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- # 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, sse2, 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,qsa,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,autowah,chorus,
- # compressor,distortion,echo,equalizer,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
- ##
- ## MIDI stuff (EXPERIMENTAL)
- ##
- [midi]
- ## soundfont:
- # A default soundfont (sf2 format). Used when an app requests the system
- # default.
- #soundfont =
- ## volume:
- # Additional attenuation applied to MIDI output, expressed in decibels. This
- # is used to help keep the mix from clipping, and so must be 0 or less. The
- # value is logarithmic, so -6 will be about half amplitude, and -12 about
- # 1/4th. The default is roughly -13.9794 (0.2, or 1/5th).
- #volume =
- ##
- ## 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
- ##
- ## PulseAudio backend stuff
- ##
- [pulse]
- ## spawn-server:
- # 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:
- # 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
- ##
- ## 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
- ##
- ## QSA backend stuff
- ##
- [qsa]
- ## device:
- # Sets the device name for the default playback device.
- #device = default
- ## capture:
- # Sets the device name for the default capture device.
- #capture = default
- ##
- ## 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
- ##
- ## 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 =
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