AES Europe 2026

Loudspeaker array beamforming technology has been widely
used; however, current frequency-domain; time-domain
design methods for calculating FIR filters face challenges,
including the need for modeling delay; high
computational complexity. To address these issues, this
paper proposes a time–frequency integrated framework. This
framework supports both pressure matching; amplitude
matching methods, enabling not only the realization of
traditional superdirective beams but also the design of
frequency-invariant beams. For the nonlinear optimization
problem in amplitude matching, an efficient solving
algorithm based on the Alternating Direction Method of
Multipliers (ADMM) is introduced. Experimental results
demonstrate that the proposed method combines the
advantages of existing frequency-domain; time-domain
approaches, directly computing FIR filter coefficients
without delay modeling while maintaining high computational
efficiency. This provides an effective solution for beam
control in loudspeaker arrays.

The inherent narrowing of directivity at high frequencies
in compact tweeters limits the spatial uniformity of sound
reproduction in modern audio systems. Conventional passive
solutions, such as waveguides; acoustic lenses,
partially mitigate this issue but typically rely on bulky
geometries; treat the diaphragm as a unitary radiator,
neglecting localized vibration behavior. This study
proposes a Matrix Wavefront Modulator (MWM), a compact
passive device that implements a differentiated
wavefront-shaping strategy based on vibration-aware
radiation control. Sound radiation from the piston-like
diaphragm dome; the breakup-prone surround is processed
independently by combining guided wavefront steering with
targeted scattering compensation. The geometry of the MWM
is optimized to adapt to the radiation characteristics of
the tweeter. Numerical simulations show that the optimized
MWM reshapes the high-frequency wavefront toward a more
spherical distribution; significantly reduces off-axis
attenuation above 10 kHz. Experimental measurements confirm
significant improvements in high-frequency directivity over
wide radiation angles.

Loudspeaker spider suspensions play a crucial role in
defining the compliance; stability of electrodynamic
transducers. Due to their woven structure impregnated with
thermosetting resins, spiders exhibit a nonlinear;
viscoelastic mechanical response, resulting in stiffness
dependence on displacement; excitation rate, as well as
energy dissipation during operation. However, viscoelastic
effects are often simplified during early loudspeaker
design stages.
This work presents a combined numerical–experimental study
aimed at characterizing the mechanical behaviour of
loudspeaker spiders; assessing its influence on
optimization choices during the pre-design phase. An
experimental campaign was conducted on spider samples with
fixed geometry; varying materials. Loading–unloading
cycle measurements were performed at different displacement
rates to capture nonlinear stiffness; hysteresis effects.
A finite element modelling framework was developed using a
2D axisymmetric formulation. Viscoelastic material
behaviour was first described through time-dependent
simulations, with model parameters identified by fitting
simulated loading–unloading curves to experimental data. A
parametric geometry optimization model based on linear
elastic assumptions was then implemented using quasi-static
simulations. Finally, the optimized spider geometries were
re-evaluated using time-dependent simulations incorporating
the identified viscoelastic material properties.
Results show that spider materials may influence its
mechanical behaviour, in particular the suspension
stiffness; hysteresis effects. Viscoelasticity mainly
affects the magnitude of the stiffness curve rather than
its overall shape, particularly at small displacements.
These findings support the use of quasi-static linear
elastic simulations for geometry optimization in early
loudspeaker design, while highlighting the importance of
material characterization for accurate performance
prediction.

Measuring the anechoic response of a loudspeaker system
requires space; facilities that are not commonly
available. The evolution of measurement instruments has
made it possible to visualize the time response of the
system under analysis, enabling the identification of
reflected signals; their elimination through time-gating
(windowing) of the impulse response. However, this comes at
the cost of a loss of resolution; characterization of
the system's response at lower frequencies. To correctly
characterize the system's response at the lowest
frequencies, the most widely used technique is the one
described by Keele in his AES paper "Low-Frequency
Loudspeaker Assessment by Nearfield Sound-Pressure
Measurement".
To obtain the overall system response, the appropriately
windowed far-field response; the near-field response are
combined, as described by Struck; Temme in their paper
"Simulated Free Field Measurements".
This operation is performed in the frequency domain, but
what happens when applied in the time domain?
The goal of this work is to use the near-field impulse
response to reconstruct the far-field portion of the
impulse response affected by environmental reflections. As
already stated, it’s quite easy to identify the first
reflection point on a far-field impulse response; this
can be used as a merging point to reconstruct the
reflections affected impulse tail using the corresponding
part of the near-field impulse measurement. Once the
far-field impulse tail is reconstructed, it is possible to
obtain the full-range frequency response of the system
under test while maintaining maximum measurement
resolution. The steps required to achieve a full-range
frequency response are fewer than those required for the
frequency-domain technique. For example, it is not
necessary to add the baffle diffraction step effect, as
demonstrated in the paper.

This paper investigates mid-to-high-frequency distortion in
traditional electrodynamic loudspeakers arising from
current-dependent nonlinearity in the magnetic circuit.
Through theoretical analysis, finite-element simulations
; experimental validation, the dominant distortion
mechanisms are identified. To mitigate distortion while
maintaining a stable frequency response, an improved
magnetic circuit is proposed, which introduces longitudinal
slits to suppress surface-concentrated eddy currents.
Experimental results demonstrate that the modified circuit
achieves greater distortion reduction compared with
conventional designs. As the improvement relies solely on
structural modifications without changing the ferromagnetic
materials, the proposed design offers a practical;
cost-effective solution for engineering applications.

