OpenAL for UT2004

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captainsnarf
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OpenAL for UT2004

Post by captainsnarf »

Download here

Windows: OpenALUT2004Fix.zip
Linux: OpenAL-UT2004-Linux64.zip

OpenAL works and sounds great in UT2004. The only problem was that voice chat did not work. Incoming voice was silent. Pushing to talk would make the game crash. Pitch changes did not work in the 64 bit version.

Those issues are all fixed in this custom version of OpenAL I hacked together. The Readme.txt file has instructions for how to install and use.

My configurations for reference

UT2004.ini

Code: Select all

[ALAudio.ALAudioSubsystem]
UseEAX=True
Use3DSound=True
UseDefaultDriver=False
CompatibilityMode=False
MaxEAXVersion=255
UsePrecache=True
ReverseStereo=False
Channels=128
MusicVolume=0.000000
AmbientVolume=0.500000
SoundVolume=1.000000
VoiceVolume=5.780000
VolumeScaleRec=0.100000
DopplerFactor=1.000000
Rolloff=0.500000
TimeBetweenHWUpdates=15.000000
DisablePitch=False
LowQualitySound=False
UseVoIP=True
UseVAD=False
UseSpatializedVoice=False
SpatializedVoiceRadius=100000.000000
EnhancedDenoiser=False
LocalZOffset=0.000000
alsoft.ini (in C:\users\{username}\AppData\Roaming folder)

Code: Select all

[General]
channels=stereo
frequency=48000
stereo-mode=headphones
;default-hrtf=irc_1037_48000
default-hrtf=ciair_48000
sample-type=float32
cf_level=0
periods=2
resampler=cubic
hrtf=true

[reverb]
boost=-12

I previously mentioned you should set your speaker setup in windows to surround71 if your headset supports it, but that is incorrect. It works better if headset is set to stereo mode.
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Anonymous.
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Joined: Sat Jun 12, 2021 10:54 pm

Re: OpenAL for UT2004

Post by Anonymous. »

I haven't yet fully played with it, but having working audio pitch + voice chat in win64 is amazing. Very nice work.
Last edited by Anonymous. on Tue Sep 12, 2023 12:03 pm, edited 2 times in total.
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pooty
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Re: OpenAL for UT2004

Post by pooty »

So what does this do for the 32-bit version? Do we get surround sound? I have Razer that supports 7.1
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captainsnarf
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Re: OpenAL for UT2004

Post by captainsnarf »

yep, you get surround sound.

It always sounded better, except voice chat and pitch changes didn't work which were kind of deal breakers. That's no longer the case.
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captainsnarf
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Re: OpenAL for UT2004

Post by captainsnarf »

All options for alsoft.ini
# OpenAL config file.
#
# Option blocks may appear multiple times, and duplicated options will take the
# last value specified. Environment variables may be specified within option
# values, and are automatically substituted when the config file is loaded.
# Environment variable names may only contain alpha-numeric characters (a-z,
# A-Z, 0-9) and underscores (_), and are prefixed with $. For example,
# specifying "$HOME/file.ext" would typically result in something like
# "/home/user/file.ext". To specify an actual "$" character, use "$$".
#
# Device-specific values may be specified by including the device name in the
# block name, with "general" replaced by the device name. That is, general
# options for the device "Name of Device" would be in the [Name of Device]
# block, while ALSA options would be in the [alsa/Name of Device] block.
# Options marked as "(global)" are not influenced by the device.
#
# The system-wide settings can be put in /etc/xdg/alsoft.conf (as determined by
# the XDG_CONFIG_DIRS env var list, /etc/xdg being the default if unset) and
# user-specific override settings in $HOME/.config/alsoft.conf (as determined
# by the XDG_CONFIG_HOME env var).
#
# For Windows, these settings should go into $AppData\alsoft.ini
#
# An additional configuration file (alsoft.ini on Windows, alsoft.conf on other
# OSs) can be placed alongside the process executable for app-specific config
# settings.
#
# Option and block names are case-senstive. The supplied values are only hints
# and may not be honored (though generally it'll try to get as close as
# possible). Note: options that are left unset may default to app- or system-
# specified values. These are the current available settings:

