AES Europe 2026

This paper introduces a novel approach for generating a
lower layer in multichannel audio upmixing, addressing a
gap in existing methods that primarily focus on mid; top
layers. Leveraging Harmonic-Percussive Separation (HPS),
the proposed framework dynamically adjusts key parameters
(separation factor, harmonic attenuation,; phase shift)
to enhance percussive components while diffusing harmonic
elements. We compared three neural network architectures
for this task: LSTM, TCN,; Transformer. Experimental
results show comparable perceptual quality; objective
metrics across all models, with the TCN being the most
balanced; suitable for deployment on edge devices.

Since 2021, 7.1.4 musical content has transitioned from a
niche specialty to a mainstream commercial deliverable
within major streaming ecosystems. However, industry
discourse indicates a disparity in how the immersive stage
is utilized across different production tiers. This paper
presents a targeted quantitative study of thirty 7.1.4
tracks (N = 30 total; 15 per category; 2021–2026),
employing a matched-pair sampling strategy driven by the
availability of 'Established Excellence' (Grammy
Award-winning/nominated immersive albums) against
genre-equivalent 'Market Dominance' (top-charting streaming
tracks). The study utilizes a multi-parameter measurement
methodology, including Inter-Channel Cross-Correlation,
hemispheric symmetry; spatial width analysis.
Furthermore, vertical spectral centroid distribution;
channel occupancy (Center; LFE) are analyzed to identify
recurring structural immersive design markers. Preliminary
findings suggest a consistent forward-facing bias; lower
activity in select channels in charting commercial releases
compared to award-recognized counterparts. By documenting
these technical indicators, such as quarter-sphere
correlation; LFE handling differences, this study
establishes a benchmark for current immersive mixing
practices; highlights the technical indicators that may
limit the transition from enhanced stereo to true immersive
envelopment.

This work presents the results of a perceptual study
investigating the influence on musicians of a virtual
acoustics system installed in the live room of a
professional recording studio. The study focused on
analyzing relationships between a selection of objective
acoustic parameters (T30, STLate, LJ); subjective
perceptions of 19 solo
musicians performing under 11 different acoustic
conditions. The experiment was conducted using the VAT
(Virtual Acoustic Technology) system; the VAT Suite
software developed at the Immersive Media Laboratory
(IMLab) in the Sound Recording Department at McGill
University. Correlations between quantitative;
qualitative analyses
show that musicians’ preferences converge on conditions
with T30 ≈ 1 s,; that late; lateral energy increases
the perception of spatiality, providing a positive balance
between clarity; acoustic support. However, longer
reverberation reduces comfort; executive control.

This study introduces a fourth-order Ambisonics-based decoding system to reproduce railway cabin running noise in a studio environment, enabling enhanced spatial impression and detailed sound field variation. Real-world operational noise was recorded using a multichannel fourth-order Ambisonics microphone (Eigenmike® EM32, mh acoustics LLC, USA), and the reproduced sound field was implemented through a multichannel loudspeaker system. The reproduced signals were quantitatively compared with the original operational noise in terms of spectral variation and waveform distortion.

Taking the premiere and reperformance of the sci-tech symphonic suite Symphonic Coding as a case study, this paper discusses audio system organization, sound diffusion, and cross-venue migration in the co-performance of symphonic and electronic music. Given the challenges of diverse live inputs, real-time control of the electronic music part, concurrent recording and live streaming, and varying acoustic conditions, the article analyzes how a single workflow handles traditional miking, electronic music generation and control, live spatial diffusion, and multi-purpose distribution. The study is structured across four levels: system design requirements, signal organization, dual-venue implementation, and engineering discussion. It illustrates the development of an interconnected workflow comprising Content, Rendering, and Distribution Layers through mixing console organization, immersive rendering, and AoIP distribution. Results indicate that the significance of this work lies not in the reproduction of the listening experience of the entire performance, but in enabling the spatial presentation of the electronic music part to remain valid across different environments based on a consistent reference. Furthermore, the project enhances reperformance capability and production flexibility through the separation of functions, roles, and systems.

