AES Europe 2026

Audio synchronization across heterogeneous media playback
devices is essential for delivering immersive sound
experiences in applications such as speaker group play;
multi-room audio playback. Existing synchronization
techniques predominantly rely on tightly coupled network
infrastructures; often embed a media sequence;
timestamp information to the media packet at the
transmitting source end, which restrict flexibility of
selecting the transmitting source; also compromises
robustness under dynamic network conditions. This paper
proposes a network; source independent audio
synchronization framework that eliminates dependency on
embedding media sequence; timestamps. The proposed
system employs an audio fingerprinting-based media
sequencing algorithm amongst the media playback devices
without relying on the type of transmitting source; the
network availability. A novel audio synchronization
algorithm is proposed which first determines a common
sequence start information given a dynamic media stream
from the transmitting source; then communicates the
fingerprint; timestamp amongst the media playback
devices without modifying the original audio packet
structure. Experimental results demonstrate that the
proposed approach achieves a high audio-audio
synchronization of less than 10ms across media playback
devices in a no network environment, thereby extending the
scope of immersive audio application irrespective of the
transmitting source.

We present a spectral-like reformulation of 2D ambisonics,
enabling an alternative representation of the sound field
in terms of amplitudes; phases. We hypothesise that it
simplifies the representation; creative manipulation of
2D ambisonics, beyond encoded directional point sources.

In 2D high-order ambisonics (HOA) of order N, a sound field
can be represented as a 2π-periodic angular function as a
combination of circular harmonics (Y_m) weighted by the
coefficients (a_m) with m ∈ [-N, N]. This representation
can be reformulated in terms of N+1 amplitudes; N
phases, similarly to a Fourier decomposition.

A simple example of this representation is the ambisonic
encoder at an angle theta. Phases are then multiples of a
phase phi = theta/2π, as frequencies are multiples of a
fundamental in harmonic sounds. Therefore, the
amplitude-phase approach can draw on the field of sound
synthesis, between harmonic; inharmonic modelling.
Operations on ambisonic vectors in amplitude-phase also
rely on Fourier representation, namely the spectral
convolution of two vectors (element-wise products of the
amplitudes, element-wise sums of the phases). Spectral
convolution has vast potential in ambisonics, allowing to
represent all the usual spatial operations (geometric;
transformative) in a simple manner.

To test this approach, we are currently developing an
ambisonic synthesiser based on Faust functions running in
Max environment. We are evaluating the scope of this
representation, both theoretical; compositional,;
then attempt to expand this approach to 3D ambisonics.

Immersive audio systems are increasingly deployed in
large-scale live music contexts, yet there is limited
research addressing how immersive concerts are perceived
; experienced by audiences. This paper presents a
practice-based; ethnographically informed study of the
immersive audio design; audience experience of the band
Heilung’s concert at Roskilde Festival, staged in the Arena
Tent where a large-scale multichannel loudspeaker system
including main, surround,; overhead arrays was used.
The study combines insights in technical system design;
pre-production methods with qualitative audience research
in order to explore how immersive sound alters perception,
embodiment,; social engagement in live concerts.
Pre-production involved scaled system simulations,
reference listening positions, timing strategies,;
power-matched test environments to translate an immersive
studio mix to a festival-scale venue. During; after the
concert, audience experience was investigated through
in-depth interviews, focus group discussions, participant
observation, binaural; ambisonic recordings,;
phenomenologically inspired interview techniques.
Findings indicate that immersive audio contributes to
heightened affective engagement, bodily involvement,; a
sense of envelopment that exceeds conventional stereo
concert experiences. Audience members described the
experience as multisensory, ritualistic,; spatially
ambiguous, often lacking technical vocabulary but
emphasizing embodied; emotional responses. Importantly,
immersion was not perceived as sound alone, but as emerging
from the interaction of sound, visuals, architecture,
social presence,; narrative framing.
The paper argues that understanding immersive concerts
requires the integration of anthropological insights with
audio engineering knowledge. While technical approaches
explain how immersive sound systems operate,
anthropological perspectives are essential for
understanding how such systems are experienced,
interpreted,; given meaning by audiences. The study
contributes to the limited body of research on the effects
of immersive concert formats by examining how audiences
perceive immersion; how they ascribe meaning to
immersive sound.

