Without clearing the "uri" field, the next Open() call attempts to
reuse the old InputStream, but it has already been closed, so Open()
always returns nullptr.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1300
With libfmt versions older than 7, this leads to an endless recursion
between Error() and FmtError(), resulting in a crash due to stack
overflow. D'oh!
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1295
During the libfmt migration, I converted "%1.3f" to just "{:1.3}"
without the "f" suffix, but libfmt defaults to scientific notation,
which can break some MPD clients.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1291
This replaces the output parameter (which is bad API design). As a
side effect, it fixes the bad [[gnu::pure]] attribute added by commit
a636d2127 which caused optimizing compilers to miscompile calls to
that function. "Pure" functions can be assumed to have no output
arguments, so the compiler can assume the function doesn't modify
them.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1282
Fixes https://github.com/MusicPlayerDaemon/MPD/discussions/1281
The problem occurred when there was libfmt-dev installed, but it was
too old (e.g. on Debian Buster), and Meson used the wrap fallback.
Those internal MPD libraries where the libfmt dependency was not
declared were still using the old system libfmt headers, which are not
ABI-compatible with MPD's own libfmt build.
From the feature request: "I generally like to have crossfade on, but
when it happens during such short tracks (e.g. 20 seconds or less) it
doesn't really sound good as those tracks are not really meant to be
crossfaded and intended to act as a bridge on their own."
Sounds reasonable. This commit doesn't add an option, but hard-codes
the limit to 20 seconds. If it turns out that users want to have it
configurable, we can still add the option.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1184
Now, "+0" means "right after the current song" and "-0" means "right
before the current song". Mnemonic: there are zero songs between the
current song and the newly added song.
A similar feature was present long ago in MPD, but was deprecated in
version 0.16 because the implementation was broken. This commit
re-adds the feature in a way that's well-defined and not broken.
Close https://github.com/MusicPlayerDaemon/MPD/issues/1221
NB: Check the sysconfdir setting to determine where mpd will look for mpd.conf; if you expect mpd to look for /etc/mpd.conf the sysconfdir must be '/etc' (i.e., not 'etc' which will result in mpd looking for /usr/local/etc/mpd.conf):
.. code-block:: none
meson configure output/release |grep sysconfdir
If this is not /etc (or another path you wish to specify):
.. code-block:: none
$ meson configure output/release -Dsysconfdir='/etc' ; meson configure output/release |grep syscon
sysconfdir /etc Sysconf data directory
The ChainFilter class is extremely complicated code, and will grow to
be even more complicated when the Filter interface gets extended.
Let's just remove it; we can easily chain many TwoFilters instead.
If DetectFilterOutputFormat() fails to determine the output format,
insert an "aformat" filter which attempts to force a specific output
format.
Fixes part 2 of of https://github.com/MusicPlayerDaemon/MPD/issues/1235
Don't truncate the FILETIME to second resolution to pass it to
std::chrono::system_clock::from_time_t(); instead, calculate the
offset between the FILETIME epoch and the
std::system_clock::time_point epoch, and use that to initialize the
time_point directly.
If draining was not requested, generate silence instead if there is no
data in the ring buffer.
The problem is that pw_stream_flush() appears to disable the stream
permanently, even though there is no state_changed callback - the
stream state remains at PW_STREAM_STATE_STREAMING, but the stream is
defunct. I have no idea why and I havn't found any documentation
about it.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1219
If UpdateStreamTag() gets called while an initial seek is pending, the
result will never be submitted to a MusicChunk. By avoiding the
UpdateStreamTag() call in that case (by moving UpdateStreamTag() to
after the PrepareInitialSeek() check), the song_tag is preserved until
UpdateStreamTag() is called again from SubmitData().
This fixes missing tags in the "httpd" output.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1137
Explanation
This adds support for DOP using the PcmExport function if the macro
ENABLE_DSD is defined. If enabled within the config-file using "dop",
the boolean dop_setting will be true. If DSD input is encountered and
the setting is on, it is checked whether the oss-device supports the
required samplerate. If that is the case, dop_active is set to true
and conversion of the input is prevented. If the sample rate is not
supported, conversion to S32 is requested. When playing back, the
PcmExport is used to pack the incoming stream into PCM. Reasoning
This is required for OSs without the required driver support for
native DSD playback that also have no ALSA. Mainly *BSD users are the
target audience for this functionality, as ALSA here is only a proxy
without full functionality. Requirements
DAC that supports the DOP standard
Building with OSS, DSD and S32-Format
Supported Formats / Required PCM Formats
DSF, DFF and WavPack-DSD will work.
