AES Europe 2026

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superseded during the following decade, the electromagnetic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.

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.

Accurate characterization of the three-dimensional sound
radiation of outdoor public-address (PA) systems is
essential for sound system engineering, environmental noise
assessment, neighbourhood protection, and the validation of
prediction models. In current practice, field measurements
around performance stages are typically restricted to
receiver heights below 5 m, limiting insight into sound
radiation at elevated positions and towards the surrounding
environment. This tutorial presents a measurement approach
using an unmanned aerial vehicle (UAV) as a platform for
Class 1 sound level measurements, enabling in-situ
characterization of large-scale PA systems sound radiation
in three dimensions.
A controlled case study was conducted at an open-air
festival site in Belgium where the sound radiation of a
professional line-array PA system was measured at heights
of 2 m and 30 m using both conventional ground-based
measurements and a drone-mounted sound level meter. To
ensure compatibility with standard sound engineering and
environmental noise practice, strict Class 1 methodology
was applied, including the use of an omnidirectional
microphone, broadband excitation signals, and background
noise correction in accordance with ISO 1996-2. Drone
self-noise was quantified under operational conditions, and
measurement data not meeting signal-to-noise validity
criteria were excluded.
The results show that reliable drone-based measurements are
achievable in the low-frequency range from 25 to 315 Hz,
which is of primary relevance for outdoor music systems and
community noise impact and disturbance. Directivity indices
derived at elevated height reveal weaker low-frequency
directivity compared to ground-level measurements. This
provides new insight into vertical sound radiation
behaviour of festival PA systems. A comparison between
measured and modelled sound levels demonstrates good
agreement in terms of angular distribution and relative
level differences.
The proposed drone-based measurement approach enables
three-dimensional sound field characterization of outdoor
PA systems that is not attainable using conventional
techniques. The method provides valuable data for sound
system engineering leading to validation of prediction
models and environmental noise assessment. This
three-dimensional decibel measurement represents a step
towards standardized UAV-based measurement methodologies
for large-scale outdoor sound reinforcement systems.
This tutorial will describe in detail the protocol to
operate a measurement drone flight. After the presentation
a practical demonstration of the drone platform will be
held outside of the building.

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.

The ECHO Project (Exploring the Cinematic Hemisphere for
Orchestra) is a collaborative initiative investigating 3D
microphone array techniques for orchestral recording.
Building on the 3D-MARCo initiative, the project provides a
platform for sound engineers, composers, researchers, and
students to explore and experiment with immersive recording
approaches. As part of this effort, an open-access database
of high-quality orchestral recordings was created from
sessions at AIR Studios, London, featuring Oscar-winning
composer Volker Bertelmann and the London Contemporary
Orchestra.The ECHO database contains recordings of four
musical pieces captured using up to 143 microphone
capsules, including seven expert-designed microphone
arrays, spot microphones, a dummy head, and a higher-order
spherical microphone system. The database enables
comparison of different recording techniques and supports
experimentation with microphone mixing, making it a
valuable resource for research, teaching, and immersive
audio production. This workshop will introduce the
microphone arrays, describe the recording process and
immersive compositional approach, and showcase selected
recordings in 7.1.4.

Multichannel audio formats require an attention to
channels' correlations and sometimes special approach. In
this workshop, we would like to continue the discussion
started at AES Show 2025 in LA and show how you can use
different measurement tools to avoid certain problems in
the final mix. For example, the mutual influence between
the upper and main beds in immersive layout or problems in
the LFE channel and how to check the mix for the
correlation issues outside the sweet spot.

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.

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.

Virtual acoustics and active acoustic systems are
increasingly used in architectural acoustics to extend the
acoustic response of performance spaces. While these
technologies have traditionally been associated with
concert halls, theaters, and multipurpose venues, their
application has recently expanded to more controlled
environments such as recording studios and music production
spaces.

This tutorial introduces the fundamental principles of
virtual acoustics implemented through active acoustic
systems, starting from their role in architectural
acoustics and room acoustics enhancement. Basic concepts
such as room impulse responses, acoustical parameters,
system architectures, and feedback control strategies are
presented at an introductory level, with emphasis on common
practices and practical limitations. The discussion then
progressively narrows to the specific case of recording
studios, where virtual acoustics are used not only to
simulate performance spaces, but also to influence musical
performance, comfort, and interaction during recording
sessions, including the use of immersive microphone
techniques.

