Previously, inode numbers were truncated to 32 bits, which could lead
to problems on XFS where inodes are 64 bit; this could lead to bogus
"recursive directory found" errors during database update.
[mk: added commit description and NEWS line]
Closes https://github.com/MusicPlayerDaemon/MPD/issues/2000
ICU 76 decided to reduce overlinking[^1] thus `icu-i18n` will no longer
add `icu-uc` when linking to shared libraries. This results in failure:
```
src/lib/icu/libicu.a.p/Converter.cxx.o: undefined reference to symbol 'ucnv_fromUnicode_76'
```
[^1]: 199bc82702
Closes https://github.com/MusicPlayerDaemon/MPD/issues/2151
This reverts commit 94b5b9f370. It was
not necessary for branch v0.23.x because there, Break() is
thread-safe; this was only changed later by commit
a3b32819b1
snd_pcm_poll_descriptors_revents() may return any error code; the
ALSA docs do not constrain the permitted values. A 'hw' device
will only ever return an error if the pfd array passed in is
invalid (-EINVAL), but other I/O plugins may return arbitary
errors. For example a network-based device may return -EPIPE etc.
The resulting exception thrown by
AlsaNonBlockPcm::DispatchSockets() must be caught to prevent the
mpd process from being aborted.
Some ALSA capture devices can have very large buffers, holding 10
seconds or more audio. Using the maximum buffer size with such
devices leads to unacceptably large, and unnecessary, latency.
Also, some ALSA drivers (e.g. HDA Intel PCH) report an invalid
maximum period size, and the period size that mpd calculates from
the maximum buffer size results in "Invalid argument" error when
applying the hw_params. Note that the "default" capture device on
many cards includes the "dsnoop" plugin which imposes a buffer
size of 16384 frames, so that "alsa://" works OK but
"alsa://plughw" or "alsa://hw" both fail.
Limit the maximum buffer time for ALSA input devices to a more useable
2 seconds, thereby avoiding both the above problems.
6d91b5c7b2 ("fix double promotions") changed
how LAME peak values are decoded, producing large incorrect values that
cause some MP3 files to play silently.
Restore the original decode from MAD fixed-point format to double and
document what it's doing.
Fixes#1823
Let OnHeaders() check the status.
The status checking code was added by commit 4f021cbced in 2011,
but in 2008, commit a8e81326d0 enabled `CURLOPT_FAILONERROR`, which
means the status checking code never had any effect.
This allows `LoadExcludeListOrLog()` to hide boring "404 Not Found"
log messages via `IsFileNotFound()`.
The `assert(!quit)` can fail if the `EventThread` gets stopped before
it enters `EventLoop::Run()`. There is a similar problem with `alive`,
which gets reset by `EventThread::Stop()`.
If that happens, then `EventLoop::Run()` should return immediately
without handling any events.
Without clearing all `in_playlist` flags, the songs will never be
revealed again if they were hidden once by a CUE sheet, not even after
the CUE sheet gets deleted or modified.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1784
This adds `MakeFfmpegError()` to the executable and fixes a linker
failure when `libavutil` is available, but `libavformat` and
`libavcodec` are not.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1786
Otherwise, building will generate these warnings:
systemd/system/meson.build:5: WARNING: Project targeting '>= 0.56.0' but tried to use feature deprecated since '0.56.0': Dependency.get_pkgconfig_variable. use Dependency.get_variable(pkgconfig : ...) instead
systemd/user/meson.build:5: WARNING: Project targeting '>= 0.56.0' but tried to use feature deprecated since '0.56.0': Dependency.get_pkgconfig_variable. use Dependency.get_variable(pkgconfig : ...) instead
libfmt 10 doesn't know how to format a StringView, and doesn't cast to
std::string_view anymore. The StringView class has been removed from
MPD 0.24 completely, and this is a stable-branch-only workaround.
Closes https://github.com/MusicPlayerDaemon/MPD/pull/1814
This should prevent ffmpeg from taking priority over the gme plugin.
The ffmpeg plugin is more buggy than gme.
One of the prominent bugs of preferring ffmpeg over gme is that ffmpeg
cannot seek SAP files while gme can. This should prevent that from
happening.
libfmt version 10 apparently doesn't know how to format a
`StringBuffer`, failing the MPD build. Since we have a formatter
specialization for `AudioFormat`, let's use that - it's better and
easier to use.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1807
Fixes hide_playlist_targets not working after server restart
Currently, `hide_playlists_targets` works by skipping songs with
`in_playlist` value set to true in
[`Directory::Walk`](a57bcd0238/src/db/plugins/simple/Directory.cxx (L237)). But
`in_playlist` is not stored and only updated in
[`UpdateWalk::PurgeDanglingFromPlaylists`](a57bcd0238/src/db/update/Playlist.cxx (L139)),
which will only be executed while updating DB.
This causes the problem that playlist target songs are correctly
hidden after database update, but will remain visible after mpd server
restarted. This pr solves the problem by storing `in_playlist` value
of songs into the `SimpleDatabase` file.
By default, if the parent of a process dies, the process gets SIGHUP
and is supposed to shut down. This however doesn't work for MPD,
because MPD redefines SIGHUP with a different meaning (like most
daemons do).
To work around this, we configure the kernel to send SIGTERM instead
of SIGHUP.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1706
Fixes a busy loop in BufferingInputStream::RunThreadLocked() because
the method never learns that seeking is ignored, even though the HTTP
stream is already broken and can never be read; nobody cared to check
for errors.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1727
If the environment variable $HOME does not exist, don't attempt to
obtain it from /etc/passwd; without $HOME, the calling process
indicates that it does not wish MPD to access the home directory.
This also prevents MPD from attempting to load
`/root/.config/mpd/mpd.conf` if MPD got started as global systemd
service. Reading from there makes no sense, only /etc/mpd.conf shall
be used then.
This piece of code was initially added by commit 5d85792178.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1687
When drop_start_samples and drop_end_samples overlap and are greater
than the actual number of samples, the `num_samples` calculation in
SubmitPCM() could underflow.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1712
Fix build with Boost 1.81.0. `<array>` was included by one of those boost headers,
however, it's no longer included as of Boost 1.81.0.
