OpenAL

My OpenAL (/etc/openal/alsoft.conf) configuration:

# OpenAL config file.

#

# Option blocks may appear multiple times, and duplicated options will take the

# last value specified. Environment variables may be specified within option

# values, and are automatically substituted when the config file is loaded.

# Environment variable names may only contain alpha-numeric characters (a-z,

# A-Z, 0-9) and underscores (_), and are prefixed with $. For example,

# specifying "$HOME/file.ext" would typically result in something like

# "/home/user/file.ext". To specify an actual "$" character, use "$$".

#

# Device-specific values may be specified by including the device name in the

# block name, with "general" replaced by the device name. That is, general

# options for the device "Name of Device" would be in the [Name of Device]

# block, while ALSA options would be in the [alsa/Name of Device] block.

# Options marked as "(global)" are not influenced by the device.

#

# The system-wide settings can be put in /etc/openal/alsoft.conf and user-

# specific override settings in $HOME/.alsoftrc.

# For Windows, these settings should go into $AppData\alsoft.ini

#

# Option and block names are case-senstive. The supplied values are only hints

# and may not be honored (though generally it'll try to get as close as

# possible). Note: options that are left unset may default to app- or system-

# specified values. These are the current available settings:

##

## General stuff

##

[general]

## disable-cpu-exts: (global)

#  Disables use of specialized methods that use specific CPU intrinsics.

#  Certain methods may utilize CPU extensions for improved performance, and

#  this option is useful for preventing some or all of those methods from being

#  used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.

#  Specifying 'all' disables use of all such specialized methods.

#disable-cpu-exts =

## drivers: (global)

#  Sets the backend driver list order, comma-seperated. Unknown backends and

#  duplicated names are ignored. Unlisted backends won't be considered for use

#  unless the list is ended with a comma (e.g. 'oss,' will try OSS first before

#  other backends, while 'oss' will try OSS only). Backends prepended with -

#  won't be considered for use (e.g. '-oss,' will try all available backends

#  except OSS). An empty list means to try all backends.

#drivers =

## 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,

#  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 = ambi3

## sample-type:

#  Sets the output sample type. Currently, all mixing is done with 32-bit float

#  and converted to the output sample type as needed. Available values are:

#  int8    - signed 8-bit int

#  uint8   - unsigned 8-bit int

#  int16   - signed 16-bit int

#  uint16  - unsigned 16-bit int

#  int32   - signed 32-bit int

#  uint32  - unsigned 32-bit int

#  float32 - 32-bit float

sample-type = int32

## frequency:

#  Sets the output frequency. If left unspecified it will try to detect a

#  default from the system, otherwise it will default to 44100.

frequency = 192000

## period_size:

#  Sets the update period size, in frames. This is the number of frames needed

#  for each mixing update. Acceptable values range between 64 and 8192.

period_size = 64

## periods:

#  Sets the number of update periods. Higher values create a larger mix ahead,

#  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 = 2

## stereo-mode:

#  Specifies if stereo output is treated as being headphones or speakers. With

#  headphones, HRTF or crossfeed filters may be used for better audio quality.

#  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 = uhj

## 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+n3d

## hrtf:

#  Controls HRTF processing. These filters provide better spatialization of

#  sounds while using headphones, but do require a bit more CPU power. The

#  default filters will only work with 44100hz or 48000hz stereo output. While

#  HRTF is used, the cf_level option is ignored. Setting this to auto (default)

#  will allow HRTF to be used when headphones are detected or the app requests

#  it, while setting true or false will forcefully enable or disable HRTF

#  respectively.

hrtf = auto

## 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)

#hrtf-paths =

## cf_level:

#  Sets the crossfeed level for stereo output. Valid values are:

#  0 - No crossfeed

#  1 - Low crossfeed

#  2 - Middle crossfeed

#  3 - High crossfeed (virtual speakers are closer to itself)

#  4 - Low easy crossfeed

#  5 - Middle easy crossfeed

#  6 - High easy crossfeed

#  Users of headphones may want to try various settings. Has no effect on non-

#  stereo modes.

cf_level = 0

## resampler: (global)

#  Selects the resampler used when mixing sources. Valid values are:

#  point - nearest sample, no interpolation

#  linear - extrapolates samples using a linear slope between samples

#  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 = cubic

## rt-prio: (global)

#  Sets real-time priority for the mixing thread. Not all drivers may use this

#  (eg. PortAudio) as they already control the priority of the mixing thread.

#  0 and negative values will disable it. Note that this may constitute a

#  security risk since a real-time priority thread can indefinitely block

#  normal-priority threads if it fails to wait. As such, the default is

#  disabled.

rt-prio = 1

