AES Europe 2025

Sound is one of the most powerful tools for accessibility in video games, enabling players with visual impairments or cognitive disabilities to navigate, interact, and fully engage with the game world. This panel will explore how sound engineers can leverage audio design to enhance accessibility, making games more inclusive without compromising artistic intent. Experts from different areas of game development will discuss practical approaches, tools, and case studies that showcase how audio can bridge gaps in accessibility.

Discussion Topics:

• Why is sound crucial for accessibility in video games? Audio cues, spatial sound, and adaptive music can replace or complement visual elements, guiding players with disabilities through complex environments and interactions.
• Designing effective spatial audio for navigation and interaction. Using 3D audio and binaural rendering to provide players with intuitive sound-based navigation, enhancing orientation and gameplay flow for blind or visually impaired users.
• Audio feedback and sonification as key accessibility tools. Implementing detailed auditory feedback for in-game actions, menu navigation, and contextual cues to improve usability and player experience.
• Case studies of games with exemplary accessible audio design. Examining how games like The Last of Us Part II, BROK: The InvestiGator, and other titles have successfully integrated sound-based accessibility features.
• Tools and middleware solutions for accessible sound design (example: InclusivityForge). Showcasing how game engines and plugins such as InclusivityForge can streamline the implementation of accessibility-focused audio solutions.
• Challenges in designing accessible game audio and overcoming them. Addressing common technical and creative challenges when designing inclusive audio experiences, including balancing accessibility with immersive design.
• Future trends in accessibility-driven audio design. Exploring how AI, procedural sound, and new hardware technologies can push the boundaries of accessibility in interactive audio environments.

Panel Guests:

• Dr Joanna Pigulak - accessibility expert in games, researcher specializing in game audio accessibility, assistant professor at the Institute of Film, Media, and Audiovisual Arts at UAM.
• Tomasz Tworek - accessibility consultant, blind gamer, and audio design collaborator specializing in improving audio cues and sonification in video games.
• Dr Tomasz Żernicki - sound engineer, creator of accessibility-focused audio technologies for games, and founder of InclusivityForge.

Target Audience:

• Sound engineers and game audio designers looking to implement accessibility features in their projects.
• Game developers interested in leveraging audio as a tool for accessibility.
• UX designers and researchers focusing on sound-based interaction in gaming.
• Middleware and tool developers aiming to create better solutions for accessible audio design.
• Industry professionals seeking to align with accessibility regulations and best practices.

This panel discussion will explore how sound engineers can enhance game accessibility through innovative audio solutions, providing insights into the latest tools, design techniques, and industry best practices.

Binaural audio is fundamental to delivering immersive spatial sound, but traditional playback has been limited to headphones. Crosstalk Cancellation (CTC) technology overcomes this limitation by enabling accurate binaural reproduction over loudspeakers, allowing for a more natural listening experience. Using a compact loudspeaker array positioned in front of the listener, CTC systems apply beamforming techniques to direct sound precisely to each ear. Combined with listener tracking, this ensures consistent and accurate binaural playback, even as the listener moves. This workshop will provide an in-depth look at the principles behind CTC technology, the role of loudspeaker array beamforming, and a live demonstration of a listener-tracked CTC soundbar.

Immersive audio has become a significant trend in music recording, reproduction, and the audio and entertainment industries. This workshop will explore microphone techniques for immersive sound recording from theoretical, aesthetic, and musical perspectives.

Capturing a music performance and its acoustic features in a specific reverberant field, such as a concert hall, requires specialized microphone techniques for immersive sound. Various microphone techniques have already been proposed for immersive music recording. Achieving a natural timbre, appropriate musical balance, wide frequency range, low distortion, and high signal-to-noise ratio are essential in music recordings for capturing the music performance, including immersive sound recording. The acoustic features of the musical performances can be naturally reproduced by appropriately capturing direct and indirect sounds in the sound field.

The first topic of this workshop will cluster and review microphone techniques based on their fundamental roles. The panelists will also introduce their immersive sound music recording concept, demonstrate their microphone techniques, and provide sound demos.

Immersive audio can expand the adequate listening area if the microphone technique is designed with this goal. This is crucial for popularizing immersive sound reproduction among music lovers. Therefore, the second topic of this workshop will discuss microphone techniques from the perspective of the listening area during reproduction. The panelist will explain his hypothesis that lower correlation values in the vertical direction contribute to the expansion of the listening area.

In immersive sound recording, various microphone techniques have been proposed to reproduce the top layer of the multichannel discrete loudspeaker layout. It is recommended to use directional microphones and position the top and middle layer microphones simultaneously to avoid phase differences that can degrade timbre. However, some reports suggest that separating the top and middle layers can enhance the perception of vertical spaciousness. Experiments conducted by the panelists also suggest that separating these layers and lowering the correlation between them can widen the listening area without altering the central listening position's impression. Comparing microphone types and installation positions in the upper layer is challenging in actual recording situations. Therefore, the panelists will compare listening impressions under various conditions and allow participants to experience these differences using virtual recording techniques (V2MA), which will be discussed as the third topic of this workshop.

Several papers have reviewed microphone techniques, but most have relied on subjective evaluation. The third topic of this workshop will attempt to evaluate microphone techniques from a physical viewpoint. The panel will introduce the Virtual Microphone Array technique (V2MA) to determine how each microphone captures a room's reflection sounds and identify the acoustical features of several microphone arrays used for immersive sound recording. V2MA generates Spatial Room Impulse Responses (SRIR) using a virtual microphone placed in a virtual room with spatial properties of dominant reflections previously sampled in an actual room.

Lectures and demos help us understand the acoustical features and intentions behind microphone techniques, but they are insufficient to grasp their spatial characteristics, especially for immersive sound recording. The panelists will provide 7.0.4ch demos to showcase the spatial features of microphone techniques using V2MA. V2MA generates the acoustic response of a microphone placed virtually in a room, calculated from spatial information of virtual sound sources, such as dominant reflections detected from sound intensities measured in the target room. This workshop will illustrate the spatial characteristics of microphone arrays, allowing us to discuss the types of reflections captured by microphones and discover the differences in spatial features between microphone techniques.

Following each panelist's presentation, a panel discussion will delve into microphone techniques from theoretical, aesthetic, and musical viewpoints. This workshop aims to review issues with microphone techniques for immersive sound and discuss potential solutions to achieve natural spatial reproduction of musical performances for home entertainment.

Artificial intelligence (AI) tools are transforming the way music is being produced. The rate of development is expeditious, and the associated metamorphosis of audio education is abrupt. Higher-level education is largely built around the objectives of knowledge transmission and skills development, evidenced by the emphasis on learning in the cognitive domain in University programmes. But the cohort of skills that music producers will require in five years’ time is unclear, making skills-based curriculum planning challenging. Audio educators require a systematic approach to integrate AI tools in ways that enhance teaching and learning.

This study uses speculative design as the underpinning research methodology. Speculative design employs design to explore and evaluate possible futures, alternative realities, and sociotechnical trends. In this study, the practical tasks in an existing university module are modified by integrating available GAI tools to replace or augment the task design. This tangible artefact is used to critique prevailing assumptions concerning the use of GAI in music production and audio education. The findings suggest that GAI tools will disrupt the existing audio education paradigm. Employing a process-centred approach to teaching and learning may represent a key progression for educators to help navigate these changes.

