AES Europe 2026

Acoustic lenses are structures that enable the focusing of
acoustic waves, with increasing applications in audio
devices like loudspeakers to concentrate energy toward a
listening position. While typically employed at higher
frequencies, achieving effective performance within the
audible frequency range remains a significant challenge due
to long acoustic wavelengths, which necessitate structures
of substantially larger dimensions.
This paper addresses the design of an acoustic lens
dedicated to operation in the audible range. The proposed
lens is composed of periodically arranged acoustic unit
cells, enabling precise control over both the sound
transmission coefficient; the phase delay. A parametric
analysis of a single acoustic unit cell was performed,
followed by global optimization of the complete lens
structure using the Particle Swarm Optimization (PSO)
algorithm. The outcome of the study is an acoustic lens
design with predefined properties that demonstrate the
desired directional characteristics. The findings highlight
the potential of this approach for effectively manipulating
the acoustic wave field; the directivity of sound
sources within the audible frequency range.

In today’s live; electronic music events there are some
sound reinforcement systems that are using horn loaded bass
speaker cabinets to provide the low-end section. Especially
for the electronic music applications the PA system is
designed to use one or multiple clusters of bass cabinets
to provide the needed SPL; impact in the low frequency
range. Despite being large; heavy the horn loaded bass
speakers have some advantages like the efficiency;
directivity which makes them a great option for electronic
music. Even more, the enthusiasts are describing them as
having a longer projection of the sound when compared with
bass reflex units. When used in clusters the bass horns
present a mutual coupling due to a larger mouth surface
area; the physics behind. This effect alters the working
parameters in a good way regarding sound reproduction;
is clearly noticed at high levels. This mechanism increases
the output close to the low edge of the frequency response
interval; changes the directivity pattern. A cluster of
four or six double 18” horn loaded bass bins placed in the
front middle of a dance area will provide good impact
described a “punchy” sound, so acclaimed in the electronic
music party scene. In this paper I will describe an
investigation of the mutual coupling between horn cabinets
using electrical; acoustical measurements to reveal the
mentioned above mechanism. Electrical impedance measurement
together with SPL; frequency response in coupled;
uncoupled scenarios are used to describe; demystify the
mutual coupling phenomena.

The development of personal sound zone systems in recent
years show great potential for low-frequency noise control
outside of noisy spaces. These approaches show promising
applications to manage noise pollution arising from
concerts in large venues or urban festivals. However, most
of the literature considered that the created sound zones
would exist in the same room or acoustic space as the noise
source. This premise hence discards all setups where the
disturbances would occur outside of concert venues (e.g in
neighboring houses). This paper presents a first
experimental study of the behavior of sound zone methods
for indoor sound zones; outdoor noise sources. These
initial results present a good efficiency of these methods
in this edge case, opening new use cases for these
approaches.

Damping in viscoelastic materials such as rubbers is often
desirable, especially in loudspeaker suspensions. Under
high strain loads however, viscoelastic materials can also
exhibit a hysteretic stiffness behavior, causing a
stiffness decrease with amplitude. In this study, we
examine the viscoelastic rubber suspension of a
loudspeaker, using the loudspeaker motor system as actuator
; sensor. From measurements we observe the hysteretic
force-displacement behavior; pronounced odd-order
harmonic distortion even at low amplitudes, in accordance
with the literature. We further explore a
macro-thermodynamic plastic flow model to model the
stiffness of viscoelastic materials. The results show that
the plastic flow suspension model explains; replicates
the observed nonlinear hysteretic behavior. We also show
that a fitted time-domain loudspeaker model including
plastic flow matches the measured distortion profile. In
contrast, models with polynomial stiffness; viscous
damping fail to explain the observed amplitude dependencies
such as odd order harmonic levels. The experiments
demonstrate that viscoelastic hysteresis occurs not only at
high but also at low amplitudes, where the elastic
stiffness is approximately linear.

Mechanical overload remains a primary limitation in
high-output loudspeaker operation, particularly at low
frequencies where large coil excursions are required.
Conventional mechanical protection strategies are typically
implemented as signal-domain limiters or filters, which act
indirectly on the loudspeaker’s mechanical state; may
introduce discontinuities, spectral modification, or
unnecessary attenuation.