##
## General stuff
##
[general]

## disable-cpu-exts: (global)
# Disables use of specialized methods that use specific CPU intrinsics.
# Certain methods may utilize CPU extensions for improved performance, and
# this option is useful for preventing some or all of those methods from being
# used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.
# Specifying 'all' disables use of all such specialized methods.
#disable-cpu-exts =

## drivers: (global)
# Sets the backend driver list order, comma-seperated. Unknown backends and
# duplicated names are ignored. Unlisted backends won't be considered for use
# unless the list is ended with a comma (e.g. 'oss,' will try OSS first before
# other backends, while 'oss' will try OSS only). Backends prepended with -
# won't be considered for use (e.g. '-oss,' will try all available backends
# except OSS). An empty list means to try all backends.
#drivers =

## channels:
# Sets the 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,
# surround3d71, ambi1, ambi2, ambi3. 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 =

## sample-type:
# Sets the default output sample type. Currently, all mixing is done with
# 32-bit float and converted to the output sample type as needed. Available
# values are:
# int8 - signed 8-bit int
# uint8 - unsigned 8-bit int
# int16 - signed 16-bit int
# uint16 - unsigned 16-bit int
# int32 - signed 32-bit int
# uint32 - unsigned 32-bit int
# float32 - 32-bit float
#sample-type = float32

## frequency:
# Sets the default output frequency. If left unspecified it will try to detect
# a default from the system, otherwise it will fallback to 48000.
#frequency =

## 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 1/50th of the frequency (20ms, or 882
# for 44100, 960 for 48000, etc).
#period_size =

## 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

## stereo-mode:
# Specifies if stereo output is treated as being headphones or speakers. With
# headphones, HRTF or crossfeed filters may be used for better audio quality.
# Valid settings are auto, speakers, and headphones.
#stereo-mode = auto

## stereo-encoding:
# Specifies the 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
# surround sound information into 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

## ambi-format:
# Specifies the channel order and normalization for the "ambi*" set of channel
# configurations. Valid settings are: fuma, acn+fuma, ambix (or acn+sn3d), or
# acn+n3d
#ambi-format = ambix

## hrtf:
# Deprecated. Consider using stereo-encoding instead. Valid values are auto,
# off, and on.
#hrtf = auto

## 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, or ambi3 will instead
# mix to a first-, second-, or third-order ambisonic buffer respectively, 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. Ambi2 and
# ambi3 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

## hrtf-size:
# Specifies the impulse response size, in samples, for the HRTF filter. Larger
# values increase the filter quality, while smaller values reduce processing
# cost. A value of 0 (default) uses the full filter size in the dataset, and
# the default dataset has a filter size of 64 samples at 48khz.
#hrtf-size = 0

## default-hrtf:
# Specifies the default HRTF to use. When multiple HRTFs are available, this
# determines the preferred one to use if none are specifically requested. Note
# that this is the enumerated HRTF name, not necessarily the filename.
#default-hrtf =

## hrtf-paths:
# Specifies a comma-separated list of paths containing HRTF data sets. The
# format of the files are described in docs/hrtf.txt. The files within the
# directories must have the .mhr file extension to be recognized. By default,
# OS-dependent data paths will be used. They will also be used if the list
# ends with a comma. On Windows this is:
# $AppData\openal\hrtf
# And on other systems, it's (in order):
# $XDG_DATA_HOME/openal/hrtf (defaults to $HOME/.local/share/openal/hrtf)
# $XDG_DATA_DIRS/openal/hrtf (defaults to /usr/local/share/openal/hrtf and
# /usr/share/openal/hrtf)
#hrtf-paths =

## cf_level:
# Sets the crossfeed level for stereo output. Valid values are:
# 0 - No crossfeed
# 1 - Low crossfeed
# 2 - Middle crossfeed
# 3 - High crossfeed (virtual speakers are closer to itself)
# 4 - Low easy crossfeed
# 5 - Middle easy crossfeed
# 6 - High easy crossfeed
# Users of headphones may want to try various settings. Has no effect on non-
# stereo modes.
#cf_level = 0

