My concept with `class CancellableOperation` doesn't work properly,
because the kernel may continue to write to the given buffer as soon
as the read finishes.
To fix this, this commit adds `class ReadOperation` which owns the
buffer and the `struct iovec`. Instances of this class persist until
the read really finishes, even if the operation is canceled.
And un-deprecate "pause" without parameter (toggles pause). I have no
idea why it was deprecated long ago; the deprecation notice was copied
from the ancient MPD wiki.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/944
Similar to commit 4e2a551f30 but for
decoder plugins. This is tailored for the FFmpeg decoder plugin which
implements some protocols (e.g. RTSP) as demuxer plugin.
MPD uses soxr with prefined resample recipes. Soxr also support defining a recipe your self.
This commit will support a custom recipe by changing the existing quality setting to "custom".
The same structs as the predefined recipes uses can now set by hand.
This will make the following settings available:
- precision 16|20|24|28|32 bits, example "28"
- phase_response - 0-100, example "45"
- passband_end - used bandwidth of source 80-99.7%, example "99.7.0"
- stopband_begin - anti aliasing 100.0+%, example "100".
- attenuation - signal reduciton in dB's, 0-30. example "3.0".
- flags "0" - additional bitmask with extra settings
The data is set in the structs soxr_quality_spec and soxr_io_spec (found in soxr.h).
install_man() is currently broken with Meson and doesn't support a
custom target argument.
The problem with this kludge is that both mpd.1 and mpd.conf.5 are
installed in /usr/share/man/man1/, but apparently, there's no solution
yet.
This allows automatic optional detection of Sphinx. This will be
useful when we start building the manpages with Sphinx, which many
users may want to have.
Since Meson 0.51, there are special build options for "native:true"
builds, prefixed with "build.". This change breaks cross builds
because `GenParseName.cxx` is no longer built with `-std=c++17`.
This patch adds defaults for "build.c_std" and "build.cpp_std".
Closes https://github.com/MusicPlayerDaemon/MPD/issues/890
Before the advent of io_uring (commit dae8da7066), this didn't
matter, because the `FileInputStream` never called this. But
`UringInputStream` is derived from `AsyncInputStream`, and needs the
handler to signal completion.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/898
Passing `length+1` to `MultiByteToWideChar()` means the function may
fill the whole buffer with output data, and could theoretically
overwrite the null terminator. In practice, this will never happen,
but this way, it's slightly more correct.
Also, null-terminate after `MultiByteToWideChar()`, after we got the
real output length. Again, this would never have been a problem, but
who knows...
Commit 60f957ed64 broken the GCC 7 build, but instead of working
around missing C++17 features in old compilers, let's update the
compiler version requirements.
This commit raises the clang requirement to version 5 because this is
the first version to support `constexpr` lambdas, to be used to
`Dsd2Pcm.cxx`.
Fixes regression from commit db93bb996c because
ParseMimeTypeParameters() assumed the items were null-terminated, but
after that commit, they were not anymore.
This is the final piece of the series to establish io_uring support on
Linux.
MPD doesn't need io_uring for its efficient bulk I/O support, but to
allow file I/O to be cancelled. This is a big problem on CIFS/NFS
mounts where processes sleep uninterruptable if the file server
disappears, deadlocking MPD.
With io_uring, a flaky NFS connection allows MPD to continue to work
(even though there are still deadlocks inside MPD which need to be
addressed).
This plugin does not yet use cancellable `open()` using
`IORING_OP_OPENAT`. This will be implemented later.
Lots of other optimization opportunities for io_uring are still
missing as well - for example the database update could benefit a lot,
but unfortunately, io_uring doesn't have `readdir()` support just yet.
Reduces unstripped size. stripped size is the same.
Also took the time to remove using std::placeholders.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The parser implemented in libmpdclient requires the first key-value pair
of the server response to be the file pair. This is due to the fact that
libmpdclient scan pairs sequentially and first attempts to extract the
file pair before parsing the currentsong response further. See:
5c751a761e/src/song.c (L559-L563)
Meta data encoded as pairs in the currentsong response will be ignored
if they are placed before the file pair in the response.
std::all_of becomes constexpr in C++20. I'm not sure it results in better
performance.