DSD64, 1 Channel -> S24:176.4kHz (untested, lack of time / missing samples)
DSD64, 2 Channel -> S24:352.8kHz
DSD64, 4 Channel -> S24:705.6kHz (untested, lmissing equipment)
DSD128, 1 Channel -> S24:352.8kHz (untested, lack of time / missing samples)
DSD128, 2 Channel -> S24:705.6kHz
DSD256, 1 Channel -> S24:705.6kHz (untested, lack of time / missing samples)
Changes
inclusion of required files
adding new domain for logging
adding dop_satisfied private function
adding required member variables for storing dop state and for dop-packing
adding dop boolean parameter to many functions that are required to act a little differently when dop is active
Testing
This has been tested to work with a Sabaj Da2 on FreeBSD, where the
red status indicator LED clearly shows that DSD playback is taking
place, instead of purple for "hi-res" which is seen when converting.
Issues
I have not tested this with S24 and right now AFMT_S32_NE is
required. If not defined, ENABLE_DSD will be undef'ed. This will be
addressed in a bit, however no DAC which supports DOP but not 32Bit is
known to me. Also, AFMT_S32_NE is not defined when building on FreeBSD
which is why this is just blatantly defined in the file at the moment.
Additionally, the new dop-option is not added into any documentation
whatsoever.
We need this even when AFMT_S24_PACKED is not available, for the
correct channel order in multi-channel files. Internally, MPD uses
FLAC channel order, but OSS uses the same channel order as ALSA.
Also, use RemoveAll() instead of directly clearing TagBuilder::items in
most cases, as its elements represent references that must be released.
Closes#1023
ProxyInputStream::Read() assigns the `offset` field, which is the
wrong offset because it does not consider Icy metadata removed from
the stream. Therefore, after every ProxyInputStream::Read() call,
IcyInputStream::Read() needs to override this offset. This was
missing at the end of the stream, when Read()==0.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1216
Fixes
../git/src/Log.hxx:121:42: error: no member named 'forward' in namespace 'std'
LogFormat(LogLevel::ERROR, e, fmt, std::forward<Args>(args)...);
Signed-off-by: Khem Raj <raj.khem@gmail.com>
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.
refactors GenerateOggSerial into a generic GenerateSerial
utility, under the util lib.
libFLAC may be encoded without Ogg support. If Ogg support is disabled,
libFLAC will still export Ogg-related methods (like setting a serial
number), and throw a runtime error when initializing an Ogg stream.
GenerateOggSerial does not depend on libogg. Refactoring it into
a generic GenerateSerial prevents having to add build-time checks
for libogg within the FLAC encoder plugin.
Add support for the following tags when using the proxy database plugin:
WORK
CONDUCTOR
LABEL
GROUPING
MUSICBRAINZ_WORKID
COMPOSERSORT
ENSEMBLE
MOVEMENT
MOVEMENTNUMBER
LOCATION
This commit adds some tags that are (mostly) interesting for listeners
of classical music.
Ensemble
--------
This is an ensemble that is playing the music, such as Wiener
Philharmoniker. The tag can be used to distinguish the ensemble from the
conductor, composer, soloist, and ensemble, that are generally all in
the "ARTIST" tag.
Movement
-------
The movement number and movement (name) of this track, i.e. "II" and
"Allegro".
ComposerSort
------------
Allows us to look for Beethoven's 9th under B, for Beethoven, not L for
Ludwig.
Location
--------
This is the location of the recording, e.g. "Wiener Musikverein".
By casting to SafeLinkIntrusiveListHook if appropriate, this fixes a
bug in the erase() method, where erase() calls
IntrusiveListHook::unlink() instead of
SafeLinkIntrusiveListHook::unlink().
Add cacert option for curl plugin
add cacert option for Curl plugin. Allows to set cacert for curl lib
Added documentation line into doc/plugins.rst with explanation for cacert option
If an InjectEvent callback schedules a timer, the loop will restart,
but the "busy" flag is still false. The fix is to move the "again"
check before the "busy" setting.
SocketEvent knows the FD is still open and is about to close it, so
it's unnecessary to rely on the kernel (via AbandonFD) to clean up the
epoll_wait list.
