3 # Option blocks may appear multiple times, and duplicated options will take the
4 # last value specified. Environment variables may be specified within option
5 # values, and are automatically substituted when the config file is loaded.
6 # Environment variable names may only contain alpha-numeric characters (a-z,
7 # A-Z, 0-9) and underscores (_), and are prefixed with $. For example,
8 # specifying "$HOME/file.ext" would typically result in something like
9 # "/home/user/file.ext". To specify an actual "$" character, use "$$".
11 # Device-specific values may be specified by including the device name in the
12 # block name, with "general" replaced by the device name. That is, general
13 # options for the device "Name of Device" would be in the [Name of Device]
14 # block, while ALSA options would be in the [alsa/Name of Device] block.
15 # Options marked as "(global)" are not influenced by the device.
17 # The system-wide settings can be put in /etc/openal/alsoft.conf and user-
18 # specific override settings in $HOME/.alsoftrc.
19 # For Windows, these settings should go into $AppData\alsoft.ini
21 # Option and block names are case-senstive. The supplied values are only hints
22 # and may not be honored (though generally it'll try to get as close as
23 # possible). Note: options that are left unset may default to app- or system-
24 # specified values. These are the current available settings:
31 ## disable-cpu-exts: (global)
32 # Disables use of specialized methods that use specific CPU intrinsics.
33 # Certain methods may utilize CPU extensions for improved performance, and
34 # this option is useful for preventing some or all of those methods from being
35 # used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.
36 # Specifying 'all' disables use of all such specialized methods.
40 # Sets the backend driver list order, comma-seperated. Unknown backends and
41 # duplicated names are ignored. Unlisted backends won't be considered for use
42 # unless the list is ended with a comma (e.g. 'oss,' will try OSS first before
43 # other backends, while 'oss' will try OSS only). Backends prepended with -
44 # won't be considered for use (e.g. '-oss,' will try all available backends
45 # except OSS). An empty list means to try all backends.
49 # Sets the output channel configuration. If left unspecified, one will try to
50 # be detected from the system, and defaulting to stereo. The available values
51 # are: mono, stereo, quad, surround51, surround51rear, surround61, surround71,
52 # ambi1, ambi2, ambi3. Note that the ambi* configurations provide ambisonic
53 # channels of the given order (using ACN ordering and SN3D normalization by
54 # default), which need to be decoded to play correctly on speakers.
58 # Sets the output sample type. Currently, all mixing is done with 32-bit float
59 # and converted to the output sample type as needed. Available values are:
60 # int8 - signed 8-bit int
61 # uint8 - unsigned 8-bit int
62 # int16 - signed 16-bit int
63 # uint16 - unsigned 16-bit int
64 # int32 - signed 32-bit int
65 # uint32 - unsigned 32-bit int
66 # float32 - 32-bit float
67 #sample-type = float32
70 # Sets the output frequency. If left unspecified it will try to detect a
71 # default from the system, otherwise it will default to 44100.
75 # Sets the update period size, in frames. This is the number of frames needed
76 # for each mixing update. Acceptable values range between 64 and 8192.
80 # Sets the number of update periods. Higher values create a larger mix ahead,
81 # which helps protect against skips when the CPU is under load, but increases
82 # the delay between a sound getting mixed and being heard. Acceptable values
83 # range between 2 and 16.
87 # Specifies if stereo output is treated as being headphones or speakers. With
88 # headphones, HRTF or crossfeed filters may be used for better audio quality.
89 # Valid settings are auto, speakers, and headphones.
93 # Specifies the encoding method for non-HRTF stereo output. 'panpot' (default)
94 # uses standard amplitude panning (aka pair-wise, stereo pair, etc) between
95 # -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ
96 # output, which encodes some surround sound information into stereo output
97 # that can be decoded with a surround sound receiver. If crossfeed filters are
98 # used, UHJ is disabled.