## resampler: (global)
# Selects the default resampler used when mixing sources. Valid values are:
# point - nearest sample, no interpolation
# linear - extrapolates samples using a linear slope between samples
# cubic - extrapolates samples using a Catmull-Rom spline
# bsinc12 - extrapolates samples using a band-limited Sinc filter (varying
# between 12 and 24 points, with anti-aliasing)
# 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
#resampler = cubic

## rt-prio: (global)
# Sets the real-time priority value for the mixing thread. Not all drivers may
# use this (eg. PortAudio) as those APIs already control the priority of the
# mixing thread. 0 and negative values will disable real-time priority. Note
# that this may constitute a security risk since a real-time priority thread
# can indefinitely block normal-priority threads if it fails to wait. Disable
# this if it turns out to be a problem.
#rt-prio = 1

## rt-time-limit: (global)
# On non-Windows systems, allows reducing the process's RLIMIT_RTTIME resource
# as necessary for acquiring real-time priority from RTKit.
#rt-time-limit = true

## sources:
# Sets the maximum number of allocatable sources. Lower values may help for
# systems with apps that try to play more sounds than the CPU can handle.
#sources = 256

## slots:
# Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
# can use a non-negligible amount of CPU time if an effect is set on it even
# if no sources are feeding it, so this may help when apps use more than the
# system can handle.
#slots = 64

## sends:
# Limits the number of auxiliary sends allowed per source. Setting this higher
# than the default has no effect.
#sends = 6

## front-stablizer:
# Applies filters to "stablize" front sound imaging. A psychoacoustic method
# is used to generate a front-center channel signal from the front-left and
# front-right channels, improving the front response by reducing the combing
# artifacts and phase errors. Consequently, it will only work with channel
# configurations that include front-left, front-right, and front-center.
#front-stablizer = false

## output-limiter:
# Applies a gain limiter on the final mixed output. This reduces the volume
# when the output samples would otherwise clamp, avoiding excessive clipping
# noise.
#output-limiter = true

## dither:
# Applies dithering on the final mix, for 8- and 16-bit output by default.
# This replaces the distortion created by nearest-value quantization with low-
# level whitenoise.
#dither = true

## dither-depth:
# Quantization bit-depth for dithered output. A value of 0 (or less) will
# match the output sample depth. For int32, uint32, and float32 output, 0 will
# disable dithering because they're at or beyond the rendered precision. The
# maximum dither depth is 24.
#dither-depth = 0

## volume-adjust:
# A global volume adjustment for source output, expressed in decibels. The
# value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will
# be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A
# value of 0 means no change.
#volume-adjust = 0

## excludefx: (global)
# Sets which effects to exclude, preventing apps from using them. This can
# help for apps that try to use effects which are too CPU intensive for the
# system to handle. Available effects are: eaxreverb,reverb,autowah,chorus,
# compressor,distortion,echo,equalizer,flanger,modulator,dedicated,pshifter,
# fshifter,vmorpher.
#excludefx =

## default-reverb: (global)
# A reverb preset that applies by default to all sources on send 0
# (applications that set their own slots on send 0 will override this).
# Available presets include: None, Generic, PaddedCell, Room, Bathroom,
# Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
# CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Mountains,
# Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
#default-reverb =

## trap-alc-error: (global)
# Generates a SIGTRAP signal when an ALC device error is generated, on systems
# that support it. This helps when debugging, while trying to find the cause
# of a device error. On Windows, a breakpoint exception is generated.
#trap-alc-error = false

## trap-al-error: (global)
# Generates a SIGTRAP signal when an AL context error is generated, on systems
# that support it. This helps when debugging, while trying to find the cause
# of a context error. On Windows, a breakpoint exception is generated.
#trap-al-error = false

##
## Ambisonic decoder stuff
##
[decoder]

## hq-mode:
# Enables a high-quality ambisonic decoder. This mode is capable of frequency-
# dependent processing, creating a better reproduction of 3D sound rendering
# over surround sound speakers.
#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