`master` doesn't use `std::array` in this file.
While we're at it, add all necessary inclusion files.
These settings do not work if CURL was compiled with
CURL_DISABLE_PROXY, and cause error "An unknown option was passed in
to libcurl".
Fixes regression by commit 7ab0dfc8ce
Of course, mode=0700 is more secure, but allowing other users access
to new directories is a choice the user should make via umask(). If
the user-chosen umask allows everybody access, MPD should probably
respect that.
The function spl_valid_name() should verify playlist names and prevent
path traversal, but it failed to do so on Windows, because it forgot
to check for backslashes.
This buggy piece of code was already present when stored playlists
were initially implemented in 2006 by commit 08003904d7, and
even during the many rounds of code refactoring, nobody ever bothered
to verify it. D'oh!
(Thanks, Paul Arzelier)
snprintf() does not return the (truncated) length actually written,
but the length that would be needed if the buffer were large enough.
This API usage mistake in FormatLastError() can lead to overflow of
the stack buffer, crashing the process (Windows only).
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1676
If no archive library was found, return from the "plugins" directory
without creating "libarchive_plugins.a". Empty static libraries are
unsupported on some operating systems such as macOS.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1650
According to the latest WebDAV specification (RFC4918),
timestamp string in the getlastmodified property is formatted
as rfc1123-date, such as "Sun, 06 Nov 1994 08:49:37 GMT".
However, to process responses from servers in the older style
format specified in RFC2518, timestamps in the HTTP-date format
had better be accepted.
As described in the libcurl api documentation, curl_getdate() can handle
timestamp strings in HTTP-date formats, including rfc1123-date.
https://www.rfc-editor.org/rfc/rfc4918#section-15.7https://www.rfc-editor.org/rfc/rfc2518.html#section-13.7https://curl.se/libcurl/c/curl_getdate.html
This fixes a bug introduced in 87fa6bca where the FLAC encoder fails to
initialize unless libFLAC is built with Ogg support. When libFLAC is
built without Ogg support, FLAC__stream_encoder_set_ogg_serial_number
unconditionally returns false.
Calling data[fill] could trigger an assertion failure if
fill==data.size(), even if we call it only to take the address.
Instead of doing that, this commit changes the code to pointer
arithmetic.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1556
occurrences of the word Master/master with Main/main. This change was test built on two different systems.
1. macos 10.15.7 with Xcode 12.4 and clang 12.0.0 on x86_64
2. macos 12.5 with Xcode 13.4.1 and clang 13.1.6 on arm64 (Apple silicon M1)
It should be noted that on macos 10.15.7 with Xcode 11.2 and clang 11.0, MPD will not build.
The MPD documentation states that clang 11.0 is the minimum requirement,
but clang 11.0 produces compile errors. Apparently the macos version
of clang 11.0 is not fully compliant.
Properly deals with iconv, unlike the current solution. have_iconv fails
when libiconv CFLAGS are passed to the compiler. Tested under OpenWrt
with its CONFIG_BUILD_NLS, which adds libiconv include flags.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
Deactivate the stream in Cancel(). This fixes stuttering after a
manual song change by refilling the whole ring buffer before
reactivating the stream.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1354
Clear not only MPD's ring buffer, but also libpipewire's buffers, to
avoid playing some audio from the previous song after a manual song
change.
Fixes part 1 of https://github.com/MusicPlayerDaemon/MPD/issues/1354
Removing the LockHasClients(); this code was copied from the "httpd"
output plugin, but unlike "httpd", the SnapCast output plugin does not
feed silence while paused, so we need to implement a delay to avoid
busy-looping the CPU.
As a side effect, this eliminates the suttering after resuming
playback, because the timer now gets reset even if there is a client.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1394
Since PipeWire 0.3.53, there is no control name anymore, therefore the
name check doesn't work anymore, breaking volume change events.
This obsoletes the crash bug fix in commit 2ee57f9b0d
If the PipeWire output has not yet been enabled and no thread_loop has
been created yet, a nullptr dereference in SetVolume() was possible
because nullptr was passed to pw_thread_loop_lock().
If a mixer is not open, rethrow the original exception each time
setting the volume is requested. This further improves error messages
sent to MPD clients.
This fixes a std::terminate() crash in the CURL storage plugin when
PropfindOperation::OnHeaders() throws an exception after receiving a
non-207 status.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1559
strlen() and strcpy() are provided by the <string.h> and <cstring>
headers (as functions in global and std namespaces, respectively).
Compilers MAY provide an implementation for either of the functions
without including the extra header but the existence of a declaration
without the header is not assured.
This issue occurs when playing HLS streaming delivered
from a server that does not support partial requests.
The issue is reproduced as follows(using Ubuntu 20.04 PC):
1. Prepare HLS example content.
$ mkdir test
$ ffmpeg -i example.flac -vn -c:a aac -b:a 128000 -f hls -hls_list_size 0 test/output.m3u8
(ffmpeg 4.2.4 is used)
2. Prepare web server without partial requests support.
(Docker version 20.10.12 and NGINX official Docker image is used)
$ docker run --name tmp-nginx-container -d nginx
$ docker cp tmp-nginx-container:/etc/nginx/conf.d/default.conf .
$ docker rm -f tmp-nginx-container
Edit default.conf and add "max_ranges 0;" to "location / {...}".
This disables partial requests support,
removes 'Accept-Ranges: bytes' header from the server response.
Then, run the server:
$ docker run --name test-nginx -v $PWD/test:/usr/share/nginx/html:ro -v $PWD/default.conf:/etc/nginx/conf.d/default.conf -d -p 8080:80 nginx
3. Setup MPD to Play the next URL.
http://address-of-the-server:8080/output.m3u8
Seeking this stream results in "exception: Not seekable".
Even if the "wave_encoder" option is disabled (and no other encoder
plugins are enabled), forcefully enable the Wave encoder (if Snapcast
is enabled).