Found with useStlAlgorithm
Signed-off-by: Rosen Penev <rosenp@gmail.com>
_exit and std::_Exit are identical, expect the latter is standard C++.
Added several functions to the std namespace as a result of headers.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
This has nothing to do with uClibc. It has everything to do with gcc's
libstdc++.
C99 math can be compile time disabled for it. Check for that and use boost
lround when std is not available.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
lrint is a configurable version of lround that behaves either as round,
floor, ceil, or trunc based on setting the proper FE_ macro using
fset/getround. Given that it's not set at all and that it defaults to
round behavior, simply replace with round.
Also removed the util/Math defines.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The entire section falls under the else path of #ifdef _WIN32. Checking
for it makes no sense. Probably some refactoring mistake.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
exp10 is a GNU function, is not part of C++, and is not available
everywhere.
pow(10,x) is an alternative that works just as well. It is used in musl as
the implementation of exp10.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The former is deprecated by C++14. The standard says they are the same:
The header defines all types and macros the same as the C standard library
header<stdint.h>.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The former is deprecated with C++14. The standard says both are the same:
The contents and meaning of the header<cstddef>are the same as the C
standard library header<stddef.h>,except that it does not declare the type
wchar_t, that it also declares the type byte and its associated
operations (21.2.5), and as noted in 21.2.3 and 21.2.4.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The former was deprecated in C++14. The Standard says they are the same:
The contents of the header<cstdarg>are the same as the C standard library
header<stdarg.h>, with the following changes: The restrictions that ISO C
places on the second parameter to the va_start macro in header<stdarg.h>
are different in this International Standard. The parameter parmN is the
rightmost parameter in the variable parameter list of the function
definition (the one just before the...).219If the parameter parmN is a
pack expansion (17.5.3) or an entity resulting from a lambda capture
(8.1.5), the program is ill-formed, no diagnostic required. If the
parameter parmN is of a reference type, or of a type that is not
compatible with the type that results when passing an argument for which
there is no parameter, the behavior is undefined.
Also changed va_list to the std:: namespace version, which is the same.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The former was deprecated with C++14. The standard says they are the same:
The contents of the header<csignal>are the same as the C standard library
header<signal.h>.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The former was deprecated with C++14. The standard says they are the same
with one exception:
The header<climits>defines all macros the same as the C standard library
header<limits.h>.
[Note:The types of the constants defined by macros in<climits>are not
required to match the types to which themacros refer.— end note]
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The former has been deprecated by C++14. They are also the same.
From the standard:
The contents and meaning of the header<cinttypes>are the same as the C
standard library header<inttypes.h>, with the following changes:
-The header<cinttypes>includes the header<cstdint>instead of<stdint.h>,and
—if and only if the typeintmax_tdesignates an extended integer type
(6.7.1), the following functionsignatures are added:intmax_t
abs(intmax_t);imaxdiv_t div(intmax_t, intmax_t);which shall have the same
semantics as the function signaturesintmax_t imaxabs(intmax_t)andimaxdiv_t
imaxdiv(intmax_t, intmax_t), respectively.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The former is deprecated by C++14. It's also functionally the same.
From the standard:
19.4
The header<cerrno>is described in Table 43. Its contents are the same as
the POSIX header<errno.h>,except that errno shall be defined as a macro.
[Note: The intent is to remain in close alignment with the POSIX
standard.] A separate errno value shall be provided for each thread.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
None of the functions in these files come from ctype.h
Also changed one instance of isdigit to the C++ variant.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
The former was deprecated with C++14.
According to the C++11 and C++17 standards, both files are identical.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
This doesn't work because IterableSplitString() returns its elements
by value.
Fixes clang warning:
loop variable 'i' is always a copy because the range of type 'IterableSplitString' (aka 'BasicIterableSplitString<char>') does not return a reference [-Werror,-Wrange-loop-analysis]
This reverts commit c84bae739a. A
configuration option is not necessary, because the PcmConvert
constructor knows already whether integer or floating point is needed.
This also reverts the previous commit which was wrong. When the
Vorbis decoder is disabled, we can't compile VorbisComments.cxx at
all.
Instead of expanding the #ifdef, this commit moves VorbisComments.cxx
to a separate library with dependencies on libvorbis (which was
missing previously, which could also lead to build failures if the
libvorbis headers were in a non-standard directory).
