View source for Sound

= 🎧 Sound Configuration Guide =

== 1️⃣ What Changed in 3374? ==
The 3374 patch uses '''OpenAL Soft''', a modern audio engine that replaces the older DirectSound system.

<big>Why this is good</big>
* Better 3D positional audio
* Proper surround sound (5.1 / 7.1)
* Headphone spatial audio (HRTF)
* More configurable and higher quality sound

If configured correctly, you’ll hear:

* Footsteps more precisely
* Better directional cues
* Clearer voice chat
* Cleaner ambient audio

== 2️⃣ How Sound Configuration Works ==
There are '''two layers''' of sound settings:
{| class="wikitable"
!File
!What It Controls
|-
|<code>UT2004.ini</code>
|In-game sound behavior
|-
|<code>alsoft.ini</code> (or <code>alsoft.conf</code>)
|How OpenAL outputs sound to your device
|}
You may only need to adjust one — but for best results, configure both.

<big>📂 Where Are the Sound Config Files?</big>

For '''Unreal Tournament 2004''' there are '''two main config files''' related to sound:

# <code>UT2004.ini</code> → game audio settings
# <code>alsoft.ini</code> (or <code>alsoft.conf</code>) → OpenAL Soft output settings

Their locations depend on your operating system.

<big>🔊 OpenAL Config</big>

<big>'''Windows location'''</big>
 C:\Users\<YourName>\AppData\Roaming\alsoft.ini
🎮 Game Config (recommended)
 C:\Games\UT2004\System\alsoft.ini
<big>'''Linux location'''</big>

Global
 /etc/xdg/alsoft.conf
User-specific
 ~/.config/alsoft.conf
🎮 Game Config (recommended)
 ~/.ut2004/System/alsoft.conf
If installed via Wine/Proton (Steam)
 ~/.steam/steam/steamapps/common/Unreal Tournament 2004/System/alsoft.ini
If it doesn’t exist, you can create it.

You can find available options and documentation [https://github.com/kcat/openal-soft/blob/master/alsoftrc.sample here]

== 3️⃣ Step-By-Step Setup ==

=== 🎮 Step A — Configure the Game (UT2004.ini) ===
Find <code>UT2004.ini</code> inside your UT2004 <code>System</code> folder.

Look for this section:
 [ALAudio.ALAudioSubsystem]
Here are the important settings:
<syntaxhighlight>[ALAudio.ALAudioSubsystem]
UseEAX=True
Use3DSound=True
UseDefaultDriver=True
CompatibilityMode=False
MaxEAXVersion=255
UsePrecache=True
ReverseStereo=False
Channels=32
MusicVolume=0.10000
AmbientVolume=0.500000
SoundVolume=0.30000
VoiceVolume=4.000000
VolumeScaleRec=0.100000
DopplerFactor=1.0
Rolloff=0.5
TimeBetweenHWUpdates=15
DisablePitch=False
LowQualitySound=False
UseVoIP=True
UseVAD=False
UseSpatializedVoice=False
SpatializedVoiceRadius=100000
EnhancedDenoiser=False
LocalZOffset=0.0</syntaxhighlight><big>'''What These Do'''</big>
* '''Use3DSound=True'''   Enables positional audio. Keep this ON.
* '''Channels=32'''   controls how many simultaneous sounds play — more = richer sound but higher CPU usage.
** 32 = good balance - default
** 64 = richer sound (uses more CPU)
** 128 = highest quality
* '''MusicVolume'''   Background music level
* '''SoundVolume'''   Weapons, effects, explosions
* '''AmbientVolume'''   Environmental sounds
* '''VoiceVolume'''   Player voice chat
* '''UseEAX=true''' - Enables EAX (Environmental Audio Extensions) - See [[Sound#🎧 What Is EAX?|🎧 What Is EAX?]]
* '''Rolloff=0.5'''   controls '''how quickly sound volume decreases with distance'''
** Higher rolloff → sound gets quiet '''faster'''
** Lower rolloff → sound stays audible '''longer'''
** Recommended range  → 0.3 - 0.5

=== 🔊 Step B — Configure OpenAL ===
This file goes in your game System folder (Windows/Linux). If it does not exist, you can create it.