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1500
This script setup a dummy android native app folder and call ndk-gdb from it.
It needs a modification in ANDROID_NDK since ndk-gdb may attach to the wrong
pid, cf. comments in the script.
Previous versions of MPD would call SetVolume on enabled outputs before
they are ready, causing all of MPD to crash. Checking the really_enabled
flag prevents this, though it also prevents setting volume before the
player starts.
Before (with the PipeWire output):
[i] ~$ mpc clear
volume: 81% repeat: off random: off single: off consume: off
[i] ~$ systemctl --user restart mpd.service
[i] ~$ mpc volume 100
MPD error: Connection closed by the server
[i] ~ 1 $
After:
[i] ~$ # mpd is freshly started w/o anything in the queue
[i] ~$ mpc
volume:100% repeat: off random: off single: off consume: off
[i] ~$ mpc volume 80
MPD error: problems setting volume
[i] ~ 1 $ mpc
volume:100% repeat: off random: off single: off consume: off
[i] ~$
Previous versions of MPD would, on parameter change, set the PipeWire
volume before clearing the restore_volume flag, causing the call to
short circuit and do nothing. Instead, clear the flag before the call.
Move audio output state check ahead of mixer check and force volume
applying even for disabled software mixed outputs.
This fixes incorrect software mixer volume that used to occur when
volume was changed while output being disabled.
This is easily reproduced with following sequence of commands on
multi-output software mixed MPD setup.
mpc volume 38; mpc disable 3; mpc volume 88; mpc enable 3
On current MPD, following commands would result in output 3 playing at
volume 38, while all other enabled outputs would play at volume
88. Moreover, global volume would display average of outputs real
volumes. In my case, it's 75.
After applying this patch, following commands would produce expected
behavior. All outputs play at expected (88) volume. And volume is
correctly displayed as 88.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1423
Signed-off-by: Vitaly Ostrosablin tmp6154@yandex.ru
Signed-off-by: Vitaly Ostrosablin <tmp6154@yandex.ru>
In libFLAC 0.3.4 (commit c9530118a4), the "dllimport" check has been
changed from "_MSC_VER" to "_WIN32", and now the MPD build is affected
by it.
Defining FLAC__NO_DLL disables the use of "dllimport", which allows
linking properly to the static libFLAC build.
Wasapi output plugin won't start playing after being paused
The cause is that the scope guard in the WASAPI work thread
(WasapiOutputPlugin.cxx, function WasapiOutputThread::Work(), in the
while (true) loop) is set up too 'late' in the execution. There is one
condition ("if (data_in_frames >= buffer_size_in_frames)") when it is
hit, the loop will continue without executing the scope guard. This
scope guard is responsible for emptying the buffer again, and if the
buffer is not emptied, the above mentioned condition will stay true.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1451
ffmpeg from current git master no longer exposes
av_malloc() nor av_free() through other included
headers. directly include libavutil/mem.h to fix
compilation with (as-yet-unreleased) ffmpeg.
The writing and reading of metadata involves lots of different programs
and libraries. Therefore it is prudent to point out how exactly MPD
receives metadata. Ideally this helps to point users to the right place
if their tags are not picked up correctly.
lib/src/libmpdclient-2.19/meson.build:1:0: ERROR: Unknown options: "needs_exe_wrapper"
The "needs_exe_wrapper" option was incorrectly set under
[built-in options] rather than [properties].
Improves the changes made in 57687779be by
using AudioManager.ACTION_AUDIO_BECOMING_NOISY rather than listening for
wired headset unplug events or Bluetooth headset disconnect events. This
method is more flexible, allowing the feature to work on other types of
audio output device, as well as Bluetooth devices that don't set their
device class correctly. This change also has the benefit of being more
responsive, pausing the audio before it is rerouted to the built-in
speaker.
https://developer.android.com/guide/topics/media-apps/volume-and-earphones
With large "max_playlist_length" settings, the "data" array can be
very large, and initializing it during MPD startup causes page faults,
resulting in allocation of physical RAM. This commit postpones the
initialization until the queue is really large, to avoid wasting
memory.
SonarLint reports the latter to be better:
std::scoped_lock basically provides the same feature as std::lock_guard,
but is more generic: It can lock several mutexes at the same time, with a
deadlock prevention mechanism (see {rule:cpp:S5524}). The equivalent code
to perform simultaneous locking with std::lock_guard is significantly more
complex. Therefore, it is simpler to use std::scoped_lock all the time,
even when locking only one mutex (there will be no performance impact).
Signed-off-by: Rosen Penev <rosenp@gmail.com>
This reverts commit 552c30eae4.
It has caused various problems; for example, MPD wasn't able to write
the pid_file (which was already mitigated by commit a4e4217204).
And apparently, the socket file created in the same directory by
mpd.socket disappears when mpd.service (re)creates the directory. I
could not reproduce this problem with 247.3, but maybe this is a bug
in older systemd versions?
Until we figure out why this happens, let's remove the
RuntimeDirectory directive. A future MPD version may be launched as
regular user, not as root, which will eliminate one major problem with
RuntimeDirectory.
Add an option to the UPnP database plugin to configure which interface
is used by upnp to discover servers.
upnp by default selects the first interface that is not loopback, which
in some cases might not be the desired interface. For example if wanting
to access a DLNA server over a VPN connection.
The "interface" option can now be set to the name of the desired
interface to achieve this.
The default behaviour remains unchanged.
Adds the Interface Name as an argument to the *Init functions to make it
possible to select which interface is used by upnp to detect servers.
Currently "nullptr" is passed in to let the upnp library select an
interface, as before.
Commit 552c30eae caused problems for those people who still had a
"pid_file" setting (even though that is obsolete with systemd),
because now /run/mpd is owned by root:root (our mpd.service has no
User=mpd directive, so systemd starts MPD as root).
To work around this problem, and to be able to keep
RuntimeDirectory=mpd (which solved a problem of other MPD users), the
best compromise seems to just ignore the "pid_file" setting when it is
of no use.
