The script seems to assume package version numbers always end in numeric versions with an optional alpha-suffix. Alas, were it only so simple... Sometimes the package is called fizzbang-1.2.3+release_info in which case the build fails. No more!
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1177
Make sure that ZeroconfDeinit() gets called even if startup fails with
an exception. Fixes an assertion failure because an Avahi TimerEvent
is still active.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1192
These were added 11 years ago in commit 766b9fd453, but I cannot find
any evidence in the FFmpeg repository that these names were ever
supported. This commit adds the tags as they are currently present in
libavformat/mov.c.
For querying tags, the real song URI should be used, because if the
(display) URI is different, requesting it will not produce a usable
response. This is a theoretical problem because none of the existing
playlist plugins sets the real_uri.
This requires changing the URI comparison in playlist::TagModified().
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1154
With the default value CURLAUTH_ANY, libcurl needs to probe for
authentication methods first, and only the second request will have an
Authorization header.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1155
This enables the JACK output plugin on Windows, but doesn't link
against libjack64.dll, instead loads the DLL at runtime with
LoadLibrary(). This kludge avoids the extremely fragile JACK shared
memory protocol by using the system's libjack64.dll, without requiring
the same DLL at build time.
This reverts commit 1532983fb5. This
optimization was bad because now all strings match if they are a
prefix of another string, and this caused collisions in the tag string
pool, corrupting the database.
The current http output doesn't provide a header for cross-origin support. This prevents to use the mpd http stream directly from an other webapplication due the origin from the webpage differs from then the audio stream.
The fix is to add the following header to the http response:
Access-Control-Allow-Origin: *
This code is complicated - and broken: the producer thread is not
allowed to call consumer methods. Also the code is not necessary
because this plugin implements Interrupt().
This finishes problems which occur early in the WasapiOutputThread;
previously, the error was ignored and the output blocked forever
without doing anything (and without reporting the error).
If an exception has been caught, the method cannot continue playback,
therefore it doesn't make sense to have the "catch" block inside the
"while" block (and not break the loop after catching an exception).
ParseTimePoint.
%Z is a glibc extension to strptime, and is a no-op there, due to the
mapping between timezone names and their definition (especially when the
name comes from a different machine) being ambiguous / impossible. Time
in HTTP headers is guaranteed to be UTC.
Passing an unknown format to strptime() implementations that don't
support it will generally cause them to return NULL, which will lead to
ParseTimePoint throwing an exception and ParseTimeStamp using an
unnecessary fallback.
Since the timezone name goes at the end of the string, we don't need to
use %Z to skip it (could be an issue in a different time stamp format),
so simply removing %Z works best.
If the constructor moves from an ExportedSong instance which refers to
somebody else's "Tag" instance, the newly constructed instance will
instead refer to its own empty "tag_buffer" field. This broke
SimpleDatabase::GetSong(), i.e. all songs on the queue restored from
the state file or added using the "addid" command.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1089
The "move" command doesn't allow open-ended ranges because they don't
make a lot of sense; moving an open-ended range is only possible if
the destination index is before the range, and in that case, the
client should be well aware how many songs there are.
Closes https://github.com/MusicPlayerDaemon/MPD/pull/1057
After commit 1afa33c3c7, an old bug was revealed:
SimpleDatabase::GetSong() constructs an ExportedSong instance by
moving the return value of Song::Export(), which causes the
LightSong::tag field to be dangling on the moved-from
ExportedSong::tag_buffer. This broke tags from CUE sheets.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1070
Commit 79b2366387 added the field `skip`
to support unaligned reads, but set the `offset` field to a wrong
value. This resulted in miscalculation of `remaining`, causing
an assertion failure.
The fix is to assign `offset` the correct value, but consider the
`skip` value in the assertion.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1067
I havn't yet figured out how to use Android's system CA certificates
with CURL/OpenSSL, so a temporary workaround is to disable verify_peer
by default. The data MPD transfers isn't extremely important, so the
servers's authenticity isn't extremely important either.
The original base relative path was introduced due to an erroneous test
where the URL started with three slashes: "https:///" instead of two,
which led to implementing handling for such cases but broke the two
slashes case.
This fix removes the base relative path handling because with two
slashes the path is anyway always relative to the host (aka absolute
URI, without host).
This reverts 216f62ea14 and part of 74b2fc7fdc
Signed-off-by: Vincent Petry <PVince81@yahoo.fr>
Use uri_has_scheme to find out if the href in Webdav responses is absolute
to use the matching base path extraction.
Signed-off-by: Vincent Petry <PVince81@yahoo.fr>
There can be more than one propstat block each with their own status
code. We're only interested in the one with the 200 status, the found
properties.
This fixes parsing to make sure we process all propstat blocks instead
of just the last one, which might have a 404 status for not-found
properties.
Signed-off-by: Vincent Petry <PVince81@yahoo.fr>
Remove additional "a:prop" in PROPFIND request to match RFC 4918 section 9.1.3.
Added Content-Type header as the body is not a true multipart POST.
Signed-off-by: Vincent Petry <PVince81@yahoo.fr>
Some FFmpeg filters change the sample format, and since MPD assumes
this never happens, this results in loud noise instead of music. This
commit finally implements the TODO comment by sending one frame of
silence to the filter and checking the output frame's format.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1009
Pass a std::string to PathTraitsUTF8::Relative(), implicitly casting
it to std::string_view. This selects the right overload which returns
std::string_view instead of `const char *`; the latter could return
`nullptr` which would cause the implicit conversion of the return
value to std::string_view to crash.
Regression caused by commits ead208987d and a98d627c0b.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/995
This allows users to disable the "CUE files as directories" feature
without having to disable the CUE playlist plugin completely. This
feature has been annoying some users.
It's been 9 years already, and there's no point in insisting on the 21
year old C standard. MPD doesn't have a lot of C code left, but why
not compile it with the latest language revision.
Since MPD is a long-running daemon, it doesn't make sense to use
dynamic binding. That allows the relocations to be read-only
("relro"), which a hardening feature.
The bug https://bugzilla.samba.org/show_bug.cgi?id=11413 makes MPD
crash after at most a minute of using the plugin. Since this bug is
five years old already and it doesn't look like it will ever be fixed,
all libsmbclient code in MPD is scheduled for removal. For now, the
plugin is disabled by default so people are less likely to hit the
crash bug.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/991
This got lost in commit 5d597a3646 (v0.21.19), but it was never
noticed because the state_file_interval was way too short due to
commit 3413d1bf23, fixed recently by commit 27cc7b352d
snprintf() is available on mingw, and the libnfs kludge broke the
build with mingw, because sprintf_s() was now both an inline function
and a "dllimport" function (because the macro renamed the inline
function snprintf() to sprintf_s() in mingw's stdio.h).
Fixes regression by commit 23d5a2b862 -
that commit always pretended that any Opus file has both track and
album gain, and thus disabled the fallback to the other if one is not
set.
This patch changes the logic to only submit ReplayGain if at least one
value is set, and apply the offset only to that value. If none is
available, then the new check in HandleAudio() will submit only the
output gain.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/977
By bit-wise ANDing the reported flags with GetScheduledFlags(),
ERROR/HANGUP always get cleared. This means the MPD event loop could
never report those conditions.
Oh no, 3413d1bf23 was broken! Instead of passing a number as
"seconds" to the duration constructor, it just abused the duration
constructor as cast operator, which caused custom state_file_interval
settings to be extremely short.
This allows canceling the blocking method LockWaitWriteAvailable(),
and thus allows breaking free of misbehaving ALSA drivers, avoiding a
MPD lockup.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/966
This allows interrupting the output thread (for some plugins which
implement this method). This way, operations can be canceled
properly, instead of waiting for some external entity.
clocale is part of C++11.
In practical terms, gcc's libstdc++ comes with its own locale defines
when the libc does not have them.
Also reworked to be dependent on !ANDROID.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
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.
This fixes a spurious "single" mode bug which occurs when using "play"
or "seek" to start playback on the song that is currently paused: in
that case, the main thread never queues the next song, and at the end
of the song, the player thread exits Run(), stopping playback, and
after that, the main thread starts the next song without considering
"single" mode.