In automotive audio playback systems, dynamically increasing driving sounds are typically taken into account by applying a generic, i.e., non-individualized, increase in overall level and low-frequency amplification to compensate increased masking. This study investigated the degree of individuality regarding the preferences of noise-dependent level and equalizer settings. A user study with 18 normal-hearing participants was conducted in which individually preferred level-dependent and frequency-dependent amplification parameters were determined using a music-based procedure in quiet and in nine different driving noise conditions. The comparison of self-adjusted parameters suggested that, on average, participants adjusted higher overall levels and more low-frequency amplification in noise than in quiet. However, preferred self-adjusted levels differedmarkedly between participants for the same listening conditions but were quite similar in a re-test session for each participant, indicating that individual preferences were stable and could be reproducibly measured with the employed personalization scheme. Furthermore, the impact of driving noise on individually preferred settings revealed strong interindividual differences, indicating that listeners can differ widely with respect to their individual optimum of how equalizer and level settings should be dynamically adapted to changes in driving conditions.

Cyclical formal reviews are essential to keep Music and Audio Technology degree programmes current. Whilst clear institutional guidance exists on the requisite documentation to be submitted, there is little guidance concerning the process used to gather the information. To address this issue, a 12 step collaborative and reflective framework was developed to review a degree programme in Music Technology.

This framework employs Walker’s ‘Naturalistic’ process model and design thinking principles to create a dynamic, stakeholder-driven review process. The framework begins with reflective analysis by faculty, helping to define program identity, teaching philosophy, and graduate attributes. Existing curricula are evaluated using Boehm et al.’s (2018) tetrad framework of Music Technology encompassing the sub-disciplines of production, technology, art, and science. Insights from industry professionals, learners, and graduates are gathered through semi-structured interviews, surveys, and focus groups to address skill gaps, learner preferences, and emerging trends. A SWOT analysis further refines the scope and limitations of the redesign process, which culminates in iterative stakeholder consultations to finalise the program’s structure, content, and delivery.

This process-centred approach emphasises adaptability, inclusivity, and relevance, thus ensuring the redesigned program is learner-centred and aligned with future professional and educational demands. By combining reflective practice and collaborative engagement, the framework offers a comprehensive, replicable model for educators redesigning degree programmes in the discipline. This case study contributes to the broader discourse on curriculum design in music and audio degree programmes, demonstrating how interdisciplinary and stakeholder-driven approaches can balance administrative requirements with pedagogical innovation.

In this contribution we present subjective tests of loudspeaker virtualization, a technique enabling the application of specific target behaviors to the physical loudspeaker system. In this work, loudspeaker virtualization is applied to virtualize a closed box car audio subwoofer to replicate the performance of a larger vented enclosure. The tests are designed to determine if any reduction in sound quality is detected by a panel of listeners when a virtualized loudspeaker is used.

Acoustic Sovereignties (2024) is a First Nations, anti-colonial spatial audio exhibition held in Naarm (Melbourne), Australia. Through curatorial and compositional practices, Acoustic Sovereignties confronts traditional soundscape and Western experimental sound disciplines by foregrounding marginalised voices.
As this research will show, the foundations of sound-based practices such as Deep Listening and Soundscape Studies consisted of romanticised notions of Indigenous spirituality, in addition to the intentional disregard for First Nations stewardship and kinship with the land and its acoustic composition. Acoustic Sovereignties aims at reclaiming Indigenous representation throughout sound-based disciplines and arts practices by providing a platform for voices, soundscapes and knowledge to be heard.

Car audio systems aim to provide information, entertainment, and acoustic comfort to drivers and passengers in cars. In addition to basic audio functions for broadcasting, playing chimes, warning sound, and music, there are special audio features such as vehicle noise compensation, spatial sound effects, individual sound zone, and active noise control. In this paper, commonly used objective measurement methods for basic sound quality and special features in cars are reviewed and discussed. All objective measurements are proposed to use the 6-unit microphone array specified in the White Paper for In-car Acoustic Measurements released by AES Technical Committee on Automotive Audio in 2023, and the main parameters to be measured are frequency responses and sound pressure levels in the car when the specially designed test signals are played back. General measurement frameworks and procedures for basic sound quality and each feature are presented. The advantages and weakness of using these parameters to characterize the basic sound quality and special features of a car audio system are discussed, and the challenges and future directions are explored.

If you are considering establishing a room for sound, i.e., recording, mixing, editing, listening, or even a room for live music, this is the crash course to attend!
Initially, we’ll walk through the essential considerations for any design of an acoustic space, (almost) no matter the purpose: Appropriate reverberation time, appropriate sound distribution, low background noise, no echoes/flutter echoes, appropriate control of early reflections, (and for stereo/surround/immersive: a degree of room symmetry).
To prevent misunderstandings, we must define the difference between room acoustics and building acoustics. This is a tutorial on room acoustics! Finding the right reverberation time for a project depends on the room's purpose. We’ll look into some relevant standards to find an appropriate target value and pay attention to the importance of the room's frequency balance, especially at low frequencies! We will take the starting point for calculation using Sabine’s equation and discuss the conditions to make it work.
The room's shape, the shape’s effect on room modes, and the distribution of the modes are mentioned (together with the term Schroeder Frequency). The acoustical properties of some conventional building materials and the consequences of choosing one in favor of another for the basic design are discussed. The membrane absorbers (plasterboard, plywood, gypsum board) and their importance in proper room design are presented here. This also involves the definition of absorption coefficients (and how to get them).
From the “raw” room and its properties, we move on to define the acoustic treatment to reach the target value. Again, the treatment often can be cheaper building materials. However, a lot of expensive specialized materials are also available. We’ll try to find a way through the jungle, keeping an eye on the spending. The tools typically are porous absorbers for the smaller rooms. Sometimes, resonance absorbers are used for larger rooms. We don’t want overkill of the high frequencies!
The placement of the sound sources in the room influences the perceived sound. A few basic rules are given. Elements to control the sound field are discussed: Absorption vs. diffusion. Some more uncomplicated principles for DYI diffusers are shown.
During the presentation, various practical solutions are presented. At the end of the tutorial, there will be some time for a minor Q&A.

Extensive studies have been made into achieving generally enjoyable sound colour in headphone listening, but few publications have been written focusing on the demanding requirements of a single audio professional, and what they actually hear.

However, headphones provide fundamentally different listening conditions, compared to our professional, in-room monitoring standards. With headphones, there is even no direct connection between measured frequency response and what a given user hears.

Media professionals from a variety of fields need awareness of such differences, and to take them into account in content production and quality control.

The paper details a recently published method and systematic steps to get to know yourself as a headphone listener. It also summarises new studies of basic listening requirements in headphone monitoring; and it explains why, even if the consumer is listening on headphones, in-room monitoring is generally the better and more relevant common denominator to base production on. The following topics and dimensions are compared across in-room and headphone monitoring: Audio format, listening level, frequency response, auditory envelopment, localisation, speech intelligibility and low frequency sensation.

New, universal headphone monitoring standards are required, before such devices may be used with a reliability and a confidence comparable to in-room monitoring adhering to, for example, ITU-R BS.1116, BS.775 and BS.2051.

Experiments testing sound for augmented reality can involve real and virtual sound sources. Paradigms are either based on rating various acoustic attributes or testing whether a virtual sound source is believed to be real (i.e., evokes an auditory illusion). This study compares four experimental designs indicating such illusions. The first is an ABX task suitable for evaluation under the authenticity paradigm. The second is a Yes/No task, as proposed to evaluate plausibility. The third is a three-alternative-forced-choice (3AFC) task using different source signals for real and virtual, proposed to evaluate transfer-plausibility. Finally, a 2AFC task was tested. The renderings compared in the tests encompassed mismatches between real and virtual room acoustics. Results confirm that authenticity is hard to achieve under nonideal conditions, and ceiling effects occur because differences are always detected. Thus, the other paradigms are better suited for evaluating practical augmented reality audio systems. Detection analysis further shows that the 3AFC transfer-plausibility test is more sensitive than the 2AFC task. Moreover, participants are more sensitive to differences between real and virtual sources in the Yes/No task than theory predicts. This contribution aims to aid in selecting experimental paradigms in future experiments regarding perceptual and technical requirements for sound in augmented reality.