On linux-rt, kernel IRQ threads are configured with priority=50, and
this change configures MPD somewhat below that priority, leaving some
room for other programs to be configured in between.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/643
This commit adds a PlaylistPlugin attribute "as_folder" which for now
is only enabled in the "CUE" playlist plugin (which handles separate
"*.cue" files). If a playlist with this flag set is being scanned
during database update, it will be parsed and its contents will be
added to the database. This allows clients to inspect them like
directories and its contents will be searchable.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/39
This attribute is not a URI; it is just the filename without its
parent directory path. To avoid confusion, let's rename it to
"filename", leaving the struct without a "uri" attribute.
No longer allocate it as a "VarSize". This used to be a clever trick
to save memory 10 years ago, but these days, keeping the code
maintainable seems more important than saving a few kilobytes of
memory.
This reverts commit 58d7804d66. It
caused a use-after-free bug when Client::OnSocketError() was called
due to a failed write, e.g. if the output buffer was full.
This optimization is useless because sane pthread_cond_signal()
implementations check the number of waiters and do not invoke a system
call if there are none.
SID files are generally collections of tunes, so a SID name field makes
sense as an MPD album. The SID tune information list (STIL) has name
and title fields for individual tunes, when such are known, but MPD is
currently not using the STIL.
Genres are not part of the SID format, so SID files are genreless. This
"default_genre" option may be used to assign a default genre to all SID
music, for example "SID", "C64", "Chiptune", etc.
This is useful in multiple mpd instances scenario, or multiple pulse outputs defined on the same mpd instance.
It is actually a more flexible way to route flows than the "sink" parameter, letting the PulseAudio routing do its job, but with the ability to isolate routing for each output.
If not specified, the role remains like it was before this commit, ie "music"
Applying software volume to S16 samples means several bits of
precision are lost; at 25% volume, two bits are lost. Additionally,
dithering adds some noise.
The problem gets worse when you apply the software volume code twice:
for the software mixer volume, and again for the replay gain. This
loses some more precision and adds even more dithering noise, which
can become audible (see
https://github.com/MusicPlayerDaemon/MPD/issues/542).
By converting everything to 24 bit, we need to shift only two bits to
the right instead of ten, losing nearly no precision, and dithering is
not needed. Even if the output device is unable to play S24 directly,
we can convert back to S16 with only one stage of dithering.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/542
Pass only the amount of data to PcmExport::Export() when its full
output fits into the ring buffer. Using only a part of the
PcmExport::Export() result may cause data corruption because
PcmExport's internal state may contain partial blocks which would need
to be rolled back when only some of its output data was used.
As a side effect, this fixes an assertion failure because
PcmExport::CalcInputSize() considered partial block data and could
cause Play() to return a number larger than the "size" parameter.
Fix src/ls.cxx to only print unique schemas.
Refactor src/ls.cxx to use src/input/InputPlugin functionality.
Add dynamic enumeration support to curl plugin.
This gives MPD more control, because attempts to avoid having partial
periods in the ALSA period buffer. For example, this means that
DrainInternal() doesn't need to generate silence to fill the partial
period.
Add a `constexpr` constructor and several `constexpr` methods to
construct a DecoderPlugin at compile time, in a way which allows
adding new methods later without having to edit each plugin.
Eliminates a number of allocations, because callers don't need to copy
the strings to a newly allocated buffer only to null-terminate them.
And most callers don't need to have a null-terminated string.
Don't call Seek() if the stream is already at the beginning. This
avoids unnecessary exceptions if seeking is not implemented by an
Inputstream implementation.
PluginUnconfigured exceptions are logged with level "info" instead of
"error". This suppresses some rather boring messages in the default
log level.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/565
This is the documented value, but for unknown reasons, "info" was
really the default.
This was never noticed because there are only very few "info" level
messages.
If a read error occurs, it is very unlikely that the InputStream will
ever recover. Removing the code removes some code complexity which
just isn't worth it. And it allows supporting multiple readers for
one buffer.
