# OpenAL config file. Options that are not under a block or are under the # [general] block are for general, non-backend-specific options. Blocks may # appear multiple times, and duplicated options will take the last value # specified. # The system-wide settings can be put in /etc/openal/alsoft.conf and user- # specific override settings in ~/.alsoftrc. # For Windows, these settings should go into %AppData%\alsoft.ini # Option and block names are case-insenstive. 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: ## disable-cpu-exts: # Disables use of the listed CPU extensions. Certain methods may utilize CPU # extensions when detected, and this option is useful for preventing those # extensions from being used. The available extensions are: sse, neon. # Specifying 'all' disables use of all extensions. #disable-cpu-exts = ## 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 #channels = stereo ## 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 ## hrtf: # Enables HRTF filters. These filters provide for better sound spatialization # while using headphones. The default filter will only work when output is # 44100hz stereo. While HRTF is active, the cf_level option is disabled. # Default is disabled since stereo speaker output quality may suffer. #hrtf = false ## hrtf_tables # Specifies a comma-separated list of files containing HRTF data sets. The # listed data sets can be used in place of or in addiiton to the the built-in # set. The format of the files are described in hrtf.txt. The filenames may # contain these markers, which will be replaced as needed: # %r - Device sampling rate # %% - Percent sign (%) # So if this is set to "kemar-%r-diffuse.mhr", it will try to open # "kemar-44100-diffuse.mhr" if the device is using 44100hz output, or # "kemar-48000-diffuse.mhr" if the device is using 48000hz output, etc. #hrtf_tables = ## 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 ## wide-stereo: # Specifies that stereo sources are given a width of about 120 degrees on each # channel, centering on -90 (left) and +90 (right), as opposed to being points # placed at -30 (left) and +30 (right). This can be useful for surround-sound # to give stereo sources a more encompassing sound. Note that the sound's # overall volume will be slightly reduced to account for the extra output. #wide-stereo = false ## frequency: # Sets the output frequency. #frequency = 44100 ## resampler: # Selects the 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 # Specifying other values will result in using the default (linear). #resampler = linear ## rt-prio: # 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 ## 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 = 4 ## 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 ## drivers: # 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 (eg. 'oss,' will list OSS first # followed by all other available backends, while 'oss' will list OSS only). # Backends prepended with - won't be available for use (eg. '-oss,' will allow # all available backends except OSS). An empty list means the default. #drivers = pulse,alsa,core,oss,solaris,sndio,mmdevapi,dsound,winmm,port,opensl,null,wave ## excludefx: # 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,echo, # flanger,modulator,dedicated #excludefx = ## 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 = 4 ## sends: # Sets the number of auxiliary sends per source. When not specified (default), # it allows the app to request how many it wants. The maximum value currently # possible is 4. #sends = ## layout: # Sets the virtual speaker layout. Values are specified in degrees, where 0 is # straight in front, negative goes left, and positive goes right. Unspecified # speakers will remain at their default positions (which are dependant on the # output format). Available speakers are back-left(bl), side-left(sl), front- # left(fl), front-center(fc), front-right(fr), side-right(sr), back-right(br), # and back-center(bc). #layout = ## layout_*: # Channel-specific layouts may be specified to override the layout option. The # same speakers as the layout option are available, and the default settings # are shown below. #layout_stereo = fl=-90, fr=90 #layout_quad = fl=-45, fr=45, bl=-135, br=135 #layout_surround51 = fl=-30, fr=30, fc=0, bl=-110, br=110 #layout_surround61 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bc=180 #layout_surround71 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bl=-150, br=150 ## default-reverb: # 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: # 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: # 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 ## ## Reverb effect stuff (includes EAX reverb) ## [reverb] ## boost: # 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: # 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 ## ## ALSA backend stuff ## [alsa] ## device: # Sets the device name for the default playback device. #device = default ## device-prefix: # 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-*: # 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: # Sets the device name for the default capture device. #capture = default ## capture-prefix: # 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-*: # 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 ## ## OSS backend stuff ## [oss] ## device: # Sets the device name for OSS output. #device = /dev/dsp ## capture: # Sets the device name for OSS capture. #capture = /dev/dsp ## ## Solaris backend stuff ## [solaris] ## device: # Sets the device name for Solaris output. #device = /dev/audio ## ## MMDevApi backend stuff ## [mmdevapi] ## ## DirectSound backend stuff ## [dsound] ## ## Windows Multimedia backend stuff ## [winmm] ## ## PortAudio backend stuff ## [port] ## device: # Sets the device index for output. Negative values will use the default as # given by PortAudio itself. #device = -1 ## capture: # Sets the device index for capture. Negative values will use the default as # given by PortAudio itself. #capture = -1 ## ## PulseAudio backend stuff ## [pulse] ## spawn-server: # Attempts to spawn a PulseAudio server when requesting to open a PulseAudio # device. Setting autospawn to false in PulseAudio's client.conf will still # prevent autospawning even if this is set to true. #spawn-server = true ## allow-moves: # 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 ## ## Wave File Writer stuff ## [wave] ## file: # 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 =