By calling OnPlayerSync(), we ensure that the main thread gets a
chance to queue the next song before the player thread exits the Run()
loop.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/850
The log levels have always been very confusing (and badly named), but
this was most confusing: if there's a log level called "default", why
is it not the default?
Closes https://github.com/MusicPlayerDaemon/MPD/issues/926
Oh the horror! This plugin cannot possibly ever have worked. It was
broken from the start, when it was added in commit 37796699cf nearly
twelve (!) years ago.
The plugin would always read at sector boundaries, so it could only
ever work at multiples of 2 kB.
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).
This fixes the Windows build. Linking failed because some packages
(e.g. libFLAC) default to enabling `_FORTIFY_SOURCE`, which is broken
in recent mingw versions
(https://github.com/msys2/MINGW-packages/issues/5803).
While libsndfile doesn't like partial reads in the middle of a file
(see commit 95ac6071b9), it allows partial reads at the end of a file.
It doesn't pay attention to the file size when issuing a read.
Commit ecb67a1ed1 (MPD 0.18.12) was a regression: previously,
partial reads at the end of a file were possible, but switching to
decoder_read_full() made this an error condition. This way, a portion
at the end of each file was lost, leading to corruption with gapless
playback (https://github.com/MusicPlayerDaemon/MPD/issues/936).
This fix switches to the newly introduced function
decoder_read_much(), which does the same as the code before commit
ecb67a1ed1.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/936
Our AudioObjectGetPropertyDataT() wrapper throws exception on error,
and calling it from OSXOutput::Disable() can cause MPD crash due to
std::terminate().
Closes https://github.com/MusicPlayerDaemon/MPD/issues/932
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.
Mounting one storage URI twice on different mount points can lead to
conflicts with the database cache file, and it doesn't make a lot of
sense.
But most importantly, our udisks storage plugin will unmount the disk
from the kernel VFS, and if two exist, they will compete with each
others. We could (and should) fix this in the udisks storage plugin,
but for now, this workaround is good enough (and useful).
When mounting something over a directory that is already a mount
point, CompositeStorage::Mount() silently overwrites the previously
mounted storage, disposing it. After that, SimpleDatabase::Mount()
will fail and handle_mount() will roll back the
CompositeStorage::Mount() command, effectively unmounting what was
there before (and also leaking memory).
Closes https://github.com/MusicPlayerDaemon/MPD/issues/918
Bug #915 is about an I/O exception thrown where none was allowed,
leading to crash via std::terminate(). However, instead of catching
and logging the error inside the decoder plugin, it should be able to
propagate the I/O error to the MPD core, so MPD can avoid trying other
decoder plugins.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/915
This will keep track of AudioOutputUnitStart() and
AudioOutputUnitStop(). This will provide some separation between "not
(yet) (re)started" and "paused".
The formula in osx_output_score_sample_rate() to detect multiples of
the source sample rate was broken: when given a 44.1 kHz input file,
it preferred 16 kHz over 48 kHz, because its `frac_portion(16)=0.75`
is smaller than `frac_portion(48)=0.91`.
That formula, introduced by commit 40a1ebee29, looks completely
wrong. It doesn't do what the code comment pretends it does.
Instead of using that `frac_portion` to calculate a score, this patch
adds to the score only if `frac_portion` is nearly `0` or `1`. This
means that the factor is nearly integer.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/904
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
A bug report (https://github.com/MusicPlayerDaemon/MPD/issues/912)
suggests that on Linux, reading on `cifs` files may rarely return 0 (=
end of file) before the end of the file has really been reached. But
that's just a theory which I need to validate, so this runtime check
shall catch this condition before the assertion in
DecoderBridge::Read() crashes MPD. Let's see.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/912
An assertion failure in UpdateQueuedSong() could trigger because the
`prev` parameter is always `nullptr`, but `queued` may be set. And in
fact, calling UpdateQueuedSong() is only necessary when the queued
song was edited, to re-queue it with the new range.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/901
When the client wants to seek, but the decoder has already finished
decoding the current song, the player restarts the decoder with an
initial seek at the new position. When this initial seek fails, MPD
pretends nothing has happened and plays this song from the start.
With this new flag, a restarted decoder marks the initial seek as
"essential" and fails the decoder if that seek fails.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/895
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...
This branch isn't yet ready for level 3 (`-Wpedantic`) due to several
C++ violations (e.g. variable length arrays). These are already
cleaned up in the master branch (0.22).
The WildMidi project added the pkg-config file in version 0.3.3, but
unfortunately, Debian still doesn't ship it 4 years later:
https://bugs.debian.org/916631
However, for cross-compiling, the pkg-config file is very helpful.
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 command is used to configure the client's connection, and this
shouldn't require any permissions. The client should be able to do
that before sending a password.
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.
libnfs is compiled with `-D_FILE_OFFSET_BITS=64`, but Meson decides
not to enable this mode. We could force this mode, but then again,
these days, nobody should be using 32-bit Windows ... so this is a
kludge only for debugging with 32-bit WINE.
This fixes a freeze bug in the NFS input/storage plugins: when libnfs
auto-reconnets after a failure, it installs the new socket on the same
file descriptor number. MPD's attempt to unregister the old socket by
calling SocketMonitor::Steal() from NfsConnection::ScheduleSocket()
fails because the new/old socket number is not registered in epoll, so
epoll_ctl() returns ENOENT. The problem is that it left
`scheduled_flags`, and so subsequent Schedule() calls will use
`EPOLL_CTL_MOD`, which will fail again and again. Instead, we need to
use `EPOLL_CTL_ADD` to register the new socket.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/806
Closes https://github.com/MusicPlayerDaemon/MPD/issues/756
LINGUISTIC_IGNORECASE is unimplemented on Wine, but since we don't
have any locale support (yet), and we're using LOCALE_NAME_INVARIANT,
NORM_IGNORECASE should essentially be the same, so why bother.
Unescape the base path and the path coming from the server (href) to fix the
comparison when the server uses different escaped characters.
The outputted name need to be unescaped. Doing that before or after the
HrefToEscapedName() call should not change the current behavior.
If the file name is "Hello & bye", 3 CharacterData events will be sent with the
State::HREF state:
- "Hello%20"
- "&"
- "%20bye"
Reproduced with files hosted on an apache2 DAV server: 2.4.38-3+deb10u3.
The _GLIBCXX_USE_C99_MATH macro is defined in glibcxx by c++config.h, which
gets included by every header. Which means a header needs to be present.
(cherry picked from commit 79e9aff338)
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>
This method gets called a lot during MPD startup, via FindChild() and
directory_load_subdir(), so this is worth optimizing at the expense of
code readability.
This speeds up MPD startup by 10%.
This reverts commit eb192137d6.
This is no longer necessary because we require FFmpeg 3.1 or newer
since MPD 0.21.2.
This fixes a deprecation warning because the implicit AVPacket copy
constructor copies the deprecated attribute `convergence_duration`.
_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 seems to be required on recent Android versions (tested with Android 10).
This is also required for android TV services (cf. next commit).
This is done using Java reflection so that the project doesn't depend on
android compat libs.
This is the case with uClibc-ng currently.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
(cherry picked from commit 769cd0ee9f0cf8ceb026aa751b5d4a390bb5dbdc)
(changed define to match master)
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>
Documentation says the limit is 5, but it was really 10 (at least
since 2004). But since MPD wants to promote using many small clients
idling around, and these clients consume only very few resources, it
seems reasonable to raise this limit's default value.
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>
Aparently, libcdio sometimes returns empty filenames, causing MPD
crashes. This shouldn't really happen, and I consider this a libcdio
bug - but if it happens, people blame MPD, so let's add a check.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/776
Fixes
../src/time/ISO8601.cxx:67:24: error: use of undeclared identifier 'strtoul'
unsigned long value = strtoul(s, &endptr, 10);
^
../src/time/ISO8601.cxx:77:14: error: use of undeclared identifier 'strtoul'
minutes = strtoul(s, &endptr, 10);
^
on NetBSD with clang 9.0.0.
Using libgme 0.6.2 on macOS, it appears that gme_info_t strings can be
empty, which creates weird track titles: (001/050)
This adds an additional check for an empty string.