On Windows, we keep using our own implementations, because GCC
implements std::mutex and std::condition_variable with pthread
emulation, which is not a good choice.
using the device "default" brings this plugin into line with the AlsaOutputPlugin; and a sample rate of 48kHz is more widely used as a native default for modern hardware than 44.1kHz
Also fixes an inconsistency between the docs and code.
MPD is a daemon for playing music. Music is played through the configured audio output(s) (which are generally local, but can be remote). The daemon stores info about all available music, and this info can be easily searched and retrieved. Player control, info retrieval, and playlist management can all be managed remotely.
MPD searches for a config file in ``$XDG_CONFIG_HOME/mpd/mpd.conf``
then ``~/.mpdconf`` then ``/etc/mpd.conf`` or uses ``CONF_FILE``.
Read more about MPD at http://www.musicpd.org/
OPTIONS
-------
..program:: mpd
..option:: --help
Output a brief help message.
..option:: --kill
Kill the currently running mpd session. The pid_file parameter must be specified in the config file for this to work.
..option:: --no-config
Don't read from the configuration file.
..option:: --no-daemon
Don't detach from console.
..option:: --stderr
Print messages to stderr.
..option:: --verbose
Verbose logging.
..option:: --version
Print version information.
FILES
-----
:file:`~/.mpdconf`
User configuration file.
:file:`/etc/mpd.conf`
Global configuration file.
SEE ALSO
--------
:manpage:`mpd.conf(5)`, :manpage:`mpc(1)`
BUGS
----
If you find a bug, please report it at https://github.com/MusicPlayerDaemon/MPD/issues/
@@ -27,7 +27,7 @@ The default plugin. Stores a copy of the database in memory. A file is used for
proxy
-----
Provides access to the database of another :program:`MPD` instance using libmpdclient. This is useful when you run mount the music directory via NFS/SMB, and the file server already runs a :program:`MPD` instance. Only the file server needs to update the database.
Provides access to the database of another :program:`MPD` instance using libmpdclient. This is useful when you mount the music directory via NFS/SMB, and the file server already runs a :program:`MPD` instance. Only the file server needs to update the database.
..list-table::
:widths:20 80
@@ -116,7 +116,7 @@ Provides a list of SMB/CIFS servers on the local network.
udisks
------
Queries the udisks2 daemon via D-Bus and obtain a list of file systems (e.g. USB sticks or other removable media).
Queries the udisks2 daemon via D-Bus and obtains a list of file systems (e.g. USB sticks or other removable media).
upnp
----
@@ -131,15 +131,39 @@ Input plugins
alsa
----
Allows :program:`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:
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:
..code-block::none
mpc add alsa:// plays audio from device hw:0,0
mpc add alsa:// plays audio from device default
..code-block::none
mpc add alsa://hw:1,0 plays audio from device hw:1,0 cdio_paranoia
mpc add alsa://hw:1,0 plays audio from device hw:1,0
..code-block::none
mpc add alsa://hw:1,0?format=44100:16:2 plays audio from device hw:1,0 sampling 16-bit stereo at 44.1kHz.
..list-table::
:widths:20 80
:header-rows:1
* - Setting
- Description
* - **default_device NAME**
- The alsa device id to use when none is specified in the URI.
* - **default_format F**
- 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.
cdio_paranoia
-------------
@@ -326,6 +350,8 @@ faad
Decodes AAC files using libfaad.
.._decoder_ffmpeg:
ffmpeg
------
@@ -395,30 +421,22 @@ Video game music file emulator based on `game-music-emu <https://bitbucket.org/m
- Description
* - **accuracy yes|no**
- Enable more accurate sound emulation.
* - **default_fade**
- The default fade-out time, in seconds. Used by songs that don't specify their own fade-out time.
is a MP4 container file (:file:`*.m4a`) which contains both ALAC and
is an MP4 container file (:file:`*.m4a`) which contains both ALAC and
DSD data. It is disabled by default, and works only if you explicitly
enable it. Without this plugin, the ALAC parts gets handled by the
`FFmpeg decoder plugin
<https://www.musicpd.org/doc/user/decoder_plugins.html#ffmpeg_decoder>`_. This
:ref:`FFmpeg decoder plugin <decoder_ffmpeg>`. This
plugin should be enabled only if you have a bit-perfect playback path
to a DSD-capable DAC; for everybody else, playing back the ALAC copy
of the file is better.