In recent years, professional and consumer audio and music technology has advanced in several areas, including sensory immersion, electronic transmission, content formats, and creation tools. The production and consumption of immersive media experiences increasingly rely on a global network of interconnected frameworks. These experiences, once confined to separate content markets like music, movies, video games, and virtual reality, are now becoming interoperable, ubiquitous, and adaptable to individual preferences, conditions, and languages. This article explores this evolution, focusing on flexible immersive audio creation and reproduction. We examine the development of object-based immersive audio technology and its role in unifying broadcast content with embodied experiences. We introduce the concept of Acoustic Objects, proposing a universal spatial audio scene representation model for creating and distributing versatile, navigable sound in music, multimedia, and virtual or extended reality applications.

Since we've moved from stereo to surround and 3D/immersive productions, many immersive music mixes still sound very much like larger stereo versions. Part of the reason for this is the record company's demands and the argument, that people don't have properly set up systems at home or only listen with headphones. But that's not the way to experience the real adventure, which is to create new, stunning sound and musical experiences. The workshop will not criticize mixes, but try to open the door to the new dimension of music and discuss the pros and cons that producers have to deal with today.

Immersive audio mix presentations involve transmitting and rendering several audio elements simultaneously. This enables next-generation applications, such as personalized playback. Using immersive loudspeaker and headphone MUSHRA tests, we investigate rate vs. quality for a typical mix presentation use case of a foreground stereo element, plus a background Ambisonics scene. For coding, we use Immersive Audio Model and Formats, a recently proposed system for Next-Generation Audio. Excellent quality is achieved at 384 kbit/s, even with reasonable amount of personalization. We also propose a framework for content-aware analysis that can significantly reduce the bitrate even when using underlying legacy audio coding instances.

This study presents an ethnographic analysis of current immersive music production workflows, examining industry trends, tools, and methodologies. Through interviews and participant observations with professionals across various sectors, the research identifies common patterns, effective strategies, and persistent obstacles in immersive audio production. Key findings highlight the ongoing struggle for standardized workflows, the financial and technological barriers faced by independent artists, and the critical role of collaboration between engineers and creatives. Despite the growing adoption of immersive formats, workflows still follow stereo conventions, treating spatialization as an afterthought and complicating the translation of mixes across playback systems. Additionally, the study explores the evolving influence of object-based and bed-based mixing techniques, monitoring inconsistencies across playback systems, and the need for improved accessibility to immersive production education. By synthesizing qualitative insights, this paper contributes to the broader discourse on immersive music production, offering recommendations for future research and industry-wide best practices to ensure the sustainable integration of spatial audio technologies.

A psychoacoustic experiment is conducted to evaluate and compared the auditory distance perception in reflected sound field by using static and dynamic VAD. The binaural signals creased by a point source at different distances in a rectangular room are simulated. The contribution of direct sound to binaural signals is simulated by near-field head-related transfer function filters and a gain factor to account for the propagation attenuation of spherical surface wave. The contribution of early reflections up to the second order and later reverberation are respectively simulated by the image source method and Schroeder reverberation algorithm. The results of psychoacoustic experiment indicates that there are still significant differences between the perceived distances created by static VAD and these created by dynamic VAD in the simulated reflected condition, although the differences are not so large as those in the simulated free-field case. The results of dynamic VAD are more consistent with these of real sound source. Therefore, simulating reflections reduces the in-head-localization and thus improves the control of perceived distance in headphone presentation, but static VAD is still less effective in creating different distance perception. Dynamic VAD is still needed in the distance perception experiment for hearing researches even if simulated reflections are included. In practical applications, dynamic VAD is advocated for recreating virtual source at different distance.

With the rapid development of virtual reality (VR) and augmented reality (AR), spatial audio recording and reproduction have gained increasing research interest. Higher Order Ambisonics (HOA) stands out for its adaptability to various playback devices and its ability to integrate head orientation. However, current HOA recordings often rely on bulky spherical microphone arrays (SMA), and portable devices like smartphones are limited by array configuration and number of microphones. We propose a method for HOA encoding using a smartphone microphone array (SPMA). By designing beamformers for each order of spherical harmonic functions based on the array manifold, the method enables HOA encoding and up-scaling. Validation on a real SPMA and its simulated free-field counterpart in noisy and reverberant conditions showed that the method successfully encodes and up-scales HOA up to the fourth order with just four irregularly arranged microphones.

Through a practice-oriented study, various coincident, near-coincident, and non-coincident immersive microphone arrays were compared during drum recordings for different contemporary popular music genres. In a preliminary study, the OCT-3D, PCMA-3D, 2L-Cube, Hamasaki Square, IRT Cross, Ambisonics A-Format, and native B-Format were informally compared, revealing that the differences between non-coincident systems were much smaller than the differences between coincident and non-coincident systems. This led to a reduction in microphone systems for the final drum recordings. Four microphone techniques were selected: OCT-3D, native B-Format, Ambisonics A-Format, and IRT Cross. These were compared within the context of two different songs – a calm pop track and an energetic rock song – where the drums were respectively recorded in a dry drum booth and a large studio hall. Through a listening test with a small sample group, it was determined which microphone technique was best suited for each song. Participants were also asked to identify the general favorite, without musical context, as well as how the spatiality, timbre, and height were perceived. It was concluded that the choice of immersive microphone technique depends on the musical context. Conclusions from more objective studies focus primarily on accurate localization, with non-coincident systems consistently performing the best. However, these studies do not take into account the musical context, where accurate localization does not always take precedence. Furthermore, it was noted that height perception in music is not solely created by speakers in the height range. The comparative drum recordings are published through https://www.immersive.pxl.be/immersive- microphone-techniques-for-drums/.

The evolution of 3D audio has significantly influenced the music and film industries, yet its full potential remains untapped. This panel will explore how immersive audio technologies, including Ambisonics, Dolby Atmos, and volumetric sound, shape new frontiers beyond traditional applications. We will focus on three key areas: accessibility in video games, the integration of 3D audio in gaming experiences, and its growing role in the automotive industry. Our panelists will discuss the state of the market, technological limitations, and emerging opportunities where spatial audio enhances user experience, safety, and engagement. This discussion aims to inspire innovation and collaboration among researchers, developers, and industry professionals.

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Room acoustics optimisation in live sound environments using signal processing techniques has captivated the minds of audio enthusiasts and researchers alike for over half a century. From analogue filters in the 1950s, to modern research efforts such as room impulse response equalisation and adaptive sound field control, this subject has exploded to life. Controlling the sound field in a static acoustic space is complex due to the high number of system variables, such as reflections, speaker crosstalk, equipment-induced coloration, room modes, reverberation, diffraction and listener positioning. These challenges are further amplified by dynamic variables such as audience presence, environmental conditions and room occupancy changes, which continuously and unpredictably reshape the sound field.
A primary objective of live sound reinforcement is to deliver uniform sound quality across the audience area. This is most critical at audience ear level, where tonal balance, clarity, and spatial imaging are most affected by variations in the sound field. While placing microphones at audience ear level positions could enable real-time monitoring, large-scale deployment is impractical due to audience interference.
This research will explore the feasibility of an adaptive virtual microphone-based approach to room acoustics optimisation. By strategically placing microphone arrays and leveraging virtual microphone technology, the system estimates the sound field dynamically at audience ear level without requiring physical microphones. By continuously repositioning focal points across listening zones, a small number of arrays could effectively monitor large audience areas. If accurate estimations can be achieved, real-time sound field control becomes more manageable and effective.