99 #stereo-encoding = panpot
102 # Specifies the channel order and normalization for the "ambi*" set of channel
103 # configurations. Valid settings are: fuma, acn+sn3d, acn+n3d
104 #ambi-format = acn+sn3d
107 # Controls HRTF processing. These filters provide better spatialization of
108 # sounds while using headphones, but do require a bit more CPU power. The
109 # default filters will only work with 44100hz or 48000hz stereo output. While
110 # HRTF is used, the cf_level option is ignored. Setting this to auto (default)
111 # will allow HRTF to be used when headphones are detected or the app requests
112 # it, while setting true or false will forcefully enable or disable HRTF
117 # Specifies the default HRTF to use. When multiple HRTFs are available, this
118 # determines the preferred one to use if none are specifically requested. Note
119 # that this is the enumerated HRTF name, not necessarily the filename.
123 # Specifies a comma-separated list of paths containing HRTF data sets. The
124 # format of the files are described in docs/hrtf.txt. The files within the
125 # directories must have the .mhr file extension to be recognized. By default,
126 # OS-dependent data paths will be used. They will also be used if the list
127 # ends with a comma. On Windows this is:
128 # $AppData\openal\hrtf
129 # And on other systems, it's (in order):
130 # $XDG_DATA_HOME/openal/hrtf (defaults to $HOME/.local/share/openal/hrtf)
131 # $XDG_DATA_DIRS/openal/hrtf (defaults to /usr/local/share/openal/hrtf and
132 # /usr/share/openal/hrtf)
136 # Sets the crossfeed level for stereo output. Valid values are:
139 # 2 - Middle crossfeed
140 # 3 - High crossfeed (virtual speakers are closer to itself)
141 # 4 - Low easy crossfeed
142 # 5 - Middle easy crossfeed
143 # 6 - High easy crossfeed
144 # Users of headphones may want to try various settings. Has no effect on non-
148 ## resampler: (global)
149 # Selects the resampler used when mixing sources. Valid values are:
150 # point - nearest sample, no interpolation
151 # linear - extrapolates samples using a linear slope between samples
152 # sinc4 - extrapolates samples using a 4-point Sinc filter
153 # bsinc - extrapolates samples using a band-limited Sinc filter (varying
154 # between 12 and 24 points, with anti-aliasing)
155 # Specifying other values will result in using the default (linear).
159 # Sets real-time priority for the mixing thread. Not all drivers may use this
160 # (eg. PortAudio) as they already control the priority of the mixing thread.
161 # 0 and negative values will disable it. Note that this may constitute a
162 # security risk since a real-time priority thread can indefinitely block
163 # normal-priority threads if it fails to wait. As such, the default is
168 # Sets the maximum number of allocatable sources. Lower values may help for
169 # systems with apps that try to play more sounds than the CPU can handle.
173 # Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
174 # can use a non-negligible amount of CPU time if an effect is set on it even
175 # if no sources are feeding it, so this may help when apps use more than the
180 # Limits the number of auxiliary sends allowed per source. Setting this higher
181 # than the default has no effect.
185 # Applies filters to "stablize" front sound imaging. A psychoacoustic method
186 # is used to generate a front-center channel signal from the front-left and
187 # front-right channels, improving the front response by reducing the combing
188 # artifacts and phase errors. Consequently, it will only work with channel
189 # configurations that include front-left, front-right, and front-center.
190 #front-stablizer = false
193 # Applies a gain limiter on the final mixed output. This reduces the volume
194 # when the output samples would otherwise clamp, avoiding excessive clipping
196 #output-limiter = true
199 # Applies dithering on the final mix, for 8- and 16-bit output by default.
200 # This replaces the distortion created by nearest-value quantization with low-
205 # Quantization bit-depth for dithered output. A value of 0 (or less) will
206 # match the output sample depth. For int32, uint32, and float32 output, 0 will
207 # disable dithering because they're at or beyond the rendered precision. The
208 # maximum dither depth is 24.
