b23f81b686
Both 5.1 Side and Rear configurations use 'surround51' to look up the appropriate decoder file. The decoder loader already handles mapping between rear and side channels, so there's no need for separate options.
452 lines
16 KiB
Plaintext
452 lines
16 KiB
Plaintext
# OpenAL config file.
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#
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# Option blocks may appear multiple times, and duplicated options will take the
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# last value specified. Environment variables may be specified within option
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# values, and are automatically substituted when the config file is loaded.
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# Environment variable names may only contain alpha-numeric characters (a-z,
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# A-Z, 0-9) and underscores (_), and are prefixed with $. For example,
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# specifying "$HOME/file.ext" would typically result in something like
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# "/home/user/file.ext". To specify an actual "$" character, use "$$".
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#
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# Device-specific values may be specified by including the device name in the
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# block name, with "general" replaced by the device name. That is, general
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# options for the device "Name of Device" would be in the [Name of Device]
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# block, while ALSA options would be in the [alsa/Name of Device] block.
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# Options marked as "(global)" are not influenced by the device.
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#
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# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
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# specific override settings in $HOME/.alsoftrc.
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# For Windows, these settings should go into $AppData\alsoft.ini
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#
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# Option and block names are case-senstive. The supplied values are only hints
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# and may not be honored (though generally it'll try to get as close as
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# possible). Note: options that are left unset may default to app- or system-
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# specified values. These are the current available settings:
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##
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## General stuff
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##
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[general]
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## disable-cpu-exts: (global)
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# Disables use of specialized methods that use specific CPU intrinsics.
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# Certain methods may utilize CPU extensions for improved performance, and
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# this option is useful for preventing some or all of those methods from being
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# used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.
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# Specifying 'all' disables use of all such specialized methods.
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#disable-cpu-exts =
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## drivers: (global)
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# Sets the backend driver list order, comma-seperated. Unknown backends and
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# duplicated names are ignored. Unlisted backends won't be considered for use
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# unless the list is ended with a comma (e.g. 'oss,' will try OSS first before
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# other backends, while 'oss' will try OSS only). Backends prepended with -
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# won't be considered for use (e.g. '-oss,' will try all available backends
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# except OSS). An empty list means to try all backends.
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#drivers =
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## channels:
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# Sets the output channel configuration. If left unspecified, one will try to
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# be detected from the system, and defaulting to stereo. The available values
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# are: mono, stereo, quad, surround51, surround51rear, surround61, surround71,
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# ambi1, ambi2, ambi3. Note that the ambi* configurations provide ambisonic
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# channels of the given order (using ACN ordering and SN3D normalization by
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# default), which need to be decoded to play correctly on speakers.
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#channels =
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## sample-type:
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# Sets the output sample type. Currently, all mixing is done with 32-bit float
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# and converted to the output sample type as needed. Available values are:
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# int8 - signed 8-bit int
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# uint8 - unsigned 8-bit int
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# int16 - signed 16-bit int
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# uint16 - unsigned 16-bit int
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# int32 - signed 32-bit int
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# uint32 - unsigned 32-bit int
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# float32 - 32-bit float
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#sample-type = float32
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## frequency:
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# Sets the output frequency. If left unspecified it will try to detect a
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# default from the system, otherwise it will default to 44100.
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#frequency =
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## period_size:
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# Sets the update period size, in frames. This is the number of frames needed
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# for each mixing update. Acceptable values range between 64 and 8192.
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#period_size = 1024
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## periods:
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# Sets the number of update periods. Higher values create a larger mix ahead,
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# which helps protect against skips when the CPU is under load, but increases
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# the delay between a sound getting mixed and being heard. Acceptable values
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# range between 2 and 16.
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#periods = 4
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## stereo-mode:
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# Specifies if stereo output is treated as being headphones or speakers. With
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# headphones, HRTF or crossfeed filters may be used for better audio quality.
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# Valid settings are auto, speakers, and headphones.