### Why this is relevant
- `AbandonFD` assumes that upon closing the socket, the FD will be automatically removed from the epoll list. That fd is associated with a reference to the `SocketEvent`, so this is an important and dangerous assumption to get wrong. In the case that the FD isn't immediately removed from the list by the kernel, the event loop can crash due to the `SocketEvent` being destroyed and it being a use-after-free bug at that point.
- If a socket FD happens to be duplicated, then closing the SocketEvent FD will not automatically remove it from epoll, and will trigger said bug/crash. It is only automatically removed when all FD references to the underlying socket/resource are closed?
- A `fork()` is one example where a socket FD can be duplicated and result in this situation.
- `CLOEXEC` might be considered mitigation for this but also introduces a race condition where the crash can occur between a `fork()` and `exec()` without additional synchronization to freeze the event loop.
One could argue the mpd event loop isn't fork-safe, and thus should be allowed to use `AbandonFD` however it likes. A decision on whether this is intended should probably be declared; but either way this fix seems appropriate in cases where `Abandon` isn't actually necessary. It also might be possible to fix `AbandonFD` to mark the `SocketEvent` as removed without using `EPOLL_CTL_DEL`?
[edit: made this dependent on HAVE_THREADED_EVENT_LOOP which is always
true for MPD, but not for ncmpc, for example - mk]
This reverts commit 7bc1c9925b. It
caused a crash with the ALSA plugin family (through
MultiSocketMonitor::ReplaceSocketList() and
MultiSocketMonitor::AddSocket()). Until we have a proper fix, the
assertion patch is reverted.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1020
Instead of using this as a base class implementing a virtual method,
the new class IdleEvent can be used as a variable, decoupling
IdleMonitor's internal state from the derived class.
This is similar to commit 30a5dd267b
which refactored TimeoutMonitor to TimerEvent.
@@ -23,11 +23,23 @@ The default plugin. Stores a copy of the database in memory. A file is used for
- The path of the cache directory for additional storages mounted at runtime. This setting is necessary for the **mount** protocol command.
* - **compress yes|no**
- Compress the database file using gzip? Enabled by default (if built with zlib).
* - **hide_playlist_targets yes|no**
- Hide songs which are referenced by playlists? Thas is,
playlist files which are represented in the database as virtual
directories (playlist plugin setting ``as_directory``). This
option is enabled by default and avoids duplicate songs; one
copy for the original file, and another copy in the virtual
directory of a CUE file referring to it.
proxy
-----
Provides access to the database of another :program:`MPD` instance using libmpdclient. This is useful when you mount the music directory via NFS/SMB, and the file server already runs a :program:`MPD` instance. Only the file server needs to update the database.
Provides access to the database of another :program:`MPD` instance
using `libmpdclient
<https://www.musicpd.org/libs/libmpdclient/>`_. This is useful when
you mount the music directory via NFS/SMB, and the file server already
runs a :program:`MPD` (0.20 or newer) instance. Only the file server
needs to update the database.
..list-table::
:widths:20 80
@@ -210,6 +222,8 @@ will be in effect.
- Verify the peer's SSL certificate? `More information <http://curl.haxx.se/libcurl/c/CURLOPT_SSL_VERIFYPEER.html>`_.
* - **verify_host yes|no**
- Verify the certificate's name against host? `More information <http://curl.haxx.se/libcurl/c/CURLOPT_SSL_VERIFYHOST.html>`_.
* - **cacert**
- Set path to Certificate Authority (CA) bundle `More information <https://curl.se/libcurl/c/CURLOPT_CAINFO.html>`_.
ffmpeg
------
@@ -295,37 +309,6 @@ in the form ``qobuz://track/ID``, e.g.:
* - **format_id N**
- 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.:
..warning::
This plugin is currently defunct because Tidal has changed the
protocol and decided not to share documentation.
..code-block::none
mpc add tidal://track/59727857
..list-table::
:widths:20 80
:header-rows:1
* - Setting
- Description
* - **token TOKEN**
- The Tidal application token. Since Tidal is unwilling to assign a token to MPD, this needs to be reverse-engineered from another (approved) Tidal client.
* - **username USERNAME**
- The Tidal user name.
* - **password PASSWORD**
- The Tidal password.
* - **audioquality Q**
- The Tidal "audioquality" parameter. Possible values: HI_RES, LOSSLESS, HIGH, LOW. Default is HIGH.
.._decoder_plugins:
Decoder plugins
@@ -482,9 +465,39 @@ Module player based on MODPlug.