One of the many applications of acoustic metamaterials is
the ability to substantially improve acoustic insulation in
the low-frequency range compared to traditional materials.
The objective of this study was to investigate a
vibroacoustic metamaterial consisting of a soft solid plane
with embedded inclusions. The analysed structure consists
of a porous layer with periodic solid elements, which
allows for enhanced insulation properties. A numerical
model considering interactions between the acoustic domain
; a solid was developed using COMSOL Multiphysics. The
influence of selected material; geometric parameters,
such as the shape of the inclusions; their placement, on
the overall effectiveness of the structure was analysed.
Based on the simulation results, a variant of the structure
was selected; used to create a prototype of the
metamaterial. The acoustic insulation of the constructed
structure was then measured in the diffuse field. The next
step is to conduct an optimization using the PSO algorithm
in order to find geometry of the structure that can achieve
the most favourable results in the selected frequency
range. The optimized structure will then be validated by
creating an additional sample; conducting another
measurement.

Generating 4pi acoustical atmosphere of a target space is
important for creating an immersive sound content. A
SBA-based reverb is a useful tool for this purpose. We
developed VSVerb, a SBA reverb that generates 4pi
reverberation from the virtual sound sources detected from
three orthogonal x-y-z sound intensities measured at the
target space. A virtual sound source, also known as a
mirror source, is an acoustic concept in geometrical
acoustics. According to this theory, many virtual sound
sources are considered to be located outside the room;
provide reflection sounds inside the room. Since the
spatial information of virtual sound sources is a kind of
fingerprint of a room’s reverberant characteristics,
correctly sampled virtual sound sources enables us to
recreate room's reverberation precisely.
Several methods have been proposed for detecting virtual
sound sources of a room, i.e., dominant reflection sounds
in a room, by using the spatial room impulse responses
(SRIRs). However, these methods have the disadvantage of
failing to detect small virtual sound sources that provide
late reflections, because they detect sources by focusing
on the peak amplitude values in SRIRs. It is difficult to
distinguish if a small peak in the latter part of SRIR
indicates the reflection or noise component. Additionally,
in low-band analysis, side robes of the band pass filter
add many large peaks to the SRIRs,; they make it
difficult to detect true reflection peaks.
To overcome these disadvantages of the conventional
methods, we developed a method that detects virtual sound
sources without using the amplitude characteristics of
SRIRs. We call this method “Speed Detection.” This method
detects virtual sound sources based on the spatial moving
speed of the sound intensity. Instead of measuring SRIRs of
the sound pressure, we measure SRIRs of x-y-z instantaneous
sound intensities. Since we can assume that the reflection
sound comes from a “certain-sized” virtual sound source
over a “certain period,” the sound intensity provided by
the virtual sound source is considered to remain within a
small area; move slowly while the source emits the
reflection sound. We focused on this behavior of sound
intensities; developed the new detection method.
First, we identify the portions of the sound intensity that
move slowly; isolate them as the “Source intensity.”
Then, we calculate the positions, strengths,; phase
characteristics of the virtual sound sources from these
Source intensities of the x, y,; z directions. We
examined Speed Detection method by generating several types
of 4pi reverbs from the virtual sound sources detected
using this method,; verified that it works well in many
cases. However, we have also found that it does not always
work well. We have realized the necessity of improving the
threshold value for classifying sound intensity into the
source intensity or other.
We have used to classify sound intensities into source
intensities; others by referring a threshold value,
vt=40(1000t+10)^1.5 [m/s], where t indicates the arrival
time [s] of the sound intensity. This equation is based on
our practical experience, rather than scientific facts. It
works well in most cases, but some adjustments are required
in very rare cases. To apply the threshold value to various
acoustical conditions of the target spaces, we propose
switching the threshold value from our conventional
equation to an averaged value using a time-varying time
window. To examine the newly proposed threshold value, we
conducted experiments on detecting virtual sound sources of
a simple rectangular room. The results demonstrated the
validity of the new threshold value. We expect this new
threshold value to improve the sound quality of VSVerb;
V2MA as well.

Recent advances in large-scale multichannel loudspeaker
systems have enabled immersive concert formats that extend
spatial control beyond conventional stereo; small
multichannel configurations. High-density loudspeaker
arrays (HDLAs) allow sound to be distributed across complex
architectural spaces, challenging established distinctions
between composition, performance,; live sound practice.
In live contexts, however, the realization of spatial
attributes is often constrained by system complexity,
limited rehearsal time,; the lack of artist-facing
spatial control interfaces. As a result, spatial
realization; sound diffusion are frequently delegated to
sound engineers, who translate artistic material to the
acoustic; architectural conditions of the venue in real
time.

This paper examines three immersive concerts presented
during Sonic Days 2025 in Denmark, realized on both
large-scale; small-scale multichannel loudspeaker
systems. The concerts represent contrasting production
contexts, including a site-specific spatial composition
conceived explicitly for a high-density loudspeaker array
; performances by artists whose practices are typically
oriented toward stereo or small multichannel formats.
Across these cases, spatialization functioned variously as
compositional material, interpretive layer,; adaptive
live-mixing practice.

