# 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/openal/alsoft.conf and user-
# specific override settings in $HOME/.alsoftrc.
# For Windows, these settings should go into $AppData\alsoft.ini
#
# 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, surround51rear, 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 frames. This is the number of frames needed
# for each mixing update. Acceptable values range between 64 and 8192.
#period_size = 1024
## 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+sn3d, acn+n3d
#ambi-format = acn+sn3d
## hrtf:
# Controls HRTF processing. These filters provide better spatialization of
# sounds while using headphones, but do require a bit more CPU power. The
# default filters will only work with 44100hz or 48000hz stereo output. 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
## 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 resampler used when mixing sources. Valid values are:
# point - nearest sample, no interpolation
# linear - extrapolates samples using a linear slope between samples
# sinc4 - extrapolates samples using a 4-point Sinc filter
# bsinc - extrapolates samples using a band-limited Sinc filter (varying
# between 12 and 24 points, with anti-aliasing)
# Specifying other values will result in using the default (linear).
#resampler = linear
## rt-prio: (global)
# Sets real-time priority for the mixing thread. Not all drivers may use this
# (eg. PortAudio) as they already control the priority of the mixing thread.
# 0 and negative values will disable it. 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. As such, the default is
# disabled.
#rt-prio = 0
## 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 = 16
## 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,chorus,compressor,
# distortion,echo,equalizer,flanger,modulator,dedicated
#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. Enabling this also requires specifying decoder
# configuration files for the appropriate speaker configuration you intend to
# use (see the quad, surround51, etc options below). Currently, up to third-
# order decoding is supported.
hq-mode = false
## 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. Requires hq-mode to be enabled.
nfc = true
## nfc-ref-delay
# Specifies the reference delay value for ambisonic output. When 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
# applies to first-order output all the same. When left unset, normal output
# is created with no near-field simulation.
nfc-ref-delay =
## quad:
# Decoder configuration file for Quadrophonic 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
## emulate-eax: (global)
# Allows the standard reverb effect to be used in place of EAX reverb. EAX
# reverb processing is a bit more CPU intensive than standard, so this option
# allows a simpler effect to be used at the loss of some quality.
#emulate-eax = 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 = false
## 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
##
## 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- =
## 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- =
## 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 whenever needed (initializing the
# backend, opening devices, etc).
#spawn-server = false
## 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.
#buffer-size = 0
##
## MMDevApi backend stuff
##
[mmdevapi]
##
## 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