The PulseAudio sound server reads configuration directives from
a file ~/.config/pulse/daemon.conf on startup and when that
file doesn't exist from
@PA_DEFAULT_CONFIG_DIR@/daemon.conf . Please note that the
server also reads a configuration script on startup
default.pa which also contains runtime configuration
directives.
The configuration file is a simple collection of variable
declarations. If the configuration file parser encounters either ;
or # it ignores the rest of the line until its end.
For the settings that take a boolean argument the values
true , yes , on and 1
are equivalent, resp. false , no ,
off , 0 .
daemonize= Daemonize after startup. Takes a
boolean value, defaults to no . The --daemonize
command line option takes precedence.
fail= Fail to start up if any of the directives
in the configuration script default.pa
fail. Takes a boolean argument, defaults to yes . The --fail command line
option takes precedence.
allow-module-loading= Allow/disallow module
loading after startup. This is a security feature that if
dsabled makes sure that no further modules may be loaded into
the PulseAudio server after startup completed. It is recommended
to disable this when system-instance is
enabled. Please note that certain features like automatic
hot-plug support will not work if this option is enabled. Takes
a boolean argument, defaults to yes . The
--disallow-module-loading command line option takes
precedence.
allow-exit= Allow/disallow exit on user
request. Defaults to yes .
resample-method= The resampling algorithm to
use. Use one of src-sinc-best-quality ,
src-sinc-medium-quality , src-sinc-fastest ,
src-zero-order-hold , src-linear ,
trivial , speex-float-N ,
speex-fixed-N , ffmpeg . See the
documentation of libsamplerate and speex for explanations of the
different src- and speex- methods, respectively. The method
trivial is the most basic algorithm implemented. If
you're tight on CPU consider using this. On the other hand it has
the worst quality of them all. The Speex resamplers take an
integer quality setting in the range 0..10 (bad...good). They
exist in two flavours: fixed and float . The former uses fixed point
numbers, the latter relies on floating point numbers. On most
desktop CPUs the float point resampler is a lot faster, and it
also offers slightly better quality. See the output of
dump-resample-methods for a complete list of all
available resamplers. Defaults to speex-float-3 . The
--resample-method command line option takes precedence.
Note that some modules overwrite or allow overwriting of the
resampler to use.
enable-remixing= If disabled never upmix or
downmix channels to different channel maps. Instead, do a simple
name-based matching only. Defaults to yes.
enable-lfe-remixing= If disabled when upmixing or
downmixing ignore LFE channels. When this option is disabled the
output LFE channel will only get a signal when an input LFE
channel is available as well. If no input LFE channel is
available the output LFE channel will always be 0. If no output
LFE channel is available the signal on the input LFE channel
will be ignored. Defaults to no .
use-pid-file= Create a PID file in the runtime directory
($XDG_RUNTIMEDIR/pulse/pid ). If this is enabled you may
use commands like --kill or --check . If
you are planning to start more than one PulseAudio process per
user, you better disable this option since it effectively
disables multiple instances. Takes a boolean argument, defaults
to yes . The --use-pid-file command line
option takes precedence.
cpu-limit= If disabled do not install the CPU load
limiter, even on platforms where it is supported. This option is
useful when debugging/profiling PulseAudio to disable disturbing
SIGXCPU signals. Takes a boolean argument, defaults to
no . The --no-cpu-limit command line
argument takes precedence.
system-instance= Run the daemon as system-wide
instance, requires root priviliges. Takes a boolean argument,
defaults to no . The --system command line
argument takes precedence.
local-server-type= Please don't use this option if
you don't have to! This option is currently only useful when you
want D-Bus clients to use a remote server. This option may be
removed in future versions. If you only want to run PulseAudio
in the system mode, use the system-instance option.
This option takes one of user , system or
none as the argument. This is essentially a duplicate
for the system-instance option. The difference is the
none option, which is useful when you want to use a
remote server with D-Bus clients. If both this and
system-instance are defined, this option takes
precedence. Defaults to whatever the system-instance
is set.
enable-shm= Enable data transfer via POSIX
shared memory. Takes a boolean argument, defaults to
yes . The --disable-shm command line
argument takes precedence.
shm-size-bytes= Sets the shared memory segment
size for the daemon, in bytes. If left unspecified or is set to 0
it will default to some system-specific default, usually 64
MiB. Please note that usually there is no need to change this
value, unless you are running an OS kernel that does not do
memory overcommit.
lock-memory= Locks the entire PulseAudio process
into memory. While this might increase drop-out safety when used
in conjunction with real-time scheduling this takes away a lot
of memory from other processes and might hence considerably slow
down your system. Defaults to no .
flat-volumes= Enable 'flat' volumes, i.e. where
possible let the sink volume equal the maximum of the volumes of
the inputs connected to it. Takes a boolean argument, defaults
to yes .
high-priority= Renice the daemon after startup to
become a high-priority process. This a good idea if you
experience drop-outs during playback. However, this is a certain
security issue, since it works when called SUID root only, or
RLIMIT_NICE is used. root is dropped immediately after gaining
the nice level on startup, thus it is presumably safe. See
yes . The --high-priority
command line option takes precedence.
realtime-scheduling= Try to acquire SCHED_FIFO
scheduling for the IO threads. The same security concerns as
mentioned above apply. However, if PA enters an endless loop,
realtime scheduling causes a system lockup. Thus, realtime
scheduling should only be enabled on trusted machines for
now. Please not that only the IO threads of PulseAudio are made
real-time. The controlling thread is left a normally scheduled
thread. Thus enabling the high-priority option is orthogonal.
