329 lines
12 KiB
Plaintext
329 lines
12 KiB
Plaintext
# OpenAL config file. Options that are not under a block or are under the
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# [general] block are for general, non-backend-specific options. Blocks may
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# appear multiple times, and duplicated options will take the last value
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# specified.
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# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
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# specific override settings in ~/.alsoftrc.
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# For Windows, these settings should go into %AppData%\alsoft.ini
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# Option and block names are case-insenstive. The supplied values are only
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# hints and may not be honored (though generally it'll try to get as close as
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# possible). Note: options that are left unset may default to app- or system-
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# specified values. These are the current available settings:
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## disable-cpu-exts:
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# Disables use of the listed CPU extensions. Certain methods may utilize CPU
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# extensions when detected, and this option is useful for preventing those
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# extensions from being used. The available extensions are: sse, neon.
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# Specifying 'all' disables use of all extensions.
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#disable-cpu-exts =
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## channels:
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# Sets the output channel configuration. If left unspecified, one will try to
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# be detected from the system, and defaulting to stereo. The available values
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# are: mono, stereo, quad, surround51, surround61, surround71
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#channels = stereo
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## sample-type:
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# Sets the output sample type. Currently, all mixing is done with 32-bit float
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# and converted to the output sample type as needed. Available values are:
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# int8 - signed 8-bit int
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# uint8 - unsigned 8-bit int
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# int16 - signed 16-bit int
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# uint16 - unsigned 16-bit int
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# int32 - signed 32-bit int
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# uint32 - unsigned 32-bit int
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# float32 - 32-bit float
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#sample-type = float32
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## hrtf:
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# Enables HRTF filters. These filters provide for better sound spatialization
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# while using headphones. The default filter will only work when output is
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# 44100hz stereo. While HRTF is active, the cf_level option is disabled.
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# Default is disabled since stereo speaker output quality may suffer.
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#hrtf = false
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## hrtf_tables
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# Specifies a comma-separated list of files containing HRTF data sets. The
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# listed data sets can be used in place of or in addiiton to the the built-in
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# set. The format of the files are described in hrtf.txt. The filenames may
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# contain these markers, which will be replaced as needed:
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# %r - Device sampling rate
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# %% - Percent sign (%)
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# So if this is set to "kemar-%r-diffuse.mhr", it will try to open
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# "kemar-44100-diffuse.mhr" if the device is using 44100hz output, or
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# "kemar-48000-diffuse.mhr" if the device is using 48000hz output, etc.
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#hrtf_tables =
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## cf_level:
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# Sets the crossfeed level for stereo output. Valid values are:
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# 0 - No crossfeed
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# 1 - Low crossfeed
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# 2 - Middle crossfeed
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# 3 - High crossfeed (virtual speakers are closer to itself)
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# 4 - Low easy crossfeed
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# 5 - Middle easy crossfeed
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# 6 - High easy crossfeed
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# Users of headphones may want to try various settings. Has no effect on non-
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# stereo modes.
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#cf_level = 0
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## wide-stereo:
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# Specifies that stereo sources are given a width of about 120 degrees on each
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# channel, centering on -90 (left) and +90 (right), as opposed to being points
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# placed at -30 (left) and +30 (right). This can be useful for surround-sound
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# to give stereo sources a more encompassing sound. Note that the sound's
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# overall volume will be slightly reduced to account for the extra output.
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#wide-stereo = false
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## frequency:
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# Sets the output frequency.
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#frequency = 44100
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## resampler:
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# Selects the resampler used when mixing sources. Valid values are:
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# point - nearest sample, no interpolation
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# linear - extrapolates samples using a linear slope between samples
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# cubic - extrapolates samples using a Catmull-Rom spline
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# Specifying other values will result in using the default (linear).
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#resampler = linear
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## rt-prio:
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# Sets real-time priority for the mixing thread. Not all drivers may use this
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# (eg. PortAudio) as they already control the priority of the mixing thread.
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# 0 and negative values will disable it. Note that this may constitute a
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# security risk since a real-time priority thread can indefinitely block
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# normal-priority threads if it fails to wait. As such, the default is
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# disabled.
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#rt-prio = 0
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## period_size:
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# Sets the update period size, in frames. This is the number of frames needed
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# for each mixing update. Acceptable values range between 64 and 8192.
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#period_size = 1024
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## periods:
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# Sets the number of update periods. Higher values create a larger mix ahead,
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# which helps protect against skips when the CPU is under load, but increases
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# the delay between a sound getting mixed and being heard. Acceptable values
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# range between 2 and 16.
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#periods = 4
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## sources:
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# Sets the maximum number of allocatable sources. Lower values may help for
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# systems with apps that try to play more sounds than the CPU can handle.
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#sources = 256
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## drivers:
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# Sets the backend driver list order, comma-seperated. Unknown backends and
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# duplicated names are ignored. Unlisted backends won't be considered for use
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# unless the list is ended with a comma (eg. 'oss,' will list OSS first
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# followed by all other available backends, while 'oss' will list OSS only).
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# Backends prepended with - won't be available for use (eg. '-oss,' will allow
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# all available backends except OSS). An empty list means the default.
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#drivers = pulse,alsa,core,oss,solaris,sndio,qsa,mmdevapi,dsound,winmm,port,opensl,null,wave
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## excludefx:
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# Sets which effects to exclude, preventing apps from using them. This can
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# help for apps that try to use effects which are too CPU intensive for the
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# system to handle. Available effects are: eaxreverb,reverb,chorus,echo,
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# flanger,modulator,dedicated
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#excludefx =
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## slots:
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# Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
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# can use a non-negligible amount of CPU time if an effect is set on it even
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# if no sources are feeding it, so this may help when apps use more than the
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# system can handle.