* - Setting
- Description
* - **resampling_mode nearest|linear|spline|fir**
- Sets the resampling mode. "nearest" disables interpolation (good for chiptunes). "linear" makes modplug use linear interpolation (fast, good quality). "spline" makes modplug use cubic spline interpolation (high quality). "fir" makes modplug use 8-tap fir filter (extremely high quality). Defaults to "fir".
* - **loop_count**
- Number of times to loop the module if it uses backward loops. Default is 0 which prevents looping. -1 loops forever.
openmpt
-------
Module player based on `libopenmpt <https://lib.openmpt.org>`_.
..list-table::
:widths:20 80
:header-rows:1
* - Setting
- Description
* - **repeat_count**
- Set how many times the module repeats. -1: repeat forever. 0: play once, repeat zero times (the default). n>0: play once and repeat n times after that.
* - **stereo_separation**
- Sets the stereo separation. The supported value range is [0,200]. Defaults to 100.
* - **interpolation_filter 0|1|2|4|8**
- Sets the interpolation filter. 0: internal default. 1: no interpolation (zero order hold). 2: linear interpolation. 4: cubic interpolation. 8: windowed sinc with 8 taps. Defaults to 0.
* - **override_mptm_interp_filter yes|no**
- If `interpolation_filter` has been changed, setting this to yes will force all MPTM modules to use that interpolation filter. If set to no, MPTM modules will play with their own interpolation filter regardless of the value of `interpolation_filter`. Defaults to no.
* - **volume_ramping**
- Sets the amount of volume ramping done by the libopenmpt mixer. The default value is -1, which indicates a recommended default value. The meaningful value range is [-1..10]. A value of 0 completely disables volume ramping. This might cause clicks in sound output. Higher values imply slower/softer volume ramps.
* - **sync_samples yes|no**
- Syncs sample playback when seeking. Defaults to yes.
* - **emulate_amiga yes|no**
- Enables the Amiga resampler for Amiga modules. This emulates the sound characteristics of the Paula chip and overrides the selected interpolation filter. Non-Amiga module formats are not affected by this setting. Defaults to yes.
* - **emulate_amiga_type**
- Configures the filter type to use for the Amiga resampler. Supported values are: "auto": Filter type is chosen by the library and might change. This is the default. "a500": Amiga A500 filter. "a1200": Amiga A1200 filter. "unfiltered": BLEP synthesis without model-specific filters. The LED filter is ignored by this setting. This filter mode is considered to be experimental and might change in the future. Defaults to "auto". Requires libopenmpt 0.5 or higher.
mpcdec
------
@@ -579,6 +592,10 @@ Encodes into `FLAC <https://xiph.org/flac/>`_ (lossless).
- Description
* - **compression**
- Sets the libFLAC compression level. The levels range from 0 (fastest, least compression) to 8 (slowest, most compression).
* - **oggflac yes|no**
- Configures if the stream should be Ogg FLAC versus native FLAC. Defaults to "no" (use native FLAC).
* - **oggchaining yes|no**
- Configures if the stream should use Ogg Chaining for in-stream metadata. Defaults to "no". Setting this to "yes" also enables Ogg FLAC.
lame
----
@@ -643,11 +660,15 @@ Encodes into `Ogg Opus <http://www.opus-codec.org/>`_.
* - Setting
- Description
* - **bitrate**
- Sets the data rate in bit per second. The special value "auto" lets libopus choose a rate (which is the default), and "max" uses the maximum possible data rate.
- Sets the data rate in bits per second. The special value "auto" lets libopus choose a rate (which is the default), and "max" uses the maximum possible data rate.
* - **complexity**
- Sets the `Opus complexity <https://wiki.xiph.org/OpusFAQ#What_is_the_complexity_of_Opus.3F>`_.
* - **signal**
- Sets the Opus signal type. Valid values are "auto" (the default), "voice" and "music".
* - **vbr yes|no|constrained**
- Sets the vbr mode. Setting to "yes" (default) enables variable bitrate, "no" forces constant bitrate and frame sizes, "constrained" uses constant bitrate analogous to CBR in AAC and MP3.
* - **packet_loss**
- Sets the expected packet loss percentage. This value can be increased from the default "0" for a more redundant stream at the expense of quality.
* - **opustags yes|no**
- Configures how metadata is interleaved into the stream. If set to yes, then metadata is inserted using ogg stream chaining, as specified in :rfc:`7845`. If set to no (the default), then ogg stream chaining is avoided and other output-dependent method is used, if available.