The paper analyzes how control over spatial attributes is
negotiated between artists; sound engineers in live
immersive concert settings,; how this negotiation
affects the interpretation of artistic intent; audience
experience. Particular attention is given to the role of
sound engineers as active mediators whose decisions shape
spatial form, listening perspective,; the relationship
between sound; architecture. The findings suggest that
immersive concert formats redistribute creative agency
across artists, technicians,; technological
infrastructures,; point toward the need for revised
conceptual frameworks for authorship, performance,;
listening in large-scale spatial audio environments.

The acoustic design of music rooms is well supported by
existing guidances which are covering recording spaces,
practice rooms, green rooms,; large-scale performance
environments. However, the direct application of these
standards to high school; college buildings is often
constrained by limitations in budget, space, client
requirements; construction timelines. As a result,
educational music spaces present various design challenges
that require specially considered solutions. This paper
examines key architectural; acoustic issues for music
teaching; performance spaces in high schools, including
wall performance between non-compatible spaces, limited
room volumes,; other acoustic challenges, i.e.
interconnecting doors; windows between the spaces. A
case study of a good design implemented at the large school
project is presented to demonstrate how strategic planning
; interdisciplinary coordination can result in
high-quality, functional,; acoustically successful
learning environments. It is highlighted that the
collaboration between the design team; acoustic
consultants was the key to resolve the major project
challenges to achieve the best possible performance results
across all spaces.

Accurate identification of audio coding artifacts is
instrumental in encoder design, audio post-processing,;
perceptual quality assessment. This paper addresses the
detection of artifacts arising from changes in the
effective bandwidth of coded audio signals caused by coarse
spectral quantization. Such bandwidth variations give rise
to two prominent artifact types: bandwidth limitation (BL)
; birdies, also referred to as spectral islands (SI).
Blind detection methods, requiring no reference signal, are
presented for both artifact types. Bandwidth limitation
is detected by analyzing variations in the zero-crossing
count across time-domain subband signals, enabling
estimation of both fixed; time-varying cutoff
frequencies. Spectral islands are identified through
analysis of the spectrogram by detecting clusters of
isolated components in the time–frequency domain,
characterized by their temporal; spectral extents. The
proposed methods are evaluated using audio material from
the ODAQ; USAC verification datasets. Results show that
the BL detection method achieves an average bandwidth
estimation error of approximately 160 Hz; demonstrates
robustness to noisy bandwidth-limited signals. In addition,
the detected birdie artifacts are perceptually validated
through listening tests, indicating an improvement in
perceived quality following detection; subsequent
suppression of the birdie artifacts.

Soundscapes; sonic atmospheres are often approached as
environmental conditions perceived; evaluated through
their acoustic properties; affective qualities. Recent
predictive; inferential accounts of perception, however,
suggest a different understanding: that perception operates
as an anticipatory process in which sensory input is
primarily used to minimise error in an ongoing predictive
model of the world, rather than to construct experience
from the bottom up. From this perspective, auditory
perception is an active, temporally extended process shaped
by expectation, memory, attention,; action.

This paper explores what such a predictive understanding
contributes to the study of everyday sonic atmospheres.
Drawing on predictive processing as a conceptual
framework—while acknowledging its contested status—the
paper situates auditory perception alongside other sensory
modalities as part of a broader inferential engagement with
environments. Classical auditory phenomena; longer-term
perceptual “illusions” motivate this reframing by
highlighting how expectations shape experience across
multiple timescales.
The main analytical focus is the case of transitioning from
one atmosphere to another. Atmospheres are approached here
as multimodal, quasi-objective phenomena that do not reside
in sound, space, or subjects alone, but emerge through
shared, situated engagement. Transitions foreground this
process by exposing how expectations, attentional
strategies,; perceptual norms are recalibrated over
time. From a predictive perspective, atmospheres are
constituted through collective anticipatory activity, in
which agents continuously negotiate environmental cues;
affordances across sensory modalities. Attunement is thus
understood as a temporally extended, socially coordinated
process shaped by prior experience; anticipated action.
By analysing atmospheric transitions through a predictive
lens, the paper argues that sonic atmospheres can be
understood as dynamically constituted; reconfigurable
achievements. This reframing challenges object-centred or
purely subjective accounts of atmospheres; opens new
ways of thinking about how sonic environments are shaped,
staged,; transformed in everyday life.

The acoustic characterisation of indoor spaces is crucial
for a wide range of applications. While global metrics
provide convenient descriptors of a room's overall
behaviour, a more spatially detailed analysis offers deeper
insight into the spatio-temporal structure of the sound
field, albeit at a higher experimental cost. This paper
proposes a methodology that leverages the predictive
capabilities of sound field reconstruction methods to
estimate room acoustic parameters as a function of
position. The approach is experimentally evaluated in an
auditorium, where it achieves accurate estimation of
temporal; energetic room acoustic parameters across the
entire audience area. In addition, the reconstructed field
yields higher intelligibility indices compared to the raw
measurements. Overall, these results highlight the
potential of sound field reconstruction techniques as a
practical tool for room acoustic characterisation; for
supporting assistive listening technologies.

