alsoftrc.sample

# OpenAL config file. Options that are not under a block or are under the
# [general] block are for general, non-backend-specific options. Blocks may
# appear multiple times, and duplicated options will take the last value
# specified.
# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
# specific override settings in ~/.alsoftrc.
# For Windows, these settings should go into %AppData%\alsoft.ini

# Option and block names are case-insenstive. 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:

## disable-cpu-exts:
#  Disables use of the listed CPU extensions. Certain methods may utilize CPU
#  extensions when detected, and this option is useful for preventing those
#  extensions from being used. The available extensions are: sse, neon.
#  Specifying 'all' disables use of all extensions.
#disable-cpu-exts =

## 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, surround61, surround71
#channels = stereo

## 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 = float32

## hrtf:
#  Enables HRTF filters. These filters provide for better sound spatialization
#  while using headphones. The default filter will only work when output is
#  44100hz stereo. While HRTF is active, the cf_level option is disabled.
#  Default is disabled since stereo speaker output quality may suffer.
#hrtf = false

## hrtf_tables
#  Specifies a comma-separated list of files containing HRTF data sets. The
#  listed data sets can be used in place of or in addiiton to the the built-in
#  set. 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 (%)
#  So if this is set to "kemar-%r-diffuse.mhr", it will try to open
#  "kemar-44100-diffuse.mhr" if the device is using 44100hz output, or
#  "kemar-48000-diffuse.mhr" if the device is using 48000hz output, etc.
#hrtf_tables =

## 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

## wide-stereo:
#  Specifies that stereo sources are given a width of about 120 degrees on each
#  channel, centering on -90 (left) and +90 (right), as opposed to being points
#  placed at -30 (left) and +30 (right). This can be useful for surround-sound
#  to give stereo sources a more encompassing sound. Note that the sound's
#  overall volume will be slightly reduced to account for the extra output.
#wide-stereo = false

## frequency:
#  Sets the output frequency.
#frequency = 44100

## resampler:
#  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
#  Specifying other values will result in using the default (linear).
#resampler = linear

## rt-prio:
#  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 = 0

## 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 = 1024

## 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 = 4

## 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

## drivers:
#  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 (eg. 'oss,' will list OSS first
#  followed by all other available backends, while 'oss' will list OSS only).
#  Backends prepended with - won't be available for use (eg. '-oss,' will allow
#  all available backends except OSS). An empty list means the default.
#drivers = pulse,alsa,core,oss,solaris,sndio,mmdevapi,dsound,winmm,port,opensl,null,wave

## excludefx:
#  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,echo,modulator,
#  dedicated
#excludefx =

## 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 = 4

## 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 =

## layout:
#  Sets the virtual speaker layout. Values are specified in degrees, where 0 is
#  straight in front, negative goes left, and positive goes right. Unspecified
#  speakers will remain at their default positions (which are dependant on the
#  output format). Available speakers are back-left(bl), side-left(sl), front-
#  left(fl), front-center(fc), front-right(fr), side-right(sr), back-right(br),
#  and back-center(bc).
#layout =

## layout_*:
#  Channel-specific layouts may be specified to override the layout option. The
#  same speakers as the layout option are available, and the default settings
#  are shown below.
#layout_stereo     = fl=-90, fr=90
#layout_quad       = fl=-45, fr=45, bl=-135, br=135
#layout_surround51 = fl=-30, fr=30, fc=0, bl=-110, br=110
#layout_surround61 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bc=180
#layout_surround71 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bl=-150, br=150

## default-reverb:
#  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:
#  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 = false

## trap-al-error:
#  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 = false

##
## Reverb effect stuff (includes EAX reverb)
##
[reverb]

## boost:
#  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

## emulate-eax:
#  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

##
## ALSA backend stuff
##
[alsa]

## device:
#  Sets the device name for the default playback device.
#device = default

## device-prefix:
#  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-*:
#  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:
#  Sets the device name for the default capture device.
#capture = default

## capture-prefix:
#  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-*:
#  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

##
## OSS backend stuff
##
[oss]

## device:
#  Sets the device name for OSS output.
#device = /dev/dsp

## capture:
#  Sets the device name for OSS capture.
#capture = /dev/dsp

##
## Solaris backend stuff
##
[solaris]

## device:
#  Sets the device name for Solaris output.
#device = /dev/audio

##
## MMDevApi backend stuff
##
[mmdevapi]

##
## DirectSound backend stuff
##
[dsound]

##
## Windows Multimedia backend stuff
##
[winmm]

##
## PortAudio backend stuff
##
[port]

## device:
#  Sets the device index for output. Negative values will use the default as
#  given by PortAudio itself.
#device = -1

## capture:
#  Sets the device index for capture. Negative values will use the default as
#  given by PortAudio itself.
#capture = -1

##
## PulseAudio backend stuff
##
[pulse]

## spawn-server:
#  Attempts to spawn a PulseAudio server when requesting to open a PulseAudio
#  device. Setting autospawn to false in PulseAudio's client.conf will still
#  prevent autospawning even if this is set to true.
#spawn-server = true

## allow-moves:
#  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 = false

##
## Wave File Writer stuff
##
[wave]

## file:
#  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 =