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.

The proposed workshop/tutorial serves as a prequel to the
presentation on the history of dynamic loudspeakers given
at the 158th Convention (Warsaw, 2025). It focuses on the
earliest phase of consumer loudspeaker technology in the
1920s, prior to the widespread adoption of dynamic
loudspeakers in the mass market.

Loudspeakers had been in use since the mid-1910s for public
address applications, and the rapid global expansion of
broadcast radio soon brought loudspeakers into domestic
use. The 1920s constituted a period of rapid innovation in
loudspeaker design, preceding the introduction of the
dynamic loudspeaker, which achieved significant commercial
impact only in the latter part of the decade.

The workshop/tutorial will examine consumer loudspeaker
technologies of the 1920s, the concurrent advancements in
audio electronics and signal sources that enabled
subsequent developments, and the earliest efforts in
systematic loudspeaker theory and measurement.

Two loudspeaker types dominated this period: horn
loudspeakers driven by electromagnetic drivers similar to
those used in headphones and telephone receivers (with
headphones, particularly Baldwin models, also serving as
the basis for do-it-yourself loudspeakers), and open-baffle
cone loudspeakers, frequently actuated by electromagnetic
reed drivers.

Although these transducer technologies were rapidly
superseded during the following decade, the electromagnetic
loudspeaker era already featured multi-way loudspeakers
employing passive crossovers. Early measurements exposed
deficiencies in frequency response, leading to the
introduction of equalisation techniques, including notch
filters, to correct these responses.

Developments in amplification were equally significant. The
1920s saw the introduction of push-pull amplifiers
(described at the time as “distortionless”) and, as a key
contributor to improved bandwidth and reduced distortion,
new audio transformers derived from Bell Labs’ telephone
research. Amplifier power limitations nevertheless remained
a dominant constraint in loudspeaker design, resulting in
the widespread use of strong resonances to achieve high
sensitivity. Improvements in signal source quality from the
mid-1920s onwards — including advances in radio
transmission and the introduction of electrical disc
recording and playback — further increased the demand for
improved loudspeaker performance, ultimately contributing
to the development of dynamic loudspeakers. In contrast,
headphone technology appears to have undergone relatively
little development during this period.

The tutorial will conclude with a brief overview of the
loudspeaker manufacturing landscape of the era, noting that
only a small proportion of manufacturers survived the
transition to dynamic loudspeaker technology.

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.

There are many types of different distortions that can be
measured from linear to non-linear distortion. Often the
two are convoluted together and the linear distortion
influences the non-linear distortion. Distortion is also
very signal and level dependent and it is hard to compare
one type of distortion measurement to another. There are
many type of non-linear distortion metrics, e.g. THD, THD+N
and IMD being the most classic ones using sine tones as the
test signal. But how can we measure distortion with real
signals such as speech and music or even noise and compare
the results to audibility? This tutorial discusses a wide
range of distortion measurements, discusses what is audible
and what distortion sounds like.

Before digital signal processing took over electronic
keyboard instruments, they were implemented using analogue
circuits that used tubes/valves, transistors, and even neon
lightbulbs! Yet using these components keyboards were
developed that could mimic string and brass ensembles,
pianos and harpsichords and many other instruments. How did
they do it?

The purpose of this tutorial is to look at both the
architecture and the circuitry of these instruments. And
show how amazing results could be achieved using
comparatively simple electronic circuitry. It will look at:

1. The basic architecture of these instruments
2. How they generated the right notes,
3. How they desired envelope,
4. And imposed them on the waveform,
5. Simulated the effect of many instruments playing
together.

It will also look at how, if it was required, touch
sensitivity could be achieved, such as in electronic
pianos. Where possible there will be audio examples
demonstrating the sounds that could be achieved.

For many people who have only ever experienced the digital
world it will be illuminating to see just how much could be
achieved by comparatively simple circuits.
In those days electronic components were expensive so
considerable ingenuity was expended in minimising the total
number of components required.

These instruments are part of our musical and audio
heritage and the circuit techniques they used are in danger
of being forgotten so this tutorial will be a timely
reminder of what used to be done.
It may also provide useful information to people who are
attempting to model these instruments using modern digital
methods.

The tutorial will be accessible to everyone, you will not
have to be an electronic engineer to understand the
principles behind these unique pieces of audio engineering
history.

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.

Headphones have become the dominant device for music
playback, and their design appears to have reached a
certain level of technical maturity. This workshop presents
an overview of the current state of the art in headphone
design and examines potential directions for future
technological development, addressing both acoustic
aspects—including transducer design—and signal-processing
approaches.

The workshop establishes a common foundation by introducing
the fundamentals of headphone acoustics and design
principles, together with a brief overview of the
historical development of headphones and the main headphone
types in use today.

Based on this foundation, the workshop addresses current
challenges and future development potential in headphone
technology, including:
• Transducer and acoustic development potential: materials,
design methodologies and simulation techniques, and
advances in measurement technology
• Characteristics of a high-quality headphone: What
differentiates an excellent headphone from a good one? To
what extent can headphone performance be characterized
using current measurement techniques, and what additional
metrics, target criteria, or perceptual considerations may
be required? What is the role of mechanical quality?
• Signal processing potential: from advanced noise
cancellation and augmented hearing to spatial audio
processing
• Challenges in realistic spatial reproduction: interaction
between auditory and visual environments
• Emerging wireless technologies: technologies such as UWB
and Bluetooth 6 offer not only increased bandwidth and
reduced latency but also the capability to localize the
playback device. What are the implications for conventional
headphone performance and for spatial audio applications?
• Changes in studio workflows: professional practice has
evolved from loudspeakers as the primary monitoring tools,
with headphones mainly used for detailed analysis, toward
headphones playing a central role in the early stages of
recording and mixing. What are the consequences of this
shift for headphone design and signal processing?
• Technically feasible but not yet commercialized
solutions: advanced headphone concepts that are achievable
with current technology but have not yet been adopted due
to economic or practical constraints

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.

The demand for wireless audio expands constantly, while the
available RF spectrum over recent decades has shrunk and
become more crowded. This session will explore strategies
for making wireless audio work cleanly and reliably,
essential information for live production, as well as TV
and film production.

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.