MPEG-4 SLS (scalable lossless coding) was published more
than 20 years ago. In the meantime several tools to improve
coding efficiency; flexibilities have been invented.
Currently, in MPEG WG6 (audio coding) there are two
standardization activities on lossless audio coding: Audio
Coding for Machines (ACoM); Biomedical; general
waveform signal coding (BWC).
ACoM phase 1 originally was targeted only towards lossless
storage formats for training of machine listening schemes,
but additional uses cases like “user generated content
analysis”, “live stream content analysis”,; “artistic
creation” have been added. The focus was extended to the
transmission of audio data from microphone (arrays) to
central processing units.
BWC is a joint activity with TU-R SG21. While ACoM started
with a large number of use cases; includes the
specification of a rich set of metadata BWC started with a
focus on medical data like electroencephalogram (EEG);
electrocardiogram (ECG). However, BWC can be used for audio
signals, too; medical data coding are on the list of use
cases for ACoM.
The call for proposals (CfP) for ACoM was completed in
January 2025. Two proposals, both outperforming MPEG-4 SLS,
had been submitted. Both proposals reused; optimized
core codecs from BWC. Currently, MPEG audio investigates
how the ACoM proposals can be merged into BWC. This merge
process must be completed end of April 2026.
The presentation will give details about ACoM use cases,
the ACoM CfP process, the results of the CfP; results
from the merge process.

This presentation develops a conceptual framework for
understanding how visitors cognize sound in museum
exhibitions. While sound increasingly features in museum
practice, research has focused primarily on measuring
visitor enjoyment; engagement rather than examining the
specific meanings sound generates. This gap reflects the
absence of a framework conceptualizing sound's
meaning-making capacities to guide empirical investigation.
Drawing on scholarship from music studies, semiotics,
phenomenology,; embodied cognition, I propose a
seven-component spectrum identifying distinct yet
interrelated meanings that sound can convey in museums:
aesthetic, representational, emotional, sensorial,
imaginative, social,; political. These meanings can be
apprehended independently or in combination, typically
through emergent, pre-conscious perception rather than
deliberate awareness.
The spectrum builds on the premise that museum sound
meaning-making unfolds through dynamics internalized from
early childhood as we attune to the world sonically. It
draws on the notion of sound as a "sonic aggregate"
(Grimshaw; Garner 2015)—encompassing social, contextual,
temporal,; embodied experiences—rather than reducing
sound to wave phenomena. Visitors actively co-produce
meanings by drawing on their moods, memories, knowledge,
; imagination during exhibition encounters.
Each meaning category is illustrated with exhibition case
studies, demonstrating the spectrum's applicability across
diverse sound-based multimodal museum practices—from
popular music exhibitions to sound art installations. The
spectrum aims to catalyze research through varied
methodological approaches; establish analytical
standards for studying sound in museums, with potential
adoption by international standardization bodies.

Variable‑curvature line arrays achieve their intended directivity and spectral balance through phase‑coherent summation across cabinets. Even small timing disparities between elements perturb the interference patterns that shape the array response, with consequences for both spatial coverage and timbre. In this work we quantify these effects end‑to‑end. Using simulations for a typical 12‑element array, we examine how inter‑element delays modify the frequency response across an audience area. We then apply an auditory coloration model to predict the perceived impact of those modifications and validate the predictions through controlled listening tests. We observe that delays of a few dozen microseconds generate pronounced spectral coloration that listeners consistently judge as degraded quality, whereas coloration becomes detectable at delays on the order of one microsecond. These results translate into synchronization accuracy targets for high‑fidelity line‑array deployments.

Metamaterials are engineered structures whose acoustic;
mechanical behavior arise from their geometric
configuration; internal architecture rather than their
material properties. Within this group, vibroacoustic
metamaterials offer the ability to influence elastic wave
propagation by introducing frequency bands in which
flexural vibrations are either suppressed or selectively
altered. The integration of such structures into musical
instruments, particularly acoustic guitars, provides a
promising approach to shaping their vibroacoustic response
; mitigating undesirable structural resonances.
The objective of this project is to design a vibroacoustic
metamaterial capable of modifying the resonance properties
of an acoustic guitar soundboard. For this purpose,
vibration measurements with Laser Doppler Vibrometer were
conducted to identify the fundamental resonant modes of the
soundboard. Based on these measurements, a coupled
structural-acoustic numerical model was developed using
COMSOL Multiphysics; subsequently calibrated with the
experimental data. In the following phase, various
vibroacoustic metamaterial configurations were designed,
; their influence on the resonance characteristics of the
soundboard was investigated. The most effective resonator
design was fabricated using 3D printing; its performance
was experimentally evaluated.
The anticipated outcome of this research is the development
of an effective method for tailoring; enhancing the
tonal response of an acoustic guitar without modifying its
conventional construction, thereby contributing to new
design strategies for stringed musical instruments.