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#stereo-mode = auto
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## stereo-panning:
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# Specifies the panning method for non-HRTF stereo output. uhj (default)
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# creates stereo-compatible two-channel UHJ output, which encodes some
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# surround sound information, while paired uses standard pair-wise panning
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# between -30 and +30 degrees. If crossfeed filters are used, uhj panning is
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# disabled.
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#stereo-panning = uhj
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## ambi-format:
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# Specifies the channel order and normalization for the "ambi*" set of channel
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# configurations. Valid settings are: fuma, acn+sn3d, acn+n3d
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#ambi-format = acn+sn3d
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## hrtf:
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# Controls HRTF processing. These filters provide better spatialization of
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# sounds while using headphones, but do require a bit more CPU power. The
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# default filters will only work with 44100hz or 48000hz stereo output. While
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# HRTF is used, the cf_level option is ignored. Setting this to auto (default)
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# will allow HRTF to be used when headphones are detected or the app requests
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# it, while setting true or false will forcefully enable or disable HRTF
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# respectively.
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#hrtf = auto
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## default-hrtf:
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# Specifies the default HRTF to use. When multiple HRTFs are available, this
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# determines the preferred one to use if none are specifically requested. Note
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# that this is the enumerated HRTF name, not necessarily the filename.
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#default-hrtf =
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## hrtf-paths:
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# Specifies a comma-separated list of paths containing HRTF data sets. The
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# format of the files are described in docs/hrtf.txt. The files within the
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# directories must have the .mhr file extension to be recognized. By default,
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# OS-dependent data paths will be used. They will also be used if the list
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# ends with a comma. On Windows this is:
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# $AppData\openal\hrtf
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# And on other systems, it's (in order):
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# $XDG_DATA_HOME/openal/hrtf (defaults to $HOME/.local/share/openal/hrtf)
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# $XDG_DATA_DIRS/openal/hrtf (defaults to /usr/local/share/openal/hrtf and
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# /usr/share/openal/hrtf)
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#hrtf-paths =
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## cf_level:
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# Sets the crossfeed level for stereo output. Valid values are:
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# 0 - No crossfeed
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# 1 - Low crossfeed
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# 2 - Middle crossfeed
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# 3 - High crossfeed (virtual speakers are closer to itself)
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# 4 - Low easy crossfeed
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# 5 - Middle easy crossfeed
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# 6 - High easy crossfeed
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# Users of headphones may want to try various settings. Has no effect on non-
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# stereo modes.
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#cf_level = 0
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## resampler: (global)
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# Selects the resampler used when mixing sources. Valid values are:
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# point - nearest sample, no interpolation
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# linear - extrapolates samples using a linear slope between samples
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# sinc4 - extrapolates samples using a 4-point Sinc filter
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# sinc8 - extrapolates samples using an 8-point Sinc filter
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# bsinc - extrapolates samples using a band-limited Sinc filter (varying
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# between 12 and 24 points, with anti-aliasing)
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# Specifying other values will result in using the default (linear).
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#resampler = linear
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## rt-prio: (global)
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# Sets real-time priority for the mixing thread. Not all drivers may use this
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# (eg. PortAudio) as they already control the priority of the mixing thread.
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# 0 and negative values will disable it. Note that this may constitute a
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# security risk since a real-time priority thread can indefinitely block
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# normal-priority threads if it fails to wait. As such, the default is
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# disabled.
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#rt-prio = 0
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## sources:
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# Sets the maximum number of allocatable sources. Lower values may help for
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# systems with apps that try to play more sounds than the CPU can handle.
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#sources = 256
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## slots:
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# Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
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# can use a non-negligible amount of CPU time if an effect is set on it even
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# if no sources are feeding it, so this may help when apps use more than the
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# system can handle.
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#slots = 4
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## sends:
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# Sets the number of auxiliary sends per source. When not specified (default),
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# it allows the app to request how many it wants. The maximum value currently
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# possible is 4.