212 # A global volume adjustment for source output, expressed in decibels. The
213 # value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will
214 # be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A
215 # value of 0 means no change.
218 ## excludefx: (global)
219 # Sets which effects to exclude, preventing apps from using them. This can
220 # help for apps that try to use effects which are too CPU intensive for the
221 # system to handle. Available effects are: eaxreverb,reverb,chorus,compressor,
222 # distortion,echo,equalizer,flanger,modulator,dedicated
225 ## default-reverb: (global)
226 # A reverb preset that applies by default to all sources on send 0
227 # (applications that set their own slots on send 0 will override this).
228 # Available presets are: None, Generic, PaddedCell, Room, Bathroom,
229 # Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
230 # CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
231 # Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
234 ## trap-alc-error: (global)
235 # Generates a SIGTRAP signal when an ALC device error is generated, on systems
236 # that support it. This helps when debugging, while trying to find the cause
237 # of a device error. On Windows, a breakpoint exception is generated.
238 #trap-alc-error = false
240 ## trap-al-error: (global)
241 # Generates a SIGTRAP signal when an AL context error is generated, on systems
242 # that support it. This helps when debugging, while trying to find the cause
243 # of a context error. On Windows, a breakpoint exception is generated.
244 #trap-al-error = false
247 ## Ambisonic decoder stuff
252 # Enables a high-quality ambisonic decoder. This mode is capable of frequency-
253 # dependent processing, creating a better reproduction of 3D sound rendering
254 # over surround sound speakers. Enabling this also requires specifying decoder
255 # configuration files for the appropriate speaker configuration you intend to
256 # use (see the quad, surround51, etc options below). Currently, up to third-
257 # order decoding is supported.
261 # Enables compensation for the speakers' relative distances to the listener.
262 # This applies the necessary delays and attenuation to make the speakers
263 # behave as though they are all equidistant, which is important for proper
264 # playback of 3D sound rendering. Requires the proper distances to be
265 # specified in the decoder configuration file.
269 # Enables near-field control filters. This simulates and compensates for low-
270 # frequency effects caused by the curvature of nearby sound-waves, which
271 # creates a more realistic perception of sound distance. Note that the effect
272 # may be stronger or weaker than intended if the application doesn't use or
273 # specify an appropriate unit scale, or if incorrect speaker distances are set
274 # in the decoder configuration file. Requires hq-mode to be enabled.
278 # Specifies the reference delay value for ambisonic output. When channels is
279 # set to one of the ambi* formats, this option enables NFC-HOA output with the
280 # specified Reference Delay parameter. The specified value can then be shared
281 # with an appropriate NFC-HOA decoder to reproduce correct near-field effects.
282 # Keep in mind that despite being designed for higher-order ambisonics, this
283 # applies to first-order output all the same. When left unset, normal output
284 # is created with no near-field simulation.
288 # Decoder configuration file for Quadrophonic channel output. See
289 # docs/ambdec.txt for a description of the file format.
293 # Decoder configuration file for 5.1 Surround (Side and Rear) channel output.
294 # See docs/ambdec.txt for a description of the file format.
298 # Decoder configuration file for 6.1 Surround channel output. See
299 # docs/ambdec.txt for a description of the file format.
303 # Decoder configuration file for 7.1 Surround channel output. See
304 # docs/ambdec.txt for a description of the file format. Note: This can be used
305 # to enable 3D7.1 with the appropriate configuration and speaker placement,
306 # see docs/3D7.1.txt.
310 ## Reverb effect stuff (includes EAX reverb)
315 # A global amplification for reverb output, expressed in decibels. The value
316 # is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a
317 # scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A
318 # value of 0 means no change.
321 ## emulate-eax: (global)
322 # Allows the standard reverb effect to be used in place of EAX reverb. EAX
323 # reverb processing is a bit more CPU intensive than standard, so this option
324 # allows a simpler effect to be used at the loss of some quality.
