diff options
Diffstat (limited to 'alsoftrc.sample')
-rw-r--r-- | alsoftrc.sample | 175 |
1 files changed, 143 insertions, 32 deletions
diff --git a/alsoftrc.sample b/alsoftrc.sample index eccc88b5..8061ed1c 100644 --- a/alsoftrc.sample +++ b/alsoftrc.sample @@ -48,7 +48,10 @@ ## channels: # Sets the output channel configuration. If left unspecified, one will try to # be detected from the system, and defaulting to stereo. The available values -# are: mono, stereo, quad, surround51, surround51rear, surround61, surround71 +# are: mono, stereo, quad, surround51, surround51rear, surround61, surround71, +# ambi1, ambi2, ambi3. Note that the ambi* configurations provide ambisonic +# channels of the given order (using ACN ordering and SN3D normalization by +# default), which need to be decoded to play correctly on speakers. #channels = ## sample-type: @@ -78,7 +81,7 @@ # which helps protect against skips when the CPU is under load, but increases # the delay between a sound getting mixed and being heard. Acceptable values # range between 2 and 16. -#periods = 4 +#periods = 3 ## stereo-mode: # Specifies if stereo output is treated as being headphones or speakers. With @@ -86,6 +89,20 @@ # Valid settings are auto, speakers, and headphones. #stereo-mode = auto +## stereo-encoding: +# Specifies the encoding method for non-HRTF stereo output. 'panpot' (default) +# uses standard amplitude panning (aka pair-wise, stereo pair, etc) between +# -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ +# output, which encodes some surround sound information into stereo output +# that can be decoded with a surround sound receiver. If crossfeed filters are +# used, UHJ is disabled. +#stereo-encoding = panpot + +## ambi-format: +# Specifies the channel order and normalization for the "ambi*" set of channel +# configurations. Valid settings are: fuma, acn+sn3d, acn+n3d +#ambi-format = acn+sn3d + ## hrtf: # Controls HRTF processing. These filters provide better spatialization of # sounds while using headphones, but do require a bit more CPU power. The @@ -96,21 +113,24 @@ # respectively. #hrtf = auto -## hrtf_tables: -# Specifies a comma-separated list of files containing HRTF data sets. The -# format of the files are described in hrtf.txt. The filenames may contain -# these markers, which will be replaced as needed: -# %r - Device sampling rate -# %% - Percent sign (%) -# The listed files are relative to system-dependant data directories. On -# Windows this is: +## default-hrtf: +# Specifies the default HRTF to use. When multiple HRTFs are available, this +# determines the preferred one to use if none are specifically requested. Note +# that this is the enumerated HRTF name, not necessarily the filename. +#default-hrtf = + +## hrtf-paths: +# Specifies a comma-separated list of paths containing HRTF data sets. The +# format of the files are described in docs/hrtf.txt. The files within the +# directories must have the .mhr file extension to be recognized. By default, +# OS-dependent data paths will be used. They will also be used if the list +# ends with a comma. On Windows this is: # $AppData\openal\hrtf # And on other systems, it's (in order): # $XDG_DATA_HOME/openal/hrtf (defaults to $HOME/.local/share/openal/hrtf) # $XDG_DATA_DIRS/openal/hrtf (defaults to /usr/local/share/openal/hrtf and # /usr/share/openal/hrtf) -# An absolute path may also be specified, if the given file is elsewhere. -#hrtf_tables = default-%r.mhr +#hrtf-paths = ## cf_level: # Sets the crossfeed level for stereo output. Valid values are: @@ -129,11 +149,11 @@ # Selects the resampler used when mixing sources. Valid values are: # point - nearest sample, no interpolation # linear - extrapolates samples using a linear slope between samples -# sinc4 - extrapolates samples using a 4-point Sinc filter -# sinc8 - extrapolates samples using an 8-point Sinc filter -# bsinc - extrapolates samples using a band-limited Sinc filter (varying -# between 12 and 24 points, with anti-aliasing) -# Specifying other values will result in using the default (linear). +# cubic - extrapolates samples using a Catmull-Rom spline +# bsinc12 - extrapolates samples using a band-limited Sinc filter (varying +# between 12 and 24 points, with anti-aliasing) +# bsinc24 - extrapolates samples using a band-limited Sinc filter (varying +# between 24 and 48 points, with anti-aliasing) #resampler = linear ## rt-prio: (global) @@ -155,19 +175,53 @@ # can use a non-negligible amount of CPU time if an effect is set on it even # if no sources are feeding it, so this may help when apps use more than the # system can handle. -#slots = 4 +#slots = 64 ## sends: -# Sets the number of auxiliary sends per source. When not specified (default), -# it allows the app to request how many it wants. The maximum value currently -# possible is 4. -#sends = +# Limits the number of auxiliary sends allowed per source. Setting this higher +# than the default has no effect. +#sends = 16 + +## front-stablizer: +# Applies filters to "stablize" front sound imaging. A psychoacoustic method +# is used to generate a front-center channel