## sources:

#  Sets the maximum number of allocatable sources. Lower values may help for

#  systems with apps that try to play more sounds than the CPU can handle.

#sources = 256

## slots:

#  Sets the maximum number of Auxiliary Effect Slots an app can create. A slot

#  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 = 64

## 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 = false

## 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 = false

## 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,autowah,chorus,

#  compressor,distortion,echo,equalizer,flanger,modulator,dedicated,pshifter,

#  fshifter

#excludefx =

## default-reverb: (global)

#  A reverb preset that applies by default to all sources on send 0

#  (applications that set their own slots on send 0 will override this).

#  Available presets are: None, Generic, PaddedCell, Room, Bathroom,

#  Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,

#  CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,

#  Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.

#default-reverb =

## trap-alc-error: (global)

#  Generates a SIGTRAP signal when an ALC device error is generated, on systems

#  that support it. This helps when debugging, while trying to find the cause

#  of a device error. On Windows, a breakpoint exception is generated.

trap-alc-error = true

## trap-al-error: (global)

#  Generates a SIGTRAP signal when an AL context error is generated, on systems

#  that support it. This helps when debugging, while trying to find the cause

#  of a context error. On Windows, a breakpoint exception is generated.

trap-al-error = true

##

## 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 = true

## 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 = false

## 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 = false

## 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]

## boost: (global)

#  A global amplification for reverb 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 about half, and -12 about 1/4th. A

#  value of 0 means no change.

#boost = 0

##

## PulseAudio backend stuff

##

[pulse]

## spawn-server: (global)

#  Attempts to autospawn a PulseAudio server whenever needed (initializing the

#  backend, enumerating devices, etc). Setting autospawn to false in Pulse's

#  client.conf will still prevent autospawning even if this is set to true.

spawn-server = true

## allow-moves: (global)

#  Allows PulseAudio to move active streams to different devices. Note that the

#  device specifier (seen by applications) will not be updated when this

#  occurs, and neither will the AL device configuration (sample rate, format,

#  etc).

allow-moves = true

## fix-rate:

#  Specifies whether to match the playback stream's sample rate to the device's

#  sample rate. Enabling this forces OpenAL Soft to mix sources and effects

#  directly to the actual output rate, avoiding a second resample pass by the

#  PulseAudio server.

fix-rate = true

##

## ALSA backend stuff

##

[alsa]

## device: (global)

#  Sets the device name for the default playback device.

#device = default

## device-prefix: (global)

#  Sets the prefix used by the discovered (non-default) playback devices. This

#  will be appended with "CARD=c,DEV=d", where c is the card id and d is the

#  device index for the requested device name.

#device-prefix = plughw:

## device-prefix-*: (global)

#  Card- and device-specific prefixes may be used to override the device-prefix

#  option. The option may specify the card id (eg, device-prefix-NVidia), or

#  the card id and device index (eg, device-prefix-NVidia-0). The card id is

#  case-sensitive.

#device-prefix- =

## capture: (global)

#  Sets the device name for the default capture device.

#capture = default

## capture-prefix: (global)

#  Sets the prefix used by the discovered (non-default) capture devices. This

#  will be appended with "CARD=c,DEV=d", where c is the card id and d is the

#  device number for the requested device name.

#capture-prefix = plughw:

## capture-prefix-*: (global)

#  Card- and device-specific prefixes may be used to override the

#  capture-prefix option. The option may specify the card id (eg,

#  capture-prefix-NVidia), or the card id and device index (eg,

#  capture-prefix-NVidia-0). The card id is case-sensitive.

#capture-prefix- =

## mmap:

#  Sets whether to try using mmap mode (helps reduce latencies and CPU

#  consumption). If mmap isn't available, it will automatically fall back to

#  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 = true

## 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

##

## WASAPI backend stuff

##

[wasapi]

##

## 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 = ~/Desktop/Ambisonic.amb1

## bformat: (global)

#  Creates AMB format files using first-order ambisonics instead of a standard

#  single- or multi-channel .wav file.

bformat = true

[eax]

enable = true