This way, MPD can reliably detect whether it was started as systemd
service, which is better than checking sd_booted(), which only checks
whether systemd manages all services, but still MPD could be started
manually.
This uncomments the code which had been present already in the first
Snapcast commit (copied from the "httpd" output plugin), but I
commented it because I did not know whether I needed to send silence
samples to all Snapcast clients.
As a side effect, this fixes playback when no Snapcast client is
connected; this was broken because Pause() always returned a positive
value when there were no clients.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1310
Unlike GetFilenameSuffix(), uri_get_suffix() removes the query string
first, which breaks file names with question marks in the name.
Therefore, uri_get_suffix() shall only be applied to remote URIs.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1316
Without clearing the "uri" field, the next Open() call attempts to
reuse the old InputStream, but it has already been closed, so Open()
always returns nullptr.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1300
With libfmt versions older than 7, this leads to an endless recursion
between Error() and FmtError(), resulting in a crash due to stack
overflow. D'oh!
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1295
During the libfmt migration, I converted "%1.3f" to just "{:1.3}"
without the "f" suffix, but libfmt defaults to scientific notation,
which can break some MPD clients.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1291
This replaces the output parameter (which is bad API design). As a
side effect, it fixes the bad [[gnu::pure]] attribute added by commit
a636d2127 which caused optimizing compilers to miscompile calls to
that function. "Pure" functions can be assumed to have no output
arguments, so the compiler can assume the function doesn't modify
them.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1282
Fixes https://github.com/MusicPlayerDaemon/MPD/discussions/1281
The problem occurred when there was libfmt-dev installed, but it was
too old (e.g. on Debian Buster), and Meson used the wrap fallback.
Those internal MPD libraries where the libfmt dependency was not
declared were still using the old system libfmt headers, which are not
ABI-compatible with MPD's own libfmt build.
From the feature request: "I generally like to have crossfade on, but
when it happens during such short tracks (e.g. 20 seconds or less) it
doesn't really sound good as those tracks are not really meant to be
crossfaded and intended to act as a bridge on their own."
Sounds reasonable. This commit doesn't add an option, but hard-codes
the limit to 20 seconds. If it turns out that users want to have it
configurable, we can still add the option.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1184
Now, "+0" means "right after the current song" and "-0" means "right
before the current song". Mnemonic: there are zero songs between the
current song and the newly added song.
A similar feature was present long ago in MPD, but was deprecated in
version 0.16 because the implementation was broken. This commit
re-adds the feature in a way that's well-defined and not broken.
Close https://github.com/MusicPlayerDaemon/MPD/issues/1221
NB: Check the sysconfdir setting to determine where mpd will look for mpd.conf; if you expect mpd to look for /etc/mpd.conf the sysconfdir must be '/etc' (i.e., not 'etc' which will result in mpd looking for /usr/local/etc/mpd.conf):
.. code-block:: none
meson configure output/release |grep sysconfdir
If this is not /etc (or another path you wish to specify):
.. code-block:: none
$ meson configure output/release -Dsysconfdir='/etc' ; meson configure output/release |grep syscon
sysconfdir /etc Sysconf data directory
The ChainFilter class is extremely complicated code, and will grow to
be even more complicated when the Filter interface gets extended.
Let's just remove it; we can easily chain many TwoFilters instead.
Don't truncate the FILETIME to second resolution to pass it to
std::chrono::system_clock::from_time_t(); instead, calculate the
offset between the FILETIME epoch and the
std::system_clock::time_point epoch, and use that to initialize the
time_point directly.
If draining was not requested, generate silence instead if there is no
data in the ring buffer.
The problem is that pw_stream_flush() appears to disable the stream
permanently, even though there is no state_changed callback - the
stream state remains at PW_STREAM_STATE_STREAMING, but the stream is
defunct. I have no idea why and I havn't found any documentation
about it.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1219
Explanation
This adds support for DOP using the PcmExport function if the macro
ENABLE_DSD is defined. If enabled within the config-file using "dop",
the boolean dop_setting will be true. If DSD input is encountered and
the setting is on, it is checked whether the oss-device supports the
required samplerate. If that is the case, dop_active is set to true
and conversion of the input is prevented. If the sample rate is not
supported, conversion to S32 is requested. When playing back, the
PcmExport is used to pack the incoming stream into PCM. Reasoning
This is required for OSs without the required driver support for
native DSD playback that also have no ALSA. Mainly *BSD users are the
target audience for this functionality, as ALSA here is only a proxy
without full functionality. Requirements
DAC that supports the DOP standard
Building with OSS, DSD and S32-Format
Supported Formats / Required PCM Formats
DSF, DFF and WavPack-DSD will work.
DSD64, 1 Channel -> S24:176.4kHz (untested, lack of time / missing samples)
DSD64, 2 Channel -> S24:352.8kHz
DSD64, 4 Channel -> S24:705.6kHz (untested, lmissing equipment)
DSD128, 1 Channel -> S24:352.8kHz (untested, lack of time / missing samples)
DSD128, 2 Channel -> S24:705.6kHz
DSD256, 1 Channel -> S24:705.6kHz (untested, lack of time / missing samples)
Changes
inclusion of required files
adding new domain for logging
adding dop_satisfied private function
adding required member variables for storing dop state and for dop-packing
adding dop boolean parameter to many functions that are required to act a little differently when dop is active
Testing
This has been tested to work with a Sabaj Da2 on FreeBSD, where the
red status indicator LED clearly shows that DSD playback is taking
place, instead of purple for "hi-res" which is seen when converting.
Issues
I have not tested this with S24 and right now AFMT_S32_NE is
required. If not defined, ENABLE_DSD will be undef'ed. This will be
addressed in a bit, however no DAC which supports DOP but not 32Bit is
known to me. Also, AFMT_S32_NE is not defined when building on FreeBSD
which is why this is just blatantly defined in the file at the moment.
Additionally, the new dop-option is not added into any documentation
whatsoever.