The occurrence of eigenmodes is one of the fundamental phenomena in the acoustics of small rooms. The modes formation results in an uneven distribution of the sound pressure level in the room. To determine the resonance frequencies and their distributions, numerical methods, analytical methods or experimental studies are used. For the purpose of this paper, an experimental study was carried out in a small room. The study analysed the results of measuring the sound pressure level distributions in the room, with a special focus on the frequency range 20 Hz - 32 Hz, below the first modal frequency in the room. The measurement were conducted in the rectangular grid 9x9 microphones, which resulted in 0.5 m microphones grid resolution. The influence of evanescent modes on the total sound field was investigated. The research takes into account several sound source locations. On the basis of the acoustic measurement carried out, frequency response curves were also plotted. This paper presents a few methods for analysing these curves based on standard deviation, the linear least squares method, coefficient of determination R^2 and root mean squared error (RMSE). The results obtained made it possible to determine the best position of the acoustic source in the room under study. The effect of evanescent modes on the total sound field was also observed.

Mono compatibility is a fundamental challenge in audio production, ensuring that stereo mixes retain clarity, balance, and spectral integrity when summed to mono. Traditional stereo widening techniques often introduce phase shifts, comb filtering, and excessive decorrelation, causing perceptual loss of critical mix elements in mono playback. Diffuse Signal Processing (DiSP) is introduced as a convolution-based method that improves mono compatibility while maintaining stereo width.

This study investigates the application of DiSP to the left and right channels of a stereo mix, leveraging MATLAB-synthesized TDI responses to introduce spectrally balanced, non-destructive acoustic energy diffusion. TDI convolution is then applied to both the left and right channels of the final stereo mix.

A dataset of stereo mixes from four genres (electronic, heavy metal, orchestral, and pop/rock) was analyzed. The study evaluated phase correlation, mono-summed frequency response deviation and amount of comb filtering to quantify improvements in mono summation. Spectral plots and wavelet transforms provided objective analysis. Results demonstrated that DiSP reduced phase cancellation, significantly decreased comb filtering artifacts, and improved spectral coherence in mono playback while preserving stereo width within the original mix. Applying this process to the final left and right channels allows an engineer to mix freely without the concern of the mono mix’s compatibility.

DiSP’s convolution-based approach offers a scalable, adaptive solution for modern mixing and mastering workflows, overcoming the limitations of traditional stereo processing. Future research includes machine learning-driven adaptive DiSP, frequency-dependent processing enhancements, and expansion to spatial audio formats (5.1, 7.1, Dolby Atmos) to optimize mono downmixing. The findings confirm DiSP as a robust and perceptually transparent method for improving mono compatibility without compromising stereo imaging.

Standing waves are a phenomenon ever-present in the reproduction of low frequencies and have a direct impact on the auditory perception of this frequency region.
This study addresses the challenges posed by standing waves which are difficult to measure accurately using conventional pressure microphones, due to their spatial and temporal characteristics. To combat these issues, a state-of-the-art sound pressure velocity probe specifically designed for measurement of intensity in the low-frequency spectrum is developed. Using this probe, the research includes the development of new energy estimation parameters to better quantify the characteristics of sound fields influenced by standing waves. Additionally, a novel "standing-wave-ness" parameter is proposed, based on two diffuseness quantities dealing with the proportion of locally confined energy and the temporal variation of the intensity vectors. The performance of the new method and probe is evaluated through both simulated and real-world measurement data. Simulations provide a controlled environment to assess the method's accuracy across a variety of scenarios, including both standing wave and non-standing wave conditions. These initial simulations are followed by validation through measurement data obtained from an anechoic chamber, ensuring that the method's capabilities are tested in highly controlled, close-to-real-world settings. Preliminary results from this dual approach show promising potential for the new method to quantify the presence of standing waves, adding a new dimension in the visualisation and understanding of low-frequency phenomena.

As spatial audio shifts from a premium feature to a mainstream expectation, significant challenges remain in delivering a uniform experience across devices, formats, and playback systems. This panel brings together industry and academic experts to explore the key technologies driving the future of immersive audio for consumers. We’ll discuss the core technological advancements, software, hardware, and ecosystem innovations necessary to enable more seamless and consistent spatial audio experiences. Additionally, we will examine the challenges of delivering perceptually accurate spatial audio across diverse playback environments and identify the most critical areas of focus for industry and academia to accelerate broader consumer adoption of spatial audio.

This study investigated whether categorical perception—a phenomenon observed in speech perception—extends to the discrimination of vowel-like timbre qualities. Categorical perception occurs when continuous acoustic variations are perceived as distinct categories, leading to better discrimination near category boundaries than within a category. To test this, discrimination thresholds for the center frequency of a one-third-octave band formant introduced into the spectrum of a pink noise burst were measured in five subjects using an adaptive psychophysical procedure. Thresholds were assessed at distinctive formant frequencies of selected Polish vowels and at boundaries between adjacent vowel categories along the formant-frequency continuum. Results showed no reduction in discrimination thresholds at category boundaries, suggesting an absence of categorical perception for vowel-like timbre. One possible explanation for this finding lies in the listening mode—a concept from ecological auditory research—describing cognitive strategies in auditory tasks. The design of both the stimuli and the experimental procedure likely encouraged an acousmatic listening mode, which focuses solely on the sensory characteristics of sound, without reference to its source or meaning. This may have suppressed cues typically used in the categorical perception of speech sounds, which are associated with the communication listening mode. These findings highlight the importance of considering listening mode in future research on categorical perception of timbre and suggest that vowel-like timbre discrimination may involve perceptual mechanisms distinct from those used in speech sound discrimination.

Individual head-related transfer functions (HRTFs) are instrumental in rendering plausible spatial audio playback over headphones as well as in understanding auditory perception. Nowadays, the numerical calculation of individual HRTFs is achievable even without high-performance computers. However, the main obstacle is the acquisition of a mesh of the pinnae with a submillimeter accuracy. One approach to this problem is the photogrammetric reconstruction (PR), which estimates a 3D shape from 2D input, e.g., photos. Albeit easy to use, this approach comes with a trade-off in the resulting mesh quality, which subsequently has a substantial impact on the HRTF's quality. In this study, we investigated the effect of PR on HRTF quality as compared to HRTFs calculated from a reference mesh acquired with a high-quality structured-light scanner. Additionally, we applied two pinna deformation methods, which registered a non-individual high-quality pinna to the individual low-quality PR pinna by means of geometric distances. We investigated the potential of these methods to improve the quality of the PR-based pinna meshes. Our evaluation involved the geometrical, acoustical, and psychoacoustical domains including a sound-localization experiment with 9 participants. Our results show that neither PR nor PR-improvement methods were able to provide individual HRTFs of sufficient quality, indicating that without extensive pre- or post-processing, PR provides too little individual detail in the HRTF-relevant pinna regions.

The rapid advancement of generative artificial intelligence has created highly realistic DeepFake multimedia content, posing significant challenges for digital security and authenticity verification. This paper presents the development of a comprehensive testbed designed to detect counterfeit audio content generated by DeepFake techniques. The proposed framework integrates forensic spectral analysis, numerical and statistical modeling, and machine learning-based detection to assess the authenticity of multimedia samples. Our study evaluates various detection methodologies, including spectrogram comparison, Euclidean distance-based analysis, pitch modulation assessment, and spectral flatness deviations. The results demonstrate that cloned and synthetic voices exhibit distinctive acoustic anomalies, with forensic markers such as pitch mean absolute error and power spectral density variations serving as effective indicators of manipulation. By systematically analyzing human, cloned, and synthesized voices, this research provides a foundation for advancing DeepFake detection strategies. The proposed testbed offers a scalable and adaptable solution for forensic audio verification, contributing to the broader effort of safeguarding multimedia integrity in digital environments.