..list-table::
:widths:20 80
:header-rows:1
* - Setting
- Description
* - **gapless yes|no**
- This specifies whether to support gapless playback of MP3s which have the necessary headers. Useful if your MP3s have headers with incorrect information. If you have such MP3s, it is highly recommended that you fix them using `vbrfix <http://www.willwap.co.uk/Programs/vbrfix.php>`_ instead of disabling gapless MP3 playback. The default is to support gapless MP3 playback.
mad
---
@@ -462,7 +480,9 @@ Decodes Musepack files using `libmpcdec <http://www.musepack.net/>`_.
mpg123
------
Decodes MP3 files using `libmpg123 <http://www.mpg123.de/>`_.
Decodes MP3 files using `libmpg123 <http://www.mpg123.de/>`_. Currently, this
decoder does not support streams (e.g. archived files, remote files over HTTP,
...), only regular local files.
opus
----
@@ -472,7 +492,7 @@ Decodes Opus files using `libopus <http://www.opus-codec.org/>`_.
pcm
---
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".
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".
sidplay
-------
@@ -486,9 +506,11 @@ C64 SID decoder based on `libsidplayfp <https://sourceforge.net/projects/sidplay
* - 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 http://www.hvsc.c64.org/download/C64Music/DOCUMENTS/Songlengths.faq.
- 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.
* - **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.
* - **default_genre GENRE**
- Optional default genre for SID songs.
* - **filter yes|no**
- Turns the SID filter emulation on or off.
* - **kernal**
@@ -724,6 +746,25 @@ Valid quality values for libsoxr:
* "medium"
* "low"
* "quick"
* "custom"
If the quality is set to custom also the following settings are available:
* - Name
- Description
* - **precision**
- The precision in bits. Valid values 16,20,24,28 and 32 bits.
* - **phase_response**
- Between the 0-100, Where 0=MINIMUM_PHASE and 50=LINEAR_PHASE.
* - **passband_end**
- The % of source bandwidth where to start filtering. Typical between the 90-99.7.
* - **stopband_begin**
- The % of the source bandwidth Where the anti aliasing filter start. Value 100+.
* - **attenuation**
- Reduction in dB's to prevent clipping from the resampling process.
* - **flags**
- Bitmask with additional option see soxr documentation for specific flags.
.._output_plugins:
@@ -868,6 +909,10 @@ The jack plugin connects to a `JACK server <http://jackaudio.org/>`_.
- 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.
* - **auto_destination_ports yes|no**
- If set to *yes*, then MPD will automatically create connections between the send ports of
MPD and receive ports of the first sound card; if set to *no*, then MPD will only create
connections to the contents of *destination_ports* if it is set. Enabled by default.
* - **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.
@@ -1002,6 +1047,8 @@ The pulse plugin connects to a `PulseAudio <http://www.freedesktop.org/wiki/Soft
- Sets the host name of the PulseAudio server. By default, :program:`MPD` connects to the local PulseAudio server.
* - **sink NAME**
- Specifies the name of the PulseAudio sink :program:`MPD` should play on.
* - **media_role ROLE**
- Specifies a custom media role that :program:`MPD` reports to PulseAudio. Default is "music". (optional).
* - **scale_volume FACTOR**
- 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.
@@ -1090,11 +1137,58 @@ The "Solaris" plugin runs only on SUN Solaris, and plays via /dev/audio.
- Sets the path of the audio device, defaults to /dev/audio.
wasapi
------
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.
..list-table::
:widths:20 80
:header-rows:1
* - 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".
.._filter_plugins:
Filter plugins
==============
ffmpeg
------
Configures a FFmpeg filter graph.
This plugin requires building with ``libavfilter`` (FFmpeg).