This article examines open-reel tape recorders marketed
under the “Yupiter” brand as a key technology of everyday
life in late Soviet Ukraine; as a material foundation
for the formation of a personal acoustic environment in the
Ukrainian SSR. The study aims to reconstruct the
“biography” of the device, including its design, serial
production ramp-up, distribution,; use. It shows how the
institutional constraints of a planned economy;
defense-sector priorities were translated into domestic
regimes of listening; recording. Methodologically, the
article combines approaches from sound studies, the history
of technology,; the history of everyday life,
supplemented by concepts of the “domestication” of
technology, DIY culture,; “phonographic labor.” The
source base includes internal documents of the Kyiv
“Kommunist” plant (annual reports, explanatory memoranda,
plans,; quality-related materials for 1968-1976),
interdepartmental reviews; programmatic materials of the
sector, technical handbooks; instructions, as well as
oral interviews with users. Bringing together the “upper”
level of managerial reporting; the “lower” level of user
experience makes it possible to identify a gap between
quality as a planning category; quality as a daily
practice: repairability, shortages of parts; tape,
re-recording,; selective choice of media were more the
norm than the exception. The article demonstrates that the
“fine-tuning” of tape recorders became institutionalized
through networks of amateur knowledge; informal service,
while fluctuating availability (shortage; overstock)
shaped the social geography of purchase. Ultimately,
“Yupiter” emerges not as a symbol of progress or nostalgia,
but as a material trace of late-socialist modernization -
one that helps integrate the Ukrainian case into
international debates on media materiality, listening,;
the politics of audibility. Particular attention is paid to
the temporality of the object: the extension of “Yupiter’s”
normative life cycle through repair; re-recording, as
well as its “outliving” of the Soviet system in the 1990s.
This makes it possible to interpret the tape recorder as a
carrier of acoustic memory; an indicator of social
hierarchies of access to technology. The findings refine
the understanding of shortage not as mere lack, but as an
everyday regime in the life of things.

Audio-over-Ethernet (AoE) protocols have become fundamental
in modern live sound reinforcement systems, yet their
real-world synchronization behavior under diverse stress
conditions, both in terms of load; configuration, is not
accurately characterized. Microsecond-scale timing
mismatches between amplifier outputs can disrupt line-array
interference patterns, reducing directivity control;
spectral consistency. Ensuring robust timing accuracy
across large, mixed-traffic network topologies is therefore
critical for predictable system performance.
This paper presents a comprehensive, application-oriented
evaluation of Dante, AES67; Milan-AVB. A representative
multi-hop architecture typical of touring deployments has
been considered. A controlled measurement campaign,
combining eight daisy-chained switches, heavy concurrent
data traffic approaching link saturation,; sub-sampled
latency tracking, assesses each protocol under ideal
conditions, typical field situations,; common
misconfigurations.
Results reveal clear performance distinctions. Dante
exhibits substantial timing variations, exceeding
100~$\mu$s under load. AES67 provides tighter
synchronization but remains vulnerable to configuration
errors, which can induce latency drift or even audio packet
loss. Milan-AVB consistently maintains sub-microsecond
accuracy across all scenarios.

Mixed-phase impulse response equalization can improve
magnitude; phase response, but conventional objectives
such as mean-squared error (MSE) can favor solutions that
introduce objectionable temporal artifacts, including
pre-echo; extended post-echo ringing. This paper
proposes a Spatial Equalization Quality Measure (SEQM) to
select a mixed-phase equalization filter that better
controls these artifacts while remaining computationally
simple; applicable across multiple listening positions.
SEQM combines (i) a temporal-domain metric that penalizes
energy preceding the main pulse of an impulse response;
energy persisting after it, while also accounting for the
decay rate of the post-response tail, with (ii) a spatial
aggregation rule that summarizes quality across measurement
positions. We use SEQM to select the modeling delay for
mixed-phase finite-impulse-response (FIR) equalization;
to compare mixed-phase FIR designs with minimum-phase FIR
; IIR alternatives under a common multi-position
measurement framework. Experiments using semi-anechoic
measurements across 34 spatial positions for two
loudspeakers show that SEQM consistently selects
substantially shorter delays than MSE-based selection;
yields impulse responses with reduced pre-echo; faster
post-response decay, while maintaining comparable
frequency-response equalization. These results suggest that
SEQM is a practical objective tool for designing
multi-position mixed-phase equalization filters.