Through practical examples and listening demonstrations
developed at the Immersive Medial Laboratory in the
Department of Music Research of McGill University, the
tutorial illustrates how different virtual acoustic
conditions can be designed and applied in studio contexts,
highlighting their perceptual effects and implications for
musicians, recording engineers, and producers. The tutorial
aims to provide attendees with a clear conceptual framework
and practical insight into how virtual acoustics and active
sound reinforcement systems can be effectively employed
across architectural and studio applications, preparing the
audience for more advanced technical discussions on these
topics.

Obsidian Neural is a novel, open-source VST3 plugin that
addresses the technical challenges of integrating
generative AI models directly into a low-latency digital
audio workstation (DAW) environment. This workshop will
provide a deep dive into the architecture designed to use
AI as a real-time performance instrument. We will cover the
C++/DSP strategies necessary for minimizing latency during
the asynchronous generation of audio loops via models like
Stable Audio Open. Crucially, we will detail the system's
ability to maintain musical coherence during a live mix,
achieved through an internal LLM "Brain" that processes
contextual session data (BPM, key, existing tracks) to
enrich generation prompts. Furthermore, we will explore the
technical solutions implemented for seamless integration
with the live mixing paradigm: quantized MIDI triggering,
multi-output routing, and the novel "Draw-to-Sound"
feature, which employs a Vision Language Model (VLM) to
translate visual input into musical parameters. This work
demonstrates a robust framework for generative AI to
function as an instantaneous, adaptable partner within
professional audio engineering workflows.

Speech intelligibility is a key factor in successful
communication across various domains, including research,
post-production for film and television, live sound
reinforcement, and audio production. Traditional assessment
methods often lack objectivity or fail to capture the
listener’s experience in real-world scenarios. In this
workshop, we introduce an innovative approach to measuring
speech intelligibility based on the concept of “Listening
Effort.” We will present the underlying technology, share
practical examples from different application areas, and
demonstrate how this method can be integrated into
workflows to optimize intelligibility. Attendees will have
the opportunity to participate in a hands-on demonstration
and discuss potential use cases relevant to their own work.
This session is designed for professionals and researchers
seeking reliable and actionable tools for evaluating and
improving speech intelligibility in diverse environments.
In this workshop, we present a new technology for measuring
speech intelligibility (“Listening Effort”). The method is
used in research, post-production (film/TV), live sound,
and audio production. The session is aimed at professionals
from both academia and industry who are interested in
objectively assessing and optimizing speech intelligibility.

Participants will be able to join a short demo/exercise and
ask questions.

Introduction & Relevance: Overview of the importance of
speech intelligibility across different fields
Technology & Methodology: Presentation of the measurement
method and underlying concepts
Practical Examples: Case studies from research,
post-production (film/TV), live sound, and production
Live Demo / Interactive Exercise: Practical demonstration
and opportunity for active participation
Discussion & Outlook: Q&A, exchange of ideas, and future
perspectives

With the omnipresence of immersive audio the loudness
agenda has been pushed out of the spotlight. While there
are important areas (like TV) where the introduction has
been a resounding success with a complete paradigm shift,
others have not yet fully embraced the "auditory cease
fire" (Radio) or even searched for ways to counteract
loudness normalisation or still gain a loudness advantage
(pop music). In this workshop, two veterans of the EBU
loudness group PLOUD will elaborate on potential
meta-reasons for the resistance in the latter areas as well
as survey recent developments and challenges.

3D audio allows music to be experienced as an immersive
spatial environment. By distributing musical elements
around the listener, the perception of width, depth, and
presence is enhanced, allowing individual sounds to have a
greater emotional impact.

In this workshop, Lasse Nipkow presents a creative concept
in which recordings of acoustic instruments are used to
create a virtual sound space via 3D audio. The workshop
illustrates how spatial hearing, auditory fusion, and
perceptual grouping allow multiple tracks of a single
instrument to be perceived as a coherent sound. Different
instruments located at the same position in space merge
into a new, unified sound, while perceptual grouping
organizes similar sounds into a spatially coherent
connection. Lasse Nipkow clearly explains the
psychoacoustic principles and demonstrates how the ear
integrates distributed sound sources into a unified musical
experience.