Content exchange and collaboration serve as catalysts for repository creation that supports creative industries and fuels model development in machine learning and AI. Despite numerous repositories, challenges persist in discoverability, rights preservation, and efficient reuse of audiovisual assets. To address these issues, the SCENE (Searchable multi-dimensional Data Lakes supporting Cognitive Film Production & Distribution for the Promotion of the European Cultural Heritage) project introduces an automated audio quality assessment toolkit integrated within its Media Assets Management (MAM) platform. This toolkit comprises a suite of advanced metrics, such as artifact detection, bandwidth estimation, compression history analysis, noise profiling, speech intelligibility, environmental sound recognition, and reverberation characterization. The metrics are extracted using dedicated Flask-based web services that interface with a data lake architecture. By streamlining the inspection of large-scale audio repositories, the proposed solution benefits both high-end film productions and smaller-scale collaborations. The pilot phase of the toolkit will involve professional filmmakers who will provide feedback to refine post-production workflows. This paper presents the motivation, design, and implementation details of the toolkit, highlighting its potential to assess content quality management and contribute to more efficient content exchange in the creative industries.

When navigating the environment, we primarily rely on sight. However, in its absence, individuals must develop precise spatial awareness using other senses. A blind person can recognize their immediate surroundings through touch, but assessing larger spaces requires auditory perception.
This project presents a method for auditory training in children with visual disabilities through structured audio plays designed to teach spatial pronouns and enhance spatial orientation via auditory stimuli. The format and structure of these audio plays allow for both guided learning with a mentor and independent exploration. Binaural recordings serve as the core component of the training exercises. The developed audio plays and their analyses are available on the YouTube platform in the form of videos and interactive exercises.
The next step of this project involves developing an application that enables students to create individual accounts and track their progress. Responses collected during exercises will help assess the impact of the audio plays on students, facilitating improvements and modifications to the training materials.
Additionally, linking vision-related questions with responses to auditory exercises will, over time, provide insights into the correlation between these senses. The application can serve multiple purposes: collecting research data, offering spatial recognition and auditory perception training, and creating a comprehensive, structured environment for auditory skill development.

This paper investigates the innovative synthesis of procedurally generated visual and auditory content through the use of Artificial Intelligence (AI) Tools, specifically focusing on Generative Pre-Trained Transformer (GPT) networks.
This research explores the process of procedurally generating an audiovisual representations of semantic context by generating images, artificially providing motion and generating corresponding multilayered sound. The process enables the generation of stopped-motion audiovisual representations of concepts.
This approach not only highlights the capacity for Generative AI to produce cohesive and semantically rich audiovisual media but also delves into the interconnections between visual art, music, sonification, and computational creativity. By examining the synergy between generated imagery and corresponding soundscapes, this research paper aims to uncover new insights into the aesthetic and technical implications of the use of AI in art.
This research embodies a direct application of AI technology across multiple disciplines creating intermodal media. Research findings propose a novel framework for understanding and advancing the use of AI in the creative processes, suggesting potential pathways for future interdisciplinary research and artistic expression.
Through this work, this study contributes to the broader discourse on the role of AI in enhancing creative practices, offering perspectives on how various modes of semantic representation can be interleaved using state-of-the-art technology.

We present G.A.D.A. (Guitar Audio Dataset for AI), a novel open-source dataset designed for advancing research in guitar audio analysis, signal processing, and machine learning applications. This comprehensive corpus comprises recordings from three main guitar categories: electric, acoustic, and bass guitars, featuring multiple instruments within each category to ensure dataset diversity and robustness.

The recording methodology employs two distinct approaches based on instrument type. Electric and bass guitars were recorded using direct recording techniques via DI boxes, providing clean, unprocessed signals ideal for further digital processing and manipulation. For acoustic guitars, where direct recording was not feasible, we utilized multiple microphone configurations at various positions to capture the complete acoustic properties of the instruments. Both recording approaches prioritize signal quality while maintaining maximum flexibility for subsequent processing and analysis.

The dataset includes standardized recordings of major and minor chords played in multiple positions and voicings across all instruments. Each recording is accompanied by detailed metadata, including instrument specifications, recording equipment details, microphone configurations (for acoustic guitars), and chord information. The clean signals from electric instruments enable various post-processing applications, including virtual amplifier modeling, effects processing, impulse response convolution, and room acoustics simulation.

To evaluate G.A.D.A.'s effectiveness in machine learning applications, we propose a comprehensive testing framework using established algorithms including k-Nearest Neighbors, Support Vector Machines, Convolutional Neural Networks, and Feed-Forward Neural Networks. These experiments will focus on instrument classification tasks using both traditional audio features and deep learning approaches.

G.A.D.A. will be freely available for academic and research purposes, complete with documentation, preprocessing scripts, example code, and usage guidelines. This resource aims to facilitate research in musical instrument classification, audio signal processing, deep learning applications in music technology, computer-aided music education, and automated music transcription systems.

The combination of standardized recording methodologies, comprehensive metadata, and the inclusion of both direct-recorded and multi-microphone captured audio makes G.A.D.A. a valuable resource for comparative studies and reproducible research in music information retrieval and audio processing.

Published studies indicate that musicians outperform non-musicians in a variety of non-musical auditory tasks, a phenomenon known as the “musicians’ hearing advantage effect.” One widely reported benefit is enhanced speech-in-noise (SIN) recognition. It was observed that musicians’ speech-in-noise (SIN) recognition thresholds (SRTs) are lower than those of non-musicians, though findings—mainly from English-language studies—are mixed; some confirm these advantage, while others do not. This study extends SRT measurements to Polish, a language with distinct phonetic characteristics. Participants completed a Polish speech intelligibility test, reconstructing sentences masked by multitalker babble noise by selecting words from a list displayed on a computer screen. Speech levels remained constant while masking noise was adjusted adaptively: increasing after each correct response and decreasing after each error. Three groups were tested: musicians, musically trained audio engineers, and non-musicians. Results showed that musicians and audio engineers had SRTs 2 and 2.7 dB lower than non-musicians, respectively. Although audio engineers exhibited slightly lower SRTs than musicians, the difference was minimal, with statistical significance just above the conventional 5% threshold. Thus, under these conditions, no clear advantage of audio engineers over musicians in SIN performance was observed.

An individual human pinna geometry can be used to achieve plausible personalized audio reproduction. However, an accurate acquisition of the pinna geometry typically requires the use of specialized equipment and often involves time-consuming post-processing to remove potential artifacts. To obtain an artifact-free but individualized mesh, a parametric pinna model based on cubic Bézier curves (BezierPPM) can be used to represent an individual pinna. However, the parameters need to be manually tuned to the acquired listener’s geometry. For increased scalability, we propose Mesh2PPM, a framework for an automatic estimation of BezierPPM parameters from an individual pinna. Mesh2PPM relies on a deep neural network (DNN) that was trained on a dataset of synthetic multi-view images rendered from BezierPPM instances. For the evaluation, unseen BezierPPM instances were presented to Mesh2PPM which inferred the BezierPPM parameters. We subsequently assessed the geometric errors between the meshes obtained from the BezierPPM parametrized with the inferred parameters and the actual pinna meshes. We investigated the effects of the camera-grid type, jittered camera positions, and additional depth information in images on the estimation quality. While depth information had no effect, the camera-grid type and the jittered camera positions both had effects. A camera grid of 3×3 provided the best estimation quality, yielding Pompeiu-Hausdorff distances of 2.05 ± 0.4 mm and 1.4 ± 0.3 mm with and without jittered camera
positions, respectively, and root-mean-square (RMS) distances of 0.92 ± 0.12 mm and 0.52 ± 0.07 mm. These results motivate further improvements of the proposed framework to be ultimately applicable for an automatic estimation of pinna geometries obtained from actual listeners.