This paper presents the perceptual evaluation of the Open Binaural Renderer (OBR), an open-source librarydeveloped for headphone-based rendering of Immersive Audio Model and Formats (IAMF) content. The evaluationfollowed an iterative framework in which findings from a pilot listening study informed the tuning of renderingprofiles, and the resulting renderer was benchmarked against established proprietary solutions. In the pilot study,19 expert listeners rated the Overall Listening Experience (OLE) of the initial prototype (OBRv1) and five externalrenderers across diverse audio content. Qualitative feedback was analysed using inductive coding to identify salientperceptual dimensions. The pilot revealed content-dependent performance and showed that a single default profilewas inadequate, yielding mixed responses in both the numerical scale and in the qualitative feedback and motivatingthe development of multiple rendering profiles in OBRv2. The main study evaluated two OBRv2 profiles targetingdifferent reverberation characteristics (Direct and Ambient) alongside three top-performing external renderers. Atotal of 39 participants, divided into expert and non-expert groups, rated five perceptual attributes: Voice Quality,Envelopment, Externalisation, Overall Listening Experience, and Timbral Balance. Mixed-design ANOVA revealedsignificant main effects of renderer condition on all attributes. Pairwise comparisons showed that OBRv2,Ambientachieved significantly higher OLE ratings than one proprietary renderer and reached statistical parity with theremaining two, representing a measurable improvement over the prototype. A trade-off between Voice Qualityand Externalisation was observed, driven by the level of reverberation in each renderer. The results demonstratethat iterative, perceptually informed tuning can yield competitive binaural rendering quality in an open-sourceframework.

Despite the growing number of hearing-impaired workers
wearing hearing-aids in occupational settings,
understanding speech in multi-talker situations remains
challenging. This difficulty is particularly pronounced in
open-plan offices, where simultaneous talkers; room
reverberation are prone to degrade speech intelligibility.
While spatial cues are essential for segregating target
speech from competing sources, hearing-aids signal
processing may alter binaural information that supports
spatial hearing.
Accurate evaluation of hearing-aids performance is
therefore crucial. Objective speech intelligibility metrics
offer an efficient alternative to time-consuming listening
tests; however, their validity in complex spatial scenarios
involving hearing-impaired listeners remains unclear.
Monaural metrics such as HASPI account for individual
hearing loss but neglect spatial information, whereas
binaural metrics such as MBSTOI incorporate spatial cues
but are primarily designed for normal-hearing listeners.
This study evaluates the ability of existing objective
metrics to predict speech intelligibility for hearing-aid
users in multi-talker spatial environments. Listening tests
are conducted on 20 hearing-impaired participants fitted
with binaural hearing-aids. Four types of multi-talker
auditory scenes representative of open-plan offices are
reproduced using a loudspeaker array. They involve a target
speech, combined with diffuse noise; a localized
competing speech source. Objective measurements are
performed using an acoustic mannequin fitted with the
participants’ hearing-aids. HASPI; MBSTOI values are
computed from the binaural signals recorded at the eardrums
; incorporating individual hearing losses.
Objective predictions are compared with subjective
intelligibility scores,; an ablation analysis is
conducted to distinguish the effects of hearing loss
modeling from those of binaural processing.

High-speed 1-bit signals generated by oversampling are
widely used in audio applications as they allow simple
demodulation via low-pass filtering while preserving
in-band spectral characteristics with high accuracy.
However, conventional FIR filtering of such signals
generally requires conversion to a multi-bit representation
at a common sampling frequency, which increases
computational cost; complicates the overall processing
flow. This paper addresses the convolution of high-speed
1-bit audio signals with multi-bit FIR impulse responses
; presents a systematic formulation of a multiplier-less
convolution approach. Based on a mathematical
reinterpretation of convolution, the proposed formulation
describes how time shifting; amplitude weighting can be
expressed through structured rearranging of 1-bit samples
without arithmetic operations. This provides a theoretical
description of previously reported 1-bit convolution
methods; however, its validity has not been fully
formalized. We examine the spectral characteristics of the
proposed convolution method; compare them with those
obtained by multi-bit convolution followed by ΔΣ
modulation. Experiments are conducted by convolving 1-bit
input signals with FIR filters having multi-band frequency
responses. Spectral analysis shows that the proposed method
achieves extremely high agreement with the standard
approach within the audible band while the differences
appear primarily at much higher frequencies outside the
audible range. These results demonstrate that convolution
of high-speed 1-bit audio signals can be achieved without
multipliers, suggesting the potential for highly efficient
hardware-oriented signal processing architectures.