Fixes
../git/src/Log.hxx:121:42: error: no member named 'forward' in namespace 'std'
LogFormat(LogLevel::ERROR, e, fmt, std::forward<Args>(args)...);
Signed-off-by: Khem Raj <raj.khem@gmail.com>
refactors GenerateOggSerial into a generic GenerateSerial
utility, under the util lib.
libFLAC may be encoded without Ogg support. If Ogg support is disabled,
libFLAC will still export Ogg-related methods (like setting a serial
number), and throw a runtime error when initializing an Ogg stream.
GenerateOggSerial does not depend on libogg. Refactoring it into
a generic GenerateSerial prevents having to add build-time checks
for libogg within the FLAC encoder plugin.
Add support for the following tags when using the proxy database plugin:
WORK
CONDUCTOR
LABEL
GROUPING
MUSICBRAINZ_WORKID
COMPOSERSORT
ENSEMBLE
MOVEMENT
MOVEMENTNUMBER
LOCATION
This commit adds some tags that are (mostly) interesting for listeners
of classical music.
Ensemble
--------
This is an ensemble that is playing the music, such as Wiener
Philharmoniker. The tag can be used to distinguish the ensemble from the
conductor, composer, soloist, and ensemble, that are generally all in
the "ARTIST" tag.
Movement
-------
The movement number and movement (name) of this track, i.e. "II" and
"Allegro".
ComposerSort
------------
Allows us to look for Beethoven's 9th under B, for Beethoven, not L for
Ludwig.
Location
--------
This is the location of the recording, e.g. "Wiener Musikverein".
By casting to SafeLinkIntrusiveListHook if appropriate, this fixes a
bug in the erase() method, where erase() calls
IntrusiveListHook::unlink() instead of
SafeLinkIntrusiveListHook::unlink().
Add cacert option for curl plugin
add cacert option for Curl plugin. Allows to set cacert for curl lib
Added documentation line into doc/plugins.rst with explanation for cacert option
If an InjectEvent callback schedules a timer, the loop will restart,
but the "busy" flag is still false. The fix is to move the "again"
check before the "busy" setting.
SocketEvent knows the FD is still open and is about to close it, so
it's unnecessary to rely on the kernel (via AbandonFD) to clean up the
epoll_wait list.
### Why this is relevant
- `AbandonFD` assumes that upon closing the socket, the FD will be automatically removed from the epoll list. That fd is associated with a reference to the `SocketEvent`, so this is an important and dangerous assumption to get wrong. In the case that the FD isn't immediately removed from the list by the kernel, the event loop can crash due to the `SocketEvent` being destroyed and it being a use-after-free bug at that point.
- If a socket FD happens to be duplicated, then closing the SocketEvent FD will not automatically remove it from epoll, and will trigger said bug/crash. It is only automatically removed when all FD references to the underlying socket/resource are closed?
- A `fork()` is one example where a socket FD can be duplicated and result in this situation.
- `CLOEXEC` might be considered mitigation for this but also introduces a race condition where the crash can occur between a `fork()` and `exec()` without additional synchronization to freeze the event loop.
One could argue the mpd event loop isn't fork-safe, and thus should be allowed to use `AbandonFD` however it likes. A decision on whether this is intended should probably be declared; but either way this fix seems appropriate in cases where `Abandon` isn't actually necessary. It also might be possible to fix `AbandonFD` to mark the `SocketEvent` as removed without using `EPOLL_CTL_DEL`?
[edit: made this dependent on HAVE_THREADED_EVENT_LOOP which is always
true for MPD, but not for ncmpc, for example - mk]
This reverts commit 7bc1c9925b. It
caused a crash with the ALSA plugin family (through
MultiSocketMonitor::ReplaceSocketList() and
MultiSocketMonitor::AddSocket()). Until we have a proper fix, the
assertion patch is reverted.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1020
Instead of using this as a base class implementing a virtual method,
the new class IdleEvent can be used as a variable, decoupling
IdleMonitor's internal state from the derived class.
This is similar to commit 30a5dd267b
which refactored TimeoutMonitor to TimerEvent.
@@ -23,11 +23,23 @@ The default plugin. Stores a copy of the database in memory. A file is used for
- 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).
* - **hide_playlist_targets yes|no**
- Hide songs which are referenced by playlists? Thas is,
playlist files which are represented in the database as virtual
directories (playlist plugin setting ``as_directory``). This
option is enabled by default and avoids duplicate songs; one
copy for the original file, and another copy in the virtual
directory of a CUE file referring to it.
proxy
-----
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.
Provides access to the database of another :program:`MPD` instance
using `libmpdclient
<https://www.musicpd.org/libs/libmpdclient/>`_. This is useful when
you mount the music directory via NFS/SMB, and the file server already
runs a :program:`MPD` (0.20 or newer) instance. Only the file server
needs to update the database.
..list-table::
:widths:20 80
@@ -49,6 +61,15 @@ upnp
Provides access to UPnP media servers.
..list-table::
:widths:20 80
:header-rows:1
* - Setting
- Description
* - **interface**
- Interface used to discover media servers. Decided by upnp if left unconfigured.
Storage plugins
===============
@@ -185,6 +206,11 @@ Plays audio CDs using libcdio. The URI has the form: "cdda://[DEVICE][/TRACK]".
- 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.
* - **speed N**
- Request CDParanoia cap the extraction speed to Nx normal CD audio rotation speed, keeping the drive quiet.
* - **mode disable|overlap|full**
- Set the paranoia mode; ``disable`` means no fixups, ``overlap``
performs overlapped reads, and ``full`` enables all options.
* - **skip yes|no**
- If set to ``no``, then never skip failed reads.
curl
----
@@ -193,8 +219,9 @@ 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 :file:`~/.curlrc`
will be in effect.
variables such as ``http_proxy``will be in effect.
User name and password are read from an optional :file:`~/.netrc`, :file:`~/.curlrc` is not read.
..list-table::
:widths:20 80
@@ -210,6 +237,8 @@ will be in effect.
- Verify the peer's SSL certificate? `More information <http://curl.haxx.se/libcurl/c/CURLOPT_SSL_VERIFYPEER.html>`_.