This paper proposes a methodological framework for
mechanical loudspeaker protection based on the
virtualization of admissible system behavior. The approach
is formulated within a nonlinear wave digital loudspeaker
model; realized using a direct–inverse–direct
architecture. Mechanical protection is embedded directly
into the virtual loudspeaker dynamics by shaping the
nonlinear suspension compliance as a function of voice-coil
displacement. As the excursion approaches a prescribed
admissible limit, the virtual compliance is progressively
reduced using a smooth raised-cosine law, resulting in a
continuous increase of the virtual mechanical stiffness.
Excessive excursion is therefore prevented as a consequence
of the system dynamics, without explicit limiting,
clipping, or signal-domain intervention.

The proposed framework is evaluated through numerical
simulations using steady-state low-frequency sinusoids;
low-frequency sine bursts under free-air loading. Results
are compared against an unprotected loudspeaker; a fixed
high-pass filter configured to meet the same excursion
constraint. The simulations verify that the proposed method
enforces a soft excursion ceiling without discontinuities,
preserves low-frequency output in the near-limit operating
region,; exhibits stable; immediate recovery
following transient excitation. Distortion behavior is
characterized; shown to increase smoothly as a result of
the introduced mechanical nonlinearity.

The results demonstrate that mechanical protection can be
realized as an emergent property of a virtual loudspeaker
model rather than as an external control action. The
proposed approach provides a physically interpretable;
numerically robust foundation for virtualization-based
loudspeaker protection.

Target curves for the sound signature of headphones are a
helpful design target during the development process. While
a lot of attention has been made to fi nd target curves that
match the listening preference of consumers, equivalents
for studio headphones date back to the 90’s. In the context
of music production a mutual target or even standard is
essential as to make mixing; mastering more
gear-independent. This becomes even more important since
the main tool for sound engineers shifts from loudspeakers
in professional environments such as acoustically treated
studios to headphones, often additionally equipped with
virtualization algorithms. This enables them to be more fl
exible; to rely less on potentially expensive
loudspeaker setups. The diffuse fi eld target curve that is
currently still the only standardized target curve for
studio headphones is often reported to not match a real
loudspeaker-equivalent of studio environments. In this
paper, we approach to find a new standard target curve for
studio headphones emulating the frequency response of a
loudspeaker setup in modern studio environments.
For this, we give an overview of current target curves;
match them to their equivalent loudspeaker setups.
Based on that we propose a new methodology for a
measurement-based target curve incorporating typical
panning paradigms of music signals based on measurements
inside multiple control rooms. To verify the results, we
conduct listening tests with professionals in multiple
studio environments.

Few studies exist on the perception; measurement of
nonlinear distortion in headphones. This paper reports the
detection thresholds; perceived sound quality from real
distortion in headphones. Five different distortion
measurements were made on the headphones to determine how
well they predict audibility; quality. Music samples
were binaurally recorded on six headphones at playback
levels ranging from 85 to +110 dBA at 3 dB increments. The
recordings were reproduced at a normal playback level (83
dBA) through a reference headphone with low distortion. The
headphone recordings were post-processed to remove both
level; frequency response differences so only nonlinear
distortions; residual noise remained. In a second test,
listeners rated the similarity in quality of headphones
relative to an undistorted reference; a hidden version
of it. The results provide evidence audible distortion in
headphones with music occurs at significantly higher
playback levels (104 to 112 dBA SPL) than what is
considered typical; safe. The percentage of measured THD
in the headphone had the highest correlation with the
detection thresholds while the non-coherent distortion with
music best predicted the similarity ratings. We discuss the
results; the practical implications they might have on
future headphone design, testing; measurement.

There are three architectural approaches to
microelectromechanical systems (MEMS) microphones,
miniature devices used in a wide range of products.
Capacitive microelectromechanical systems (MEMS)
microphones are embedded in billions of consumer
electronics. Solder-compatible; providing tight
part-to-part sensitivity matching—all in a small
footprint—capacitive MEMS microphones have demonstrated
improved performance in recent years. State-of-the-art
digital capacitive MEMS microphones can now achieve up to
72dB signal-to-noise ratio (SNR), with a 22dBA noise floor
; overall dynamic range in the order of 106 dB.

