antkeeper
/
superbuild


								# OpenAL config file.

								#

								# Option blocks may appear multiple times, and duplicated options will take the

								# last value specified. Environment variables may be specified within option

								# values, and are automatically substituted when the config file is loaded.

								# Environment variable names may only contain alpha-numeric characters (a-z,

								# A-Z, 0-9) and underscores (_), and are prefixed with $. For example,

								# specifying "$HOME/file.ext" would typically result in something like

								# "/home/user/file.ext". To specify an actual "$" character, use "$$".

								#

								# Device-specific values may be specified by including the device name in the

								# block name, with "general" replaced by the device name. That is, general

								# options for the device "Name of Device" would be in the [Name of Device]

								# block, while ALSA options would be in the [alsa/Name of Device] block.

								# Options marked as "(global)" are not influenced by the device.

								#

								# The system-wide settings can be put in /etc/xdg/alsoft.conf (as determined by

								# the XDG_CONFIG_DIRS env var list, /etc/xdg being the default if unset) and

								# user-specific override settings in $HOME/.config/alsoft.conf (as determined

								# by the XDG_CONFIG_HOME env var).

								#

								# For Windows, these settings should go into $AppData\alsoft.ini

								#

								# An additional configuration file (alsoft.ini on Windows, alsoft.conf on other

								# OSs) can be placed alongside the process executable for app-specific config

								# settings.

								#

								# Option and block names are case-senstive. The supplied values are only hints

								# and may not be honored (though generally it'll try to get as close as

								# possible). Note: options that are left unset may default to app- or system-

								# specified values. These are the current available settings:


								##

								## General stuff

								##

								[general]