## quad:
# Decoder configuration file for Quadraphonic channel output. See
# docs/ambdec.txt for a description of the file format.
#quad =

## surround51:
# Decoder configuration file for 5.1 Surround (Side and Rear) channel output.
# See docs/ambdec.txt for a description of the file format.
#surround51 =

## surround61:
# Decoder configuration file for 6.1 Surround channel output. See
# docs/ambdec.txt for a description of the file format.
#surround61 =

## surround71:
# Decoder configuration file for 7.1 Surround channel output. See
# docs/ambdec.txt for a description of the file format.
#surround71 =

## surround3d71:
# Decoder configuration file for 3D7.1 Surround channel output. See
# docs/ambdec.txt for a description of the file format. See also
# docs/3D7.1.txt for information about 3D7.1.
#surround3d71 =

##
## UHJ and Super Stereo stuff
##
[uhj]

## decode-filter: (global)
# Specifies the all-pass filter type for UHJ decoding and Super Stereo
# processing. Valid values are:
# iir - utilizes dual IIR filters, providing a wide pass-band with low CPU
# use, but causes additional phase shifts on the signal.
# fir256 - utilizes a 256-point FIR filter, providing more stable results but
# exhibiting attenuation in the lower and higher frequency bands.
# fir512 - utilizes a 512-point FIR filter, providing a wider pass-band than
# fir256, at the cost of more CPU use.
#decode-filter = iir

## encode-filter: (global)
# Specifies the all-pass filter type for UHJ output encoding. Valid values are
# the same as for decode-filter.
#encode-filter = iir

##
## 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

##
## 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 = true

##
## 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

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

## device: (global)
# Sets the device name for the default playback device.
#device = default

## device-prefix: (global)
# Sets the prefix used by the discovered (non-default) playback devices. This
# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
# device index for the requested device name.
#device-prefix = plughw:

## device-prefix-*: (global)
# Card- and device-specific prefixes may be used to override the device-prefix
# option. The option may specify the card id (eg, device-prefix-NVidia), or
# the card id and device index (eg, device-prefix-NVidia-0). The card id is
# case-sensitive.
#device-prefix- =

## custom-devices: (global)
# Specifies a list of enumerated playback devices and the ALSA devices they
# refer to. The list pattern is "Display Name=ALSA device;...". The display
# names will be returned for device enumeration, and the ALSA device is the
# device name to open for each enumerated device.
#custom-devices =

## capture: (global)
# Sets the device name for the default capture device.
#capture = default

## capture-prefix: (global)
# Sets the prefix used by the discovered (non-default) capture devices. This
# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
# device number for the requested device name.
#capture-prefix = plughw:

## capture-prefix-*: (global)
# Card- and device-specific prefixes may be used to override the
# capture-prefix option. The option may specify the card id (eg,
# capture-prefix-NVidia), or the card id and device index (eg,
# capture-prefix-NVidia-0). The card id is case-sensitive.
#capture-prefix- =

## custom-captures: (global)
# Specifies a list of enumerated capture devices and the ALSA devices they
# refer to. The list pattern is "Display Name=ALSA device;...". The display
# names will be returned for device enumeration, and the ALSA device is the
# device name to open for each enumerated device.
#custom-captures =

## mmap:
# Sets whether to try using mmap mode (helps reduce latencies and CPU
# consumption). If mmap isn't available, it will automatically fall back to
# non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
# and anything else will force mmap off.
#mmap = true

## allow-resampler:
# Specifies whether to allow ALSA's built-in resampler. Enabling this will
# allow the playback device to be set to a different sample rate than the
# actual output, causing ALSA to apply its own resampling pass after OpenAL
# Soft resamples and mixes the sources and effects for output.
#allow-resampler = false

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

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

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

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

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

##
## QSA backend stuff
##
[qsa]

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

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

## custom-devices: (global)
# Specifies a list of enumerated devices and the ports they connect to. The
# list pattern is "Display Name=ports regex;Display Name=ports regex;...". The
# display names will be returned for device enumeration, and the ports regex
# is the regular expression to identify the target ports on the server (as
# given by the jack_get_ports function) for each enumerated device.
#custom-devices =