It lets you choose:

* '''Stereo, headphones, or surround setups'''
* '''HRTF on/off (for 3D sound on headphones)'''
* '''Best resampler for quality vs performance''' 

== 4️⃣ Choose Your Audio Setup ==
The wiki has '''preset blocks''' you can copy/paste depending on your output device:

📌 ''Stereo (speakers)''

📌 ''Stereo with HRTF (headphones)''

📌 ''Surround 5.1 / 7.1''

 …and variants for both Windows and Linux<div class="mw-collapsible mw-collapsed">
====         <span><span>🔈 Stereo Speakers         <div class="mw-collapsible-toggle-placeholder"></div>     ====
    <div class="mw-collapsible-content">
        <p>Basic stereo setup</p>
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Windows</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
<syntaxhighlight copy="">
## CONFIG - Default
## General stuff
##
[General]

## 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=speakers

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=stereo

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=basic

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
#hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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


##
## 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.
#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=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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
#nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
#speaker-dist=1


##
## 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

##
## WASAPI backend stuff
##
[wasapi]

## spatial-api:
#  Specifies whether to use a Spatial Audio stream for playback. This may
#  provide expanded capabilities for surround sound and with-height speaker
#  configurations. Very experimental.
#spatial-api=false

## exclusive-mode:
#  Enables Exlusive mode for playback devices. This uses the device directly,
#  allowing lower latencies but prevents the device from being used multiple
#  times simultaneously. Ignores the periods setting when enabled, as WASAPI
#  automatically sets a buffer size based on the period size.
#exclusive-mode=true

## allow-resampler:
#  Specifies whether to allow an extra resampler pass on the output. Enabling
#  this will allow the playback device to be set to a different sample rate
#  than the actual output can accept, causing the backend to apply its own
#  resampling pass after OpenAL Soft mixes the sources and processes effects
#  for output.
allow-resampler=true

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

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

##
## EAX extensions stuff
##
[eax]

## enable: (global)
#  Sets whether to enable EAX extensions or not.
enable=true</syntaxhighlight>
            </div>
        </div>
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Linux</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
<syntaxhighlight copy="">
## CONFIG - Default
## General stuff
##
[General]

## 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=speakers

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=stereo

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=basic

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
#hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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


##
## 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.
#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=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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
#nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
#speaker-dist=1


##
## 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

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

## 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=true

## 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

##
## PipeWire backend stuff
##
[pipewire]

## assume-audio: (global)
#  Causes the backend to succeed initialization even if PipeWire reports no
#  audio support. Currently, audio support is detected by the presence of audio
#  source or sink nodes, although this can cause false negatives in cases where
#  device availability during library initialization is spotty. Future versions
#  of PipeWire are expected to have a more robust method to test audio support,
#  but in the mean time this can be set to true to assume PipeWire has audio
#  support even when no nodes may be reported at initialization time.
#assume-audio=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
#rt-mix=false

##
## 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=false

## 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=false

## adjust-latency:
#  Attempts to adjust the overall latency of device playback. Note that this
#  may have adverse effects on the resulting internal buffer sizes and mixing
#  updates, leading to performance problems and drop-outs. However, if the
#  PulseAudio server is creating a lot of latency, enabling this may help make
#  it more manageable.
#adjust-latency=false

##
## JACK backend stuff
##
[jack]

## spawn-server: (global)
#  Attempts to autospawn a JACK server when initializing.
#spawn-server=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
rt-mix=true

## connect-ports:
#  Attempts to automatically connect the client ports to physical server ports.
#  Client ports that fail to connect will leave the remaining channels
#  unconnected and silent (the device format won't change to accommodate).
connect-ports=true
</syntaxhighlight>
            </div>
        </div>
    </div>
</div>

----
<div class="mw-collapsible mw-collapsed">
====         <span>🎧 Headphones</span>          <div class="mw-collapsible-toggle-placeholder"></div>     ====
    <div class="mw-collapsible-content">
        <p>Basic stereo setup for headphones</p>
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Windows</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
                <syntaxhighlight copy>## CONFIG - Headphones - Basic Stereo
## General stuff
##
[General]

## 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=headphones

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=stereo

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=basic

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
#hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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


##
## 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.
#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=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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
#nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
#speaker-dist=1


##
## 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

##
## WASAPI backend stuff
##
[wasapi]

## spatial-api:
#  Specifies whether to use a Spatial Audio stream for playback. This may
#  provide expanded capabilities for surround sound and with-height speaker
#  configurations. Very experimental.
#spatial-api=false

## exclusive-mode:
#  Enables Exlusive mode for playback devices. This uses the device directly,
#  allowing lower latencies but prevents the device from being used multiple
#  times simultaneously. Ignores the periods setting when enabled, as WASAPI
#  automatically sets a buffer size based on the period size.
#exclusive-mode=true