"The issue is that ParseCommandArgSignedSongTime parses with
SongTime::FromS, not SignedSongTime::FromS, before casting back to a
SignedSongTime for the return. With x86 overflow rules this doesn't
matter, but on ARM the first cast turns negative values to zero."
Closes https://github.com/MusicPlayerDaemon/MPD/issues/757
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]
When calling OggSeekFindEOS() from inside a OggVisitor callback, then
the #InputStream may be in the middle of an Ogg packet, and the newly
initialized #ogg_sync_state will not be able to load it without the
help of ogg_sync_pageseek(). By passing "synced=false" to
OggSeekFindEOS(), we force the use of ogg_sync_pageseek() even when
not actually seeking.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/719
This reverts commit c84bae739a. A
configuration option is not necessary, because the PcmConvert
constructor knows already whether integer or floating point is needed.
It appears that [[fallthrough]] is valid in C++ but not in C. And
in some Clang versions (e.g. Clang 11 on macOS), Clang is pedantic
about this and considers it an error to use [[fallthrough]] in a
.c file such as src/util/format.c.
This changes makes gcc_fallthrough a no-op under Clang in C files.
This reverts commit 4475b8ca04. Further
testing revealed that the threaded resolver still uses a timeout of
0ms. This revert however lowers the bound to a minimum of 1ms instead
of 10ms.
This prepares the migration away from strptime() for Windows
portability.
But the real reason I'm doing this is that strptime() on Apple is
buggy: strptime("14", "%H%M%S") (without separating colons) succeeds
even though only the hour has been parsed. This fixes recent Travis
failures in the ParseISO8601() unit test.
Older clang versions don't support the GCC __attribute__ syntax. For
those, don't use anything at all, and new clang versions shall use the
standard syntax.
Expand $PATH at evaluation and not at assignment, which fixes the
problem that /usr/lib/ccache was added to $PATH between the
MATRIX_EVAL assignment and its evaluation.
This can cause request completion in the I/O thread before this
constructor returns, leaving the object in an abstract state, causing
a crash due to pure virtual method call. We should not start the
request until this object is fully constructed.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/665
The ALSA "null" driver opens /dev/null and returns the file handle
from snd_pcm_poll_descriptors(), but /dev/null cannot be used with
epoll, the epoll_ctl() system call returns -EPERM. This means that
the ALSA output hangs, eventually freezing the whole MPD process.
This commit adds a workaround to the MultiSocketMonitor class which is
used by the ALSA output plugin.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/695
Most importantly, this commit translates ZZIP_ENOENT to
std::system_error(ENOENT) so IsFileNotFound() returns true and
find_stream_art() can suppress the log line.
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).
Commit 13208bf5a7 added range support to
the `move` command, but applied the wrong offset to the `to` variable.
When the source range is before the current song, and the song thus
gets decremented by the range size, then the final destination offset
must also be decremented by the range size.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/663
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
Without requesting the property, "good" WebDAV servers would not send
it, and so MPD could never recognize a directory, failing the database
update.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/660
Apparently, libmpcdec sets gain/peak variables to zero if they are not
present. This clashes with our formula and results in bogus values
which cause noise during playback.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/640
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 header had been available for a long time on Linux, but was
removed in glibc 2.30. This commit moves the `#include` line inside
the `#ifdef __sun` block and adds a fake declaration of `I_FLUSH` for
the Linux build.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/630
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.
The new Response instance in the `catch` block didn't have the
`command` attribute set, so the error response didn't indicate which
command had failed, which however is required in the MPD protocol.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/628
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.
High Voltage SID Collection (HVSC) metadata fields are encoded in
windows-1252, as described in DOCUMENTS/SID_file_format.txt:
https://www.hvsc.c64.org/download/C64Music/DOCUMENTS/SID_file_format.txt
If utf-8 transcoding fails, or the ICU library is unavailable, fall
back to plain ASCII and replace other characters with '?'.
Version 10.2+0.93+1 was released five years ago in 2014 and is the
first version to feature cdio_cddap_free_messages(). There is no way
to check the libcdio-paranoia version at compile time, so let's just
remove support for older versions instead of attempting to fix the
cdio_cddap_free_messages() check at build time.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/613
This reverts commit f7ed7446ae. It was
a bad idea, because MAD_F_MIN and MAD_F_MAX do not represent the
clamping limits, but the theoretical minimum and maximum values of the
mad_fixed_t data type.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/617
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"
The check IsSeekableCurrentSong() was added by commit
44b200240f in version 0.20.19, but it
caused a regression: by doing the branch only if the current song is
seekable, the player would restart the current song if it was not
seekable, and later the initial seek would fail; but we already know
it's not seekable, and so we should fail early.
libmad has a hard-coded maximum PCM buffer size; if we make our
output_buffer just as large, we can avoid the loop, because any
possible size will fit.
libmad requires padding the input buffer with "MAD_BUFFER_GUARD" zero
bytes at the end of the file, or else it is unable to decode the last
frame.
This fixes yet another bug which prevented this plugin from decoding
the last frame, see
https://github.com/MusicPlayerDaemon/MPD/issues/601
The Xing/LAME frame indicates how many frames there are, but that
excludes the initial Xing/LAME frame. Therefore, it should not be
counted.
This fixes an off-by-one bug which caused the last frame to be
skipped, fixing one part of
https://github.com/MusicPlayerDaemon/MPD/issues/601
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.
MPD used to do that when this code lived in the player thread, but it
was removed by commit 98a7c62d7a4f716d90af6d78e18d1a3b10bc54b3; and
the replacement code in the ALSA output plugin didn't have it.
Without this timer, DispatchSockets() may disable the
MultiSocketMonitor and if Play() doesn't get called soon, it never
gets a chance to generate silence. However if Play() gets called,
generating silence isn't necessary anymore...
Resulting from this misdesign (added by commit ccafe3f3cf in 0.21.3),
the silence generator didn't work reliably.
In DispatchSockets(), when there was not enough data, but enough for
current playback, the method would disable the "active" flag so the
next Play() call would re-enable the MultiSocketMonitor.
This was an abuse of the flag which could result in a crash
in Cancel(), because that method asserts that the period_buffer is
empty, which it may be not.
The solution is to add anther flag called "waiting" which shares some
behavior with the old flag.
Apparently, if snd_pcm_drain() returns EAGAIN, it does not actually
want to be called again; the next call will snd_pcm_drain() will also
return EAGAIN, forever, even though the PCM state has meanwhile
switched to SND_PCM_STATE_SETUP. This causes a busy loop; to fix
this, we should always check snd_pcm_state() to see if draining is
really required.
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.
Commit 1eae9339f2 added support for
multiple "groups" in the "list" command, and this change allows
clients to detect that this behavior, which had been documented for
several years, is now implemented properly.
Workaround for a regression caused by commit
a06bf388d9, revealing a problem with
discarding odd numer of frames in the DSD_U32 and DoP converters,
causing distortions with DSD_U32 and DoP on 32 bit CPUs.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/469
This can happen if the DoP converter doesn't get enough source samples
for one destination quad. This isn't a critical bug, because the OSS
plugin doesn't support DoP yet, but it's good to be prepared.
Instead of passing tag and group, pass an array of tags. To support a
nested return value, return a nested std::map of std::maps. Each key
specifies the tag value, and each value may be another nesting level.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/408
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.
Since we now don't duplicate all items, we can easily remove the 64kB
limit from OpusReader::ReadString() and instead silently ignore and
skip all strings which are longer than 4 kB.
This fixes a tag duplication bug with Opus file containing a very long
`METADATA_BLOCK_PICTURE` tag, which occurred because the Opus plugin
returned false after parsing all tags, and then the MPD core fell back
to FFmpeg which scanned the tags again.
Return `404 not found` for some common well-known paths, as clients requesting them usually do that automatically and don't expect endless audio stram.
Closes#572
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.
This reverts commit ff3e2c0514. The
check was necessary, after all, because this is what checked whether
the decoder had finished the current or the next song.
> The "queued" flag can only possibly be set if the decoder is still
> decoding the current song or if the decoder is stopped.
That was wrong because ProcessCommand() sets `queued=true` and also
starts the decoder (if it was idle).
> This is also what the following assert() checks.
That was also wrong, because the assert() has two conditions.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/566
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 the decoder finishes decoding the current song between the two
IsIdle() checks, MPD stops playback instead of starting the decoder
for the next song.