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#slots = 4
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## sends:
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# Sets the number of auxiliary sends per source. When not specified (default),
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# it allows the app to request how many it wants. The maximum value currently
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# possible is 4.
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#sends =
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## layout:
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# Sets the virtual speaker layout. Values are specified in degrees, where 0 is
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# straight in front, negative goes left, and positive goes right. Unspecified
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# speakers will remain at their default positions (which are dependant on the
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# output format). Available speakers are back-left(bl), side-left(sl), front-
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# left(fl), front-center(fc), front-right(fr), side-right(sr), back-right(br),
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# and back-center(bc).
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#layout =
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## layout_*:
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# Channel-specific layouts may be specified to override the layout option. The
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# same speakers as the layout option are available, and the default settings
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# are shown below.
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#layout_stereo = fl=-90, fr=90
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#layout_quad = fl=-45, fr=45, bl=-135, br=135
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#layout_surround51 = fl=-30, fr=30, fc=0, bl=-110, br=110
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#layout_surround61 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bc=180
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#layout_surround71 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bl=-150, br=150
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## default-reverb:
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# A reverb preset that applies by default to all sources on send 0
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# (applications that set their own slots on send 0 will override this).
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# Available presets are: None, Generic, PaddedCell, Room, Bathroom,
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# Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
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# CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
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# Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
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#default-reverb =
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## trap-alc-error:
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# Generates a SIGTRAP signal when an ALC device error is generated, on systems
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# that support it. This helps when debugging, while trying to find the cause
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# of a device error. On Windows, a breakpoint exception is generated.
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#trap-alc-error = false
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## trap-al-error:
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# Generates a SIGTRAP signal when an AL context error is generated, on systems
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# that support it. This helps when debugging, while trying to find the cause
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# of a context error. On Windows, a breakpoint exception is generated.
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#trap-al-error = false
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##
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## Reverb effect stuff (includes EAX reverb)
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##
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[reverb]
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## boost:
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# A global amplification for reverb output, expressed in decibels. The value
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# is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a
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# scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A
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# value of 0 means no change.
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#boost = 0
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## emulate-eax:
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# Allows the standard reverb effect to be used in place of EAX reverb. EAX
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# reverb processing is a bit more CPU intensive than standard, so this option
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# allows a simpler effect to be used at the loss of some quality.
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#emulate-eax = false
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##
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## ALSA backend stuff
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##
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[alsa]
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## device:
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# Sets the device name for the default playback device.
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#device = default
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## device-prefix:
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# Sets the prefix used by the discovered (non-default) playback devices. This
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# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
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# device index for the requested device name.
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#device-prefix = plughw:
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## device-prefix-*:
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# Card- and device-specific prefixes may be used to override the device-prefix
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# option. The option may specify the card id (eg, device-prefix-NVidia), or
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# the card id and device index (eg, device-prefix-NVidia-0). The card id is
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# case-sensitive.
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#device-prefix- =
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## capture:
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# Sets the device name for the default capture device.
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#capture = default
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## capture-prefix:
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# Sets the prefix used by the discovered (non-default) capture devices. This
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# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
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# device number for the requested device name.
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#capture-prefix = plughw:
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## capture-prefix-*:
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# Card- and device-specific prefixes may be used to override the
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# capture-prefix option. The option may specify the card id (eg,
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# capture-prefix-NVidia), or the card id and device index (eg,
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# capture-prefix-NVidia-0). The card id is case-sensitive.
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#capture-prefix- =
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## mmap:
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# Sets whether to try using mmap mode (helps reduce latencies and CPU
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# consumption). If mmap isn't available, it will automatically fall back to
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# non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
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# and anything else will force mmap off.
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#mmap = true
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##
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## OSS backend stuff
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##
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[oss]
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## device:
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# Sets the device name for OSS output.
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#device = /dev/dsp
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## capture:
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# Sets the device name for OSS capture.
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#capture = /dev/dsp
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##
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## Solaris backend stuff
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##
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[solaris]
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## device:
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# Sets the device name for Solaris output.
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#device = /dev/audio
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##
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## MMDevApi backend stuff
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##
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[mmdevapi]
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##
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## DirectSound backend stuff
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##
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[dsound]
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##
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## Windows Multimedia backend stuff
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##
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[winmm]
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##
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## PortAudio backend stuff
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##
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[port]
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## device:
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# Sets the device index for output. Negative values will use the default as
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# given by PortAudio itself.
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#device = -1
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## capture:
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# Sets the device index for capture. Negative values will use the default as
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# given by PortAudio itself.
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#capture = -1
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##
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## PulseAudio backend stuff
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##
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[pulse]
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## spawn-server:
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# Attempts to spawn a PulseAudio server when requesting to open a PulseAudio
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# device. Setting autospawn to false in PulseAudio's client.conf will still
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# prevent autospawning even if this is set to true.
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#spawn-server = true
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## allow-moves:
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# Allows PulseAudio to move active streams to different devices. Note that the
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# device specifier seen by applications will not be updated when this occurs,
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# and neither will the AL device configuration (sample rate, format, etc).
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#allow-moves = false
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##
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## Wave File Writer stuff
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##
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[wave]
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## file:
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# Sets the filename of the wave file to write to. An empty name prevents the
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# backend from opening, even when explicitly requested.
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# THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
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#file =
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