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alsoftrc.sample 18 KB

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  1. # OpenAL config file.
  2. #
  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 "$$".
  10. #
  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.
  16. #
  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
  20. #
  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:
  25. ##
  26. ## General stuff
  27. ##
  28. [general]
  29. ## disable-cpu-exts: (global)
  30. # Disables use of specialized methods that use specific CPU intrinsics.
  31. # Certain methods may utilize CPU extensions for improved performance, and
  32. # this option is useful for preventing some or all of those methods from being
  33. # used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.
  34. # Specifying 'all' disables use of all such specialized methods.
  35. #disable-cpu-exts =
  36. ## drivers: (global)
  37. # Sets the backend driver list order, comma-seperated. Unknown backends and
  38. # duplicated names are ignored. Unlisted backends won't be considered for use
  39. # unless the list is ended with a comma (e.g. 'oss,' will try OSS first before
  40. # other backends, while 'oss' will try OSS only). Backends prepended with -
  41. # won't be considered for use (e.g. '-oss,' will try all available backends
  42. # except OSS). An empty list means to try all backends.
  43. #drivers =
  44. ## channels:
  45. # Sets the output channel configuration. If left unspecified, one will try to
  46. # be detected from the system, and defaulting to stereo. The available values
  47. # are: mono, stereo, quad, surround51, surround51rear, surround61, surround71,
  48. # ambi1, ambi2, ambi3. Note that the ambi* configurations provide ambisonic
  49. # channels of the given order (using ACN ordering and SN3D normalization by
  50. # default), which need to be decoded to play correctly on speakers.
  51. #channels =
  52. ## sample-type:
  53. # Sets the output sample type. Currently, all mixing is done with 32-bit float
  54. # and converted to the output sample type as needed. Available values are:
  55. # int8 - signed 8-bit int
  56. # uint8 - unsigned 8-bit int
  57. # int16 - signed 16-bit int
  58. # uint16 - unsigned 16-bit int
  59. # int32 - signed 32-bit int
  60. # uint32 - unsigned 32-bit int
  61. # float32 - 32-bit float
  62. #sample-type = float32
  63. ## frequency:
  64. # Sets the output frequency. If left unspecified it will try to detect a
  65. # default from the system, otherwise it will default to 44100.
  66. #frequency =
  67. ## period_size:
  68. # Sets the update period size, in frames. This is the number of frames needed
  69. # for each mixing update. Acceptable values range between 64 and 8192.
  70. #period_size = 1024
  71. ## periods:
  72. # Sets the number of update periods. Higher values create a larger mix ahead,
  73. # which helps protect against skips when the CPU is under load, but increases
  74. # the delay between a sound getting mixed and being heard. Acceptable values
  75. # range between 2 and 16.
  76. #periods = 3
  77. ## stereo-mode:
  78. # Specifies if stereo output is treated as being headphones or speakers. With
  79. # headphones, HRTF or crossfeed filters may be used for better audio quality.
  80. # Valid settings are auto, speakers, and headphones.
  81. #stereo-mode = auto
  82. ## stereo-encoding:
  83. # Specifies the encoding method for non-HRTF stereo output. 'panpot' (default)
  84. # uses standard amplitude panning (aka pair-wise, stereo pair, etc) between
  85. # -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ
  86. # output, which encodes some surround sound information into stereo output
  87. # that can be decoded with a surround sound receiver. If crossfeed filters are
  88. # used, UHJ is disabled.
  89. #stereo-encoding = panpot
  90. ## ambi-format:
  91. # Specifies the channel order and normalization for the "ambi*" set of channel
  92. # configurations. Valid settings are: fuma, acn+sn3d, acn+n3d
  93. #ambi-format = acn+sn3d
  94. ## hrtf:
  95. # Controls HRTF processing. These filters provide better spatialization of
  96. # sounds while using headphones, but do require a bit more CPU power. The
  97. # default filters will only work with 44100hz or 48000hz stereo output. While
  98. # HRTF is used, the cf_level option is ignored. Setting this to auto (default)
  99. # will allow HRTF to be used when headphones are detected or the app requests
  100. # it, while setting true or false will forcefully enable or disable HRTF
  101. # respectively.
  102. #hrtf = auto
  103. ## default-hrtf:
  104. # Specifies the default HRTF to use. When multiple HRTFs are available, this
  105. # determines the preferred one to use if none are specifically requested. Note
  106. # that this is the enumerated HRTF name, not necessarily the filename.