..list-table::
:widths:20 80
:header-rows:1
* - Setting
- Description
* - **graph "..."**
- Specifies the ``libavfilter`` graph; read the `FFmpeg
@@ -44,7 +44,9 @@ ALSA is not available on Android; only the :ref:`OpenSL ES
Compiling from source
---------------------
Download the source tarball from the `MPD home page <https://musicpd.org>`_ and unpack it:
`Download the source tarball <https://www.musicpd.org/download.html>`_
and unpack it (or `clone the git repository
<https://github.com/MusicPlayerDaemon/MPD>`_):
..code-block::none
@@ -53,7 +55,7 @@ Download the source tarball from the `MPD home page <https://musicpd.org>`_ and
In any case, you need:
* a C++14 compiler (e.g. gcc 6.0 or clang 3.9)
* a C++17 compiler (e.g. GCC 8 or clang 5)
*`Meson 0.49.0 <http://mesonbuild.com/>`__ and `Ninja
<https://ninja-build.org/>`__
* Boost 1.58
@@ -185,47 +187,6 @@ ABI is the Android ABI to be built, e.g. ":code:`arm64-v8a`".
This downloads various library sources, and then configures and builds :program:`MPD`.
systemd socket activation
-------------------------
Using systemd, you can launch :program:`MPD` on demand when the first client attempts to connect.
:program:`MPD` comes with two systemd unit files:a "service" unit and
a "socket" unit. These will be installed to the directory specified
with :code:`-Dsystemd_system_unit_dir=...`,
e.g. :file:`/lib/systemd/system`.
To enable socket activation, type:
..code-block::none
systemctl enable mpd.socket
systemctl start mpd.socket
In this configuration, :program:`MPD` will ignore the :ref:`listener
settings <listeners>` (``bind_to_address`` and ``port``).
systemd user unit
-----------------
You can launch :program:`MPD` as a systemd user unit. These will be
installed to the directory specified with
:code:`-Dsystemd_user_unit_dir=...`,
e.g. :file:`/usr/lib/systemd/user` or
:file:`$HOME/.local/share/systemd/user`.
Once the user unit is installed, you can start and stop :program:`MPD` like any other service:
..code-block::none
systemctl --user start mpd
To auto-start :program:`MPD` upon login, type:
..code-block::none
systemctl --user enable mpd
Configuration
*************
@@ -260,6 +221,13 @@ another file; the given file name is relative to the current file:
include "other.conf"
You can use :code:`include_optional` instead if you want the included file
to be optional; the directive will be ignored if the file does not exist:
..code-block::none
include_optional "may_not_exist.conf"
Configuring the music directory
-------------------------------
@@ -336,6 +304,44 @@ The following table lists the input options valid for all plugins:
More information can be found in the :ref:`input_plugins` reference.
.._input_cache:
Configuring the Input Cache
^^^^^^^^^^^^^^^^^^^^^^^^^^^
The input cache prefetches queued song files before they are going to
be played. This has several advantages:
- risk of buffer underruns during playback is reduced because this
decouples playback from disk (or network) I/O
- bulk transfers may be faster and more energy efficient than loading
small chunks on-the-fly
- by prefetching several songs at a time, the hard disk can spin down
for longer periods of time
This comes at a cost:
- memory usage
- bulk transfers may reduce the performance of other applications
which also want to access the disk (if the kernel's I/O scheduler
isn't doing its job properly)
To enable the input cache, add an ``input_cache`` block to the
configuration file:
..code-block::none
input_cache {
size "1 GB"
}
This allocates a cache of 1 GB. If the cache grows larger than that,
older files will be evicted.
You can flush the cache at any time by sending ``SIGHUP`` to the
:program:`MPD` process, see :ref:`signals`.
Configuring decoder plugins
---------------------------
@@ -419,7 +425,7 @@ The following table lists the audio_output options valid for all plugins:
:ref:`oss_plugin` and PulseAudio :ref:`pulse_plugin`), the
software mixer, the ":samp:`null`" mixer (allows setting the
volume, but with no effect; this can be used as a trick to
implement an external mixer :ref:`external_mixer`) or no mixer
implement an external mixer, see:ref:`external_mixer`) or no mixer
(:samp:`none`). By default, the hardware mixer is used for
devices which support it, and none for the others.
* - **filters "name,...**"
@@ -714,8 +720,9 @@ 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).
* - **audio_buffer_size SIZE**
- Adjust the size of the internal audio buffer. Default is
:samp:`4 MB` (4 MiB).
Zeroconf
^^^^^^^^
@@ -768,17 +775,17 @@ Real-Time Scheduling
On Linux, :program:`MPD` attempts to configure real-time scheduling for some threads that benefit from it.