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#sends =
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## volume-adjust:
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# A global volume adjustment for source output, expressed in decibels. The
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# value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will
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# be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A
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# value of 0 means no change.
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#volume-adjust = 0
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## excludefx: (global)
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# Sets which effects to exclude, preventing apps from using them. This can
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# help for apps that try to use effects which are too CPU intensive for the
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# system to handle. Available effects are: eaxreverb,reverb,chorus,compressor,
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# distortion,echo,equalizer,flanger,modulator,dedicated
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#excludefx =
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## default-reverb: (global)
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# A reverb preset that applies by default to all sources on send 0
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# (applications that set their own slots on send 0 will override this).
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# Available presets are: None, Generic, PaddedCell, Room, Bathroom,
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# Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
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# CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
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# Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
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#default-reverb =
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## trap-alc-error: (global)
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# Generates a SIGTRAP signal when an ALC device error is generated, on systems
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# that support it. This helps when debugging, while trying to find the cause
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# of a device error. On Windows, a breakpoint exception is generated.
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#trap-alc-error = false
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## trap-al-error: (global)
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# Generates a SIGTRAP signal when an AL context error is generated, on systems
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# that support it. This helps when debugging, while trying to find the cause
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# of a context error. On Windows, a breakpoint exception is generated.
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#trap-al-error = false
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##
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## Ambisonic decoder stuff
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##
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[decoder]
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## hq-mode:
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# Enables a high-quality ambisonic decoder. This mode is capable of frequency-
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# dependent processing, creating a better reproduction of 3D sound rendering
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# over surround sound speakers. Enabling this also requires specifying decoder
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# configuration files for the appropriate speaker configuration you intend to
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# use (see the quad, surround51, etc options below). Currently, up to third-
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# order decoding is supported.
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hq-mode = false
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## distance-comp:
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# Enables compensation for the speakers' relative distances to the listener.
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# This applies the necessary delays and attenuation to make the speakers
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# behave as though they are all equidistant, which is important for proper
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# playback of 3D sound rendering. Requires the high-quality ambisonic decoder,
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# as well as the proper distances to be specified in the decoder configuration
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# file.
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distance-comp = true
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## quad:
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# Decoder configuration file for Quadrophonic channel output. See
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# docs/ambdec.txt for a description of the file format.
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quad =
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## surround51:
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# Decoder configuration file for 5.1 Surround (Side and Rear) channel output.
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# See docs/ambdec.txt for a description of the file format.
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surround51 =
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## surround61:
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# Decoder configuration file for 6.1 Surround channel output. See
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# docs/ambdec.txt for a description of the file format.
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surround61 =
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## surround71:
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# Decoder configuration file for 7.1 Surround channel output. See
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# docs/ambdec.txt for a description of the file format. Note: This can be used
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# to enable 3D7.1 with the appropriate configuration and speaker placement,
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# see docs/3D7.1.txt.
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surround71 =
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##
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## Reverb effect stuff (includes EAX reverb)
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##
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[reverb]
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## boost: (global)
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# A global amplification for reverb output, expressed in decibels. The value
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# is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a
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# scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A
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# value of 0 means no change.
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#boost = 0
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## emulate-eax: (global)
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# Allows the standard reverb effect to be used in place of EAX reverb. EAX
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# reverb processing is a bit more CPU intensive than standard, so this option
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# allows a simpler effect to be used at the loss of some quality.
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#emulate-eax = false
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##
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## PulseAudio backend stuff
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##
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[pulse]
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## spawn-server: (global)
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# Attempts to autospawn a PulseAudio server whenever needed (initializing the
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# backend, enumerating devices, etc). Setting autospawn to false in Pulse's
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# client.conf will still prevent autospawning even if this is set to true.
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#spawn-server = true
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## allow-moves: (global)
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# Allows PulseAudio to move active streams to different devices. Note that the
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# device specifier (seen by applications) will not be updated when this
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# occurs, and neither will the AL device configuration (sample rate, format,
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# etc).
