The name of the plugin.

A resampler built into MPD. Its quality is very poor, but its CPU usage is low. This is the fallback if MPD was compiled without an external resampler.

A resampler using libsamplerate a.k.a. Secret Rabbit Code (SRC). Name Description type The interpolator type. See below for a list of known types. The following converter types are provided by libsamplerate: Type Description "Best Sinc Interpolator" or "0" Band limited sinc interpolation, best quality, 97dB SNR, 96% BW. "Medium Sinc Interpolator" or "1" Band limited sinc interpolation, medium quality, 97dB SNR, 90% BW. "Fastest Sinc Interpolator" or "2" Band limited sinc interpolation, fastest, 97dB SNR, 80% BW. "ZOH Sinc Interpolator" or "3" Zero order hold interpolator, very fast, very poor quality with audible distortions. "Linear Interpolator" or "4" Linear interpolator, very fast, poor quality.

A resampler using libsoxr, the SoX Resampler library Name Description quality The libsoxr quality setting. Valid values are: "very high" "high" (the default) "medium" "low" "quick" threads The number of libsoxr threads. "0" means "automatic". The default is "1" which disables multi-threading.

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, ...). 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 sndio plugin uses the sndio library. It should normally be used on OpenBSD. Setting Description device NAME The audio output device libsndio will attempt to use. The default is "default" which causes libsndio to select the first output device. buffer_time MS Set the application buffer time in milliseconds.

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, VLC, and mpv 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. Setting Description device NAME Sets the device which should be used. Uses device names as listed in the "Audio Devices" window of "Audio MIDI Setup". hog_device yes|no Hog the device. This means that it takes exclusive control of the audio output device it is playing through, and no other program can access it. sync_sample_rate yes|no Synchronize the sample rate. It will try to set the output device sample rate to the corresponding sample rate of the file playing. If the output device does not support the sample rate the track has, it will try to select the best possible for each file. channel_map SOURCE,SOURCE,... Specifies a channel map. If your audio device has more than two outputs this allows you to route audio to auxillary outputs. For predictable results you should also specify a "format" with a fixed number of channels, e.g. "*:*:2". The number of items in the channel map must match the number of output channels of your output device. Each list entry specifies the source for that output channel; use "-1" to silence an output. For example, if you have a four-channel output device and you wish to send stereo sound (format "*:*:2") to outputs 3 and 4 while leaving outputs 1 and 2 silent then set the channel map to "-1,-1,0,1". In this example '0' and '1' denote the left and right channel respectively. The channel map may not refer to outputs that do not exist according to the format. If the format is "*:*:1" (mono) and you have a four-channel sound card then "-1,-1,0,0" (dual mono output on the second pair of sound card outputs) is a valid channel map but "-1,-1,0,1" is not because the second channel ('1') does not exist when the output is mono.

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. Requires libpulse. 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. format_path P An alternative to path which provides a format string referring to tag values. The special tag iso8601 emits the current date and time in ISO8601 format (UTC). Every time a new song starts or a new tag gets received from a radio station, a new file is opened. If the format does not render a file name, nothing is recorded. A tag name enclosed in percent signs ('%') is replaced with the tag value. Example: ~/.mpd/recorder/%artist% - %title%.ogg Square brackets can be used to group a substring. If none of the tags referred in the group can be found, the whole group is omitted. Example: [~/.mpd/recorder/[%artist% - ]%title%.ogg] (this omits the dash when no artist tag exists; if title also doesn't exist, no file is written) The operators "|" (logical "or") and "&" (logical "and") can be used to select portions of the format string depending on the existing tag values. Example: ~/.mpd/recorder/[%title%|%name%].ogg (use the "name" tag if no title exists) 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 using libshout. 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 .asx playlist files.

Reads .cue files.

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

Reads .m3u playlist files.

Reads extended .m3u playlist files.

Reads the cuesheet metablock from a FLAC file.

Reads .pls playlist files.

Reads music links from .rss files.

Download playlist from SoundCloud. It accepts URIs starting with soundcloud://. Setting Description apikey KEY An API key to access the SoundCloud servers.

Reads XSPF playlist files.