signal from the front-left and +# front-right channels, improving the front response by reducing the combing +# artifacts and phase errors. Consequently, it will only work with channel +# configurations that include front-left, front-right, and front-center. +#front-stablizer = false + +## output-limiter: +# Applies a gain limiter on the final mixed output. This reduces the volume +# when the output samples would otherwise clamp, avoiding excessive clipping +# noise. +#output-limiter = true + +## dither: +# Applies dithering on the final mix, for 8- and 16-bit output by default. +# This replaces the distortion created by nearest-value quantization with low- +# level whitenoise. +#dither = true + +## dither-depth: +# Quantization bit-depth for dithered output. A value of 0 (or less) will +# match the output sample depth. For int32, uint32, and float32 output, 0 will +# disable dithering because they're at or beyond the rendered precision. The +# maximum dither depth is 24. +#dither-depth = 0 + +## volume-adjust: +# A global volume adjustment for source output, expressed in decibels. The +# value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will +# be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A +# value of 0 means no change. +#volume-adjust = 0 ## excludefx: (global) # Sets which effects to exclude, preventing apps from using them. This can # help for apps that try to use effects which are too CPU intensive for the -# system to handle. Available effects are: eaxreverb,reverb,chorus,compressor, -# distortion,echo,equalizer,flanger,modulator,dedicated +# system to handle. Available effects are: eaxreverb,reverb,autowah,chorus, +# compressor,distortion,echo,equalizer,flanger,modulator,dedicated,pshifter, +# fshifter #excludefx = ## default-reverb: (global) @@ -192,6 +246,69 @@ #trap-al-error = false ## +## Ambisonic decoder stuff +## +[decoder] + +## hq-mode: +# Enables a high-quality ambisonic decoder. This mode is capable of frequency- +# dependent processing, creating a better reproduction of 3D sound rendering +# over surround sound speakers. Enabling this also requires specifying decoder +# configuration files for the appropriate speaker configuration you intend to +# use (see the quad, surround51, etc options below). Currently, up to third- +# order decoding is supported. +hq-mode = false + +## distance-comp: +# Enables compensation for the speakers' relative distances to the listener. +# This applies the necessary delays and attenuation to make the speakers +# behave as though they are all equidistant, which is important for proper +# playback of 3D sound rendering. Requires the proper distances to be +# specified in the decoder configuration file. +distance-comp = true + +## nfc: +# Enables near-field control filters. This simulates and compensates for low- +# frequency effects caused by the curvature of nearby sound-waves, which +# creates a more realistic perception of sound distance. Note that the effect +# may be stronger or weaker than intended if the application doesn't use or +# specify an appropriate unit scale, or if incorrect speaker distances are set +# in the decoder configuration file. Requires hq-mode to be enabled. +nfc = true + +## nfc-ref-delay +# Specifies the reference delay value for ambisonic output. When channels is +# set to one of the ambi* formats, this option enables NFC-HOA output with the +# specified Reference Delay parameter. The specified value can then be shared +# with an appropriate NFC-HOA decoder to reproduce correct near-field effects. +# Keep in mind that despite being designed for higher-order ambisonics, this +# applies to first-order output all the same. When left unset, normal output +# is created with no near-field simulation. +nfc-ref-delay = + +## quad: +# Decoder configuration file for Quadraphonic channel output. See +# docs/ambdec.txt for a description of the file format. +quad = + +## surround51: +# Decoder configuration file for 5.1 Surround (Side and Rear) channel output. +# See docs/ambdec.txt for a description of the file format. +surround51 = + +## surround61: +# Decoder configuration file for 6.1 Surround channel output. See +# docs/ambdec.txt for a description of the file format. +surround61 = + +## surround71: +# Decoder configuration file for 7.1 Surround channel output. See +# docs/ambdec.txt for a description of the file format. Note: This can be used +# to enable 3D7.1 with the appropriate configuration and speaker placement, +# see docs/3D7.1.txt. +surround71 = + +## ## Reverb effect stuff (includes EAX reverb) ## [reverb] @@ -203,12 +320,6 @@ # value of 0 means no change. #boost = 0 -## emulate-eax: (global) -# Allows the standard reverb effect to be used in place of EAX reverb. EAX -# reverb processing is a bit more CPU intensive than standard, so this option -# allows a simpler effect to be used at the loss of some quality. -#emulate-eax = false - ## ## PulseAudio backend stuff ## @@ -334,9 +445,9 @@ #buffer-size = 0 ## -## MMDevApi backend stuff +## WASAPI backend stuff ## -[mmdevapi] +[wasapi] ## ## DirectSound backend stuff |