								## disable-cpu-exts: (global)

								#  Disables use of specialized methods that use specific CPU intrinsics.

								#  Certain methods may utilize CPU extensions for improved performance, and

								#  this option is useful for preventing some or all of those methods from being

								#  used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.

								#  Specifying 'all' disables use of all such specialized methods.

								#disable-cpu-exts =


								## drivers: (global)

								#  Sets the backend driver list order, comma-seperated. Unknown backends and

								#  duplicated names are ignored. Unlisted backends won't be considered for use

								#  unless the list is ended with a comma (e.g. 'oss,' will try OSS first before

								#  other backends, while 'oss' will try OSS only). Backends prepended with -

								#  won't be considered for use (e.g. '-oss,' will try all available backends

								#  except OSS). An empty list means to try all backends.

								#drivers =


								## channels:

								#  Sets the output channel configuration. If left unspecified, one will try to

								#  be detected from the system, and defaulting to stereo. The available values

								#  are: mono, stereo, quad, surround51, surround61, surround71, ambi1, ambi2,

								#  ambi3. Note that the ambi* configurations provide ambisonic channels of the

								#  given order (using ACN ordering and SN3D normalization by default), which

								#  need to be decoded to play correctly on speakers.

								#channels =


								## sample-type:

								#  Sets the output sample type. Currently, all mixing is done with 32-bit float

								#  and converted to the output sample type as needed. Available values are:

								#  int8    - signed 8-bit int

								#  uint8   - unsigned 8-bit int

								#  int16   - signed 16-bit int

								#  uint16  - unsigned 16-bit int

								#  int32   - signed 32-bit int

								#  uint32  - unsigned 32-bit int

								#  float32 - 32-bit float

								#sample-type = float32


								## frequency:

								#  Sets the output frequency. If left unspecified it will try to detect a

								#  default from the system, otherwise it will default to 44100.

								#frequency =


								## period_size:

								#  Sets the update period size, in sample frames. This is the number of frames

								#  needed for each mixing update. Acceptable values range between 64 and 8192.

								#  If left unspecified it will default to 1/50th of the frequency (20ms, or 882

								#  for 44100, 960 for 48000, etc).

								#period_size =


								## periods:

								#  Sets the number of update periods. Higher values create a larger mix ahead,

								#  which helps protect against skips when the CPU is under load, but increases

								#  the delay between a sound getting mixed and being heard. Acceptable values

								#  range between 2 and 16.

								#periods = 3


								## stereo-mode:

								#  Specifies if stereo output is treated as being headphones or speakers. With

								#  headphones, HRTF or crossfeed filters may be used for better audio quality.

								#  Valid settings are auto, speakers, and headphones.

								#stereo-mode = auto


								## stereo-encoding:

								#  Specifies the encoding method for non-HRTF stereo output. 'panpot' (default)

								#  uses standard amplitude panning (aka pair-wise, stereo pair, etc) between

								#  -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ

								#  output, which encodes some surround sound information into stereo output

								#  that can be decoded with a surround sound receiver. If crossfeed filters are

								#  used, UHJ is disabled.

								#stereo-encoding = panpot


								## ambi-format:

								#  Specifies the channel order and normalization for the "ambi*" set of channel

								#  configurations. Valid settings are: fuma, acn+fuma, ambix (or acn+sn3d), or

								#  acn+n3d

								#ambi-format = ambix


								## hrtf:

								#  Controls HRTF processing. These filters provide better spatialization of

								#  sounds while using headphones, but do require a bit more CPU power. While

								#  HRTF is used, the cf_level option is ignored. Setting this to auto (default)

								#  will allow HRTF to be used when headphones are detected or the app requests

								#  it, while setting true or false will forcefully enable or disable HRTF

								#  respectively.

								#hrtf = auto


								## hrtf-mode:

								#  Specifies the rendering mode for HRTF processing. Setting the mode to full

								#  (default) applies a unique HRIR filter to each source given its relative

								#  location, providing the clearest directional response at the cost of the

								#  highest CPU usage. Setting the mode to ambi1, ambi2, or ambi3 will instead

								#  mix to a first-, second-, or third-order ambisonic buffer respectively, then

								#  decode that buffer with HRTF filters. Ambi1 has the lowest CPU usage,

								#  replacing the per-source HRIR filter for a simple 4-channel panning mix, but

								#  retains full 3D placement at the cost of a more diffuse response. Ambi2 and

								#  ambi3 increasingly improve the directional clarity, at the cost of more CPU

								#  usage (still less than "full", given some number of active sources).

								#hrtf-mode = full


								## hrtf-size:

								#  Specifies the impulse response size, in samples, for the HRTF filter. Larger

								#  values increase the filter quality, while smaller values reduce processing

								#  cost. A value of 0 (default) uses the full filter size in the dataset, and

								#  the default dataset has a filter size of 32 samples at 44.1khz.

								#hrtf-size = 0


								## default-hrtf:

								#  Specifies the default HRTF to use. When multiple HRTFs are available, this

								#  determines the preferred one to use if none are specifically requested. Note

								#  that this is the enumerated HRTF name, not necessarily the filename.

								#default-hrtf =


								## hrtf-paths:

								#  Specifies a comma-separated list of paths containing HRTF data sets. The

								#  format of the files are described in docs/hrtf.txt. The files within the

								#  directories must have the .mhr file extension to be recognized. By default,

								#  OS-dependent data paths will be used. They will also be used if the list

								#  ends with a comma. On Windows this is:

								#  $AppData\openal\hrtf

								#  And on other systems, it's (in order):

								#  $XDG_DATA_HOME/openal/hrtf  (defaults to $HOME/.local/share/openal/hrtf)

								#  $XDG_DATA_DIRS/openal/hrtf  (defaults to /usr/local/share/openal/hrtf and

								#                               /usr/share/openal/hrtf)

								#hrtf-paths =


								## cf_level:

								#  Sets the crossfeed level for stereo output. Valid values are:

								#  0 - No crossfeed

								#  1 - Low crossfeed

								#  2 - Middle crossfeed

								#  3 - High crossfeed (virtual speakers are closer to itself)

								#  4 - Low easy crossfeed

								#  5 - Middle easy crossfeed

								#  6 - High easy crossfeed

								#  Users of headphones may want to try various settings. Has no effect on non-

								#  stereo modes.

								#cf_level = 0


								## resampler: (global)

								#  Selects the default resampler used when mixing sources. Valid values are:

								#  point - nearest sample, no interpolation

								#  linear - extrapolates samples using a linear slope between samples

								#  cubic - extrapolates samples using a Catmull-Rom spline

								#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying

								#            between 12 and 24 points, with anti-aliasing)

								#  fast_bsinc12 - same as bsinc12, except without interpolation between down-

								#                 sampling scales

								#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying

								#            between 24 and 48 points, with anti-aliasing)

								#  fast_bsinc24 - same as bsinc24, except without interpolation between down-

								#                 sampling scales

								#resampler = linear


								## rt-prio: (global)

								#  Sets the real-time priority value for the mixing thread. Not all drivers may

								#  use this (eg. PortAudio) as those APIs already control the priority of the

								#  mixing thread. 0 and negative values will disable real-time priority. Note

								#  that this may constitute a security risk since a real-time priority thread

								#  can indefinitely block normal-priority threads if it fails to wait. Disable

								#  this if it turns out to be a problem.

								#rt-prio = 1


								## rt-time-limit: (global)

								#  On non-Windows systems, allows reducing the process's RLIMIT_RTTIME resource

								#  as necessary for acquiring real-time priority from RTKit.

								#rt-time-limit = true


								## sources:

								#  Sets the maximum number of allocatable sources. Lower values may help for

								#  systems with apps that try to play more sounds than the CPU can handle.

								#sources = 256


								## slots:

								#  Sets the maximum number of Auxiliary Effect Slots an app can create. A slot

								#  can use a non-negligible amount of CPU time if an effect is set on it even

								#  if no sources are feeding it, so this may help when apps use more than the

								#  system can handle.

								#slots = 64


								## sends:

								#  Limits the number of auxiliary sends allowed per source. Setting this higher

								#  than the default has no effect.

								#sends = 6


								## front-stablizer:

								#  Applies filters to "stablize" front sound imaging. A psychoacoustic method

								#  is used to generate a front-center channel signal from the front-left and

								#  front-right channels, improving the front response by reducing the combing

								#  artifacts and phase errors. Consequently, it will only work with channel

								#  configurations that include front-left, front-right, and front-center.

								#front-stablizer = false


								## output-limiter:

								#  Applies a gain limiter on the final mixed output. This reduces the volume

								#  when the output samples would otherwise clamp, avoiding excessive clipping

								#  noise.

								#output-limiter = true


								## dither:

								#  Applies dithering on the final mix, for 8- and 16-bit output by default.

								#  This replaces the distortion created by nearest-value quantization with low-

								#  level whitenoise.

								#dither = true


								## dither-depth:

								#  Quantization bit-depth for dithered output. A value of 0 (or less) will

								#  match the output sample depth. For int32, uint32, and float32 output, 0 will

								#  disable dithering because they're at or beyond the rendered precision. The

								#  maximum dither depth is 24.

								#dither-depth = 0


								## volume-adjust:

								#  A global volume adjustment for source output, expressed in decibels. The

								#  value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will

								#  be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A

								#  value of 0 means no change.

								#volume-adjust = 0


								## excludefx: (global)

								#  Sets which effects to exclude, preventing apps from using them. This can

								#  help for apps that try to use effects which are too CPU intensive for the

								#  system to handle. Available effects are: eaxreverb,reverb,autowah,chorus,

								#  compressor,distortion,echo,equalizer,flanger,modulator,dedicated,pshifter,

								#  fshifter,vmorpher.

								#excludefx =


								## default-reverb: (global)

								#  A reverb preset that applies by default to all sources on send 0

								#  (applications that set their own slots on send 0 will override this).

								#  Available presets are: None, Generic, PaddedCell, Room, Bathroom,

								#  Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,

								#  CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,

								#  Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.

								#default-reverb =


								## trap-alc-error: (global)

								#  Generates a SIGTRAP signal when an ALC device error is generated, on systems

								#  that support it. This helps when debugging, while trying to find the cause

								#  of a device error. On Windows, a breakpoint exception is generated.

								#trap-alc-error = false


								## trap-al-error: (global)

								#  Generates a SIGTRAP signal when an AL context error is generated, on systems

								#  that support it. This helps when debugging, while trying to find the cause

								#  of a context error. On Windows, a breakpoint exception is generated.

								#trap-al-error = false


								##

								## Ambisonic decoder stuff

								##

								[decoder]