  107. #default-hrtf =
  108. ## hrtf-paths:
  109. # Specifies a comma-separated list of paths containing HRTF data sets. The
  110. # format of the files are described in docs/hrtf.txt. The files within the
  111. # directories must have the .mhr file extension to be recognized. By default,
  112. # OS-dependent data paths will be used. They will also be used if the list
  113. # ends with a comma. On Windows this is:
  114. # $AppData\openal\hrtf
  115. # And on other systems, it's (in order):
  116. # $XDG_DATA_HOME/openal/hrtf (defaults to $HOME/.local/share/openal/hrtf)
  117. # $XDG_DATA_DIRS/openal/hrtf (defaults to /usr/local/share/openal/hrtf and
  118. # /usr/share/openal/hrtf)
  119. #hrtf-paths =
  120. ## cf_level:
  121. # Sets the crossfeed level for stereo output. Valid values are:
  122. # 0 - No crossfeed
  123. # 1 - Low crossfeed
  124. # 2 - Middle crossfeed
  125. # 3 - High crossfeed (virtual speakers are closer to itself)
  126. # 4 - Low easy crossfeed
  127. # 5 - Middle easy crossfeed
  128. # 6 - High easy crossfeed
  129. # Users of headphones may want to try various settings. Has no effect on non-
  130. # stereo modes.
  131. #cf_level = 0
  132. ## resampler: (global)
  133. # Selects the resampler used when mixing sources. Valid values are:
  134. # point - nearest sample, no interpolation
  135. # linear - extrapolates samples using a linear slope between samples
  136. # cubic - extrapolates samples using a Catmull-Rom spline
  137. # bsinc12 - extrapolates samples using a band-limited Sinc filter (varying
  138. # between 12 and 24 points, with anti-aliasing)
  139. # bsinc24 - extrapolates samples using a band-limited Sinc filter (varying
  140. # between 24 and 48 points, with anti-aliasing)
  141. #resampler = linear
  142. ## rt-prio: (global)
  143. # Sets real-time priority for the mixing thread. Not all drivers may use this
  144. # (eg. PortAudio) as they already control the priority of the mixing thread.
  145. # 0 and negative values will disable it. Note that this may constitute a
  146. # security risk since a real-time priority thread can indefinitely block
  147. # normal-priority threads if it fails to wait. As such, the default is
  148. # disabled.
  149. #rt-prio = 0
  150. ## sources:
  151. # Sets the maximum number of allocatable sources. Lower values may help for
  152. # systems with apps that try to play more sounds than the CPU can handle.
  153. #sources = 256
  154. ## slots:
  155. # Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
  156. # can use a non-negligible amount of CPU time if an effect is set on it even
  157. # if no sources are feeding it, so this may help when apps use more than the
  158. # system can handle.
  159. #slots = 64
  160. ## sends:
  161. # Limits the number of auxiliary sends allowed per source. Setting this higher
  162. # than the default has no effect.
  163. #sends = 16
  164. ## front-stablizer:
  165. # Applies filters to "stablize" front sound imaging. A psychoacoustic method
  166. # is used to generate a front-center channel signal from the front-left and
  167. # front-right channels, improving the front response by reducing the combing
  168. # artifacts and phase errors. Consequently, it will only work with channel
  169. # configurations that include front-left, front-right, and front-center.
  170. #front-stablizer = false
  171. ## output-limiter:
  172. # Applies a gain limiter on the final mixed output. This reduces the volume
  173. # when the output samples would otherwise clamp, avoiding excessive clipping
  174. # noise.
  175. #output-limiter = true
  176. ## dither:
  177. # Applies dithering on the final mix, for 8- and 16-bit output by default.
  178. # This replaces the distortion created by nearest-value quantization with low-
  179. # level whitenoise.
  180. #dither = true
  181. ## dither-depth:
  182. # Quantization bit-depth for dithered output. A value of 0 (or less) will
  183. # match the output sample depth. For int32, uint32, and float32 output, 0 will
  184. # disable dithering because they're at or beyond the rendered precision. The
  185. # maximum dither depth is 24.
  186. #dither-depth = 0
  187. ## volume-adjust:
  188. # A global volume adjustment for source output, expressed in decibels. The
  189. # value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will
  190. # be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A
  191. # value of 0 means no change.
  192. #volume-adjust = 0
  193. ## excludefx: (global)
  194. # Sets which effects to exclude, preventing apps from using them. This can
  195. # help for apps that try to use effects which are too CPU intensive for the
  196. # system to handle. Available effects are: eaxreverb,reverb,chorus,compressor,
  197. # distortion,echo,equalizer,flanger,modulator,dedicated,pshifter
  198. #excludefx =
  199. ## default-reverb: (global)
  200. # A reverb preset that applies by default to all sources on send 0
  201. # (applications that set their own slots on send 0 will override this).
