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488 lines
18 KiB
Plaintext
488 lines
18 KiB
Plaintext
6 years ago
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# 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 = 3
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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-encoding:
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# Specifies the encoding method for non-HRTF stereo output. 'panpot' (default)
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# uses standard amplitude panning (aka pair-wise, stereo pair, etc) between
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# -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ
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# output, which encodes some surround sound information into stereo output
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# that can be decoded with a surround sound receiver. If crossfeed filters are
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# used, UHJ is disabled.
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#stereo-encoding = panpot
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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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# 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 = 64
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## sends:
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# Limits the number of auxiliary sends allowed per source. Setting this higher
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# than the default has no effect.
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#sends = 16
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## output-limiter:
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# Applies a gain limiter on the final mixed output. This reduces the volume
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# when the output samples would otherwise clamp, avoiding excessive clipping
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# noise.
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#output-limiter = true
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## dither:
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# Applies dithering on the final mix, for 8- and 16-bit output by default.
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# This replaces the distortion created by nearest-value quantization with low-
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# level whitenoise.
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#dither = true
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## dither-depth:
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# Quantization bit-depth for dithered output. A value of 0 (or less) will
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# match the output sample depth. For int32, uint32, and float32 output, 0 will
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# disable dithering because they're at or beyond the rendered precision. The
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# maximum dither depth is 24.
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#dither-depth = 0
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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 proper distances to be
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# specified in the decoder configuration file.
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distance-comp = true
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## nfc:
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# Enables near-field control filters. This simulates and compensates for low-
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# frequency effects caused by the curvature of nearby sound-waves, which
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# creates a more realistic perception of sound distance. Note that the effect
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# may be stronger or weaker than intended if the application doesn't use or
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# specify an appropriate unit scale, or if incorrect speaker distances are set
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# in the decoder configuration file. Requires hq-mode to be enabled.
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nfc = true
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## nfc-ref-delay
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# Specifies the reference delay value for ambisonic output. When channels is
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# set to one of the ambi* formats, this option enables NFC-HOA output with the
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# specified Reference Delay parameter. The specified value can then be shared
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# with an appropriate NFC-HOA decoder to reproduce correct near-field effects.
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# Keep in mind that despite being designed for higher-order ambisonics, this
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# applies to first-order output all the same. When left unset, normal output
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# is created with no near-field simulation.
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nfc-ref-delay =
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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
|
||
|
# 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
|