Individualized head-related transfer functions (HRTFs)
require accurate pinna geometry, yet commodity multi-view
captures leave the ear region self-occluded; weakly
textured. We present a practical pipeline that couples
ear-centric acquisition with 3D Gaussian splatting (3DGS)
; the boundary element method (BEM) for complete HRTF
computation. The protocol augments horizontal views with
per-ear elevated captures under directional lighting; 3DGS
training with depth-distortion regularization yields
watertight meshes via truncated signed distance function
(TSDF) fusion. Standardized head coordinates; ear-canal
annotations interface the mesh with BEM. Experimental
evaluations demonstrate that our method achieves lower
ear-region geometric error; lower full-band spectral
distortion compared to existing image-based personalized
reconstruction baselines including AudioEar, NeuS,;
Metashape MVS.

Situational awareness is a multisensory ability that
enables individuals to perceive; appropriately take into
account their immediate environment. This perception of the
world through our senses is carried out continuously;
unconsciously throughout the day. When auditory perception
is degraded, an individual may no longer correctly perceive
a doorbell, a water leak, or an alarm signal, which
negatively affects quality of life; may lead to
dangerous situations. Auditory perception can in particular
be degraded by hearing loss, a common; widespread
condition. The most common treatment consists of wearing
hearing aids, which are mainly designed to improve speech
intelligibility, especially in noisy environments. Feedback
from hearing-impaired people; hearing-aid users
indicates that, although auditory situational awareness has
been recognised as an essential component of well-being, it
remains insufficiently studied; requires further
investigation. There is currently no standard method for
assessing to which extent one's situational awareness is
affected by hearing impairment; the use of hearing aids.
This is a complex process that requires assessing the
perception of relevant sound events within a continuous
stream of multisensorial information, by individuals who
have different subjective preferences. Most existing
methods are limited to evaluating only a subset of the
problem, such as identification; localisation of
non-speech sound events. The rise of new technologies, such
as virtual reality, enables the development of assessment
methods within more realistic yet controlled environments.
This study aims to review existing methods in order to
highlight their limitations in addressing the issue at hand.

Headphone listening has become an integral part of everyday
life, spanning music consumption, communication, online
media,; increasingly, computer gaming. These diverse
listening contexts make individual sound exposure highly
variable; difficult to quantify. While music listening
; occupational headphone use have been widely studied,
sound exposure from gaming remains comparatively
undocumented. This study investigated the relationship
between self‑reported exposure through headphones;
cochlear function assessed using transient evoked
otoacoustic emissions (TEOAE). Forty‑one university
students completed a detailed questionnaire on listening
habits,; TEOAEs were recorded in both ears across five
half‑octave frequency bands. Estimated weekly exposure
levels were derived from participants’ reported durations
; contexts of use. TEOAE amplitude, signal‑to‑noise ratio
(SNR),; reproducibility showed clear frequency‑dependent
patterns; small ear asymmetries, consistent with typical
OAE behaviour. Only limited associations were found between
self‑reported exposure; TEOAE measures, with significant
effects emerging primarily for SNR; reproducibility in
the highest‑exposure group. No consistent differences were
observed between long‑term gamers; non‑gamers. These
findings suggest that self‑reported exposure alone may be
insufficient to detect subtle cochlear changes in young
adults,; underscore the need for more precise
exposure‑monitoring methods when evaluating recreational
sound exposure risks.

This paper presents a multichannel adaptive filtering
algorithm for real-time full-band adaptive transaural
reproduction on general-purpose hardware. It is based on a
multichannel frequency-domain FxLMS algorithm using an
overlap-save framework for both filtering; adaptation,
; is extended with (i) online plant identification for
fully adaptive operation, (ii) frequency-dependent
normalization for faster convergence,; (iii)
frequency-dependent regularization to stabilize adaptation.
The proposed algorithm is implemented in C language on a
standard desktop PC; evaluated on a 4x2 transaural
configuration running in real time at 48 kHz with 2048-tap
control filters. Two evaluation tests are conducted. The
first test consists of reproducing two uncorrelated
white-noise signals at the ears of a manikin using
crosstalk cancellation as the performance metric. An
average crosstalk cancellation of 32 dB over 100 Hz–20 kHz
is demonstrated. The second experiment considers binaural
signal reproduction as a more realistic use case of the
algorithm. In both cases, performance is assessed for both
a static listener; a moving listener scenario,
demonstrating the algorithm’s ability to rapidly re-adapt.