@@ -715,7 +736,7 @@ 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:
@@ -990,6 +1011,8 @@ On Linux, OSS has been superseded by ALSA. Use the ALSA output plugin :ref:`alsa
- Description
* - **device PATH**
- Sets the path of the PCM device. If not specified, then MPD will attempt to open /dev/sound/dsp and /dev/dsp.
* - **dop yes|no**
- If set to yes, then DSD over PCM according to the `DoP standard <http://dsd-guide.com/dop-open-standard>`_ is enabled. This wraps DSD samples in fake 24 bit PCM, and is understood by some DSD capable products, but may be harmful to other hardware. Therefore, the default is no and you can enable the option at your own risk.
The according hardware mixer plugin understands the following settings:
@@ -1052,6 +1075,28 @@ The pipe plugin starts a program and writes raw PCM data into its standard input
* - **command CMD**
- This command is invoked with the shell.
pipewire
--------
Connect to a `PipeWire <https://pipewire.org/>`_ server. Requires
``libpipewire``.
..list-table::
:widths:20 80
:header-rows:1
* - Setting
- Description
* - **target NAME**
- Link to the given target. If not specified, let the PipeWire
manager select a target. To get a list of available targets,
type ``pw-cli dump short Node``
* - **remote NAME**
- The name of the remote to connect to. The default is
``pipewire-0``.
* - **dsd yes|no**
- Enable DSD playback. This requires PipeWire 0.38.
.._pulse_plugin:
pulse
@@ -1107,8 +1152,6 @@ You must set a format.
- Sets the host name of the `ShoutCast <http://www.shoutcast.com/>`_ / `IceCast <http://icecast.org/>`_ server.
* - **port PORTNUMBER**
- Connect to this port number on the specified host.
* - **timeout SECONDS**
- Set the timeout for the shout connection in seconds. Defaults to 2 seconds.
* - **protocol icecast2|icecast1|shoutcast**
- Specifies the protocol that wil be used to connect to the server. The default is "icecast2".
@@ -110,6 +111,19 @@ The following command shows a list of compile-time options:
meson configure output/release
NB: Check the sysconfdir setting to determine where mpd will look for mpd.conf; if you expect mpd to look for /etc/mpd.conf the sysconfdir must be '/etc' (i.e., not 'etc' which will result in mpd looking for /usr/local/etc/mpd.conf):
..code-block::none
meson configure output/release |grep sysconfdir
If this is not /etc (or another path you wish to specify):
*`Meson 0.56.0 <http://mesonbuild.com/>`__ and `Ninja
<https://ninja-build.org/>`__
* cmake
* pkg-config
@@ -625,13 +639,20 @@ By default, all clients are unauthenticated and have a full set of permissions.
- Allows reading of the database, displaying the current playlist, and current status of :program:`MPD`.
* - **add**
- Allows adding songs and loading playlists.
* - **player**
- Allows any player and queue manipulation (start/pause/stop
playback etc.).
* - **control**
- Allows all other player and playlist manipulations.
* - **admin**
- Allows database updates and allows shutting down :program:`MPD`.
- Allows manipulating outputs, stickers and partitions,
mounting/unmounting storage and shutting down :program:`MPD`.
:code:`local_permissions` may be used to assign other permissions to clients connecting on a local socket.
:code:`host_permissions` may be used to assign permissions to clients
with a certain IP address.
:code:`password` allows the client to send a password to gain other permissions. This option may be specified multiple times with different passwords.
Note that the :code:`password` option is not secure: passwords are sent in clear-text over the connection, and the client cannot verify the server's identity.
- 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
^^^^^^^^^^^^^^^^^^^^
@@ -1120,7 +1145,7 @@ 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
^^^^^^^^^^^^^^^
@@ -1202,6 +1227,34 @@ Your bug report should contain:
* relevant portions of the log file (:option:`--verbose`)
* be clear about what you expect MPD to do, and what is actually happening
.._profiler:
Too Much CPU Usage
^^^^^^^^^^^^^^^^^^
If you believe MPD consumes too much CPU, `write a bug report
<https://github.com/MusicPlayerDaemon/MPD/issues>`_ with a profiling
information.
On Linux, this can be obtained with :program:`perf` (on Debian,
installed the package :file:`linux-perf`), for example::
perf record -p `pidof mpd`
Run this command while MPD consumes much CPU, let it run for a minute
or so, and stop it by pressing ``Ctrl-C``. Then type::
perf report >mpd_perf.txt
Upload the output file to the bug report.
..note::
This requires having debug symbols for MPD and all relevant
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