## allow-resampler:
#  Specifies whether to allow an extra resampler pass on the output. Enabling
#  this will allow the playback device to be set to a different sample rate
#  than the actual output can accept, causing the backend to apply its own
#  resampling pass after OpenAL Soft mixes the sources and processes effects
#  for output.
allow-resampler=true

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

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

##
## EAX extensions stuff
##
[eax]

## enable: (global)
#  Sets whether to enable EAX extensions or not.
enable=true</syntaxhighlight>
            </div>
        </div>
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Linux</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
                <syntaxhighlight copy>## CONFIG - Headphones - Basic Stereo
## General stuff
##
[General]

## 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=headphones

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=stereo

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=basic

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
#hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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

##
## 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.
# 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=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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
#nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
# speaker-dist=1


##
## 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

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

## 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=true

## 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

##
## PipeWire backend stuff
##
[pipewire]

## assume-audio: (global)
#  Causes the backend to succeed initialization even if PipeWire reports no
#  audio support. Currently, audio support is detected by the presence of audio
#  source or sink nodes, although this can cause false negatives in cases where
#  device availability during library initialization is spotty. Future versions
#  of PipeWire are expected to have a more robust method to test audio support,
#  but in the mean time this can be set to true to assume PipeWire has audio
#  support even when no nodes may be reported at initialization time.
#assume-audio=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
#rt-mix=false

##
## 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=false

## 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=false

## adjust-latency:
#  Attempts to adjust the overall latency of device playback. Note that this
#  may have adverse effects on the resulting internal buffer sizes and mixing
#  updates, leading to performance problems and drop-outs. However, if the
#  PulseAudio server is creating a lot of latency, enabling this may help make
#  it more manageable.
#adjust-latency=false

##
## JACK backend stuff
##
[jack]

## spawn-server: (global)
#  Attempts to autospawn a JACK server when initializing.
#spawn-server=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
rt-mix=true

## connect-ports:
#  Attempts to automatically connect the client ports to physical server ports.
#  Client ports that fail to connect will leave the remaining channels
#  unconnected and silent (the device format won't change to accommodate).
connect-ports=true</syntaxhighlight>
            </div>
        </div>
    </div>
</div>

----

<div class="mw-collapsible mw-collapsed">

====         <span>🎧 Headphones + HRTF (Recommended for Competitive Play)</span>         <div class="mw-collapsible-toggle-placeholder"></div>     ====
    <div class="mw-collapsible-content">
For more information about HRTF, see the documentation page [https://github.com/kcat/openal-soft/blob/master/docs/hrtf.txt here]
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Windows</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
                <syntaxhighlight copy>## CONFIG - Headphones + HRTF
## General stuff
##
[General]

## 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=headphones

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=stereo

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=hrtf

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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


##
## 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.
# 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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
#speaker-dist=1


##
## 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

##
## WASAPI backend stuff
##
[wasapi]

## spatial-api:
#  Specifies whether to use a Spatial Audio stream for playback. This may
#  provide expanded capabilities for surround sound and with-height speaker
#  configurations. Very experimental.
#spatial-api=false

## exclusive-mode:
#  Enables Exlusive mode for playback devices. This uses the device directly,
#  allowing lower latencies but prevents the device from being used multiple
#  times simultaneously. Ignores the periods setting when enabled, as WASAPI
#  automatically sets a buffer size based on the period size.
#exclusive-mode=true

## allow-resampler:
#  Specifies whether to allow an extra resampler pass on the output. Enabling
#  this will allow the playback device to be set to a different sample rate
#  than the actual output can accept, causing the backend to apply its own
#  resampling pass after OpenAL Soft mixes the sources and processes effects
#  for output.
allow-resampler=true

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

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

##
## EAX extensions stuff
##
[eax]

## enable: (global)
#  Sets whether to enable EAX extensions or not.
enable=true</syntaxhighlight>
            </div>
        </div>
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Linux</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
                <syntaxhighlight copy>## CONFIG - Headphones + HRTF
## General stuff
##
[General]

## 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=headphones

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=stereo

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=hrtf

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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


##
## 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.
# 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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
#speaker-dist=1