This is usually not visible problem, because the main thread restarts
it via playlist::ResumePlayback(), but that way it, ignores "single"
mode.
As a workaround, this commit adds another "queued" check which
re-enters the player loop and checks again whether to start the
decoder.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/556
The "queued" flag can only possibly be set if the decoder is still
decoding the current song or if the decoder is stopped. This is also
what the following assert() checks. This check was not necessary.
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.
ALSA is just one out of many output plugins, and detailed plugin
documentation should only live in the user manual, without having
duplicates in the (brief) manpage.
Also move "mixer_type" to the "optional audio output parameters"
section; it is a generic option, not specific to ALSA.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/552
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.
Commit b3a458338a added a LocateUri()
call to several playlist commands, which applied InputPlugin URI
scheme verification to playlist URIs. This broke the SoundCloud
playlist plugin which uses "soundcloud://" URIs for which no input
plugin exists.
This commit allows the caller to specify the kind of plugin which
shall be used to verify the URI. Right now, only "input" is
implemented; "storage" uses the "input" verification for now; and
"playlist" has no verification at all (for now).
Closes https://github.com/MusicPlayerDaemon/MPD/issues/528
Expect OnSocketReady() to cancel events. If it returns false, the
SocketMonitor may be destructed already. This fixes a use-after-free
bug in the "httpd" output plugin.
This missing piece probably never really hurt, because
HttpdClient::OnSocketClosed() would be called right after a socket
error, but it's better to be explicit about closing on error.
Fixes:
src/net/IPv4Address.hxx: In member function 'constexpr IPv4Address::operator SocketAddress() const':
src/net/IPv4Address.hxx:171:24: error: a reinterpret_cast is not a constant expression
171 | return SocketAddress((const struct sockaddr *)&address,
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
src/net/IPv6Address.hxx: In member function 'constexpr IPv6Address::operator SocketAddress() const':
src/net/IPv6Address.hxx:138:24: error: a reinterpret_cast is not a constant expression
138 | return SocketAddress((const struct sockaddr *)&address,
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Closes https://github.com/MusicPlayerDaemon/MPD/issues/522
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.
This is a guide for those who wish to hack on the MPD source code. MPD is an open project, and we are always happy about contributions. So far, more than 150 people have contributed patches. This document is work in progress. Most of it may be incomplete yet. Please help!
Code Style
==========
**********
* indent with tabs (width 8)
* don't write CPP when you can write C++: use inline functions and constexpr instead of macros
* comment your code, document your APIs
* the code should be C++14 compliant, and must compile with :program:`GCC`6.0 and :program:`clang`3.4
* report error conditions with C++ exceptions, preferable derived from :envvar:`std::runtime_error`
* the code should be C++17 compliant, and must compile with :program:`GCC`8 and :program:`clang`5
* all code must be exception-safe
* classes and functions names use CamelCase; variables are lower-case with words separated by underscore
Some example code:
~~~~~~~~~~~~~~~~~~
..code-block::c
@@ -32,8 +30,58 @@ Some example code:
returnxyz;
}
Error handling
==============
If an error occurs, throw a C++ exception, preferably derived from
:code:`std::runtime_error`. The function's API documentation should
mention that. If a function cannot throw exceptions, add
:code:`noexcept` to its prototype.
Some parts of MPD use callbacks to report completion; the handler
classes usually have an "error" callback which receives a
:code:`std::exception_ptr`
(e.g. :code:`BufferedSocket::OnSocketError()`). Wrapping errors in
:code:`std::exception_ptr` allows propagating details about the error
across thread boundaries to the entity which is interested in handling
it (e.g. giving the MPD client details about an I/O error caught by
the decoder thread).
Out-of-memory errors (i.e. :code:`std::bad_alloc`) do not need to be
handled. Some operating systems such as Linux do not report
out-of-memory to userspace, and instead kill a process to recover.
Even if we know we are out of memory, there is little we can do except
for aborting the process quickly. Any other attempts to give back
memory may cause page faults on the way which make the situation
worse.
Error conditions which are caused by a bug do not need to be handled
at runtime; instead, use :code:`assert()` to detect them in debug
builds.
git Branches
************
There are two active branches in the git repository:
- the "unstable" branch called ``master`` where new features are
merged. This will become the next major release eventually.
- the "stable" branch (currently called ``v0.22.x``) where only bug
fixes are merged.
Once :program:`MPD` 0.23 is released, a new branch called ``v0.23.x``
will be created for 0.23 bug-fix releases; after that, ``v0.22.x``
will eventually cease to be maintained.
After bug fixes have been added to the "stable" branch, it will be
merged into ``master``. This ensures that all known bugs are fixed in
@@ -59,7 +107,7 @@ possible, to be sure that you don't break any disabled code.
Don't mix several changes in one single patch. Create a separate patch for every change. Tools like :program:`stgit` help you with that. This way, we can review your patches more easily, and we can pick the patches we like most first.
Basic stgit usage
-----------------
=================
stgit allows you to create a set of patches and refine all of them: you can go back to any patch at any time, and re-edit it (both the code and the commit message). You can reorder patches and insert new patches at any position. It encourages creating separate patches for tiny changes.
@@ -94,35 +142,7 @@ When the whole patch series is finished, convert stgit patches to git commits:
stg commit
Submitting Patches
==================
******************
Send your patches to the mailing list:
Email: `mpd-devel <mpd-devel@musicpd.org>`_
:program:`git pull` requests are preferred.
Development Tools
=================
Clang Static Analyzer
---------------------
The `static analyzer <http://clang-analyzer.llvm.org/>`_ is a tool that helps find bugs. To run it on the MPD code base, install LLVM and clang. configure MPD to use clang:
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 ``~/.mpd/mpd.conf`` 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:`$XDG_CONFIG_HOME/mpd/mpd.conf`
User configuration file (usually :file:`~/.config/mpd/mpd.conf`).
: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/
The default plugin. Stores a copy of the database in memory. A file is used for permanent storage.
@@ -25,9 +25,9 @@ The default plugin. Stores a copy of the database in memory. A file is used for
- Compress the database file using gzip? Enabled by default (if built with zlib).
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
@@ -42,46 +42,51 @@ Provides access to the database of another :program:`MPD` instance using libmpdc
* - **password**
- The password used to log in to the "master" :program:`MPD` instance.
* - **keepalive yes|no**
- Send TCP keepalive packets to the "master" :program:`MPD` instance? This option can help avoid certain firewalls dropping inactive connections, at the expensive of a very small amount of additional network traffic. Disabled by default.
- Send TCP keepalive packets to the "master" :program:`MPD` instance? This option can help avoid certain firewalls dropping inactive connections, at the expense of a very small amount of additional network traffic. Disabled by default.
upnp
~~~~
----
Provides access to UPnP media servers.
Storage plugins
---------------
===============
local
~~~~~
-----
The default plugin which gives :program:`MPD` access to local files. It is used when music_directory refers to a local directory.
curl
~~~~
----
A WebDAV client using libcurl. It is used when :code:`music_directory` contains a http:// or https:// URI, for example :samp:`https://the.server/dav/`.
A WebDAV client using libcurl. It is used when :code:`music_directory`
contains a ``http://`` or ``https://`` URI, for example
:samp:`https://the.server/dav/`.
smbclient
~~~~~~~~~
---------
Load music files from a SMB/CIFS server. It is used when:code:`music_directory` contains a smb:// URI, for example :samp:`smb://myfileserver/Music`.
Load music files from a SMB/CIFS server. It is used when
:code:`music_directory` contains a ``smb://`` URI, for example
:samp:`smb://myfileserver/Music`.
Note that :file:`libsmbclient` has a serious bug which causes MPD to
crash, and therefore this plugin is disabled by default and should not
be used until the bug is fixed:
https://bugzilla.samba.org/show_bug.cgi?id=11413
nfs
~~~
---
Load music files from a NFS server. It is used when:code:`music_directory` contains a nfs:// URI according to RFC2224, for example :samp:`nfs://servername/path`.
Load music files from a NFS server. It is used when
:code:`music_directory` contains a ``nfs://`` URI according to
RFC2224, for example :samp:`nfs://servername/path`.