  202. # Available presets are: None, Generic, PaddedCell, Room, Bathroom,
  203. # Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
  204. # CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
  205. # Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
  206. #default-reverb =
  207. ## trap-alc-error: (global)
  208. # Generates a SIGTRAP signal when an ALC device error is generated, on systems
  209. # that support it. This helps when debugging, while trying to find the cause
  210. # of a device error. On Windows, a breakpoint exception is generated.
  211. #trap-alc-error = false
  212. ## trap-al-error: (global)
  213. # Generates a SIGTRAP signal when an AL context error is generated, on systems
  214. # that support it. This helps when debugging, while trying to find the cause
  215. # of a context error. On Windows, a breakpoint exception is generated.
  216. #trap-al-error = false
  217. ##
  218. ## Ambisonic decoder stuff
  219. ##
  220. [decoder]
  221. ## hq-mode:
  222. # Enables a high-quality ambisonic decoder. This mode is capable of frequency-
  223. # dependent processing, creating a better reproduction of 3D sound rendering
  224. # over surround sound speakers. Enabling this also requires specifying decoder
  225. # configuration files for the appropriate speaker configuration you intend to
  226. # use (see the quad, surround51, etc options below). Currently, up to third-
  227. # order decoding is supported.
  228. hq-mode = false
  229. ## distance-comp:
  230. # Enables compensation for the speakers' relative distances to the listener.
  231. # This applies the necessary delays and attenuation to make the speakers
  232. # behave as though they are all equidistant, which is important for proper
  233. # playback of 3D sound rendering. Requires the proper distances to be
  234. # specified in the decoder configuration file.
  235. distance-comp = true
  236. ## nfc:
  237. # Enables near-field control filters. This simulates and compensates for low-
  238. # frequency effects caused by the curvature of nearby sound-waves, which
  239. # creates a more realistic perception of sound distance. Note that the effect
  240. # may be stronger or weaker than intended if the application doesn't use or
  241. # specify an appropriate unit scale, or if incorrect speaker distances are set
  242. # in the decoder configuration file. Requires hq-mode to be enabled.
  243. nfc = true
  244. ## nfc-ref-delay
  245. # Specifies the reference delay value for ambisonic output. When channels is
  246. # set to one of the ambi* formats, this option enables NFC-HOA output with the
  247. # specified Reference Delay parameter. The specified value can then be shared
  248. # with an appropriate NFC-HOA decoder to reproduce correct near-field effects.
  249. # Keep in mind that despite being designed for higher-order ambisonics, this
  250. # applies to first-order output all the same. When left unset, normal output
  251. # is created with no near-field simulation.
  252. nfc-ref-delay =
  253. ## quad:
  254. # Decoder configuration file for Quadraphonic channel output. See
  255. # docs/ambdec.txt for a description of the file format.
  256. quad =
  257. ## surround51:
  258. # Decoder configuration file for 5.1 Surround (Side and Rear) channel output.
  259. # See docs/ambdec.txt for a description of the file format.
  260. surround51 =
  261. ## surround61:
  262. # Decoder configuration file for 6.1 Surround channel output. See
  263. # docs/ambdec.txt for a description of the file format.
  264. surround61 =
  265. ## surround71:
  266. # Decoder configuration file for 7.1 Surround channel output. See
  267. # docs/ambdec.txt for a description of the file format. Note: This can be used
  268. # to enable 3D7.1 with the appropriate configuration and speaker placement,
  269. # see docs/3D7.1.txt.
  270. surround71 =
  271. ##
  272. ## Reverb effect stuff (includes EAX reverb)
  273. ##
  274. [reverb]
  275. ## boost: (global)
  276. # A global amplification for reverb output, expressed in decibels. The value
  277. # is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a
  278. # scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A
  279. # value of 0 means no change.
  280. #boost = 0
  281. ##
  282. ## PulseAudio backend stuff
  283. ##
  284. [pulse]
  285. ## spawn-server: (global)
  286. # Attempts to autospawn a PulseAudio server whenever needed (initializing the
  287. # backend, enumerating devices, etc). Setting autospawn to false in Pulse's
  288. # client.conf will still prevent autospawning even if this is set to true.
  289. #spawn-server = true
  290. ## allow-moves: (global)
  291. # Allows PulseAudio to move active streams to different devices. Note that the
  292. # device specifier (seen by applications) will not be updated when this
  293. # occurs, and neither will the AL device configuration (sample rate, format,
  294. # etc).
  295. #allow-moves = false
  296. ## fix-rate:
  297. # Specifies whether to match the playback stream's sample rate to the device's
  298. # sample rate. Enabling this forces OpenAL Soft to mix sources and effects
  299. # directly to the actual output rate, avoiding a second resample pass by the
  300. # PulseAudio server.