The multi-stimulus test with hidden reference and anchor (MUSHRA) is a prevalent method for the subjective audio quality evaluation. Despite its popularity, the technique is not immune to biases. Empirical evidence indicates that the presence of labels (quality descriptors) equidistantly distributed along the rating scale may be the cause of its non-linear warping; however, other factors could evoke even stronger non-linear effects. This study aims to investigate the hypothesis that stimulus spacing bias may induce a greater magnitude of non-linear warping of the quality scale compared to that caused by the presence of labels. To this end, a group of more than 120 naïve listeners participated in MUSHRA-compliant listening tests using labeled and unlabeled graphic scales. The audio excerpts, representing two highly skewed distributions of quality levels, were reproduced over headphones in an acoustically treated room. The findings of this study verify the postulated hypothesis and shed new light on the mechanisms biasing results of the MUSHRA-conformant listening tests.

In order to reproduce audio over headphones as in-
tended, it is essential to have well-defined and con-
sistent references of how headphones should sound.
With the aim of stereo reproduction in mind, the field
has established a de-facto reference target curve called
the Harman Target Curve to which headphone transfer
functions are commonly compared. This contribution
questions if the same target curve is suitable when used
for the reproduction of spatial audio. First, the ori-
gins the Harman Curve are revisited; it is motivated by
the frequency response of loudspeaker playback in a
specific listening room. The necessary measurement
procedures are described in detail. Then, the paper
discusses the applicability of existing targets to spa-
tial audio. Therein, it is possible to embed convincing
spatial room information directly into the production,
thereby calling into question the motivation for incor-
porating a listening room in the headphone target. The
paper concludes with a listening experiment that com-
pares the preference of different target curves for both
spatial audio and stereo

How can Audio Augmented Reality (AAR) serve as a storytelling medium? Sound designer Matias Harju shares insights from The Reign Union, an experimental interactive AAR story currently exhibited at WHS Union Theatre in Helsinki, Finland.

This workshop addresses the challenges and breakthroughs of creating an immersive, headphone-based 6DoF AAR experience. In The Reign Union, two simultaneous participants experience the same bio-fictional story from different points of audition. Narrative design considerations and approaches are discussed and demonstrated through video clips featuring binaural sound recorded from the experience. References to other AAR experiences around the world are included to provide a broader context. A central theme is how reality anchors the narrative, while virtual sounds reveal new perspectives and interpretations.

The session also briefly examines the development of an in-house 6DoF AAR prototype platform, used for The Reign Union story as well as other narrative research conducted by the author and his team. This has been a journey through various pose tracking, virtual acoustic, and authoring solutions, resulting in a scalable system potentially suited for complex indoor spaces.

Matias, author of the forthcoming book Audio Augmented Reality: Concepts, Technologies, and Narratives (Routledge, June 2025), invites attendees to discuss and discover the possibilities of AAR as a tool for storytelling and artistic expression.

This research aims to explore the impact of variations in apparent sound source width and position on emotional and physiological responses among listeners, with a particular focus on the domain of virtual reality applications. While sound is recognized as a potent elicitor of strong emotions, the specific role of spatial characteristics, such as apparent sound source width, has not been systematically analyzed. The authors’ previous study has indicated that the spatial distribution of sound can alter perceptions of scariness. In contrast, the current study explores whether adjustments in apparent sound source width can significantly affect emotional valence and arousal, as well as human physiological metrics. The objective of this study was to investigate the impact of a single sound source width and its horizontal position on emotional engagement, thereby providing valuable insights for advancements in immersive audio experiences. Our experiments involved conducting listening tests in a spatial sound laboratory, utilizing a circular setup of sixteen loudspeakers to present a range of audio stimuli drawn from five selected recordings. The stimuli were manipulated based on two key parameters: the apparent sound source width and the spatial positioning of the sound source (front, back, left, or right). Participants assessed their emotional reactions using the Self-Assessment Manikin (SAM) pictogram method. Physiological data, including electroencephalogram, blood volume pressure, and electrodermal activity was collected in real-time via wearable sensors consisting of an EEG headset and a finger-attached device.

In recent years, applications such as virtual reality (VR) systems and room acoustics simulations have brought the modeling of sound source directivity into focus. An accurate simulation of directional responses of sound sources is essential in immersive audio applications.

Real sound sources have directional properties that are different from simple sources such as monopoles, which are sources frequently used for modeling more complex acoustic fields. For instance, the sound level of human speech as a sound source varies considerably depending on where the sound is recorded with respect to the talker’s head. The same is true for loudspeakers, which are considered linear and time-independent sources. When the sound is recorded behind the speaker, it is normal to observe differences of up to 20 dB SPL at some frequencies. The directional characteristics of sound sources become particularly pronounced at high frequencies. The propagation of real sound sources, such as human voices or musical instruments, differs from simple source models like monopoles, dipoles, and quadrupoles due to their physical structures.

The common approach to measuring directivity patterns of sound sources involves surrounding a sound source in an anechoic chamber with a high number of pressure microphones on a spherical grid and registering the sound power at these positions. Apart from the prohibitive hardware requirements, such measurement setups are mostly impractical and costly. Audio system manufacturers have developed various methods for measuring sound source directionality over the years. These methods are generally of high technical complexity.

This article proposes a new, reduced-complexity directivity measurement approach based on the spherical harmonic decomposition of the sound field. The method estimates the directional characteristics of sound sources using fewer measurement points with spherical microphone arrays. The spherical harmonic transform allows for the calculation of directivity using data collected from spherical microphone arrays instead of pressure sensors. The proposed method uses both the pressure component and spatial derivatives of the sound field and successfully determines directivity with sparse measurements.

An estimation model based on the spherical Fourier transform was developed, measurements were carried out to test this model, and preliminary results obtained from the estimation model are presented. Experiments conducted at the METU Spatial Audio Research Laboratory demonstrated the effectiveness of the proposed method. The directivity characteristics of Genelec 6010A loudspeaker are measured using eight 3rd-order spherical microphone arrays. The directivity functions obtained were highly consistent with the data provided by the loudspeaker manufacturer. The results, especially in low and mid-frequency bands, indicate the utility of the proposed method.

This paper describes ongoing research on integrating algorithmic reverberation tools designed for audio post-production into virtual acoustics, focusing on using Impulse Responses (IRs) captured from the legendary Lexicon 960L hardware reverberation unit. While previous research from the McGill University Virtual Acoustics Technology (VAT) Lab has utilized room impulse responses (RIRs) captured from various performance halls to create active acoustic environments in the recording studio, this study analyzes the perceived differences between the two listening environments and the effect of the VATLab speakers and effect of room acoustics on IRs captured from 5.0 multichannel reverb presets. Three of these multichannel IRs have been chosen to simulate a Lexicon 960L “environment” in a physical space.

Objective measurements in McGill University’s Immersive Media Laboratory (IMLAB) Control Room and in VATLab following the ISO 3382 standard measure the effect of the physical room and the omnidirectional dodecahedral speakers used for auralization. Through a subjective pilot study, subjective analysis investigates the perceived differences between the Lexicon IRs in VATLab and a control condition, the IMLAB control room. The results of an attribute rating test on perceived immersion, soundfield continuity, tone color, and overall listening experience between the two spaces helps us better understand how reverberation algorithms designed for multichannel mixing/post-production translate to a virtual acoustics system.
In conclusion, we discuss the perceptual differences between the IMLAB Control Room and VATLab and results of objective measurements.

Immersive sound reinforcement aims to create a balanced perception of sounds arriving from different directions, establishing an impression of envelopment over the audience area. Current perceptual research shows that coverage designs featuring nearly constant decay (0dB per distance doubling) preserve the level balance among audio objects in the mix. In contrast, a -3dB decay supports a more uniform sensation of envelopment, especially for off-center listening positions. For practical reasons, point-source loudspeakers remain widely used for immersive audio playback in mid-sized venues. However, point-source loudspeakers inherently decay by -6dB per distance doubling, and using them can conflict with the design goals outlined above. In this paper, we investigate the perceived differences between point-source and line-source setups using eight surrounding loudspeakers side-by-side covering a 10m x 7m audience area. The perceptual qualities of object level balance, spatial definition, and envelopment were compared in a MUSHRA listening experiment, and acoustic measurements were carried out to capture room impulse responses and binaural room impulse responses (BRIRs) of the experimental setup. The BRIRs were used to check whether the results of the listening experiment were reproducible on headphones. Both the loudspeaker and headphone-based experiments delivered highly correlated results. Also, regression models devised based on the acoustic measurements are highly correlated to the perceptual results. The results confirm that elevated line sources, exhibiting a practically realizable decay of -2dB per distance doubling, help preserve object-level balance, increase spatial definition, and provide a uniform envelopment experience throughout the audience area compared to point-source loudspeakers.