## 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

## buffer-size:
# Sets the update buffer size, in samples, that the backend will keep buffered
# to handle the server's real-time processing requests. This value must be a
# power of 2, or else it will be rounded up to the next power of 2. If it is
# less than JACK's buffer update size, it will be clamped. This option may
# be useful in case the server's update size is too small and doesn't give the
# mixer time to keep enough audio available for the processing requests.
# Ignored when rt-mix is true.
#buffer-size = 0

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

## 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]

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

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

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

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

## file: (global)
# Sets the filename of the wave file to write to. An empty name prevents the
# backend from opening, even when explicitly requested.
# THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
#file =

## bformat: (global)
# Creates AMB format files using first-order ambisonics instead of a standard
# single- or multi-channel .wav file.
#bformat = false

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

## enable: (global)
# Sets whether to enable EAX extensions or not.
#enable = true

##
## Per-game compatibility options (these should only be set in per-game config
## files, *NOT* system- or user-level!)
##
[game_compat]

## default-error: (global)
# An error value returned by alGetError when there's no current context. The
# default value is AL_INVALID_OPERATION, which lets the caller know the
# operation could not be executed. Some applications may erroneously call
# alGetError without a current context and expect 0 (AL_NO_ERROR), however
# that may cause other applications to think earlier AL calls succeeded when
# they actually failed.
#default-error = 0xA004

## nfc-scale: (global)
# A meters-per-unit distance scale applied to NFC filters. If a game doesn't
# use real-world meters for in-game units, the filters may create a too-near
# or too-distant effect. For instance, if the game uses 1 foot per unit, a
# value of 0.3048 will correctly adjust the filters. Or if the game uses 1
# kilometer per unit, a value of 1000 will correctly adjust the filters.
#nfc-scale = 1

## enable-sub-data-ext: (global)
# Enables the AL_SOFT_buffer_sub_data extension, disabling the
# AL_EXT_SOURCE_RADIUS extension. These extensions are incompatible, so only
# one can be available. The latter extension is more commonly used, but this
# option can be enabled for older apps that want the former extension.
#enable-sub-data-ext = false

## reverse-x: (global)
# Reverses the local X (left-right) position of 3D sound sources.
#reverse-x = false

## reverse-y: (global)
# Reverses the local Y (up-down) position of 3D sound sources.
#reverse-y = false

## reverse-z: (global)
# Reverses the local Z (front-back) position of 3D sound sources.
#reverse-z = false
FuriousRabbit
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Re: OpenAL for UT2004

Post by FuriousRabbit »

Wow, I've been without hearing voice chat for about 5 years because of this. I always thought it was some problem with the interplay between the various audio systems in Linux. But every other game worked other than ut2004 ... and there's probably only me, Roger, and FragMaster playing on Linux so not much hope for a fix. Did you modify the source or just change the configuration? If the former, do you have a patch? I'd need to compile for Linux.

Should I risk hearing Enyo again? :lol:
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captainsnarf
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Re: OpenAL for UT2004

Post by captainsnarf »

If you can compile for linux, go for it :) I had to modify OpenAL, source is here: https://github.com/zenakuten/openal-sof ... -vchat-fix

It's a bug in UT2004 OpenAL implementation, that's why only UT2004 is broken. I worked around it.
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trans_lux
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Re: OpenAL for UT2004

Post by trans_lux »

Very cool! Thanks!
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captainsnarf
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Re: OpenAL for UT2004

Post by captainsnarf »

OpenAL-UT2004-Linux64.zip

I built it for linux. Put libopenal.so in your UT2004/System folder and you should be good to go.
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Anonymous.
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Re: OpenAL for UT2004

Post by Anonymous. »

captainsnarf wrote: Sat Sep 09, 2023 11:09 pm OpenAL-UT2004-Linux64.zip

I built it for linux. Put libopenal.so in your UT2004/System folder and you should be good to go.
Voice chat works now, very nice.
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