See yes . The
--realtime command line option takes precedence.
realtime-priority= The realtime priority to
acquire, if realtime-scheduling is enabled. Note: JACK uses 10
by default, 9 for clients. Thus it is recommended to choose the
PulseAudio real-time priorities lower. Some PulseAudio threads
might choose a priority a little lower or higher than the
specified value. Defaults to 5 .
nice-level= The nice level to acquire for the
daemon, if high-priority is enabled. Note: on some
distributions X11 uses -10 by default. Defaults to -11.
log-target= The default log target. Use either
stderr , syslog , auto ,
file:PATH or newfile:PATH . auto is
equivalent to sylog in case daemonize is enabled,
otherwise to stderr . If set to file:PATH , logging
is directed to the file indicated by PATH. newfile:PATH is
otherwise the same as file:PATH , but existing files are never
overwritten. If the specified file already exists, a suffix is added to
the file name to avoid overwriting. Defaults to auto . The
--log-target command line option takes precedence.
log-level= Log level, one of debug ,
info , notice , warning ,
error . Log messages with a lower log level than
specified here are not logged. Defaults to
notice . The --log-level command line
option takes precedence. The -v command line option
might alter this setting.
log-meta= With each logged message log the code
location the message was generated from. Defaults to
no .
log-time= With each logged messages log the
relative time since startup. Defaults to no .
log-backtrace= When greater than 0, with each
logged message log a code stack trace up the specified
number of stack frames. Defaults to 0 .
See
rlimit-as Defaults to -1.
rlimit-rss Defaults to -1.
rlimit-core Defaults to -1.
rlimit-data Defaults to -1.
rlimit-fsize Defaults to -1.
rlimit-nofile Defaults to 256.
rlimit-stack Defaults to -1.
rlimit-nproc Defaults to -1.
rlimit-locks Defaults to -1.
rlimit-sigpending Defaults to -1.
rlimit-msgqueue Defaults to -1.
rlimit-memlock Defaults to 16 KiB. Please note
that the JACK client libraries may require more locked
memory.
rlimit-nice Defaults to 31. Please make sure that
the default nice level as configured with nice-level
fits in this resource limit, if high-priority is
enabled.
rlimit-rtprio Defaults to 9. Please make sure that
the default real-time priority level as configured with
realtime-priority= fits in this resource limit, if
realtime-scheduling is enabled. The JACK client
libraries require a real-time prority of 9 by default.
rlimit-rttime Defaults to 1000000.
Most drivers try to open the audio device with these settings
and then fall back to lower settings. The default settings are CD
quality: 16bit native endian, 2 channels, 44100 Hz sampling.
default-sample-format= The default sampling
format. Specify one of u8 , s16le ,
s16be , s24le , s24be ,
s24-32le , s24-32be , s32le ,
s32be float32le , float32be ,
ulaw , alaw . Depending on the endianness of
the CPU the formats s16ne , s16re ,
s24ne , s24re , s24-32ne ,
s24-32re , s32ne , s32re ,
float32ne , float32re (for native,
resp. reverse endian) are available as aliases.
default-sample-rate= The default sample frequency.
default-sample-channels The default number of channels.
default-channel-map The default channel map.
alternate-sample-rate The alternate sample
frequency. Sinks and sources will use either the
default-rate-rate value or this alternate value, typically 44.1
or 48kHz. Switching between default and alternate values is
enabled only when the sinks/sources are suspended. This option
is ignored in passthrough mode where the stream rate will be used.
If set to the same as the default sample rate, this feature is
disabled.
With the flat volume feature enabled, the sink HW volume is set
to the same level as the highest volume input stream. Any other streams
(with lower volumes) have the appropriate adjustment applied in SW to
bring them to the correct overall level. Sadly hadware mixer changes
cannot be timed accurately and thus this change of volumes can somtimes
cause the resulting output sound to be momentarily too loud or too soft.
So to ensure SW and HW volumes are applied concurrently without any
glitches, their application needs to be synchronized. The sink
implementation needs to support deferred volumes. The following
parameters can be used to refine the process.
enable-deferred-volume= Enable deferred volume for the sinks that
support it. This feature is enabled by default.
deferred-volume-safety-margin-usec= The amount of time (in
usec) by which the HW volume increases are delayed and HW volume
decreases are advanced. Defaults to 8000 usec.
deferred-volume-extra-delay-usec= The amount of time (in usec)
by which HW volume changes are delayed. Negative values are also allowed.
Defaults to 0.