Compelling listening examples are presented during the
workshop, both as complete 3D audio mixes and as groups and
individual tracks. This allows participants to understand
how the distribution of sounds in space, fusion, and
grouping work together in immersive music production.

This tutorial lecture is intended to present the basic
concepts of Physical Modelling (PM) sound synthesis. The
motivation behind the presentation is to increase the
awareness of the audio community of the existence of this
synthesis method, as an alternative to the much more
widespread subtractive synthesis approach.

Physical modelling synthesis allows for the creation of
realistic acoustic instrument timbres, as well as
avant-garde sound design constructs, in a way that is much
more approachable and intuitive than FM synthesis.

The lecture is intended for the beginner adept of sound
design, as well as the general audio public, so no prior
knowledge is required beyond a basic, intuitive
understanding of sound. Auditory examples will be presented
using a software modular synthesiser environment, as well
as dedicated physical modelling virtual instruments/effects
plugins within a Digital Audio Workstation application.

The presented concepts are applicable to the field of sound
design for music, film, gaming, and audiovisual arts, as
well as to teaching acoustics and musical instrument
studies in the classroom.

Topics covered:
-PM synthesis premise
-Comparison to subtractive synthesis
-Concept of a resonant model (Karplus–Strong oscillator)
-Simple and complex resonators, resonator coupling,
exciters, modifiers, pickups, and instrument body resonance
-Typical PM synthesiser architecture and workflow
-Signal processors/effects utilising physical modelling
(e.g. modal reverb)
-Dedicated virtual instruments for realistic acoustic and
electroacoustic instrument synthesis

All sound and workflow examples will be presented in a
technology-neutral way (using a variety of physical
modelling plugins from different vendors).

Most contemporary immersive audio production workflows are
centered on discrete channel-based loudspeaker formats such
as 7.1.4. These formats are rarely experienced by most
consumers and listeners, particularly in music playback. In
practice, spatial audio is predominantly delivered via
binaural reproduction. Beyond headphones, head-tracked
loudspeaker array systems now enable convincing binaural
reproduction in a practical, listener-centric manner,
unlocking spatial audio over loudspeakers for ordinary
listeners. This positions binaural reproduction not as a
secondary translation, but as the core delivery format for
immersive audio consumption.

Creating primarily for fixed speaker layouts can impose
creative and technical constraints often resulting in
restrained spatial design when content is later rendered
binaurally. This workshop advocates a binaural-centric
approach to spatial audio creation, treating binaural as
the main deliverable, while preserving compatibility with
discrete channel-based systems. Through discussion and
practical examples, we will explore how designing with
binaural in mind enables more expressive, perceptually
robust, and immersive experiences across both headphone and
loudspeaker-based binaural playback, without relying on
traditional 7.1.4-centric production models.

Binaural audio is fundamental to delivering immersive
spatial audio, however traditionally playback has been
limited to headphones. A combination of cross-talk
cancellation (CTC) technology and beamforming allows for
binaural audio playback over loudspeaker arrays by focusing
sound exactly at the listener’s ears. This enables binaural
audio to be reproduced using just compact loudspeaker
arrays in front of the listener.

The approach relies upon accurate knowledge of the
listener’s instantaneous head position, thus combining a
head-tracking system with position-adaptive loudspeaker
beamforming. This workshop explores CTC-based spatial audio
as a system: introducing the theory of CTC-based binaural
audio reproduction, and the practicality of introducing
head-tracking into such a system. The workshop will include
a live-demonstration of head-tracked binaural audio
delivery over loudspeakers using a soundbar.

Imagine that you just finished designing and are now
managing your dream immersive audio mix room for a client
with an array of 64 speakers and it functions beautifully -
then CoVid19 wreaks global havoc. You find yourself
suddenly isolated in a new country, forced into retirement
with its budgetary restrictions, and your dream studio has
become an early victim to the pandemic. What would be your
next move?

In this real-life story, follow the adventures of an
intrepid audio engineer and his quest to build a personal
version of that immersive studio that was lost – all within
a fixed-income retiree’s budget.

In this tutorial, an immersive studio design and
construction will be described including:

Inspiration from prior work by the author and colleagues
Room design goals
Equipment choices
Custom electronics design
Speaker design considerations
Speaker support and position alignment
Construction steps
VBAP, Ambisonics, and WFS approaches
Test mixes

Immersive mix examples will be demonstrated.