However, capacitive MEMS microphone technology has now
reached the limits of its architecture, which constrains
the key audio performance metrics: SNR; acoustic
overload point (AOP).

Piezoelectric MEMS microphones have not demonstrated SNR
performance exceeding 65dB,; require new materials to be
developed to increase their performance.
Optical MEMS microphones—a new architectural approach that
combines a laser optical subsystem, a MEMS; advanced
CMOS circuit design—has exceeded the limits of capacitive
technology. With 80dB SNR supporting a 14 dBA noise floor,
132 dB dynamic range,; a 146dB AOP, optical MEMS
microphones accomplish studio-quality performance in a tiny
form factor that supports semiconductor-level yields in
high-volume manufacturing.

This presentation will explain the architectural
advancements of optical MEMS microphones in comparison to
capacitive MEMS microphones. It will provide example use
cases of high-SNR; high-AOP microphones in high volume
applications.

This work presents a measurement uncertainty evaluation of
the free-field sensitivity of a MEMS microphone using a
substitution comparison method. The measurement setup is
based on principles used in secondary microphone
calibration, with sensitivity determined relative to a
calibrated reference microphone. The uncertainty analysis
follows the Guide to the Expression of Uncertainty in
Measurement (GUM), where Type A; Type B uncertainty
evaluations are propagated through a defined measurement
model to obtain the final measurement result. The MEMS
microphone sensitivity is estimated together with an
expanded uncertainty, where the calibration uncertainty of
the reference microphone is identified as the dominant
contributor. Broadband results show that the measured
sensitivity is close to the typical manufacturer
sensitivity over a wide frequency range; follows a
similar frequency trend. The proposed approach enables
reproducible estimation of the free-field sensitivity of
MEMS microphones; provides a clear framework for
uncertainty evaluation.

This paper presents an improved method for characterizing
integrated microphone arrays for Device‑Related Transfer
Function (DRTF) synthesis. A probe‑array extension of the
IMPro technique is introduced to measure all device
microphones simultaneously, eliminating unknown timing
offsets that arise in asynchronous device–probe recordings.
A custom four‑element probe array; modular test jig were
developed to evaluate relative inter‑channel propagation
delay (RIPD) accuracy across varied microphone‑port
geometries. Hybrid free‑field DRTFs were synthesized by
combining IMPro data with Boundary Element Method (BEM)
acoustic scattering simulations, demonstrating that the
probe‑array measurements capture small delay variations
essential for precise spatial‑audio modeling. The extended
IMPro method offers a practical, scalable alternative to
anechoic‑chamber measurements for modern multi‑microphone
devices.

This paper presents Part 2 of our study on personalized
timbre optimization for stereophonic sound reproduction via
earphones, following our previous work presented at the AES
International Conference on Headphone Technology in 2025.
While Part 1 established a novel auditory-model-based
framework for reproducing a listener’s natural timbre
reference; demonstrated its perceptual validity under
controlled conditions, the present study focuses on the
practical implementation; validation of this approach
for real-world use with consumer True Wireless Stereo (TWS)
earphones.

Conventional headphone; earphone personalization
techniques primarily target spatial audio reproduction or
rely on preference-based equalization, often overlooking
the accurate reproduction of natural timbre in stereophonic
content. Our approach explicitly addresses this limitation
by isolating; optimizing perceptually relevant timbral
cues while excluding spatial encoding components, thereby
improving timbral fidelity without degrading stereo imaging.

The proposed method originally consists of four stages:
high-resolution anatomical scanning of the listener’s upper
body, including the pinnae, individualized HRTF computation
using the boundary element method, selective removal of
spatial encoding components to derive a personalized
reference target response curve (PR-TRC),; perceptual
optimization using a listener-specific weighting
coefficient grounded in auditory reference fidelity rather
than preference. In this paper, each stage is simplified
; automated using smartphone-based scanning;
AI-assisted processing, enabling end users to complete the
entire personalization process via a smartphone connected
to a cloud-based server. The resulting personalized target
response curve is implemented within the computational;
memory constraints of the DSP pipeline of commercial
consumer TWS earphones.