* - **verify_host yes|no**
- Verify the certificate's name against host? `More information <http://curl.haxx.se/libcurl/c/CURLOPT_SSL_VERIFYHOST.html>`_.
* - **cacert**
- Set path to Certificate Authority (CA) bundle `More information <https://curl.se/libcurl/c/CURLOPT_CAINFO.html>`_.
ffmpeg
------
@@ -451,9 +480,39 @@ Module player based on MODPlug.
* - Setting
- Description
* - **resampling_mode nearest|linear|spline|fir**
- 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".
* - **loop_count**
- Number of times to loop the module if it uses backward loops. Default is 0 which prevents looping. -1 loops forever.
openmpt
-------
Module player based on `libopenmpt <https://lib.openmpt.org>`_.
..list-table::
:widths:20 80
:header-rows:1
* - Setting
- Description
* - **repeat_count**
- 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.
* - **stereo_separation**
- 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.
* - **emulate_amiga_type**
- 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.
mpcdec
------
@@ -548,6 +607,10 @@ Encodes into `FLAC <https://xiph.org/flac/>`_ (lossless).
- Description
* - **compression**
- Sets the libFLAC compression level. The levels range from 0 (fastest, least compression) to 8 (slowest, most compression).
* - **oggflac yes|no**
- Configures if the stream should be Ogg FLAC versus native FLAC. Defaults to "no" (use native FLAC).
* - **oggchaining yes|no**
- Configures if the stream should use Ogg Chaining for in-stream metadata. Defaults to "no". Setting this to "yes" also enables Ogg FLAC.
lame
----
@@ -612,11 +675,15 @@ Encodes into `Ogg Opus <http://www.opus-codec.org/>`_.
* - 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.
- 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.
* - **complexity**
- Sets the `Opus complexity <https://wiki.xiph.org/OpusFAQ#What_is_the_complexity_of_Opus.3F>`_.
* - **signal**
- Sets the Opus signal type. Valid values are "auto" (the default), "voice" and "music".
* - **vbr yes|no|constrained**
- 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.
* - **opustags yes|no**
- 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.
@@ -784,6 +851,16 @@ The `Advanced Linux Sound Architecture (ALSA) <http://www.alsa-project.org/>`_ p
- 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.
* - **stop_dsd_silence yes|no**
- If enabled, silence is played before manually stopping playback
("stop" or "pause") in DSD mode (native DSD or DoP). This is a
workaround for some DACs which emit noise when stopping DSD
playback.
* - **thesycon_dsd_workaround yes|no**
- If enabled, enables a workaround for a bug in Thesycon USB
audio receivers. On these devices, playing DSD512 or PCM
causes all subsequent attempts to play other DSD rates to fail,
which can be fixed by briefly playing PCM at 44.1 kHz.
* - **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.
@@ -959,6 +1036,8 @@ On Linux, OSS has been superseded by ALSA. Use the ALSA output plugin :ref:`alsa
- 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.
* - **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.
The according hardware mixer plugin understands the following settings:
@@ -1021,6 +1100,28 @@ The pipe plugin starts a program and writes raw PCM data into its standard input
* - **command CMD**
- This command is invoked with the shell.
pipewire
--------
Connect to a `PipeWire <https://pipewire.org/>`_ server. Requires
``libpipewire``.
..list-table::
:widths:20 80
:header-rows:1
* - Setting
- Description
* - **target NAME**
- Link to the given target. If not specified, let the PipeWire
manager select a target. To get a list of available targets,
type ``pw-cli ls Node``
* - **remote NAME**
- The name of the remote to connect to. The default is
``pipewire-0``.
* - **dsd yes|no**
- Enable DSD playback. This requires PipeWire 0.38.
.._pulse_plugin:
pulse
@@ -1076,8 +1177,6 @@ You must set a format.
- Sets the host name of the `ShoutCast <http://www.shoutcast.com/>`_ / `IceCast <http://icecast.org/>`_ 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".
@@ -286,10 +286,15 @@ The following tags are supported by :program:`MPD`:
***date**: the song's release date. This is usually a 4-digit year.
***originaldate**: the song's original release date.
***composer**: the artist who composed the song.
***composersort**: same as composer, but for sorting.
***performer**: the artist who performed the song.
***conductor**: the conductor who conducted the song.
***work**: `"a work is a distinct intellectual or artistic creation,
which can be expressed in the form of one or more audio recordings" <https://musicbrainz.org/doc/Work>`_
***ensemble**: the ensemble performing this song, e.g. "Wiener Philharmoniker".
***movement**: name of the movement, e.g. "Andante con moto".
***movementnumber**: movement number, e.g. "2" or "II".
***location**: location of the recording, e.g. "Royal Albert Hall".
***grouping**: "used if the sound belongs to a larger category of
sounds/music" (`from the IDv2.4.0 TIT1 description
<http://id3.org/id3v2.4.0-frames>`_).
@@ -474,7 +479,7 @@ Querying :program:`MPD`'s status
current song in seconds, but with higher resolution.
-``duration``[#since_0_20]_: Duration of the current song in seconds.
-``bitrate``: instantaneous bitrate in kbps
-``xfade``: ``crossfade`` in seconds
-``xfade``: ``crossfade`` in seconds (see :ref:`crossfading`)
-``mixrampdb``: ``mixramp`` threshold in dB
-``mixrampdelay``: ``mixrampdelay`` in seconds
-``audio``: The format emitted by the decoder plugin during
@@ -514,17 +519,19 @@ Playback options
.._command_crossfade:
:command:`crossfade {SECONDS}`
Sets crossfading between songs.
Sets crossfading between songs. See :ref:`crossfading`.
.._command_mixrampdb:
:command:`mixrampdb {deciBels}`
Sets the threshold at which songs will be overlapped. Like crossfading but doesn't fade the track volume, just overlaps. The songs need to have MixRamp tags added by an external tool. 0dB is the normalized maximum volume so use negative values, I prefer -17dB. In the absence of mixramp tags crossfading will be used. See http://sourceforge.net/projects/mixramp
Sets the threshold at which songs will be overlapped.