##
## 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

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

## 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=true

## 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

##
## PipeWire backend stuff
##
[pipewire]

## assume-audio: (global)
#  Causes the backend to succeed initialization even if PipeWire reports no
#  audio support. Currently, audio support is detected by the presence of audio
#  source or sink nodes, although this can cause false negatives in cases where
#  device availability during library initialization is spotty. Future versions
#  of PipeWire are expected to have a more robust method to test audio support,
#  but in the mean time this can be set to true to assume PipeWire has audio
#  support even when no nodes may be reported at initialization time.
#assume-audio=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
#rt-mix=false

##
## 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=false

## 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=false

## adjust-latency:
#  Attempts to adjust the overall latency of device playback. Note that this
#  may have adverse effects on the resulting internal buffer sizes and mixing
#  updates, leading to performance problems and drop-outs. However, if the
#  PulseAudio server is creating a lot of latency, enabling this may help make
#  it more manageable.
#adjust-latency=false

##
## JACK backend stuff
##
[jack]

## spawn-server: (global)
#  Attempts to autospawn a JACK server when initializing.
#spawn-server=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
rt-mix=true

## connect-ports:
#  Attempts to automatically connect the client ports to physical server ports.
#  Client ports that fail to connect will leave the remaining channels
#  unconnected and silent (the device format won't change to accommodate).
connect-ports=true</syntaxhighlight>
            </div>
        </div>
    </div>
</div>
----

<div class="mw-collapsible mw-collapsed">
====         <span>🔊 5.1  Surround Sound</span>         <div class="mw-collapsible-toggle-placeholder"></div>     ====
    <div class="mw-collapsible-content">
        <p></p>
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Windows</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
                <syntaxhighlight copy>
## Surround 5.1
## General stuff
##
[General]

## 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=speakers

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=surround51

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=uhj

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
#hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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


##
## 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.
#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=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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
#nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
#speaker-dist=1


##
## 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

##
## WASAPI backend stuff
##
[wasapi]

## spatial-api:
#  Specifies whether to use a Spatial Audio stream for playback. This may
#  provide expanded capabilities for surround sound and with-height speaker
#  configurations. Very experimental.
spatial-api=false

## exclusive-mode:
#  Enables Exlusive mode for playback devices. This uses the device directly,
#  allowing lower latencies but prevents the device from being used multiple
#  times simultaneously. Ignores the periods setting when enabled, as WASAPI
#  automatically sets a buffer size based on the period size.
#exclusive-mode=true

## allow-resampler:
#  Specifies whether to allow an extra resampler pass on the output. Enabling
#  this will allow the playback device to be set to a different sample rate
#  than the actual output can accept, causing the backend to apply its own
#  resampling pass after OpenAL Soft mixes the sources and processes effects
#  for output.
allow-resampler=true

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

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

##
## EAX extensions stuff
##
[eax]

## enable: (global)
#  Sets whether to enable EAX extensions or not.
enable=true
</syntaxhighlight>
            </div>
        </div>
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Linux</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
                <syntaxhighlight copy>
## Surround 5.1
## General stuff
##
[General]

## 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=speakers

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=surround51

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=uhj

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
#hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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


##
## 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.
#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=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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
#nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
#speaker-dist=1


##
## 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

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

## 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=true

## 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

##
## PipeWire backend stuff
##
[pipewire]

## assume-audio: (global)
#  Causes the backend to succeed initialization even if PipeWire reports no
#  audio support. Currently, audio support is detected by the presence of audio
#  source or sink nodes, although this can cause false negatives in cases where
#  device availability during library initialization is spotty. Future versions
#  of PipeWire are expected to have a more robust method to test audio support,
#  but in the mean time this can be set to true to assume PipeWire has audio
#  support even when no nodes may be reported at initialization time.
#assume-audio=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
#rt-mix=false

##
## 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=false

## 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=false

## adjust-latency:
#  Attempts to adjust the overall latency of device playback. Note that this
#  may have adverse effects on the resulting internal buffer sizes and mixing
#  updates, leading to performance problems and drop-outs. However, if the
#  PulseAudio server is creating a lot of latency, enabling this may help make
#  it more manageable.
#adjust-latency=false

##
## JACK backend stuff
##
[jack]

## spawn-server: (global)
#  Attempts to autospawn a JACK server when initializing.
#spawn-server=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
rt-mix=true

## connect-ports:
#  Attempts to automatically connect the client ports to physical server ports.
#  Client ports that fail to connect will leave the remaining channels
#  unconnected and silent (the device format won't change to accommodate).
connect-ports=true
</syntaxhighlight>
            </div>
        </div>
    </div>
</div>