A subjective evaluation using the Semantic Differential
Method was conducted to assess the perceptual impact of the
simplified implementation. Twenty-four listeners evaluated
personalized target curves generated by both the original
; simplified methods, as well as two non-personalized
target curves commonly used in commercial TWS earphones.
The results show that both personalized methods
consistently outperform non-personalized conditions in
overall sound quality; listener preference. Importantly,
no statistically significant degradation in perceived
timbral naturalness was observed between the simplified;
original methods.

These findings demonstrate that auditory-model-based
personalized timbre optimization can be effectively
translated into a practical, consumer-ready technology. The
proposed approach represents a foundational contribution to
future audio personalization; has broad applicability
across headphone; earphone systems for stereophonic
sound reproduction.

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.

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.

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.

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.

The bone-conducted occlusion effect (OE) is a major source
of acoustic discomfort for users of hearing aids, earbuds,
earplugs,; related devices. Conventional objective OE
measurements rely on in-ear microphones in human subjects,
which are time-consuming, invasive,; difficult to
control during product development. The aim of this paper
is to present a new artificial ear, specifically designed
for objective OE measurements under bone-conducted
excitation, coupled with a finite element analysis (FEA)
model developed in COMSOL Multiphysics. Both the model;
the artificial ear demonstrate good agreement regarding the
sound pressure found at the tympanic membrane for a
conventional dome at shallow, medium; deep insertions.
The validated FEA model is then used to perform a virtual
test of the bone-conducted objective OE for different
occluding devices, including plastic; foam earplugs;
a conventional closed dome for hearing aids. This is to
investigate the relative contributions; phases of the
ear-canal; device surfaces govern the resulting occluded
sound pressure. The proposed artificial ear; modeling
approach provide a controlled; repeatable platform for
studying the OE; for evaluating occluding devices during
the development process.

Accurate; efficient measurement of sound pressure levels
around the ears of occupants in cars is essential for
objective evaluation of basic sound quality; automotive
audio features such as personal sound zones; active
noise control. In this paper, the uncertainties of sound
pressure measurements obtained with 5 commonly used methods
are compared, which are the AES 6-microphone method, the
single-microphone method, the two-microphone method with
occupants presented, the head-and-torso simulator method,
; the human binaural method. Measurements were conducted
in the front-right seat of a 4-door electric Sedan, using
either all car body loudspeakers or a pair of headrest
loudspeakers driven by a two-channel uncorrelated pink
noise to generate an average sound pressure level of 70 dBA
in the seat. Each method underwent 3 complete
install–measure–remove cycles, a total of 54 recordings
were collected,; the standard deviation of the measured
average sound pressure levels was adopted to quantify
measurement uncertainty. The test results show that all the
5 methods have good repeatability; low uncertainty below
200 Hz; above 15 kHz, but have large uncertainty between
200 Hz; 15 kHz. The AES 6-microphone method demonstrates
the best repeatability with the lowest uncertainty across
most frequency resolutions,; its maximum uncertainty in
1/3 octave bands is less than 2.0 dB for sound pressure
measurements in the car. Therefore, the AES 6-microphone
method is recommended for use in engineering comparison;
reporting.

Linear loudspeaker parameters are often estimated via
fitting of transferfunctions, under the assumption of
linearity. This paper investigates the corruption of the
measurement caused by nonlinearities in the system;
presents a new; improved method for separating the true
linear response from the nonlinear components by analyzing
a sequence of measurements done at different levels. The
method is improved by analyzing the influence of the chosen
measurement levels as well as the measurement time at each
level; presents numerically optimal values for the most
typical cases of nonlinear behaviour. While the influence
of noise; nonlinear distortion can be eliminated
completely in the case of finite orders of nonlinearities
on the system, the method is also shown to provide improved
accuracy in the more realistic case where all orders are
present but only a finite number of them dominate.