								## hq-mode:

								#  Enables a high-quality ambisonic decoder. This mode is capable of frequency-

								#  dependent processing, creating a better reproduction of 3D sound rendering

								#  over surround sound speakers.

								#hq-mode = true


								## distance-comp:

								#  Enables compensation for the speakers' relative distances to the listener.

								#  This applies the necessary delays and attenuation to make the speakers

								#  behave as though they are all equidistant, which is important for proper

								#  playback of 3D sound rendering. Requires the proper distances to be

								#  specified in the decoder configuration file.

								#distance-comp = true


								## nfc:

								#  Enables near-field control filters. This simulates and compensates for low-

								#  frequency effects caused by the curvature of nearby sound-waves, which

								#  creates a more realistic perception of sound distance. Note that the effect

								#  may be stronger or weaker than intended if the application doesn't use or

								#  specify an appropriate unit scale, or if incorrect speaker distances are set

								#  in the decoder configuration file.

								#nfc = false


								## nfc-ref-delay

								#  Specifies the reference delay value for ambisonic output when NFC filters

								#  are enabled. If channels is set to one of the ambi* formats, this option

								#  enables NFC-HOA output with the specified Reference Delay parameter. The

								#  specified value can then be shared with an appropriate NFC-HOA decoder to

								#  reproduce correct near-field effects. Keep in mind that despite being

								#  designed for higher-order ambisonics, this also applies to first-order

								#  output. When left unset, normal output is created with no near-field

								#  simulation. Requires the nfc option to also be enabled.

								#nfc-ref-delay =


								## quad:

								#  Decoder configuration file for Quadraphonic channel output. See

								#  docs/ambdec.txt for a description of the file format.

								#quad =


								## surround51:

								#  Decoder configuration file for 5.1 Surround (Side and Rear) channel output.

								#  See docs/ambdec.txt for a description of the file format.

								#surround51 =


								## surround61:

								#  Decoder configuration file for 6.1 Surround channel output. See

								#  docs/ambdec.txt for a description of the file format.

								#surround61 =


								## surround71:

								#  Decoder configuration file for 7.1 Surround channel output. See

								#  docs/ambdec.txt for a description of the file format. Note: This can be used

								#  to enable 3D7.1 with the appropriate configuration and speaker placement,

								#  see docs/3D7.1.txt.

								#surround71 =


								##

								## Reverb effect stuff (includes EAX reverb)

								##

								[reverb]


								## boost: (global)

								#  A global amplification for reverb output, expressed in decibels. The value

								#  is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a

								#  scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A

								#  value of 0 means no change.

								#boost = 0


								##

								## PipeWire backend stuff

								##

								[pipewire]


								## assume-audio: (global)

								#  Causes the backend to succeed initialization even if PipeWire reports no

								#  audio support. Currently, audio support is detected by the presence of audio

								#  source or sink nodes, although this can cause false negatives in cases where

								#  device availability during library initialization is spotty. Future versions

								#  of PipeWire are expected to have a more robust method to test audio support,

								#  but in the mean time this can be set to true to assume PipeWire has audio

								#  support even when no nodes may be reported at initialization time.

								#assume-audio = false


								##

								## PulseAudio backend stuff

								##

								[pulse]


								## spawn-server: (global)

								#  Attempts to autospawn a PulseAudio server whenever needed (initializing the

								#  backend, enumerating devices, etc). Setting autospawn to false in Pulse's

								#  client.conf will still prevent autospawning even if this is set to true.

								#spawn-server = true


								## allow-moves: (global)

								#  Allows PulseAudio to move active streams to different devices. Note that the

								#  device specifier (seen by applications) will not be updated when this

								#  occurs, and neither will the AL device configuration (sample rate, format,

								#  etc).

								#allow-moves = true


								## fix-rate:

								#  Specifies whether to match the playback stream's sample rate to the device's

								#  sample rate. Enabling this forces OpenAL Soft to mix sources and effects

								#  directly to the actual output rate, avoiding a second resample pass by the

								#  PulseAudio server.

								#fix-rate = false


								## adjust-latency:

								#  Attempts to adjust the overall latency of device playback. Note that this

								#  may have adverse effects on the resulting internal buffer sizes and mixing

								#  updates, leading to performance problems and drop-outs. However, if the

								#  PulseAudio server is creating a lot of latency, enabling this may help make

								#  it more manageable.

								#adjust-latency = false


								##

								## ALSA backend stuff

								##

								[alsa]


								## device: (global)

								#  Sets the device name for the default playback device.

								#device = default


								## device-prefix: (global)

								#  Sets the prefix used by the discovered (non-default) playback devices. This

								#  will be appended with "CARD=c,DEV=d", where c is the card id and d is the

								#  device index for the requested device name.

								#device-prefix = plughw:


								## device-prefix-*: (global)

								#  Card- and device-specific prefixes may be used to override the device-prefix

								#  option. The option may specify the card id (eg, device-prefix-NVidia), or

								#  the card id and device index (eg, device-prefix-NVidia-0). The card id is

								#  case-sensitive.

								#device-prefix- =


								## custom-devices: (global)

								#  Specifies a list of enumerated playback devices and the ALSA devices they

								#  refer to. The list pattern is "Display Name=ALSA device;...". The display

								#  names will be returned for device enumeration, and the ALSA device is the

								#  device name to open for each enumerated device.

								#custom-devices =


								## capture: (global)

								#  Sets the device name for the default capture device.

								#capture = default


								## capture-prefix: (global)

								#  Sets the prefix used by the discovered (non-default) capture devices. This

								#  will be appended with "CARD=c,DEV=d", where c is the card id and d is the

								#  device number for the requested device name.

								#capture-prefix = plughw:


								## capture-prefix-*: (global)

								#  Card- and device-specific prefixes may be used to override the

								#  capture-prefix option. The option may specify the card id (eg,

								#  capture-prefix-NVidia), or the card id and device index (eg,

								#  capture-prefix-NVidia-0). The card id is case-sensitive.

								#capture-prefix- =


								## custom-captures: (global)

								#  Specifies a list of enumerated capture devices and the ALSA devices they

								#  refer to. The list pattern is "Display Name=ALSA device;...". The display

								#  names will be returned for device enumeration, and the ALSA device is the

								#  device name to open for each enumerated device.

								#custom-captures =


								## mmap:

								#  Sets whether to try using mmap mode (helps reduce latencies and CPU

								#  consumption). If mmap isn't available, it will automatically fall back to

								#  non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0

								#  and anything else will force mmap off.

								#mmap = true


								## allow-resampler:

								#  Specifies whether to allow ALSA's built-in resampler. Enabling this will

								#  allow the playback device to be set to a different sample rate than the

								#  actual output, causing ALSA to apply its own resampling pass after OpenAL

								#  Soft resamples and mixes the sources and effects for output.

								#allow-resampler = false


								##

								## OSS backend stuff

								##

								[oss]


								## device: (global)

								#  Sets the device name for OSS output.

								#device = /dev/dsp


								## capture: (global)

								#  Sets the device name for OSS capture.

								#capture = /dev/dsp


								##

								## Solaris backend stuff

								##

								[solaris]


								## device: (global)

								#  Sets the device name for Solaris output.

								#device = /dev/audio


								##

								## QSA backend stuff

								##

								[qsa]


								##

								## JACK backend stuff

								##

								[jack]


								## spawn-server: (global)

								#  Attempts to autospawn a JACK server when initializing.

								#spawn-server = false


								## custom-devices: (global)

								#  Specifies a list of enumerated devices and the ports they connect to. The

								#  list pattern is "Display Name=ports regex;Display Name=ports regex;...". The

								#  display names will be returned for device enumeration, and the ports regex

								#  is the regular expression to identify the target ports on the server (as

								#  given by the jack_get_ports function) for each enumerated device.

								#custom-devices =


								## rt-mix:

								#  Renders samples directly in the real-time processing callback. This allows

								#  for lower latency and less overall CPU utilization, but can increase the

								#  risk of underruns when increasing the amount of work the mixer needs to do.

								#rt-mix = true


								## connect-ports:

								#  Attempts to automatically connect the client ports to physical server ports.

								#  Client ports that fail to connect will leave the remaining channels

								#  unconnected and silent (the device format won't change to accommodate).

								#connect-ports = true


								## buffer-size:

								#  Sets the update buffer size, in samples, that the backend will keep buffered

								#  to handle the server's real-time processing requests. This value must be a

								#  power of 2, or else it will be rounded up to the next power of 2. If it is

								#  less than JACK's buffer update size, it will be clamped. This option may

								#  be useful in case the server's update size is too small and doesn't give the

								#  mixer time to keep enough audio available for the processing requests.

								#  Ignored when rt-mix is true.

								#buffer-size = 0


								##

								## WASAPI backend stuff

								##

								[wasapi]


								##

								## DirectSound backend stuff

								##

								[dsound]


								##

								## Windows Multimedia backend stuff

								##

								[winmm]


								##

								## PortAudio backend stuff

								##

								[port]


								## device: (global)

								#  Sets the device index for output. Negative values will use the default as

								#  given by PortAudio itself.

								#device = -1


								## capture: (global)

								#  Sets the device index for capture. Negative values will use the default as

								#  given by PortAudio itself.

								#capture = -1


								##

								## Wave File Writer stuff

								##

								[wave]


								## file: (global)

								#  Sets the filename of the wave file to write to. An empty name prevents the

								#  backend from opening, even when explicitly requested.

								#  THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!

								#file =


								## bformat: (global)

								#  Creates AMB format files using first-order ambisonics instead of a standard

								#  single- or multi-channel .wav file.

								#bformat = false


								##

								## EAX extensions stuff

								##

								[eax]


								## enable: (global)

								#  Sets whether to enable EAX extensions or not.

								#enable = true


								##

								## Per-game compatibility options (these should only be set in per-game config

								## files, *NOT* system- or user-level!)

								##

								[game_compat]


								## reverse-x: (global)

								#  Reverses the local X (left-right) position of 3D sound sources.

								#reverse-x = false


								## reverse-y: (global)

								#  Reverses the local Y (up-down) position of 3D sound sources.

								#reverse-y = false


								## reverse-z: (global)

								#  Reverses the local Z (front-back) position of 3D sound sources.

								#reverse-z = false