  301. #fix-rate = false
  302. ##
  303. ## ALSA backend stuff
  304. ##
  305. [alsa]
  306. ## device: (global)
  307. # Sets the device name for the default playback device.
  308. #device = default
  309. ## device-prefix: (global)
  310. # Sets the prefix used by the discovered (non-default) playback devices. This
  311. # will be appended with "CARD=c,DEV=d", where c is the card id and d is the
  312. # device index for the requested device name.
  313. #device-prefix = plughw:
  314. ## device-prefix-*: (global)
  315. # Card- and device-specific prefixes may be used to override the device-prefix
  316. # option. The option may specify the card id (eg, device-prefix-NVidia), or
  317. # the card id and device index (eg, device-prefix-NVidia-0). The card id is
  318. # case-sensitive.
  319. #device-prefix- =
  320. ## capture: (global)
  321. # Sets the device name for the default capture device.
  322. #capture = default
  323. ## capture-prefix: (global)
  324. # Sets the prefix used by the discovered (non-default) capture devices. This
  325. # will be appended with "CARD=c,DEV=d", where c is the card id and d is the
  326. # device number for the requested device name.
  327. #capture-prefix = plughw:
  328. ## capture-prefix-*: (global)
  329. # Card- and device-specific prefixes may be used to override the
  330. # capture-prefix option. The option may specify the card id (eg,
  331. # capture-prefix-NVidia), or the card id and device index (eg,
  332. # capture-prefix-NVidia-0). The card id is case-sensitive.
  333. #capture-prefix- =
  334. ## mmap:
  335. # Sets whether to try using mmap mode (helps reduce latencies and CPU
  336. # consumption). If mmap isn't available, it will automatically fall back to
  337. # non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
  338. # and anything else will force mmap off.
  339. #mmap = true
  340. ## allow-resampler:
  341. # Specifies whether to allow ALSA's built-in resampler. Enabling this will
  342. # allow the playback device to be set to a different sample rate than the
  343. # actual output, causing ALSA to apply its own resampling pass after OpenAL
  344. # Soft resamples and mixes the sources and effects for output.
  345. #allow-resampler = false
  346. ##
  347. ## OSS backend stuff
  348. ##
  349. [oss]
  350. ## device: (global)
  351. # Sets the device name for OSS output.
  352. #device = /dev/dsp
  353. ## capture: (global)
  354. # Sets the device name for OSS capture.
  355. #capture = /dev/dsp
  356. ##
  357. ## Solaris backend stuff
  358. ##
  359. [solaris]
  360. ## device: (global)
  361. # Sets the device name for Solaris output.
  362. #device = /dev/audio
  363. ##
  364. ## QSA backend stuff
  365. ##
  366. [qsa]
  367. ##
  368. ## JACK backend stuff
  369. ##
  370. [jack]
  371. ## spawn-server: (global)
  372. # Attempts to autospawn a JACK server whenever needed (initializing the
  373. # backend, opening devices, etc).
  374. #spawn-server = false
  375. ## buffer-size:
  376. # Sets the update buffer size, in samples, that the backend will keep buffered
  377. # to handle the server's real-time processing requests. This value must be a
  378. # power of 2, or else it will be rounded up to the next power of 2. If it is
  379. # less than JACK's buffer update size, it will be clamped. This option may
  380. # be useful in case the server's update size is too small and doesn't give the
  381. # mixer time to keep enough audio available for the processing requests.
  382. #buffer-size = 0
  383. ##
  384. ## WASAPI backend stuff
  385. ##
  386. [wasapi]
  387. ##
  388. ## DirectSound backend stuff
  389. ##
  390. [dsound]
  391. ##
  392. ## Windows Multimedia backend stuff
  393. ##
  394. [winmm]
  395. ##
  396. ## PortAudio backend stuff
  397. ##
  398. [port]
  399. ## device: (global)
  400. # Sets the device index for output. Negative values will use the default as
  401. # given by PortAudio itself.
  402. #device = -1
  403. ## capture: (global)
  404. # Sets the device index for capture. Negative values will use the default as
  405. # given by PortAudio itself.
  406. #capture = -1
  407. ##
  408. ## Wave File Writer stuff
  409. ##
  410. [wave]
  411. ## file: (global)
  412. # Sets the filename of the wave file to write to. An empty name prevents the
  413. # backend from opening, even when explicitly requested.
  414. # THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
  415. #file =
  416. ## bformat: (global)
  417. # Creates AMB format files using first-order ambisonics instead of a standard
  418. # single- or multi-channel .wav file.
  419. #bformat = false