# 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, surround3d71, # 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 48000. #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 include: None, Generic, PaddedCell, Room, Bathroom, # Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar, # CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Mountains, # 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 with surround sound # output. 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. For HRTF output, hrtf-mode must be set # to one of the ambi* values for this to function. #nfc = false ## speaker-dist: # Specifies the speaker distance in meters, used by the near-field control # filters with surround sound output. For ambisonic output modes, this value # is the basis for the NFC-HOA Reference Delay parameter (calculated as # delay_seconds = speaker_dist/343.3). This value is not used when a decoder # configuration is set for the output mode (since they specify the per-speaker # distances, overriding this setting), or when the NFC filters are off. Valid # values range from 0.1 to 10. #speaker-dist = 1 ## 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. #surround71 = ## surround3d71: # Decoder configuration file for 3D7.1 Surround channel output. See # docs/ambdec.txt for a description of the file format. See also # docs/3D7.1.txt for information about 3D7.1. #surround3d71 = ## ## 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] ## nfc-scale: (global) # A meters-per-unit distance scale applied to NFC filters. If a game doesn't # use real-world meters for in-game units, the filters may create a too-near # or too-distant effect. For instance, if the game uses 1 foot per unit, a # value of 0.3048 will correctly adjust the filters. Or if the game uses 1 # kilometer per unit, a value of 1000 will correctly adjust the filters. #nfc-scale = 1 ## 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