- Send TCP keepalive packets to the "master" :program:`MPD` instance? This option can help avoid certain firewalls dropping inactive connections, at the expense of a very small amount of additional network traffic. Disabled by default.
Allows :program:`MPD` on Linux to play audio directly from a soundcard using the scheme alsa://. Audio is by default formatted as 48 kHz 16-bit stereo, but this default can be overidden by a config file setting or by the URI. Examples:
- The sampling rate, size and channels to use. Wildcards are not allowed.
Example - "44100:16:2"
* - **auto_resample yes|no**
- If set to no, then libasound will not attempt to resample. In this case, the user is responsible for ensuring that the requested sample rate can be produced natively by the device, otherwise an error will occur.
* - **auto_channels yes|no**
- If set to no, then libasound will not attempt to convert between different channel numbers. The user must ensure that the device supports the requested channels when sampling.
* - **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, ...). Again the user must ensure that the requested format is available natively from the device.
- The `Qobuz format identifier <https://github.com/Qobuz/api-documentation/blob/master/endpoints/track/getFileUrl.md#parameters>`_, i.e. a number which chooses the format and quality to be requested from Qobuz. The default is "5" (320 kbit/s MP3).
- Sets the FFmpeg muxer option analyzeduration, which specifies how many microseconds are analyzed to probe the input. The `FFmpeg formats documentation <https://ffmpeg.org/ffmpeg-formats.html>`_ 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 <https://ffmpeg.org/ffmpeg-formats.html>`_ has more information.
- Sets the resampling mode. "nearest" disables interpolation (good for chiptunes). "linear" makes modplug use linear interpolation (fast, good quality). "spline" makes modplug use cubic spline interpolation (high quality). "fir" makes modplug use 8-tap fir filter (extremely high quality). Defaults to "fir".
- Set how many times the module repeats. -1: repeat forever. 0: play once, repeat zero times (the default). n>0: play once and repeat n times after that.
- Sets the stereo separation. The supported value range is [0,200]. Defaults to 100.
* - **interpolation_filter 0|1|2|4|8**
- Sets the interpolation filter. 0: internal default. 1: no interpolation (zero order hold). 2: linear interpolation. 4: cubic interpolation. 8: windowed sinc with 8 taps. Defaults to 0.
* - **override_mptm_interp_filter yes|no**
- If `interpolation_filter` has been changed, setting this to yes will force all MPTM modules to use that interpolation filter. If set to no, MPTM modules will play with their own interpolation filter regardless of the value of `interpolation_filter`. Defaults to no.
* - **volume_ramping**
- Sets the amount of volume ramping done by the libopenmpt mixer. The default value is -1, which indicates a recommended default value. The meaningful value range is [-1..10]. A value of 0 completely disables volume ramping. This might cause clicks in sound output. Higher values imply slower/softer volume ramps.
* - **sync_samples yes|no**
- Syncs sample playback when seeking. Defaults to yes.
* - **emulate_amiga yes|no**
- Enables the Amiga resampler for Amiga modules. This emulates the sound characteristics of the Paula chip and overrides the selected interpolation filter. Non-Amiga module formats are not affected by this setting. Defaults to yes.
- Configures the filter type to use for the Amiga resampler. Supported values are: "auto": Filter type is chosen by the library and might change. This is the default. "a500": Amiga A500 filter. "a1200": Amiga A1200 filter. "unfiltered": BLEP synthesis without model-specific filters. The LED filter is ignored by this setting. This filter mode is considered to be experimental and might change in the future. Defaults to "auto". Requires libopenmpt 0.5 or higher.
Reads 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 `libsidplayfp <https://sourceforge.net/projects/sidplay-residfp/>`_ or `libsidplay2 <https://sourceforge.net/projects/sidplay2/>`_.
- Location of your songlengths file, as distributed with the HVSC. The sidplay plugin checks this for matching MD5 fingerprints. See http://www.hvsc.c64.org/download/C64Music/DOCUMENTS/Songlengths.faq. New songlength format support requires libsidplayfp 2.0 or later.
- 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.