This plugin uses libnfs, which supports only NFS version 3. Since :program:`MPD` is not allowed to bind to "privileged ports", the NFS server needs to enable the "insecure" setting; example :file:`/etc/exports`:
..code-block::none
/srv/mp3 192.168.1.55(ro,insecure)
Don't fear: "insecure" does not mean that your NFS server is insecure. A few decades ago, people thought the concept of "privileged ports" would make network services "secure", which was a fallacy. The absence of this obsolete "security" measure means little.
See :ref:`input_nfs` for more information.
udisks
~~~~~~
------
Mount file systems (e.g. USB sticks or other removable media) using
the udisks2 daemon via D-Bus. To obtain a valid udisks2 URI, consult
@@ -106,42 +111,67 @@ MPD user.
.._neighbor_plugin:
Neighbor plugins
----------------
================
smbclient
~~~~~~~~~
---------
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
~~~~
----
Provides a list of UPnP servers on the local network.
.._input_plugins:
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
~~~~~~~~~~~~~
-------------
Plays audio CDs using libcdio. The URI has the form: "cdda://[DEVICE][/TRACK]". The simplest form cdda:// plays the whole disc in the default drive.
@@ -157,11 +187,14 @@ Plays audio CDs using libcdio. The URI has the form: "cdda://[DEVICE][/TRACK]".
- Request CDParanoia cap the extraction speed to Nx normal CD audio rotation speed, keeping the drive quiet.
curl
~~~~
----
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.
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.
..list-table::
:widths:20 80
@@ -179,44 +212,67 @@ Note that unless overridden by the below settings (e.g. by setting them to a bla
- Verify the certificate's name against host? `More information <http://curl.haxx.se/libcurl/c/CURLOPT_SSL_VERIFYHOST.html>`_.
ffmpeg
~~~~~~
------
Access to various network protocols implemented by the FFmpeg library: gopher://, rtp://, rtsp://, rtmp://, rtmpt://, rtmps://
Access to various network protocols implemented by the FFmpeg library:
Plays streams with the MMS protocol using `libmms <https://launchpad.net/libmms>`_.
nfs
~~~
.._input_nfs:
Allows :program:`MPD` to access files on NFSv3 servers without actually mounting them (i.e. in userspace, without help from the kernel's VFS layer). All URIs with the nfs:// scheme are used according to RFC2224. Example:
nfs
---
Allows :program:`MPD` to access files on NFS servers without actually
mounting them (i.e. with :program:`libnfs` in userspace, without help
from the kernel's VFS layer). All URIs with the ``nfs://`` scheme are
used according to RFC2224. Example:
..code-block::none
mpc add nfs://servername/path/filename.ogg
Note that this usually requires enabling the "insecure" flag in the server's /etc/exports file, because :program:`MPD` cannot bind to so-called "privileged" ports. Don't fear: this will not make your file server insecure; the flag was named in a time long ago when privileged ports were thought to be meaningful for security. By today's standards, NFSv3 is not secure at all, and if you believe it is, you're already doomed.
This plugin uses :program:`libnfs`, which supports only NFS version 3.
Since :program:`MPD` is not allowed to bind to so-called "privileged
ports", the NFS server needs to enable the ``insecure`` setting;
example :file:`/etc/exports`:
..code-block::none
/srv/mp3 192.168.1.55(ro,insecure)
Don't fear: this will not make your file server insecure; the flag was
named a time long ago when privileged ports were thought to be
meaningful for security. By today's standards, NFSv3 is not secure at
all, and if you believe it is, you're already doomed.
smbclient
~~~~~~~~~
---------
Allows :program:`MPD` to access files on SMB/CIFS servers (e.g. Samba or Microsoft Windows). All URIs with the smb:// scheme are used. Example:
Allows :program:`MPD` to access files on SMB/CIFS servers (e.g. Samba
or Microsoft Windows). All URIs with the ``smb://`` scheme are
Play songs from the commercial streaming service Qobuz. It plays URLs in the form qobuz://track/ID, e.g.:
Play songs from the commercial streaming service Qobuz. It plays URLs
in the form ``qobuz://track/ID``, e.g.:
..code-block::none
@@ -240,9 +296,16 @@ Play songs from the commercial streaming service Qobuz. It plays URLs in the for
- The `Qobuz format identifier <https://github.com/Qobuz/api-documentation/blob/master/endpoints/track/getFileUrl.md#parameters>`_, i.e. a number which chooses the format and quality to be requested from Qobuz. The default is "5" (320 kbit/s MP3).
tidal
~~~~~
-----
Play songs from the commercial streaming service `Tidal<http://tidal.com/>`_. It plays URLs in the form tidal://track/ID, e.g.:
Play songs from the commercial streaming service `Tidal
<http://tidal.com/>`_. It plays URLs in the form ``tidal://track/ID``,
e.g.:
..warning::
This plugin is currently defunct because Tidal has changed the
protocol and decided not to share documentation.
..code-block::none
@@ -266,10 +329,10 @@ Play songs from the commercial streaming service `Tidal <http://tidal.com/>`_. I
.._decoder_plugins:
Decoder plugins
---------------
===============
adplug
~~~~~~
------
Decodes AdLib files using libadplug.
@@ -283,17 +346,19 @@ Decodes AdLib files using libadplug.
- The sample rate that shall be synthesized by the plugin. Defaults to 48000.
audiofile
~~~~~~~~~
---------
Decodes WAV and AIFF files using libaudiofile.
faad
~~~~
----
Decodes AAC files using libfaad.
.._decoder_ffmpeg:
ffmpeg
~~~~~~
------
Decodes various codecs using FFmpeg.
@@ -309,14 +374,18 @@ Decodes various codecs using FFmpeg.
- Sets the FFmpeg muxer option probesize, which specifies probing size in bytes, i.e. the size of the data to analyze to get stream information. The `FFmpeg formats documentation <https://ffmpeg.org/ffmpeg-formats.html>`_ has more information.
flac
~~~~
----
Decodes FLAC files using libFLAC.
dsdiff
~~~~~~
.._decoder_dsdiff:
Decodes DFF files containing DSDIFF data (e.g. SACD rips).
dsdiff
------
Decodes DSDIFF (`Direct Stream Digital Interchange File Format
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
~~~
---
Decodes MP3 files using `libmad <http://www.underbit.com/products/mad/>`_.
mikmod
~~~~~~
------
Module player based on `MikMod <http://mikmod.sourceforge.net/>`_.
@@ -407,7 +472,7 @@ Module player based on `MikMod <http://mikmod.sourceforge.net/>`_.
- Sets the sample rate generated by libmikmod. Default is 44100.
modplug
~~~~~~~
-------
Module player based on MODPlug.
@@ -421,27 +486,29 @@ Module player based on MODPlug.
- Number of times to loop the module if it uses backward loops. Default is 0 which prevents looping. -1 loops forever.
mpcdec
~~~~~~
------
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
~~~~
----
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
~~~~~~~
-------
C64 SID decoder based on `libsidplayfp <https://sourceforge.net/projects/sidplay-residfp/>`_ or `libsidplay2 <https://sourceforge.net/projects/sidplay2/>`_.
@@ -452,9 +519,11 @@ 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**
@@ -463,23 +532,23 @@ C64 SID decoder based on `libsidplayfp <https://sourceforge.net/projects/sidplay
- Only libsidplayfp. Absolute path to basic rom image file.
sndfile
~~~~~~~
-------
Decodes WAV and AIFF files using `libsndfile <http://www.mega-nerd.com/libsndfile/>`_.
vorbis
~~~~~~
------
Decodes Ogg-Vorbis files using `libvorbis <http://www.xiph.org/ogg/vorbis/>`_.
wavpack
~~~~~~~
-------
Decodes WavPack files using `libwavpack <http://www.wavpack.com/>`_.
wildmidi
~~~~~~~~
--------
MIDI decoder based on `libwildmidi <http://www.mindwerks.net/projects/wildmidi/>`_.
@@ -495,10 +564,11 @@ MIDI decoder based on `libwildmidi <http://www.mindwerks.net/projects/wildmidi/>
.._encoder_plugins:
Encoder plugins
---------------
===============
flac
~~~~
----
Encodes into `FLAC <https://xiph.org/flac/>`_ (lossless).