A “phantom image” is the illusion of an independent sound
source created by two or more loudspeakers. Most often
created by manipulating level differences between
stereophonic channels (aka, “panning”), the effect is used
to create a sense of auditory space between loudspeakers
; is largely taken for granted. In recent years,
surround; immersive audio systems have attempted to
utilize phantom image processing to render audio objects in
desired positions across multiple loudspeaker arrays. This
research examined the efficacy of phantom image perception
horizontally; vertically from an active listener
perspective. After listening to a target loudspeaker,
listeners (n = 442) were asked to move a phantom sound to a
position to match that of the target loudspeaker. The
listener’s phantom placement was then compared to the
target,; subjects were allowed “correct” their phantom
position. The horizontal experiment was based on a
standard stereophonic 60° loudspeaker array with the target
loudspeaker at 15° off center. The vertical experiment
utilized elevated loudspeakers in a 60° arc with the target
loudspeaker elevated 10° above the horizon (lower
loudspeaker). Results show nearly universal “undershoot” in
horizontal placement error on first attempts with gradual
improvement over trials that coalesced around the projected
target location. However, after repeated tries, final
perceptual image locations were spread over 2/3 of the
sound-field around the target loudspeaker. In the vertical
trials perceptual locations were spread across the entire
sound field in all three trials; failed to show any
patterns of coalescence around the target loudspeaker.

EMORSION is an exploratory study examining how film audio
design shapes audience emotion; immersion. It was
conducted using scenes from four films in the horror (2)
; drama (2) genres, with two mainstream; two
independent productions. For each scene, multiple
alternative audio mixes were created by systematically
manipulating three core aspects of audio design; frequency
(pitch), dynamics (loudness),; directionality (spatial
placement). Three audience groups were exposed to the
scenes in a cinema setting, with each group experiencing
either one manipulated audio mix; a control mix.
Audience responses were assessed through a multimodal
framework combining self-reported emotion; immersion via
a questionnaire,; physiological measures, including
heart rate monitoring; video-based motion tracking.
Results show that subtle changes in audio design
significantly affect emotional perception; immersion.
Unconventional mixes produced greater variability in
interpretation, while conventional immersive mixes led to
stronger agreement across audiences. Notably, participants
often reported perceived visual changes despite no
alterations to the visual content.

Live music environments can be simulated; evaluated
through spatial audio; augmented reality (AR)
technology. However, conducting perceptual studies on AR
environments can be challenging, as multiple design
considerations; uncontrolled variables come into play.
Hence, we developed Naviqual, a tool to create a spatial
audio quality map for a virtual live music environment. We
generated objective quality contour; polar maps to
predict the quality of experience (QoE) across listener
locations; directions respectively. We found that these
maps strongly aligned with perceptual evaluations by
normal-hearing listeners through listening tests. We also
found that binaural objective metrics; signal-to-noise
ratio both strongly predict QoE across listener
translations, with the former outperforming the latter in
predicting QoE across listener directions. Overall,
Naviqual provides a QoE map for virtual live music
environments robust across various listener locations;
directions, noise locations, music content,; room
acoustics.

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.

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 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.

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.

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.

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.

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.

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.

This paper presents the production of a fully volumetric
audiovisual music composition with six degrees of
freedom, featuring a dance performance. The project
realizes the artistic potential enabled by recent
technological
advancements in volumetric video capture; spatial audio
rendering. The interdisciplinary production team
consisted of music composers, dancers, sound engineers,;
experts in 4D Gaussian Splats. An existing 3D body
scanner consisting of 112 cameras was used to capture a
dance performance in high definition video. For the
visual scene, custom 4D Gaussian Splat algorithms were
developed; employed to create the dynamic model.
Additional static 3D Gaussian Splats were captured with the
same scanner; integrated into the scene in Unity.
The acoustic scene is dynamically binauralized via SPAT
Revolution, depending on the position of the listener
in the virtual space. Audio; video scenes are run on
separate PCs, synchronized via OSC; presented via
commercially available head-mounted displays (HMDs).
Audiences report a high level of immersion at the initial
presentation at an exhibition event. A detailed evaluation
is planned in the near future. Furthermore, a unified
application for both visual; audio scenes is planned in
order to reach a wider audience.