- Only libsidplayfp. Roms are not embedded in libsidplayfp - please note https://sourceforge.net/p/sidplay-residfp/news/2013/01/released-libsidplayfp-100beta1/ But some SID tunes require rom images to play. Make C64 rom dumps from your own vintage gear or use rom files from Frodo or VICE emulation software tarballs. Absolute path to kernal rom image file.
* - **basic**
- Only libsidplayfp. Absolute path to basic rom image file.
- Sets the data rate in bits per second. The special value "auto" lets libopus choose a rate (which is the default), and "max" uses the maximum possible data rate.
- Sets the vbr mode. Setting to "yes" (default) enables variable bitrate, "no" forces constant bitrate and frame sizes, "constrained" uses constant bitrate analogous to CBR in AAC and MP3.
* - **packet_loss**
- Sets the expected packet loss percentage. This value can be increased from the default "0" for a more redundant stream at the expense of quality.
- Configures how metadata is interleaved into the stream. If set to yes, then metadata is inserted using ogg stream chaining, as specified in :rfc:`7845`. If set to no (the default), then ogg stream chaining is avoided and other output-dependent method is used, if available.
A resampler built into :program:`MPD`. Its quality is very poor, but its CPU usage is low. This is the fallback if :program:`MPD` was compiled without an external resampler.
The `Advanced Linux Sound Architecture (ALSA) <http://www.alsa-project.org/>`_ plugin uses libasound. It is recommended if you are using Linux.
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* - 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 <http://dsd-guide.com/dop-open-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.
* - **allowed_formats F1 F2 ...**
- Specifies a list of allowed audio formats, separated by a space. All items may contain asterisks as a wild card, and may be followed by "=dop" to enable DoP (DSD over PCM) for this particular format. The first matching format is used, and if none matches, MPD chooses the best fallback of this list.
The ao plugin uses the portable `libao <https://www.xiph.org/ao/>`_ library. Use only if there is no native plugin for your operating system.
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* - 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.
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* - 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 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.
The httpd plugin creates a HTTP server, similar to `ShoutCast <http://www.shoutcast.com/>`_ / `IceCast <http://icecast.org/>`_. 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.
The null plugin does nothing. It discards everything sent to it.
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* - 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 "OpenAL" plugin uses `libopenal <http://kcat.strangesoft.net/openal.html>`_. It is supported on many platforms. Use only if there is no native plugin for your operating system.
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* - 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.
- 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.
* - **dop yes|no**
- If set to yes, then DSD over PCM according to the `DoP standard <http://dsd-guide.com/dop-open-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. Under macOS you must make sure to select a physical mode on the output device which supports at least 24 bits per channel as the Mac OS X plugin only changes the sample rate.
* - **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.
- Specifies a linear scaling coefficient (ranging from 0.5 to 5.0) to apply when adjusting volume through :program:`MPD`. For example, chosing a factor equal to ``"0.7"`` means that setting the volume to 100 in :program:`MPD` will set the PulseAudio volume to 70%, and a factor equal to ``"3.5"`` means that volume 100 in :program:`MPD` corresponds to a 350% PulseAudio volume.
- 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 <https://en.wikipedia.org/wiki/ISO_8601>`_ 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: :file:`-/.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)
- Mounts the :program:`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 :ref:`vorbis_plugin`. A list of encoder plugins can be found in the encoder plugin reference :ref:`encoder_plugins`.
The `Windows Audio Session API <https://docs.microsoft.com/en-us/windows/win32/coreaudio/wasapi>`_ plugin uses WASAPI, which is supported started from Windows Vista. It is recommended if you are using Windows.
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* - Setting
- Description
* - **device NAME**
- Sets the device which should be used. This can be any valid audio device name, or index number. The default value is "", which makes WASAPI choose the default output device.
* - **enumerate yes|no**
- Enumerate all devices in log while playing started. Useful for device configuration. The default value is "no".
* - **exclusive yes|no**
- Exclusive mode blocks all other audio source, and get best audio quality without resampling. Stopping playing release the exclusive control of the output device. The default value is "no".