See :ref:`mixramp`.
.._command_mixrampdelay:
:command:`mixrampdelay {SECONDS}`
Additional time subtracted from the overlap calculated by mixrampdb. A value of "nan" disables MixRamp overlapping and falls back to crossfading.
See :ref:`mixramp`.
.._command_random:
@@ -538,12 +545,31 @@ Playback options
Sets repeat state to ``STATE``,
``STATE`` should be 0 or 1.
If enabled, MPD keeps repeating the whole queue (:ref:`single mode
<command_single>` disabled) or the current song (:ref:`single mode
<command_single>` enabled).
If :ref:`random mode <command_random>` is also enabled, the
playback order will be shuffled each time the queue gets repeated.
.._command_setvol:
:command:`setvol {VOL}`
Sets volume to ``VOL``, the range of
volume is 0-100.
.._command_getvol:
:command:`getvol`[#since_0_23]_
Read the volume. The result is a ``volume:`` line like in
:ref:`status <command_status>`. If there is no mixer, MPD will
emit an empty response. Example::
getvol
volume: 42
OK
.._command_single:
:command:`single {STATE}`[#since_0_15]_
@@ -665,18 +691,22 @@ Song ids on the other hand are stable: an id is assigned to a song
when it is added, and will stay the same, no matter how much it is
moved around. Adding the same song twice will assign different ids to
them, and a deleted-and-readded song will have a new id. This way, a
client can always be sure the correct song is being used.
client can always be sure the correct song is being used. Song ids are not
preserved across :program:`MPD` restarts.
Many commands come in two flavors, one for each address type.
Whenever possible, ids should be used.
.._command_add:
:command:`add {URI}`
:command:`add {URI} [POSITION]`
Adds the file ``URI`` to the playlist
(directories add recursively). ``URI``
can also be a single file.
The position parameter is the same as in :ref:`addid
<command_addid>`. [#since_0_23_3]_
Clients that are connected via local socket may add arbitrary
local files (URI is an absolute path). Example::
@@ -692,6 +722,13 @@ Whenever possible, ids should be used.
Id: 999
OK
If the second parameter is given, then the song is inserted at the
specified position. If the parameter starts with ``+`` or ``-``,
then it is relative to the current song [#since_0_23]_; e.g. ``+0``
inserts right after the current song and ``-0`` inserts right
before the current song (i.e. zero songs between the current song
and the newly added song).
.._command_clear:
:command:`clear`
@@ -715,14 +752,22 @@ Whenever possible, ids should be used.
at ``START:END``[#since_0_15]_ to ``TO``
in the playlist.
If ``TO`` starts with ``+`` or ``-``, then it is relative to the
current song; e.g. ``+0`` moves to right after the current song
and ``-0`` moves to right before the current song (i.e. zero songs
between the current song and the moved range).
.._command_moveid:
:command:`moveid {FROM} {TO}`
Moves the song with ``FROM`` (songid) to
``TO`` (playlist index) in the
playlist. If ``TO`` is negative, it
is relative to the current song in the playlist (if
there is one).
playlist.
If ``TO`` starts with ``+`` or ``-``, then it is relative to the
current song; e.g. ``+0`` moves to right after the current song
and ``-0`` moves to right before the current song (i.e. zero songs
between the current song and the moved song).
.._command_playlist:
@@ -736,8 +781,8 @@ Whenever possible, ids should be used.
.._command_playlistfind:
:command:`playlistfind {FILTER}`
Finds songs in the queue with strict
matching.
Search the queue for songs matching
``FILTER`` (see :ref:`Filters <filter_syntax>`).
.._command_playlistid:
@@ -757,8 +802,10 @@ Whenever possible, ids should be used.
.._command_playlistsearch:
:command:`playlistsearch {FILTER}`
Searches case-insensitively for partial matches in the
queue.
Search the queue for songs matching
``FILTER`` (see :ref:`Filters <filter_syntax>`).
Parameters have the same meaning as for :ref:`find
<command_playlistfind>`, except that search is not case sensitive.
.._command_plchanges:
@@ -885,19 +932,28 @@ remote playlists (absolute URI with a supported scheme).
.._command_load:
:command:`load {NAME} [START:END]`
:command:`load {NAME} [START:END] [POSITION]`
Loads the playlist into the current queue. Playlist
plugins are supported. A range may be specified to load
only a part of the playlist.
The ``POSITION`` parameter specifies where the songs will be
inserted into the queue; it can be relative as described in
:ref:`addid <command_addid>`. (This requires specifying the range
as well; the special value `0:` can be used if the whole playlist
shall be loaded at a certain queue position.) [#since_0_23_1]_
.._command_playlistadd:
:command:`playlistadd {NAME} {URI}`
:command:`playlistadd {NAME} {URI} [POSITION]`
Adds ``URI`` to the playlist
`NAME.m3u`.
`NAME.m3u` will be created if it does
not exist.
The ``POSITION`` parameter specifies where the songs will be
inserted into the playlist. [#since_0_23_3]_
.._command_playlistclear:
:command:`playlistclear {NAME}`
@@ -909,6 +965,8 @@ remote playlists (absolute URI with a supported scheme).
Deletes ``SONGPOS`` from the
playlist `NAME.m3u`.
The second parameter can be a range. [#since_0_23_3]_
An experimental Android build is available on Google Play. After installing and launching it, :program:`MPD` will scan the music in your Music directory and you can control it as usual with a :program:`MPD` client.
If you need to tweak the configuration, you can create a file called:file:`mpd.conf` on the data partition (the directory which is returned by Android's :dfn:`getExternalStorageDirectory()` API function).