Historically, music has developed primarily as a frontal
phenomenon, thus limiting the expressive; perceptual
potential related to sound space. The recent development of
immersive audio systems opens new creative possibilities by
expanding the artistic action space from a narrow frontal
area to a complete sphere around the listener. The
Ambisonic system (Scene-Based Audio), together with
Object-Based formats; hybrid solutions, represents
fertile ground for creative experimentation; the
redefinition of workflows in the field of spatialized sound.
In this new context, what is the role of the sound
engineer, as an electroacoustic interpreter, in immersive
musical artistic creation?
The research is based on a multidisciplinary analysis that
combines an in-depth study of current immersive audio
technologies; their performance, with observations of
existing compositional; production approaches.
Additionally, a comparative study is conducted on the
design choices of the sound engineer as an interpreter,
investigating workflows, emerging musical semantics,
available tools,; the recovery of the historical
repertoire.
Particular attention is paid to the experiment aimed at
investigating a correlation between the position of a sound
; an emotional trigger in the listener.
New directions emerge in the creative role of the sound
engineer, who goes beyond the mere technical aspect to
become an integral part of the compositional;
interpretative process, harmonizing the relationship
between technique; art.

The low-frequency performance of cross-talk cancellation
(CTC) systems is fundamentally limited by the condition
number of the plant matrix, which indicates the robustness
of the inverse system in the absence of regularisation.
This condition number, in turn, depends on the relationship
between loudspeaker spacing, listener distance,;
acoustic wavelength.
This paper derives a simple approximate expression for the
low-frequency limit of CTC performance, defined for a given
maximum affordable condition number as a function of these
parameters. The increase in condition number is also shown
to be directly related to the increase in array effort
relative to the minimum achievable array effort. The
formulation is derived for a centered listener; can be
extended to the case of off-center listener positions,
demonstrating the method's applicability to
listener-position-adaptive cross-talk cancellation systems.

The MPEG-I Immersive Audio standard for Virtual;
Augmented Reality (VR/AR) audio with six degrees of freedom
(6DoF) was completed in November 2025 by the MPEG Audio
group (ISO/IEC JTC 1/SC 29/WG 6)). It offers compressed
representation of virtual audio scenes as well as an
efficient; acoustically sophisticated rendering to both
head-tracked binaural headphones; loudspeaker setups.
The latter is a unique feature among VR/AR audio
specifications; enables convincing reproduction of
conventional stereo, surround; 3D material with a large
sweet area in home entertainment setups for a single
tracked user, without the need for a head-mounted
display. This paper describes the technology of MPEG-I
Audio listener-tracked loudspeaker rendering as a
stand-alone application with a special focus on practical
considerations for optimal room calibration.

Sound plays a critical role in virtual reality (VR),
shaping attention, narrative comprehension, emotional
engagement,; experiential plausibility under conditions
of embodiment; user agency. Although a growing body of
research addresses VR audio techniques, perceptual effects,
; sound taxonomies, existing approaches remain fragmented
; largely descriptive. In particular, they do not provide
a unifying, VR-specific account of how sound meaning;
emotional intent are operationally linked to user agency
; non-linear narrative progression. This paper presents a
narrative review of selected literature spanning game audio
frameworks, immersive sound design, narrative theory,;
plausibility-related research in games; VR. Through
synthesis of these perspectives, the review identifies a
conceptual gap in current research, namely the absence of a
VR-specific, agency-coupled sound design framework for
structuring sound meaning; emotional intent in support
of experiential plausibility as users actively shape events
in interactive VR environments.

Most of music contents available are stereo which cause
inadequate spatial treatment; listeners feel
disconnected from the music, failing to transport them into
the intended sonic environment. Insufficient separation
between instruments can lead to an unbalanced mix, where
certain elements dominate others; disrupt the overall
harmony. Instruments may appear flat; confined to a
narrow area, reducing the sense of dimensionality in the
mix. Stereo audio offers limited spatial information,
restricting its adaptability to immersive sound
environments. This research presents a novel approach for
converting stereo audio into a personalized immersive
experience by leveraging object-based audio rendering,
sound stage of listener; surround speaker capability.
The proposed system separates audio signals into individual
objects (such as instruments or vocals); dynamically
maps these objects to specific speakers based on
personalized preferences; spatial configurations. This
method improves audio localization; enhances the
listener's engagement by delivering a tailored auditory
experience.