..list-table::
@@ -511,7 +581,7 @@ Encodes into `FLAC <https://xiph.org/flac/>`_ (lossless).
- Sets the libFLAC compression level. The levels range from 0 (fastest, least compression) to 8 (slowest, most compression).
lame
~~~~
----
Encodes into MP3 using the `LAME <http://lame.sourceforge.net/>`_ library.
@@ -527,12 +597,12 @@ Encodes into MP3 using the `LAME <http://lame.sourceforge.net/>`_ library.
- Sets the bit rate in kilobit per second. Cannot be used with quality.
null
~~~~
----
Does not encode anything, passes the input PCM data as-is.
shine
~~~~~
-----
Encodes into MP3 using the `Shine <https://github.com/savonet/shine>`_ library.
@@ -546,7 +616,7 @@ Encodes into MP3 using the `Shine <https://github.com/savonet/shine>`_ library.
- Sets the bit rate in kilobit per second.
twolame
~~~~~~~
-------
Encodes into MP2 using the `TwoLAME <http://www.twolame.org/>`_ library.
@@ -562,7 +632,7 @@ Encodes into MP2 using the `TwoLAME <http://www.twolame.org/>`_ library.
- Sets the bit rate in kilobit per second. Cannot be used with quality.
opus
~~~~
----
Encodes into `Ogg Opus <http://www.opus-codec.org/>`_.
@@ -584,7 +654,7 @@ Encodes into `Ogg Opus <http://www.opus-codec.org/>`_.
.._vorbis_plugin:
vorbis
~~~~~~
------
Encodes into `Ogg Vorbis <http://www.vorbis.com/>`_.
@@ -600,13 +670,13 @@ Encodes into `Ogg Vorbis <http://www.vorbis.com/>`_.
- Sets the bit rate in kilobit per second. Cannot be used with quality.
wave
~~~~
----
Encodes into WAV (lossless).
.._resampler_plugins:
Resampler plugins
-----------------
=================
The resampler can be configured in a block named resampler, for example:
@@ -629,12 +699,12 @@ The following table lists the resampler options valid for all plugins:
- The name of the plugin.
internal
~~~~~~~~
--------
A resampler built into :program:`MPD`. Its quality is very poor, but its CPU usage is low. This is the fallback if :program:`MPD` was compiled without an external resampler.
libsamplerate
~~~~~~~~~~~~~
-------------
A resampler using `libsamplerate <http://www.mega-nerd.com/SRC/>`_ a.k.a. Secret Rabbit Code (SRC).
@@ -645,7 +715,7 @@ A resampler using `libsamplerate <http://www.mega-nerd.com/SRC/>`_ a.k.a. Secret
* - Name
- Description
* - **type**
- The interpolator type. See below for a list of known types.
- The interpolator type. Defaults to :samp:`2`. See below for a list of known types.
The following converter types are provided by libsamplerate:
@@ -667,7 +737,7 @@ The following converter types are provided by libsamplerate:
- Linear interpolator, very fast, poor quality.
soxr
~~~~
----
A resampler using `libsoxr <http://sourceforge.net/projects/soxr/>`_, the SoX Resampler library
@@ -689,16 +759,39 @@ Valid quality values for libsoxr:
* "medium"
* "low"
* "quick"
* "custom"
If the quality is set to custom also the following settings are available:
..list-table::
:widths:20 80
:header-rows:1
* - 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:
Output plugins
--------------
==============
.._alsa_plugin:
alsa
~~~~
----
The `Advanced Linux Sound Architecture (ALSA) <http://www.alsa-project.org/>`_ plugin uses libasound. It is recommended if you are using Linux.
@@ -757,7 +850,7 @@ The following attributes can be configured at runtime using the outputset comman
ao
~~
--
The ao plugin uses the portable `libao <https://www.xiph.org/ao/>`_ library. Use only if there is no native plugin for your operating system.
..list-table::
@@ -774,7 +867,8 @@ The ao plugin uses the portable `libao <https://www.xiph.org/ao/>`_ library. Use
- This specifies how many bytes to write to the audio device at once. This parameter is to work around a bug in older versions of libao on sound cards with very small buffers. The default is 1024.
sndio
~~~~~
-----
The sndio plugin uses the `sndio <http://www.sndio.org/>`_ library. It should normally be used on OpenBSD.
..list-table::
@@ -789,7 +883,7 @@ The sndio plugin uses the `sndio <http://www.sndio.org/>`_ library. It should no
- Set the application buffer time in milliseconds.
fifo
~~~~
----
The fifo plugin writes raw PCM data to a FIFO (First In, First Out) file. The data can be read by another program.
@@ -803,7 +897,7 @@ The fifo plugin writes raw PCM data to a FIFO (First In, First Out) file. The da
- This specifies the path of the FIFO to write to. Must be an absolute path. If the path does not exist, it will be created when MPD is started, and removed when MPD is stopped. The FIFO will be created with the same user and group as MPD is running as. Default permissions can be modified by using the builtin shell command umask. If a FIFO already exists at the specified path it will be reused, and will not be removed when MPD is stopped. You can use the "mkfifo" command to create this, and then you may modify the permissions to your liking.
haiku
~~~~~
-----
Use the SoundPlayer API on the Haiku operating system.
@@ -812,9 +906,14 @@ removed soon, unless there is a new maintainer.
jack
~~~~
----
The jack plugin connects to a `JACK server <http://jackaudio.org/>`_.
On Windows, this plugin loads :file:`libjack64.dll` at runtime. This
means you need to `download and install the JACK windows build
<https://jackaudio.org/downloads/>`_.
..list-table::
:widths:20 80
:header-rows:1
@@ -831,11 +930,16 @@ 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.
httpd
~~~~~
-----
The httpd plugin creates a HTTP server, similar to `ShoutCast <http://www.shoutcast.com/>`_ / `IceCast <http://icecast.org/>`_. HTTP streaming clients like mplayer, VLC, and mpv can connect to it.
It is highly recommended to configure a fixed format, because a stream cannot switch its audio format on-the-fly when the song changes.
@@ -856,7 +960,8 @@ It is highly recommended to configure a fixed format, because a stream cannot sw
- Sets a limit, number of concurrent clients. When set to 0 no limit will apply.
null
~~~~
----
The null plugin does nothing. It discards everything sent to it.
..list-table::
@@ -871,7 +976,8 @@ The null plugin does nothing. It discards everything sent to it.
.._oss_plugin:
oss
~~~
---
The "Open Sound System" plugin is supported on most Unix platforms.
On Linux, OSS has been superseded by ALSA. Use the ALSA output plugin :ref:`alsa_plugin` instead of this one on Linux.
@@ -899,7 +1005,7 @@ The according hardware mixer plugin understands the following settings:
- Choose a mixer control, defaulting to PCM.
openal
~~~~~~
------
The "OpenAL" plugin uses `libopenal <http://kcat.strangesoft.net/openal.html>`_. It is supported on many platforms. Use only if there is no native plugin for your operating system.
..list-table::
@@ -912,7 +1018,7 @@ The "OpenAL" plugin uses `libopenal <http://kcat.strangesoft.net/openal.html>`_.
- Sets the device which should be used. This can be any valid OpenAL device name. If not specified, then libopenal will choose a default device.
osx
~~~
---
The "Mac OS X" plugin uses Apple's CoreAudio API.
..list-table::
@@ -933,7 +1039,7 @@ The "Mac OS X" plugin uses Apple's CoreAudio API.
The channel map may not refer to outputs that do not exist according to the format. If the format is "*:*:1" (mono) and you have a four-channel sound card then "-1,-1,0,0" (dual mono output on the second pair of sound card outputs) is a valid channel map but "-1,-1,0,1" is not because the second channel ('1') does not exist when the output is mono.
pipe
~~~~
----
The pipe plugin starts a program and writes raw PCM data into its standard input.
@@ -949,7 +1055,7 @@ The pipe plugin starts a program and writes raw PCM data into its standard input
.._pulse_plugin:
pulse
~~~~~
-----
The pulse plugin connects to a `PulseAudio <http://www.freedesktop.org/wiki/Software/PulseAudio/>`_ server. Requires libpulse.