Numerous studies highlight the role of transient behavior in musical sounds and its impact on sound identification. This study compares these findings with established psychoacoustic measurements of detection thresholds for asynchrony in onset and offset transients, obtained using synthesized stimuli that allowed precise control of stimulus parameters. Results indicated that onset asynchrony can be detected at thresholds as low as 1 ms—even half a cycle of the component frequency. In contrast, offset asynchrony detection was found to be less precise, with thresholds ranging from 5 to 10 ms. Sensitivity improves when multiple harmonics are asynchronous. Additionally, component phase significantly influences onset asynchrony detection: at 1000 Hz and above, phase shifts raise thresholds from below 1 ms to around 50 ms, while having little effect on offset detection. Although these findings were based on controlled artificial stimuli, they can provide valuable insight into asynchrony in natural musical sounds. In many cases, detection thresholds are well below the variations observed in music, yet under certain conditions and frequencies, some temporal variations may become not perceptible.

The ECHO Project (Exploring the Cinematic Hemisphere for Orchestra) is a collaborative research initiative that explores 3D microphone array techniques for orchestral recording, involving eight experts in immersive sound recording: Kellogg Boynton, Anthony Caruso, Hyunkook Lee, Morten Lindberg, Simon Ratcliffe, Katarzyna Sochaczewska, Mark Willsher, and Nick Wollage. Building on the 3D-MARCo initiative, this project aims to provide a platform for sound engineers, composers, researchers, and students to experiment with various immersive recording techniques. To this end, an open-access database of high-quality orchestral recordings was created from a recording session at AIR Studios, London, featuring a Oscar-winning composer Volker Bertelmann and the London Contemporary Orchestra.

The ECHO database includes recordings of four pieces, captured using up to 143 microphone capsules per piece. This setup includes seven different microphone arrays designed by the experts, spot microphones, a dummy head, and a higher-order spherical microphone system. The database allows users to not only compare different techniques but also to experiment with mixing different microphones, helping them develop their own techniques. It also serves as a useful resource for research, teaching and learning in immersive audio.

This workshop will present the rationale behind each microphone array used in the project, detail the recording process, discuss the immersive approach to composition and recording methods, and present some of the recordings in 7.1.4.

Wide Dynamic Range Compression in hearing aids is becoming increasingly more complex as the number of channels and adjustable parameters grow. At the same time, there is growing demand for customization and user self-adjustment of hearing aids, necessitating a balance between complexity and user accessibility. Compression in hearing aids is governed by the input-output transfer function, which relates input magnitude to output magnitude, and is typically defined as a combination of linear piecewise segments resembling logarithmic behavior. This work presents an alternative to the conventional compression transfer function that consolidates multiple compression parameters and revisits expansion in hearing aids. The
curvilinear transfer function is a continuous curve with logarithm-like behavior, governed by two parameters—gain and compression ratio. Experimental results show that curvilinear compression reduces the amplification of low-level noise, improves signal-to-noise ratio by up to 1.0 dB, improves sound quality as measured by the Hearing Aids Speech Quality Index by up to 6.7%, and provides comparable intelligibility as measured by the Hearing Aids Speech Perception Index, with simplified parameterization compared to conventional compression.
The consolidated curvilinear transfer function is highly applicable to over-the-counter hearing aids and offers more capabilities for customization than current prominent over-the-counter and self-adjusted hearing aids.

As immersive audio content becomes more prevalent across streaming and broadcast platforms, creators and engineers face the challenge of making spatial audio accessible to listeners using legacy codecs and traditional playback systems, particularly headphones. With multiple binaural encoding methods available, choosing the right approach for a given project can be complex.

This workshop is designed as an exploration for audio professionals to better understand the strengths and applications of various binaural encoding systems. By comparing different techniques and their effectiveness in real-world scenarios, attendees will gain insights into how binaural processing can serve as a bridge between legacy and modern formats, preserving spatial cues while maintaining compatibility with existing distribution channels.

As the first in a series of workshops, this session will help define key areas for real-world testing between this convention and the next. Attendee insights and discussions will directly influence which encoding methods are explored further, ensuring that the most effective solutions are identified for different content types and delivery platforms.

Participants will gain an understanding of processing methods, and implementation strategies for various distribution platforms. By integrating these approaches, content creators can enhance accessibility and ensure that immersive audio reaches a wider audience, possibly encouraging consumers to explore how to enjoy immersive content using a variety of playback systems.

The increasing demand for spatial audio in applications such as virtual reality, immersive media, and spatial audio research necessitates robust solutions for binaural audio dataset generation for testing and validation. Binamix is an open-source Python library designed to facilitate programmatic binaural mixing using the extensive SADIE II Database, which provides HRIR and BRIR data for 20 subjects. The Binamix library provides a flexible and repeatable framework for creating large-scale spatial audio datasets, making it an invaluable resource for codec evaluation, audio quality metric development, and machine learning model training. A range of pre-built example scripts, utility functions, and visualization plots further streamline the process of custom pipeline creation. This paper presents an overview of the library's capabilities, including binaural rendering, impulse response interpolation, and multi-track mixing for various speaker layouts. The tools utilize a modified Delaunay triangulation technique to achieve accurate HRIR/BRIR interpolation where desired angles are not present in the data. By supporting a wide range of parameters such as azimuth, elevation, subject IRs, speaker layouts, mixing controls, and more, the library enables researchers to create large binaural datasets for any downstream purpose. Binamix empowers researchers and developers to advance spatial audio applications with reproducible methodologies by offering an open-source solution for binaural rendering and dataset generation.

In this work, we introduce a Neural 3D Audio Renderer (N3DAR) - a conceptual solution for creating acoustic digital twins of arbitrary spaces. We propose a workflow that consists of several stages including:
1. Simulation of high-fidelity Spatial Room Impulse Responses (SRIR) based on the 3D model of a digitalized space,
2. Building an ML-based model of this space for interpolation and reconstruction of SRIRs,
3. Development of a real-time 3D audio renderer that allows the deployment of the digital twin of a space with accurate spatial audio effects consistent with the actual acoustic properties of this space.
The first stage consists of preparation of the 3D model and running the SRIR simulations using the state-of-the-art wave-based method for arbitrary pairs of source-receiver positions. This stage provides a set of learning data being used in the second stage - training the SRIR reconstruction model. The training stage aims to learn the model of the acoustic properties of the digitalized space using the Acoustic Volume Rendering approach (AVR). The last stage is the construction of a plugin with a dedicated 3D audio renderer where rendering comprises reconstruction of the early part of the SRIR, estimation of the reverb part, and HOA-based binauralization.
N3DAR allows the building of tailored audio rendering plugins that can be deployed along with visual 3D models of digitalized spaces, where users can freely navigate through the space with 6 degrees of freedom and experience high-fidelity binaural playback in real time.
We provide a detailed description of the challenges and considerations for each of the stages. We also conduct an extensive evaluation of the audio rendering capabilities with both, objective metrics and subjective methods using a dedicated evaluation platform.