----

<div class="mw-collapsible mw-collapsed">
====         <span>🔊 7.1 Surround Sound</span>         <div class="mw-collapsible-toggle-placeholder"></div>     ====
    <div class="mw-collapsible-content">
        <p></p>
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Windows</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
<syntaxhighlight copy="">
## Surround 7.1
## General stuff
##
[General]

## 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=speakers

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=surround71

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=uhj

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
#hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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


##
## 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.
#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=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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
#nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
#speaker-dist=1


##
## 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

##
## WASAPI backend stuff
##
[wasapi]

## spatial-api:
#  Specifies whether to use a Spatial Audio stream for playback. This may
#  provide expanded capabilities for surround sound and with-height speaker
#  configurations. Very experimental.
spatial-api=false

## exclusive-mode:
#  Enables Exlusive mode for playback devices. This uses the device directly,
#  allowing lower latencies but prevents the device from being used multiple
#  times simultaneously. Ignores the periods setting when enabled, as WASAPI
#  automatically sets a buffer size based on the period size.
#exclusive-mode=true

## allow-resampler:
#  Specifies whether to allow an extra resampler pass on the output. Enabling
#  this will allow the playback device to be set to a different sample rate
#  than the actual output can accept, causing the backend to apply its own
#  resampling pass after OpenAL Soft mixes the sources and processes effects
#  for output.
allow-resampler=true

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

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

##
## EAX extensions stuff
##
[eax]

## enable: (global)
#  Sets whether to enable EAX extensions or not.
enable=true
</syntaxhighlight>
            </div>
        </div>
        <div class="mw-collapsible mw-collapsed">
            <h5>
                <span style="color: purple">For Linux</span>
                <div class="mw-collapsible-toggle-placeholder"></div>
            </h5>
            <div class="mw-collapsible-content">
<syntaxhighlight copy="">
## Surround 7.1
## General stuff
##
[General]

## 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=speakers

#  Sets the default output channel configuration. If left unspecified, one will
#  try to be detected from the system, with a fallback to stereo. The available
#  values are: mono, stereo, quad, surround51, surround61, surround71,
#  surround714, 3d71, ambi1, ambi2, ambi3, ambi4. Note that the ambi*
#  configurations output 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=surround71

## stereo-encoding:
#  Specifies the default encoding method for stereo output. Valid values are:
#  basic - Standard amplitude panning (aka pair-wise, stereo pair, etc) between
#          -30 and +30 degrees.
#  uhj - Creates a stereo-compatible two-channel UHJ mix, which encodes some
#        continuous surround sound information into stereo output that can be
#        decoded with a surround sound receiver.
#  tsme - Creates a stereo-compatible two-channel surround matrix encoded mix,
#         which encodes some discrete surround sound and height information
#         into the stereo output that can be decoded with a surround sound
#         receiver.
#  hrtf - Uses filters to provide better spatialization of sounds while using
#         stereo headphones.
#  If crossfeed filters are used, basic stereo mixing is used.
stereo-encoding=uhj

## hrtf-mode:
#  Specifies the rendering mode for HRTF processing. Setting the mode to full
#  (default) applies a unique HRIR filter to each source given its relative
#  location, providing the clearest directional response at the cost of the
#  highest CPU usage. Setting the mode to ambi1, ambi2, ambi3, or ambi4 will
#  instead mix to an ambisonic buffer of the given order, then decode that
#  buffer with HRTF filters. Ambi1 has the lowest CPU usage, replacing the per-
#  source HRIR filter for a simple 4-channel panning mix, but retains full 3D
#  placement at the cost of a more diffuse response. Higher ambisonic orders
#  increasingly improve the directional clarity, at the cost of more CPU usage
#  (still less than "full", given some number of active sources).
#hrtf-mode=full

## resampler: (global)
#  Selects the default resampler used when mixing sources. Valid values are:
#  point - nearest sample, no interpolation
#  linear - extrapolates samples using a linear slope between samples
#  spline - extrapolates samples using a Catmull-Rom spline
#  gaussian - extrapolates samples using a 4-point Gaussian filter
#  bsinc12 - extrapolates samples using a band-limited Sinc filter (varying between 12 and 24 points, with anti-aliasing)
#  fast_bsinc12 - same as bsinc12, except without interpolation between down-sampling scales
#  bsinc24 - extrapolates samples using a band-limited Sinc filter (varying between 24 and 48 points, with anti-aliasing)
#  fast_bsinc24 - same as bsinc24, except without interpolation between down-sampling scales
#  bsinc48 - extrapolates samples using a band-limited Sinc filter (48 points, with anti-aliasing)
#  fast_bsinc48 - same as bsinc48, except without interpolation between down-sampling scales
resampler=spline

