AES Europe 2026

Loudspeaker monitoring is the reference when audio
professionals evaluate content. Headphones are also
important quality-checking tools; and many consumers enjoy
music using “close-fitting listening devices”, as all
different flavours of headphones are known in recent
standards writing.

We discuss the two reproduction methods from perceptual,
recording and mastering perspectives; especially
differences in timbre, imaging and auditory envelopment
when listening to stereo. Applications of headphones in
recording, when setting up and trimming stereo or 3D
microphone arrays, are also practically detailed.

In the last part of the workshop, attendees are invited to
personally compare the two domains on the qualities and
applications discussed; with guided listening to audio
examples between a pair of precision nearfield monitors,
Genelec 8351B, and a pair of excellent headphones, Audeze
CRBN2.

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.

The Saul Walker Student Design Competition is a long-running event of the Audio Engineering Society that highlights practical and creative work in audio design. It brings together experienced judges and a wide range of strong student submissions each year.

During this session, students from around the world will present their projects and bring their hardware designs for hands-on inspection by the judges. The format encourages open discussion, giving attendees a chance to hear how ideas are evaluated and improved in a professional setting.

Sponsored by API, the competition includes cash prizes for the winners. More importantly, it offers students valuable feedback and the opportunity to connect with people working in the industry. The session is open to everyone—students and non-students alike—who are interested in seeing what participants have created and learning more about current work in audio design.

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.

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.

Join us for a panel discussion about audio design featuring some of the industry’s leading audio designers and educators. This session is meant to inspire upcoming designers and encourage dialogue with established audio designers.
 
The panelists will give a brief overview of their designs, their roles in the AES, and how and why educators and students should participate in the various design competitions that the AES has to offer. The panel discussion is followed by a Q&A session that allows for questions and exchange with the panelists.

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.