The Panel-shaped Bending Wave Loudspeaker was proposed
recently by Kawahara. The authors conducted an objective
evaluation of the diffusion characteristics of Bending Wave
Loudspeakers (BWL) using the degree of interaural
cross-correlation (DICC) in this paper.
Conventional speakers exhibit strong directionality;
rely on room reflections to create a spatial impression. In
contrast, BWLs are considered less susceptible to room
reflections due to complex mode vibrations across the
entire diaphragm.
To quantify this characteristic, the authors recorded sound
in a real-world environment using a head-and-torso
simulator (HATS); compared the BWL's DICC with that of a
conventional speaker.
The results showed that the BWL exhibited significantly
lower DICC values than conventional loudspeaker at the
front position (Center) under both broadband noise;
music conditions, confirming its high diffusivity.
Furthermore, this difference exceeded the Just Noticeable
Difference (JND) for spatial perception, suggesting it is
also significant to the human ear. In addition, analysis
separating early reflections; late reflections suggested
differences in diffusion characteristics between
conventional speakers; BWL.

The Zylia ZM-1 (19 MEMS capsules, spherical array, 88 mm
diameter, 3rd-order); Harpex Spcmic (84 MEMS capsules,
planar array, 230 mm diameter, 5th-order capable) represent
two distinct geometrical approaches to higher-order
Ambisonics capture. Despite widespread adoption in research
; production, systematic comparison of their performance
in real-world recordings remains absent from published
literature. This case study presents a controlled
comparison through simultaneous recordings of piano
recitals in the same concert hall.

Two arrays—Zylia ZM-1; Harpex Spcmic—were mounted on a
single stereo bar (17 cm apart) ensuring acoustically
identical capture positions. Recording sessions occurred in
Aula Politechniki Gdańskiej (370-seat hall, RT60 = 1.97 s)
on two dates: August 15, 2024 (Franck: Prélude, Choral et
Fugue; Prokofiev: Piano Sonata No. 4, 35.6 minutes
total); April 30, 2024 (Ginastera: Sonata No. 1, Op. 22,
15.4 minutes). Both arrays recorded simultaneously; files
were processed through manufacturer A-to-B conversion
software; peak-normalized to −0.5 dBTP. The Spcmic was
encoded to both native 5th-order; truncated 3rd-order
formats for direct comparison with the ZM-1.

Four metrics were analyzed: (1) W-channel spectral
response, (2) integrated loudness (LUFS-I per ITU-R
BS.1770-5), (3) spatial energy distribution across
Ambisonics orders,; (4) first-order directional
component ratios.

Spectral analysis reveals the ZM-1 exhibits 5–8 dB
elevation at 200–600 Hz relative to the Spcmic. Loudness
measurements show the Spcmic 3rd-order yields 2.3–3.3 dB
higher LUFS-I than the ZM-1 despite identical peak
normalization.

The primary finding concerns spatial energy: the ZM-1
exhibits 27.4 dB attenuation from 0th to 3rd order, while
the Spcmic shows only 8.4 dB—a 19 dB difference despite
both producing "3rd-order Ambisonics" format. Analysis of
both recording sessions confirms consistency across
different repertoire (romantic, 20th-century,
contemporary). Directional analysis shows the Spcmic
exhibits stronger first-order components (X/Y/Z ratios
0.68–0.83) versus the ZM-1 (0.42–0.55).

Results demonstrate that nominal Ambisonics order
inadequately characterizes spatial resolution in real
recordings. The substantial higher-order energy deficit in
compact spherical arrays has implications for reproduction
quality, decoder design,; archival standards. Arrays
with steeper rolloff may require order-dependent gain
compensation to match spatial impression of larger systems.

This case study complements existing anechoic validation by
demonstrating performance differences in authentic
recording conditions. Recordings are part of a publicly
available HOA corpus (Gdańsk University of Technology
repository).

String ageing is a familiar; perceptually important
phenomenon for guitarists; players of other stringed
instruments. From the moment a new set of strings is
installed, the sound they produce when excited begins to
change due to a combination of chemical degradation,
corrosion,; mechanical wear arising from playing.
Musicians commonly report that aged strings sound dull,
lack sustain,; feel less responsive compared to new
strings. String ageing is a function of both elapsed time
; accumulated playing time, with repeated playing
accelerating degradation through contamination; repeated
mechanical stress.

Previous studies have investigated individual aspects of
string ageing by artificially accelerating wear;
performing controlled acoustic measurements, identifying
effects such as increased damping of higher partials;
increased inharmonicity. While these approaches provide
valuable physical insight, the tightly constrained
experimental conditions differ significantly from
real-world playing conditions.

This paper presents a dataset of audio recordings of guitar
playing over a four-week period, starting from the point of
new strings being installed.
Audio performance data from different sets of electric
guitar strings is recorded daily over a four-week period,
using strictly fixed musical exercises that are repeated
multiple times per session. By collecting many takes of
identical material at each stage of string age, the dataset
enables statistical analysis of ageing-related changes
while accounting for natural performance variability.

The dataset is intended to support exploratory machine
learning investigations into string ageing, including
questions of how ageing manifests over time; playing
duration, whether string age can be predicted from audio
alone,; which audio features or learned representations
capture perceptually relevant aspects of the ageing process.

Tape recording of audio programme produces significant
noise signals underlying the audio signal. Measurements
show that total modulation noise is significant; often
around 25 dB down from a sinusoidal audio signal, although
historical measurement methods give numbers that may exceed
50 dB. The persistent popularity of tape in the audio
industry may indicate a preference for some of the more
salient tape characteristics; perhaps even modulation
noise. Measurements on a variety of tapes; machines are
presented in an attempt to understand the basic principles.
A model of modulation noise is developed which provides a
broad steepening spectral peak centred on the signal
frequency; captures much of the tape noise character.
This could be the basis of a plug-in to simulate such
noise. A new measurement method is presented culminating
in a single plot which gives a useful more complete picture
of modulation noise.