If you need to tweak the configuration, you can create a file called
:file:`mpd.conf` in MPD's data directory on the external storage
@@ -110,6 +113,19 @@ The following command shows a list of compile-time options:
meson configure output/release
NB: Check the sysconfdir setting to determine where mpd will look for mpd.conf; if you expect mpd to look for /etc/mpd.conf the sysconfdir must be '/etc' (i.e., not 'etc' which will result in mpd looking for /usr/local/etc/mpd.conf):
..code-block::none
meson configure output/release |grep sysconfdir
If this is not /etc (or another path you wish to specify):
Setting this to ``album`` will adjust volume using the album's
ReplayGain tags, while setting it to ``track`` will adjust it using
the "track" ReplayGain tags. ``auto`` uses the track ReplayGain tags
if random play is activated otherwise the album ReplayGain
tags.
If ReplayGain is enabled, then the setting ``replaygain_preamp`` is
set to a value (in dB) between ``-15`` and ``15``. This is the gain
applied to songs with ReplayGain tags.
On songs without ReplayGain tags, the setting
``replaygain_missing_preamp`` is used instead. If this setting is not
configured, then no ReplayGain is applied to such songs, and they will
appear too loud.
ReplayGain is usually implemented with a software volume filter (which
prevents `Bit-perfect playback`_). To use a hardware mixer, set
``replay_gain_handler`` to ``mixer`` in the ``audio_output`` section
(see :ref:`config_audio_output` for details).
Simple Volume Normalization
^^^^^^^^^^^^^^^^^^^^^^^^^^^
MPD implements a very simple volume normalization method which can be
enabled by setting ``volume_normalization`` to ``yes``. It supports
16 bit PCM only.
.._crossfading:
Cross-Fading
------------
If ``crossfade`` is set to a positive number, then adjacent songs are
cross-faded by this number of seconds. This is a run-time setting
:ref:`which can be controlled by clients <command_crossfade>`,
e.g. with :program:`mpc`::
mpc crossfade 10
mpc crossfade 0
Zero means cross-fading is disabled.
Cross-fading is only possible if both songs have the same audio
format. At the cost of quality loss and higher CPU usage, you can
make sure this is always given by configuring
:ref:`audio_output_format`.
.._mixramp:
MixRamp
^^^^^^^
MixRamp tags describe the loudness levels at start and end of a song
and can be used by MPD to find the best time to begin cross-fading.
MPD enables MixRamp if:
- Cross-fade is enabled
-:ref:`mixrampdelay <command_mixrampdelay>` is set to a positive
value, e.g.::
mpc mixrampdelay 1
-:ref:`mixrampdb <command_mixrampdb>` is set to a reasonable value,
e.g.::
mpc mixrampdb -17
- both songs have MixRamp tags
- both songs have the same audio format (or :ref:`audio_output_format`
is configured)
The `MixRamp <http://sourceforge.net/projects/mixramp>`__ tool can be
used to add MixRamp tags to your song files.
Client Connections
------------------
@@ -625,6 +741,9 @@ By default, all clients are unauthenticated and have a full set of permissions.
- Allows reading of the database, displaying the current playlist, and current status of :program:`MPD`.
* - **add**
- Allows adding songs and loading playlists.
* - **player**
- Allows any player and queue manipulation (start/pause/stop
playback etc.).
* - **control**
- Allows all other player and playlist manipulations.
* - **admin**
@@ -633,6 +752,9 @@ By default, all clients are unauthenticated and have a full set of permissions.
:code:`local_permissions` may be used to assign other permissions to clients connecting on a local socket.
:code:`host_permissions` may be used to assign permissions to clients
with a certain IP address.
:code:`password` allows the client to send a password to gain other permissions. This option may be specified multiple times with different passwords.
Note that the :code:`password` option is not secure: passwords are sent in clear-text over the connection, and the client cannot verify the server's identity.
@@ -948,7 +1072,19 @@ The "music directory" is where you store your music files. :program:`MPD` stores
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:file:`.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.
To exclude a file from the update, create a file called
:file:`.mpdignore` in its parent directory. Each line of that file
may contain a list of shell wildcards. Matching files (or
directories) in the current directory and all subdirectories are
excluded. Example::
*.opus
99*
Subject to pattern matching is the file/directory name. It is (not
yet) possible to match nested path names, e.g. something like
``foo/*.flac`` is not possible.
Mounting other storages into the music directory
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
@@ -988,6 +1124,15 @@ See :ref:`tags` for a list of supported tags.
The :ref:`metadata_to_use <metadata_to_use>` setting can be used to
enable or disable certain tags.
Note that :program:`MPD` may not necessarily read metadata itself,
instead relying on data reported by the decoder that was used to read
a file. For example, this is the case for the FFmpeg decoder: both
:program:`MPD` and FFmpeg need to support a given metadata format in
order for metadata to be picked up correctly.
Only if a decoder does not have metadata support will :program:`MPD`
attempt to parse a song's metadata itself.
The queue
---------
@@ -1045,6 +1190,7 @@ Check list for bit-perfect playback:
* Disable sound processing inside ALSA by configuring a "hardware"
device (:samp:`hw:0,0` or similar).
* Don't use software volume (setting :code:`mixer_type`).
* Don't use :ref:`replay_gain`.
* Don't force :program:`MPD` to use a specific audio format (settings
* Verify that you are really doing bit-perfect playback using :program:`MPD`'s verbose log and :file:`/proc/asound/card*/pcm*p/sub*/hw_params`. Some DACs can also indicate the audio format.
@@ -1205,6 +1351,34 @@ Your bug report should contain:
* relevant portions of the log file (:option:`--verbose`)
* be clear about what you expect MPD to do, and what is actually happening
.._profiler:
Too Much CPU Usage
^^^^^^^^^^^^^^^^^^
If you believe MPD consumes too much CPU, `write a bug report
<https://github.com/MusicPlayerDaemon/MPD/issues>`_ with a profiling
information.
On Linux, this can be obtained with :program:`perf` (on Debian,
installed the package :file:`linux-perf`), for example::
perf record -p `pidof mpd`
Run this command while MPD consumes much CPU, let it run for a minute
or so, and stop it by pressing ``Ctrl-C``. Then type::
perf report >mpd_perf.txt
Upload the output file to the bug report.
..note::
This requires having debug symbols for MPD and all relevant
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