This is only possible you allow :program:`MPD` to do it. This privilege is controlled by :envvar:`RLIMIT_RTPRIO`:envvar:`RLIMIT_RTTIME`. You can configure this privilege with :command:`ulimit` before launching :program:`MPD`:
This is only possible if you allow :program:`MPD` to do it. This privilege is controlled by :envvar:`RLIMIT_RTPRIO`:envvar:`RLIMIT_RTTIME`. You can configure this privilege with :command:`ulimit` before launching :program:`MPD`:
..code-block::none
ulimit -HS -r 50; mpd
ulimit -HS -r 40; mpd
Or you can use the :command:`prlimit` program from the util-linux package:
..code-block::none
prlimit --rtprio=50 --rttime=unlimited mpd
prlimit --rtprio=40 --rttime=unlimited mpd
The systemd service file shipped with :program:`MPD` comes with this setting.
@@ -796,10 +803,10 @@ You can verify whether the real-time scheduler is active with the ps command:
PID TID CLS RTPRIO COMMAND
16257 16257 TS - mpd
16257 16258 TS - io
16257 16259 FF 50 rtio
16257 16259 FF 40 rtio
16257 16260 TS - player
16257 16261 TS - decoder
16257 16262 FF 50 output:ALSA
16257 16262 FF 40 output:ALSA
16257 16263 IDL 0 update
The CLS column shows the CPU scheduler; TS is the normal scheduler; FF and RR are real-time schedulers. In this example, two threads use the real-time scheduler: the output thread and the rtio (real-time I/O) thread; these two are the important ones. The database update thread uses the idle scheduler ("IDL in ps), which only gets CPU when no other process needs it.
@@ -814,6 +821,89 @@ The CLS column shows the CPU scheduler; TS is the normal scheduler; FF and RR ar
Using MPD
*********
Starting and Stopping MPD
-------------------------
The simplest (but not the best) way to start :program:`MPD` is to
simply type::
mpd
This will start :program:`MPD` as a daemon process (which means it
detaches from your terminal and continues to run in background). To
stop it, send ``SIGTERM`` to the process; if you have configured a
``pid_file``, you can use the ``--kill`` option::
mpd --kill
The best way to manage :program:`MPD` processes is to use a service
manager such as :program:`systemd`.
systemd
^^^^^^^
:program:`MPD` ships with :program:`systemd` service units.
If you have installed :program:`MPD` with your operating system's
package manager, these are probably preinstalled, so you can start and
stop :program:`MPD` this way (like any other service)::
systemctl start mpd
systemctl stop mpd
systemd socket activation
^^^^^^^^^^^^^^^^^^^^^^^^^
Using systemd, you can launch :program:`MPD` on demand when the first client attempts to connect.
:program:`MPD` comes with two systemd unit files:a "service" unit and
a "socket" unit. These will be installed to the directory specified
with :code:`-Dsystemd_system_unit_dir=...`,
e.g. :file:`/lib/systemd/system`.
To enable socket activation, type:
..code-block::none
systemctl enable mpd.socket
systemctl start mpd.socket
In this configuration, :program:`MPD` will ignore the :ref:`listener
settings <listeners>` (``bind_to_address`` and ``port``).
systemd user unit
^^^^^^^^^^^^^^^^^
You can launch :program:`MPD` as a systemd user unit. These will be
installed to the directory specified with
:code:`-Dsystemd_user_unit_dir=...`,
e.g. :file:`/usr/lib/systemd/user` or
:file:`$HOME/.local/share/systemd/user`.
Once the user unit is installed, you can start and stop :program:`MPD` like any other service:
..code-block::none
systemctl --user start mpd
To auto-start :program:`MPD` upon login, type:
..code-block::none
systemctl --user enable mpd
.._signals:
Signals
-------
:program:`MPD` understands the following UNIX signals:
-``SIGTERM``, ``SIGINT``: shut down MPD
-``SIGHUP``: reopen log files (send this after log rotation) and
flush caches (see :ref:`input_cache`)
The client
----------
@@ -953,6 +1043,22 @@ is no way for :program:`MPD` to find out whether the DAC supports
it. DSD to PCM conversion is the fallback if DSD cannot be used
directly.
ICY-MetaData
------------
Some MP3 streams send information about the current song with a
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