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# 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 = 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#### 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