## period_size:
#  Sets the update period size, in sample frames. This is the number of frames
#  needed for each mixing update. Acceptable values range between 64 and 8192.
#  If left unspecified it will default to 512 sample frames (~10.7ms).
period_size=512

## 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=3

## 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

## 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

## 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

## 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


##
## 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.
#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=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 with surround sound
#  output. 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. For HRTF output, hrtf-mode must be set
#  to one of the ambi* values for this to function.
#nfc=false

## speaker-dist:
#  Specifies the speaker distance in meters, used by the near-field control
#  filters with surround sound output. For ambisonic output modes, this value
#  is the basis for the NFC-HOA Reference Delay parameter (calculated as
#  delay_seconds=speaker_dist/343.3). This value is not used when a decoder
#  configuration is set for the output mode (since they specify the per-speaker
#  distances, overriding this setting), or when the NFC filters are off. Valid
#  values range from 0.1 to 10.
#speaker-dist=1


##
## 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

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

## 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=true

## 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

##
## PipeWire backend stuff
##
[pipewire]

## assume-audio: (global)
#  Causes the backend to succeed initialization even if PipeWire reports no
#  audio support. Currently, audio support is detected by the presence of audio
#  source or sink nodes, although this can cause false negatives in cases where
#  device availability during library initialization is spotty. Future versions
#  of PipeWire are expected to have a more robust method to test audio support,
#  but in the mean time this can be set to true to assume PipeWire has audio
#  support even when no nodes may be reported at initialization time.
#assume-audio=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
#rt-mix=false

##
## 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=false

## 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=false

## adjust-latency:
#  Attempts to adjust the overall latency of device playback. Note that this
#  may have adverse effects on the resulting internal buffer sizes and mixing
#  updates, leading to performance problems and drop-outs. However, if the
#  PulseAudio server is creating a lot of latency, enabling this may help make
#  it more manageable.
#adjust-latency=false

##
## JACK backend stuff
##
[jack]

## spawn-server: (global)
#  Attempts to autospawn a JACK server when initializing.
#spawn-server=false

## rt-mix:
#  Renders samples directly in the real-time processing callback. This allows
#  for lower latency and less overall CPU utilization, but can increase the
#  risk of underruns when increasing the amount of work the mixer needs to do.
rt-mix=true

## connect-ports:
#  Attempts to automatically connect the client ports to physical server ports.
#  Client ports that fail to connect will leave the remaining channels
#  unconnected and silent (the device format won't change to accommodate).
connect-ports=true
</syntaxhighlight>
            </div>
</div>
</div>
</div>

== 5️⃣ Recommended Settings by Player Type ==

=== 🏆 Competitive Players ===

* Headphones + HRTF
* Channels = 32 or 64
* Lower music volume
* Higher sound effects

=== 🎮 Casual Players ===

* Stereo or surround
* Balanced volume levels
* Keep defaults unless something sounds wrong

=== ✔️ Quick Setup (for headphones) ===

# Create/edit <code>alsoft.ini</code>:
#* Choose headphone preset from wiki.
# Open <code>UT2004.ini</code>
#* Set <code>Use3DSound=True</code>
#* Raise <code>Channels</code> to at least 32
# Restart the game and test.

=== ✔️ Quick Surround Setup ===

# Pick the '''5.1 or 7.1 preset''' from the wiki for <code>alsoft.ini</code>
# Test in game with maps that have lots of sound effects (like crowded DM or Freon).

== 🎧 What Is EAX? ==
'''EAX (Environmental Audio Extensions)''' was a Creative Labs technology that added:

* Reverb effects (echo in large rooms)
* Environmental acoustics (muffled sounds behind walls)
* More immersive indoor/outdoor audio

''Originally, it required specific Sound Blaster hardware.''


✅ Should You Enable It?

* Better environmental immersion
* More realistic reverb indoors
* No modern hardware downside
* Minimal performance cost

❌ When to Turn It Off

* You want the cleanest, driest competitive audio
* You dislike environmental reverb
* You're troubleshooting weird sound artifacts

Competitive players sometimes disable it to reduce “audio clutter.”