..list-table::
@@ -962,11 +1068,13 @@ 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.
recorder
~~~~~~~~
--------
The recorder plugin writes the audio played by :program:`MPD` to a file. This may be useful for recording radio streams.
..list-table::
@@ -978,13 +1086,13 @@ The recorder plugin writes the audio played by :program:`MPD` to a file. This ma
* - **path P**
- Write to this file.
* - **format_path P**
- An alternative to path which provides a format string referring to tag values. The special tag iso8601 emits the current date and time in `ISO8601 <https://en.wikipedia.org/wiki/ISO_8601>`_ format (UTC). Every time a new song starts or a new tag gets received from a radio station, a new file is opened. If the format does not render a file name, nothing is recorded. A tag name enclosed in percent signs ('%') is replaced with the tag value. Example: :file:`~/.mpd/recorder/%artist% - %title%.ogg`. Square brackets can be used to group a substring. If none of the tags referred in the group can be found, the whole group is omitted. Example: [~/.mpd/recorder/[%artist% - ]%title%.ogg] (this omits the dash when no artist tag exists; if title also doesn't exist, no file is written). The operators "|" (logical "or") and "&" (logical "and") can be used to select portions of the format string depending on the existing tag values. Example: ~/.mpd/recorder/[%title%|%name%].ogg (use the "name" tag if no title exists)
- An alternative to path which provides a format string referring to tag values. The special tag iso8601 emits the current date and time in `ISO8601 <https://en.wikipedia.org/wiki/ISO_8601>`_ format (UTC). Every time a new song starts or a new tag gets received from a radio station, a new file is opened. If the format does not render a file name, nothing is recorded. A tag name enclosed in percent signs ('%') is replaced with the tag value. Example: :file:`-/.mpd/recorder/%artist% - %title%.ogg`. Square brackets can be used to group a substring. If none of the tags referred in the group can be found, the whole group is omitted. Example: [-/.mpd/recorder/[%artist% - ]%title%.ogg] (this omits the dash when no artist tag exists; if title also doesn't exist, no file is written). The operators "|" (logical "or") and "&" (logical "and") can be used to select portions of the format string depending on the existing tag values. Example: -/.mpd/recorder/[%title%|%name%].ogg (use the "name" tag if no title exists)
* - **encoder NAME**
- Chooses an encoder plugin. A list of encoder plugins can be found in the encoder plugin reference :ref:`encoder_plugins`.
shout
~~~~~
-----
The shout plugin connects to a ShoutCast or IceCast server using libshout. It forwards tags to this server.
You must set a format.
@@ -1028,15 +1136,16 @@ You must set a format.
.._sles_output:
sles
~~~~
----
Plugin using the `OpenSL ES <https://www.khronos.org/opensles/>`__
audio API. Its primary use is local playback on Android, where
:ref:`ALSA <alsa_plugin>` is not available.
:ref:`ALSA <alsa_plugin>` is not available. It supports 16 bit and
floating point samples.
solaris
~~~~~~~
-------
The "Solaris" plugin runs only on SUN Solaris, and plays via /dev/audio.
..list-table::
@@ -1049,25 +1158,74 @@ 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".
* - **dop yes|no**
- Enable DSD over PCM. Require exclusive mode. 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
*`Meson 0.49.0 <http://mesonbuild.com/>`__ and `Ninja
<https://ninja-build.org/>`__
* cmake
* pkg-config
* quilt
Just like with the native build, unpack the :program:`MPD` source
tarball and change into the directory. Then, instead of
@@ -183,47 +201,6 @@ 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
*************
@@ -258,6 +235,13 @@ 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
-------------------------------
@@ -334,6 +318,44 @@ 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
---------------------------
@@ -402,15 +424,10 @@ The following table lists the audio_output options valid for all plugins:
- The name of the plugin
* - **name**
- The name of the audio output. It is visible to the client. Some plugins also use it internally, e.g. as a name registered in the PULSE server.
* - **format**
- Always open the audio output with the specified audio format samplerate:bits:channels), regardless of the format of the input file. This is optional for most plugins.
Any of the three attributes may be an asterisk to specify that this attribute should not be enforced, example: 48000:16:*. *:*:* is equal to not having a format specification.
The following values are valid for bits: 8 (signed 8 bit integer samples), 16, 24 (signed 24 bit integer samples padded to 32 bit), 32 (signed 32 bit integer samples), f (32 bit floating point, -1.0 to 1.0), "dsd" means DSD (Direct Stream Digital). For DSD, there are special cases such as "dsd64", which allows you to omit the sample rate (e.g. dsd512:2 for stereo DSD512, i.e. 22.5792 MHz).
The sample rate is special for DSD: :program:`MPD` counts the number of bytes, not bits. Thus, a DSD "bit" rate of 22.5792 MHz (DSD512) is 2822400 from :program:`MPD`'s point of view (44100*512/8).
* - **enabed yes|no**
* - **format samplerate:bits:channels**
- Always open the audio output with the specified audio format, regardless of the format of the input file. This is optional for most plugins.
See :ref:`audio_output_format` for a detailed description of the value.
* - **enabled yes|no**
- Specifies whether this audio output is enabled when :program:`MPD` is started. By default, all audio outputs are enabled. This is just the default setting when there is no state file; with a state file, the previous state is restored.
* - **tags yes|no**
- If set to no, then :program:`MPD` will not send tags to this output. This is only useful for output plugins that can receive tags, for example the httpd output plugin.
@@ -422,7 +439,7 @@ 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,...**"
@@ -497,6 +514,11 @@ The following table lists the playlist_plugin options valid for all plugins:
- The name of the plugin
* - **enabled yes|no**
- Allows you to disable a playlist plugin without recompiling. By default, all plugins are enabled.
* - **as_directory yes|no**
- With this option, a playlist file of this type is parsed during
database update and converted to a virtual directory, allowing
MPD clients to access individual entries. By default, this is
only enabled for the :ref:`cue plugin <cue_playlist>`.
More information can be found in the :ref:`playlist_plugins`
reference.
@@ -504,13 +526,34 @@ reference.
Audio Format Settings
---------------------
Global Audio Format
~~~~~~~~~~~~~~~~~~~
.._audio_output_format:
The setting audio_output_format forces :program:`MPD` to use one audio format for all outputs. Doing that is usually not a good idea. The values are the same as in format in the audio_output section.
Global Audio Format
^^^^^^^^^^^^^^^^^^^
The setting ``audio_output_format`` forces :program:`MPD` to use one
audio format for all outputs. Doing that is usually not a good idea.
The value is specified as ``samplerate:bits:channels``.
Any of the three attributes may be an asterisk to specify that this
attribute should not be enforced, example: ``48000:16:*``.
``*:*:*`` is equal to not having a format specification.
The following values are valid for bits: ``8`` (signed 8 bit integer
samples), ``16``, ``24`` (signed 24 bit integer samples padded to 32
bit), ``32`` (signed 32 bit integer samples), ``f`` (32 bit floating
point, -1.0 to 1.0), ``dsd`` means DSD (Direct Stream Digital). For
DSD, there are special cases such as ``dsd64``, which allows you to
omit the sample rate (e.g. ``dsd512:2`` for stereo DSD512,
i.e. 22.5792 MHz).
The sample rate is special for DSD: :program:`MPD` counts the number
of bytes, not bits. Thus, a DSD "bit" rate of 22.5792 MHz (DSD512) is
2822400 from :program:`MPD`'s point of view (44100*512/8).
Resampler
~~~~~~~~~
^^^^^^^^^
Sometimes, music needs to be resampled before it can be played; for example, CDs use a sample rate of 44,100 Hz while many cheap audio chips can only handle 48,000 Hz. Resampling reduces the quality and consumes a lot of CPU. There are different options, some of them optimized for high quality and others for low CPU usage, but you can't have both at the same time. Often, the resampler is the component that is responsible for most of :program:`MPD`'s CPU usage. Since :program:`MPD` comes with high quality defaults, it may appear that :program:`MPD` consumes more CPU than other software.
@@ -523,7 +566,7 @@ Client Connections
.._listeners:
Listeners
~~~~~~~~~
^^^^^^^^^
The setting :code:`bind_to_address` specifies which addresses
:program:`MPD` listens on for connections from clients. It can be
@@ -566,7 +609,7 @@ used.