This paper presents an objective method for estimating the performance of 3D microphone arrays, which is also applicable to 2D arrays. The method incorporates the physical characteristics and relative positions of the microphones, merging these elements through a weighted summation to derive the arrays' directional patterns. These patterns are represented as a "Modified Steering Vector." Additionally, leveraging the spatial properties of spherical harmonics, we transform the array's directional pattern into the spherical harmonic domain. This transformation enables a quantitative analysis of the physical properties of each component, providing a comprehensive understanding of the array's performance. Overall, the proposed method offers a deeply insightful and versatile framework for evaluating the performance of both 2D and 3D microphone arrays by fully exploiting their inherent physical characteristics.

The reconstruction of sound fields is a critical component in a range of applications, including spatial audio for augmented, virtual, and mixed reality (AR/VR/XR) environments, as well as for optimizing acoustics in physical spaces. Traditional approaches to sound field reconstruction predominantly rely on interpolation techniques, which estimate sound fields based on a limited number of spatial and temporal measurements. However, these methods often struggle with issues of accuracy and realism, particularly in complex and dynamic environments. Recent advancements in deep learning have provided promising alternatives, particularly with the introduction of Physics-Informed Neural Networks (PINNs), which integrate physical laws directly into the model training process. This study aims to explore the application of PINNs for sound field reconstruction, focusing on the challenge of predicting acoustic fields in unmeasured areas. The experimental setup involved the collection of impulse response data from the Promenadikeskus concert hall in Pori, Finland, using various source and receiver positions. The PINN framework is then utilized to simulate the hall’s acoustic behavior, with parameters incorporated to model sound propagation across different frequencies and source-receiver configurations. Despite challenges arising from computational load, pre-processing strategies were implemented to optimize the model's efficiency. The results demonstrate that PINNs can accurately reconstruct sound fields in complex acoustic environments, offering significant potential for real-time sound field control and immersive audio applications.

3D recordings seem to be an attractive solution when trying to achieve the immersion effect. Recently, Dolby Atmos is an increasingly popular format for distributing three-dimensional music recordings. Although currently the main format for producing music recordings is still stereophony.

How to optimally extend traditional microphone techniques when recording classical music to obtain both stereo recordings and three-dimensional formats (e.g. Dolby Atmos) in the post-production process? The author is trying to answer this question using the example of a recording of Dietrich Buxtehude work "Membra Jesu Nostri", BuxWV 75. The cycle of seven cantatas composed in 1680 is one of the most important and most popular compositions of the early Baroque era. The first Polish recording was made by the Arte Dei Suonatori conducted by Bartłomiej Stankowiak, accompanied by soloists and choral parts performed by the choir Cantus Humanus.

The author will present his concept of a set of microphones for 3D recordings. In addition to the detailed setup of microphones, it will cover the method of post-production of the recording, combining stereo with a mix of the recording into the Dolby Atmos system in a 7.2.4 speaker configuration. A workflow will be proposed to facilitate the change between different formats.

This paper investigates the subjective evaluation of two prominent three-dimensional spatialization techniques—Vector Base Amplitude Panning (VBAP) and High-Order Ambisonics (HOA)—using IRCAM’s Spat in an immersive concert setting. The listening test was conducted in the New Hall at the Royal Danish Academy of Music, which features a 44-speaker immersive audio system. The musical stimuli included electronic compositions and modern orchestral recordings, providing a diverse range of temporal and spectral content. The participants comprised experienced Tonmeisters and non-experienced musicians, who were seated in off-center positions to simulate real-world audience conditions. This study provides an ecologically valid subjective evaluation methodology.
The results indicated that VBAP excelled in spatial clarity and sound quality, while HOA demonstrated superior envelopment. The perceptual differences between the two techniques were relatively minor, influenced by room acoustics and suboptimal listening positions. Furthermore, music genre had no significant impact on the evaluation outcomes.
The study highlights VBAP’s strength in precise localization and HOA's capability for creating immersive soundscapes, aiming to bridge the gap between ideal and real-world applications in immersive sound reproduction and perception. The findings suggest the need to balance trade-offs when selecting spatialization techniques for specific purposes, venues, and audience positions. Future research will focus on evaluating a wider range of spatialization methods in concert environments and optimizing them to improve the auditory experience for distributed audiences.

#N/A

Hearing loss is a global public health issue due to its high prevalence and negative impact on various aspects of one’s life, including well being and cognition. Despite their crucial role in auditory rehabilitation, hearing aids remain inaccessible to many due to their high costs, particularly in low- and middle-income countries. Existing open-source solutions often rely on high-power, bulky platforms rather than compact, low-power wearables suited for real-world applications. This work introduces Tiresias, an open-source hearing aid development board designed for real-time audio processing using low-cost electronics. Integrating key hearing aid functionalities into a compact six-layer printed circuit board (PCB), Tiresias features multichannel compression, digital filtering, beamforming, Bluetooth connectivity, and physiological data monitoring, fostering modularity and accessibility through publicly available hardware and firmware resources based on the Nordic nRF Connect and Zephyr real-time operating system (RTOS). By addressing technological and accessibility challenges, this work advances open-source hearing aid development, enabling research in hearing technologies, while also supporting future refinements and real-world validation.

Join a panel of professionals from a variety of fields in the industry as we discuss topics including how to enter the audio industry, how they each got started in their own careers and the path their careers took, and give advice geared towards students and recent graduates. Bring your questions for the panelists – most of this workshop will be focused the information YOU want to hear!

*Introduction
With the growing market of immersive audio, both new and exciting production possibilities are emerging, alongside the resurfacing of existing surround sound production techniques. As audio production continues to evolve, understanding the impact of temporal properties on spatial perception becomes increasingly critical. One of the most effective ways to create a sense of space and depth, as well as to enhance listener envelopment, is through precise manipulation of temporal characteristics of sound.

*Temporal Adjustments in Audio Production
In stereophonic recording techniques, spatialization is often achieved by carefully controlling both each microphone’s distance from the sound source and the distance between microphones, in conjunction with leveraging variations in microphone sensitivity through polar patterns and directional rejection.
These distance-based variations introduce time delays, which are fundamental to spatial localization and depth perception. Similarly, in post-production workflows, delaying and applying differentiated effects to signals serve as powerful tools for enhancing immersion and spatiality. The controlled use of delay, reflections, and micro-temporal variations plays a significant role in shaping perceived auditory space. These techniques are widely used in both as mixing approaches with music and also sound design where artificially introducing delays helps simulate the propagation of sound in physical spaces, creating a more authentic and immersive auditory experience.

*Psychoacoustic Phenomena and Spatial Perception
Closely delayed or slightly altered signals give rise to psychoacoustic effects that influence spatial perception rather than purely temporal perception.
For instance, the number, spectral characteristics, and temporal distribution of reflections can lead a listener to perceive an auditory environment akin to a concert hall, even in the absence of an actual reverberant space.
The well-known Haas effect (precedence effect) provides insights into how human perception prioritizes the first-arriving sound over subsequent delayed versions, influencing localization and clarity. Additionally, the concepts of Temporal Integration Window (auditory signal fusion) describe how multiple signals originating from the same source are perceptually fused into a single event, affecting spatial coherence and envelopment.

*Workshop and Study Overview
This workshop presents and exemplifies findings from an ongoing semester-long study, which is currently being prepared as a submission to the Journal of the Audio Engineering Society. The study investigates whether sensation, timbral perception, and temporal integration windows are influenced when the delayed signal's spatial position is altered. By showcasing how spatial modifications of delayed signals affect auditory perception, the workshop aims to contribute insights to the field of immersive audio production.

*Conclusion
This research underscores the importance of temporal manipulation in immersive audio, bridging psychoacoustics with production techniques. By examining spatial perception through the lens of delay-based processing, the study offers new perspectives on designing more effective immersive sound experiences. The workshop will provide participants with theoretical insights and practical examples, encouraging further exploration of the intersection between temporal properties and spatial audio design.