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#allow-moves = false
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## fix-rate:
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# Specifies whether to match the playback stream's sample rate to the device's
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# sample rate. Enabling this forces OpenAL Soft to mix sources and effects
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# directly to the actual output rate, avoiding a second resample pass by the
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# PulseAudio server.
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#fix-rate = false
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##
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## ALSA backend stuff
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##
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[alsa]
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## device: (global)
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# Sets the device name for the default playback device.
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#device = default
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## device-prefix: (global)
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# Sets the prefix used by the discovered (non-default) playback devices. This
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# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
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# device index for the requested device name.
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#device-prefix = plughw:
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## device-prefix-*: (global)
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# Card- and device-specific prefixes may be used to override the device-prefix
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# option. The option may specify the card id (eg, device-prefix-NVidia), or
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# the card id and device index (eg, device-prefix-NVidia-0). The card id is
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# case-sensitive.
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#device-prefix- =
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## capture: (global)
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# Sets the device name for the default capture device.
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#capture = default
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## capture-prefix: (global)
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# Sets the prefix used by the discovered (non-default) capture devices. This
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# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
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# device number for the requested device name.
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#capture-prefix = plughw:
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## capture-prefix-*: (global)
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# Card- and device-specific prefixes may be used to override the
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# capture-prefix option. The option may specify the card id (eg,
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# capture-prefix-NVidia), or the card id and device index (eg,
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# capture-prefix-NVidia-0). The card id is case-sensitive.
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#capture-prefix- =
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## mmap:
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# Sets whether to try using mmap mode (helps reduce latencies and CPU
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# consumption). If mmap isn't available, it will automatically fall back to
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# non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
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# and anything else will force mmap off.
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#mmap = true
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## allow-resampler:
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# Specifies whether to allow ALSA's built-in resampler. Enabling this will
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# allow the playback device to be set to a different sample rate than the
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# actual output, causing ALSA to apply its own resampling pass after OpenAL
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# Soft resamples and mixes the sources and effects for output.
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#allow-resampler = false
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##
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## OSS backend stuff
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##
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[oss]
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## device: (global)
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# Sets the device name for OSS output.
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#device = /dev/dsp
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## capture: (global)
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# Sets the device name for OSS capture.
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#capture = /dev/dsp
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##
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## Solaris backend stuff
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##
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[solaris]
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## device: (global)
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# Sets the device name for Solaris output.
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#device = /dev/audio
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##
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## QSA backend stuff
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##
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[qsa]
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##
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## JACK backend stuff
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##
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[jack]
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## spawn-server: (global)
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# Attempts to autospawn a JACK server whenever needed (initializing the
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# backend, opening devices, etc).
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#spawn-server = false
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## buffer-size:
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# Sets the update buffer size, in samples, that the backend will keep buffered
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# to handle the server's real-time processing requests. This value must be a
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# power of 2, or else it will be rounded up to the next power of 2. If it is
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# less than JACK's buffer update size, it will be clamped. This option may
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# be useful in case the server's update size is too small and doesn't give the
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# mixer time to keep enough audio available for the processing requests.
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#buffer-size = 0
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##
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## MMDevApi backend stuff
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##
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[mmdevapi]
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##
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## DirectSound backend stuff
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##
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[dsound]
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##
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## Windows Multimedia backend stuff
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##
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[winmm]
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##
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## PortAudio backend stuff
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##
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[port]
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## device: (global)
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# Sets the device index for output. Negative values will use the default as
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# given by PortAudio itself.
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#device = -1
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## capture: (global)
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# Sets the device index for capture. Negative values will use the default as
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# given by PortAudio itself.
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#capture = -1
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##
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## Wave File Writer stuff
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##
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[wave]
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## file: (global)
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# Sets the filename of the wave file to write to. An empty name prevents the
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# backend from opening, even when explicitly requested.
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# THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
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#file =
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## bformat: (global)
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# Creates AMB format files using first-order ambisonics instead of a standard
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# single- or multi-channel .wav file.
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#bformat = false
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