Permissions and Passwords
~~~~~~~~~~~~~~~~~~~~~~~~~
^^^^^^^^^^^^^^^^^^^^^^^^^
By default, all clients are unauthenticated and have a full set of permissions. This can be restricted with the settings :code:`default_permissions` and :code:`password`.
@@ -629,7 +672,7 @@ Other Settings
Section :ref:`tags` contains a list of supported tags.
The State File
~~~~~~~~~~~~~~
^^^^^^^^^^^^^^
The state file is a file where :program:`MPD` saves and restores its state (play queue, playback position etc.) to keep it persistent across restarts and reboots. It is an optional setting.
@@ -645,9 +688,11 @@ The State File
- Specify the state file location. The parent directory must be writable by the :program:`MPD` user (+wx).
* - **state_file_interval SECONDS**
- Auto-save the state file this number of seconds after each state change. Defaults to 120 (2 minutes).
* - **restore_paused yes|no**
- If set to :samp:`yes`, then :program:`MPD` is put into pause mode instead of starting playback after startup. Default is :samp:`no`.
The Sticker Database
~~~~~~~~~~~~~~~~~~~~
^^^^^^^^^^^^^^^^^^^^
"Stickers" are pieces of information attached to songs. Some clients
use them to store ratings and other volatile data. This feature
These settings are various limitations to prevent :program:`MPD` from using too many resources (denial of service).
@@ -677,7 +722,7 @@ These settings are various limitations to prevent :program:`MPD` from using too
* - **connection_timeout SECONDS**
- If a client does not send any new data in this time period, the connection is closed. Clients waiting in "idle" mode are excluded from this. Default is 60.
* - **max_connections NUMBER**
- This specifies the maximum number of clients that can be connected to :program:`MPD` at the same time. Default is 5.
- This specifies the maximum number of clients that can be connected to :program:`MPD` at the same time. Default is 100.
* - **max_playlist_length NUMBER**
- The maximum number of songs that can be in the playlist. Default is 16384.
* - **max_command_list_size KBYTES**
@@ -686,7 +731,7 @@ These settings are various limitations to prevent :program:`MPD` from using too
- The maximum size of the output buffer to a client (maximum response size). Default is 8192 (8 MiB).
Buffer Settings
~~~~~~~~~~~~~~~
^^^^^^^^^^^^^^^
Do not change these unless you know what you are doing.
@@ -696,11 +741,12 @@ 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
~~~~~~~~
^^^^^^^^
If Zeroconf support (`Avahi <http://avahi.org/>`_ or Apple's Bonjour)
was enabled at compile time with :code:`-Dzeroconf=...`,
@@ -745,22 +791,24 @@ The :code:`music_directory` setting tells :program:`MPD` to read files from the
The database setting tells :program:`MPD` to pass all database queries on to the :program:`MPD` instance running on the file server (using the proxy plugin).
.._realtime:
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.
@@ -778,22 +826,107 @@ 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.
Note
~~~~
..note::
There is a rumor that real-time scheduling improves audio quality. That is not true. All it does is reduce the probability of skipping (audio buffer xruns) when the computer is under heavy load.
There is a rumor that real-time scheduling improves audio
quality. That is not true. All it does is reduce the probability of
skipping (audio buffer xruns) when the computer is under heavy
load.
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
----------
@@ -817,7 +950,7 @@ Depending on the size of your music collection and the speed of the storage, thi
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.
Mounting other storages into the music directory
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
:program:`MPD` has various storage plugins of which multiple instances can be "mounted" into the music directory. This way, you can use local music, file servers and USB sticks at the same time. Example:
@@ -912,15 +1045,23 @@ Check list for bit-perfect playback:
device (:samp:`hw:0,0` or similar).
* Don't use software volume (setting :code:`mixer_type`).
* 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.
.._dsd:
Direct Stream Digital (DSD)
---------------------------
DSD (`Direct Stream Digital<https://en.wikipedia.org/wiki/Direct_Stream_Digital>`_) is a digital format that stores audio as a sequence of single-bit values at a very high sampling rate.
DSD (`Direct Stream Digital
<https://en.wikipedia.org/wiki/Direct_Stream_Digital>`_) is a digital
format that stores audio as a sequence of single-bit values at a very
high sampling rate. It is the sample format used on `Super Audio CDs
<https://en.wikipedia.org/wiki/Super_Audio_CD>`_.
:program:`MPD` understands the file formats dff and dsf. There are three ways to play back DSD:
:program:`MPD` understands the file formats :ref:`DSDIFF
<decoder_dsdiff>` and :ref:`DSF <decoder_dsf>`. There are three ways
to play back DSD:
* Native DSD playback. Requires ALSA 1.0.27.1 or later, a sound driver/chip that supports DSD and of course a DAC that supports DSD.
@@ -933,6 +1074,22 @@ 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
@@ -957,47 +1114,79 @@ Sometimes, it is helpful to run :program:`MPD` in a terminal and follow what hap
..code-block::none
mpd --stdout --no-daemon --verbose
mpd --stderr --no-daemon --verbose
Support
-------
Getting Help
~~~~~~~~~~~~
^^^^^^^^^^^^
The :program:`MPD` project runs a `forum <https://forum.musicpd.org/>`_ and an IRC channel (#mpd on Freenode) for requesting help. Visit the MPD help page for details on how to get help.
The :program:`MPD` project runs a `forum <https://forum.musicpd.org/>`_ and an IRC channel (#mpd on Libera.Chat) for requesting help. Visit the MPD help page for details on how to get help.
Common Problems
~~~~~~~~~~~~~~~
^^^^^^^^^^^^^^^
1. Database
^^^^^^^^^^^
Startup
"""""""
Question: I can't see my music in the MPD database!
This happens on Linux when :file:`/proc/sys/net/ipv6/bindv6only` is
disabled. MPD first binds to IPv6, and this automatically binds to
IPv4 as well; after that, MPD binds to IPv4, but that fails. You can
safely ignore this, because MPD works on both IPv4 and IPv6.
Database
""""""""
I can't see my music in the MPD database
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
* Check your :code:`music_directory` setting.
* Does the MPD user have read permission on all music files, and read+execute permission on all music directories (and all of their parent directories)?
* Did you update the database? (mpc update)
* Did you enable all relevant decoder plugins at compile time? :command:`mpd --version` will tell you.
Question: MPD doesn't read ID3 tags!
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
MPD doesn't read ID3 tags!
~~~~~~~~~~~~~~~~~~~~~~~~~~
* You probably compiled :program:`MPD` without libid3tag. :command:`mpd --version` will tell you.
2.Playback
^^^^^^^^^^^
Playback
""""""""
Question: I can't hear music on my client!
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
I can't hear music on my client
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
* That problem usually follows a misunderstanding of the nature of :program:`MPD`. :program:`MPD` is a remote-controlled music player, not a music distribution system. Usually, the speakers are connected to the box where :program:`MPD` runs, and the :program:`MPD` client only sends control commands, but the client does not actually play your music.
:program:`MPD` has output plugins which allow hearing music on a remote host (such as httpd), but that is not :program:`MPD`'s primary design goal.
Question: "Device or resource busy"
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
Error "Device or resource busy"
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
* This ALSA error means that another program uses your sound hardware exclusively. You can stop that program to allow :program:`MPD` to use it.
@@ -1016,7 +1205,7 @@ Your bug report should contain:
* be clear about what you expect MPD to do, and what is actually happening
MPD crashes
~~~~~~~~~~~
^^^^^^^^^^^
All :program:`MPD` crashes are bugs which must be fixed by a developer, and you should write a bug report. (Many crash bugs are caused by codec libraries used by :program:`MPD`, and then that library must be fixed; but in any case, the :program:`MPD``bug tracker <https://github.com/MusicPlayerDaemon/MPD/issues>`_ is a good place to report it first if you don't know.)
@@ -1037,7 +1226,7 @@ You can extract the backtrace from a core dump, or by running :program:`MPD` in
..code-block::none
gdb --args mpd --stdout --no-daemon --verbose
gdb --args mpd --stderr --no-daemon --verbose
run
As soon as you have reproduced the crash, type ":command:`bt`" on the
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