328 ## PulseAudio backend stuff
332 ## spawn-server: (global)
333 # Attempts to autospawn a PulseAudio server whenever needed (initializing the
334 # backend, enumerating devices, etc). Setting autospawn to false in Pulse's
335 # client.conf will still prevent autospawning even if this is set to true.
338 ## allow-moves: (global)
339 # Allows PulseAudio to move active streams to different devices. Note that the
340 # device specifier (seen by applications) will not be updated when this
341 # occurs, and neither will the AL device configuration (sample rate, format,
346 # Specifies whether to match the playback stream's sample rate to the device's
347 # sample rate. Enabling this forces OpenAL Soft to mix sources and effects
348 # directly to the actual output rate, avoiding a second resample pass by the
353 ## ALSA backend stuff
358 # Sets the device name for the default playback device.
361 ## device-prefix: (global)
362 # Sets the prefix used by the discovered (non-default) playback devices. This
363 # will be appended with "CARD=c,DEV=d", where c is the card id and d is the
364 # device index for the requested device name.
365 #device-prefix = plughw:
367 ## device-prefix-*: (global)
368 # Card- and device-specific prefixes may be used to override the device-prefix
369 # option. The option may specify the card id (eg, device-prefix-NVidia), or
370 # the card id and device index (eg, device-prefix-NVidia-0). The card id is
375 # Sets the device name for the default capture device.
378 ## capture-prefix: (global)
379 # Sets the prefix used by the discovered (non-default) capture devices. This
380 # will be appended with "CARD=c,DEV=d", where c is the card id and d is the
381 # device number for the requested device name.
382 #capture-prefix = plughw:
384 ## capture-prefix-*: (global)
385 # Card- and device-specific prefixes may be used to override the
386 # capture-prefix option. The option may specify the card id (eg,
387 # capture-prefix-NVidia), or the card id and device index (eg,
388 # capture-prefix-NVidia-0). The card id is case-sensitive.
392 # Sets whether to try using mmap mode (helps reduce latencies and CPU
393 # consumption). If mmap isn't available, it will automatically fall back to
394 # non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
395 # and anything else will force mmap off.
399 # Specifies whether to allow ALSA's built-in resampler. Enabling this will
400 # allow the playback device to be set to a different sample rate than the
401 # actual output, causing ALSA to apply its own resampling pass after OpenAL
402 # Soft resamples and mixes the sources and effects for output.
403 #allow-resampler = false
411 # Sets the device name for OSS output.
415 # Sets the device name for OSS capture.
419 ## Solaris backend stuff
424 # Sets the device name for Solaris output.
433 ## JACK backend stuff
437 ## spawn-server: (global)
438 # Attempts to autospawn a JACK server whenever needed (initializing the
439 # backend, opening devices, etc).
440 #spawn-server = false
443 # Sets the update buffer size, in samples, that the backend will keep buffered
444 # to handle the server's real-time processing requests. This value must be a
445 # power of 2, or else it will be rounded up to the next power of 2. If it is
446 # less than JACK's buffer update size, it will be clamped. This option may
447 # be useful in case the server's update size is too small and doesn't give the
448 # mixer time to keep enough audio available for the processing requests.
452 ## MMDevApi backend stuff
457 ## DirectSound backend stuff
462 ## Windows Multimedia backend stuff
467 ## PortAudio backend stuff
472 # Sets the device index for output. Negative values will use the default as
473 # given by PortAudio itself.
477 # Sets the device index for capture. Negative values will use the default as
478 # given by PortAudio itself.
482 ## Wave File Writer stuff
487 # Sets the filename of the wave file to write to. An empty name prevents the
488 # backend from opening, even when explicitly requested.
489 # THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
493 # Creates AMB format files using first-order ambisonics instead of a standard
494 # single- or multi-channel .wav file.