Binaural rendering is typically assessed via timbre;
localization accuracy, while its intrinsic spatial
resolution remains rarely quantified. This paper proposes a
perceptual evaluation method based on Minimum Audible Angle
(MAA) measurements to estimate the azimuthal
just-noticeable difference (JND) introduced by binaural
rendering algorithms. We systematically compared several
rendering algorithms across eight reference azimuths using
two participant-allocation paradigms. The results show that
spatial resolution is significantly influenced by Ambisonic
order; choice of the rendering alrorithm, with MAA
thresholds systematically decreasing as the truncation
order increases. Furthermore, the propsed method
successfully captures physiological spatial characteristics
; identifies resolution limits imposed by reference
angles. While both participant-allocation paradigms yield
consistent qualitative trends, the repeated-measures design
provides superior data stability. These findings
demonstrate that the proposed MAA-based method is an
effective tool for quantifying the spatial resolution of
binaural rendering algorithms.

Personal sound zones aim to reproduce distinct audio
contents in separate spatial regions using loudspeaker
arrays, while minimizing acoustic interference between
zones. Although well established theoretically, their
real-time implementation remains challenging due to the
long impulse responses involved; the latency constraints
of audio processing systems.
This work presents a real-time implementation of personal
sound zones based on the pressure matching method in a
static context, i.e. transfer functions between the
loudspeakers; the zones are assumed to remain constant.
Sound zone filters are computed in the frequency domain
from experimentally measured impulse responses between an
array of 18 loudspeakers; two microphone arrays of 9
microphones defining a bright zone; a dark zone. The
system performance is then evaluated in terms of acoustic
contrast, reproduction error,; effective frequency
range. To meet real-time constraints, a fast partitioned
convolution algorithm has been used, namely the
Uniformly-Partitioned Overlap Save (UPOLS). This methods
has been implemented in C++ as an external block for the
Purr Data real-time audio environment. Experimental
results, obtained in a semi-anechoic environment,
demonstrate that it enables stable real-time multichannel
convolution with negligible numerical error compared to
offline convolution. The proposed system results in a
functional real-time sound zones demonstrator, suitable for
experimental; interactive spatial audio applications.
The codes are shared in a GitHub repository so that the
scientific community can benefit from them.

The evaluation of audio quality is important in the
development of immersive audio algorithms; reproduction
systems,; binaural models are often used for this as a
quick alternative to listening tests. Coloration (i.e.,
perceived loudness differences integrated across ears;
frequency) is one key quality aspect; however, the majority
of models used to predict coloration are often
oversimplified or are missing a dedicated binaural stage to
consider the relative contribution of the left; right
ear signals. A binaural coloration model is presented that
builds upon previous work; tests three different
approaches for its binaural stage. The proposed model is
evaluated in comparison with nine models that are
frequently used to predict coloration by using data from
five listening tests totaling 252 stimuli with various
audio contents; source positions. The proposed model
performed best with 85% of explained variance, followed by
predictions based on ISO 532-1 loudness, yielding 78%
explained variance. The commonly used log-spectral distance
performed worst, with only 44% explained variance. The
three tested binaural stages had little influence on the
performance of the proposed model. The model is made freely
available to download.

This study evaluates three Next-Generation Audio (NGA)
rendering systems through listening tests using real-life
audio content. The testing paradigm prioritized subjective
preference over adherence to a ground-truth reference.
Participants assessed perceptual spatial audio attributes
in both 5.1; 7.1.4 loudspeaker setups. The findings
suggest that strict adherence to the rendering algorithm
used during content creation is not mandatory in terms of
listener preference. While not advocating disregarding
artistic intent without consideration, this study proposes
that such flexibility in reproduction can be an acceptable
compromise.

Immersive audio continues to expand beyond traditional
studio; terrestrial field-recording environments, yet
underwater soundscapes—particularly those involving marine
mammals—remain largely documented in mono or stereo
formats. This paper presents a practical; low-cost
approach for capturing immersive underwater audio using a
newly developed wideband hydrophone; a multichannel
array optimized for marine environments. The hydrophones,
designed by the author, feature a low noise floor, extended
frequency response exceeding 100 kHz,; direct
compatibility with standard P48 phantom-powered audio
recorders, enabling deployment without specialized
underwater preamplifiers or power systems.

To translate established immersive recording techniques
into the ocean environment, an array architecture was
developed based on a compact eight-element cube geometry.
Two array variants were constructed to account for the
significantly higher speed of sound in water compared to
air, allowing the spatial characteristics of underwater
sources to be captured with appropriate inter-element
spacing. Field recordings were conducted off the coast of
Hawaii in January during the peak season for humpback whale
song. Recordings were made at multiple depths; positions
to explore variations in reverberation, propagation,;
ambient biological activity.

Preliminary results indicate that the system captures
detailed spatial cues from humpback whale vocalizations
while simultaneously preserving the rich ambient marine
soundscape. The extended ultrasonic response further allows
slowed or pitch-shifted playback to reveal fine temporal
structures not typically audible. This work demonstrates a
feasible method for immersive underwater recording;
provides a foundation for both scientific research;
creative content production.