Archives of Acoustics
type: channel
description: Archives of Acoustics is an English-language peer-reviewed online quarterly journal publishing original research papers from all areas of acoustics and abstracts from some specialized acoustical conferences. It gives free internet access to its full content to current issues. Archives of Acoustics, the peer-reviewed online quarterly journal, publishes original research papers from areas of acoustics like: linear and nonlinear acoustics: theory, acoustical measurements, and instrumentation acoustics of music: music theory, signal processing, and instrument building physiological acoustics: hearing, audiology, and psychoacoustics speech processing and communication systems: speech production and perception, voice recognition and control architectural, building, and environmental acoustics and soundscape computational and numerical acoustics sound and ultrasound in biology and medicine: ultrasonic imaging, ultrasonography, high-intensity ultrasound, therapeutic ultrasound electroacoustics: research in the field of microphones, loudspeakers, headphones, accelerometers, acoustic calibrators noise, noise control, and vibration: theory and instrumentation for noise determination and sound quality criteria, audio signal processing and transducers, acoustic materials sonochemistry: acoustic cavitation, sonoprocessing, and all aspects of physical and chemical effects of sound ultrasound and ultrasonic transducers underwater acoustics: acoustical oceanography and fisheries acoustics acousto-optics artificial intelligence in acoustic research This journal provides immediate open access to its content on the principle that making research freely available to the public supports a greater global exchange of knowledge. All articles are published on Creative Commons licence CC BY 4.0. Archives of Acoustics is available in Diamond Open Access and no fees are charged to the authors or readers. Journal Metrics MNiSW (2024): 100 | Impact Factor 2024: 0.9 (2023: 0.6) | CiteScore 2024: 1…
publisher: Institute of Fundamental Technological Research (IPPT) of the Polish Academy of Sciences
language: en
publication name: Archives of Acoustics
issn: 0137-5075
copyright: Copyright on any open access article in the Archives of Acoustics published by the Polish Academy of Sciences and the Institute of Fundamental Technological Research Polish Academy of Sciences is retained by the author(s). The authors grant the Polish Academy of Sciences and the Institute of Fundamental Technological Research Polish Academy of Sciences a license to publish the article and identify themselves as the original publisher. The authors also grant any user the right to use the article freely as long as its integrity is maintained and its original authors, citation details, and publisher are identified. The Creative Commons Attribution 4.0 International CC BY 4.0 formalizes these and other terms and conditions of publishing articles. Exceptions to copyright policy For the articles that were published between 2019– 2023 (including No. 1 and 2, 2023), the policies are as follows: The Creative Commons Attribution License-ShareAlike 4.0 formalizes these and other terms and conditions of publishing articles: https://creativecommons.org/licenses/by-sa/4.0/. For the articles which were previously published, before year 2019, policies that are different from the above. In all such, access to these articles is free from fees or any other access restrictions. Permissions for the use of the texts published in that journal may be sought directly from the Editorial Office of Archives of Acoustics
items: In Memoriam Professor Eugeniusz Kozaczka
items: Subwavelength Underwater Imaging of a Wire Array Metamaterial Based on Fabry–Perot Resonance
items: Single-Sensor Passive Ranging of Underwater Monopoles Using Near-Field/Far-Field Energy Contrasts
items: Analysis of Decision Fusion in Speech Detection
items: Potential Applications of Ultrasonic Parametric Array Loudspeakers (PALs) in Room Acoustic Measurements
items: Simulation Analysis of Beam Intensity Attenuation Patterns and Source Depth Estimation Using a Vertical Long Line Array
items: Comparison of Methods for Determining the Airflow Resistivity of Porous and Covering Materials
items: The Impact of Training Strategies on Overfitting in Vowel Classification Using PS-HFCC Parametrization for Automatic Speech Recognition
items: Comparative Study of the Acoustic Efficiency of Prototype Sound Absorbing Panels Used in the Railway Track
items: Review of Microphone-Based Contactless Vital Signs Monitoring Systems
items: Experimental and Computational Analysis on the Acoustic Performance of Side Outlet Muffler with Semi-Circular Baffles
items: Effect of Listener Head Position on Speech Intelligibility in an Automotive Cabin
items: Multi-label Bird Species Classification Using Transfer Learning Network
items: Aeroacoustic Effect of Savonius Rotor Segmentation
items: Structural Parametric Study of a Piezoelectric Energy Harvester for a Specific Excitation Frequency of an Electric Motor, Considering Fatigue Life
items: Simulation and Experiment of a Medium-Distance Underwater Ultrasonic Wireless Power Transfer System
items: Experimental Results of Diver Detection in Harbor Environments Using Single Acoustic Vector Sensor
items: CAPSE-ViT: A Lightweight Framework for Underwater Acoustic Vessel Classification Using Coherent Spectral Estimation and Modified Vision Transformer
items: Benchmarking the First Realistic Dataset for Speech Separation
items: Assessment of Noise from Different Types of Rounds Fired from a Saluting Gun
items: Comparison of Plate Vibration and Structural Sound Reduction Using Square-Based Sensor–Actuator Piezoelectric Hybrids with Different Shapes of Sensor Part
items: Sensitivity Analysis of Acoustic Parameters in a Theatre Hall: A Case Study of the Maria Zankovetska Theatre in Lviv
items: Sound Source Localisation in Digital Hearing Aids: A Review of Critical Factors
items: The Impact of HVAC Systems on Speech Intelligibility in University Classrooms
items: A Hybrid Method for Predicting the Bistatic Target Strength Based on the Monostatic Scattering Sound Pressure Data
items: Localization of Virtual Sound Source Reproduced by the Crosstalk Cancellation System Under Different Reflective Conditions
items: Listening Effort in Reverberant Rooms: A Comparative Study of Subjective Perception and Objective Acoustic Metrics
items: Determination of the Elastic Constant of the Top Plate of a Cello in the Interaction with the Bridge
items: Single Vector Hydrophone DOA Estimation: Leveraging Deep Learning with CNN-CBAM
items: An IT System for Assessing Noise Level Distribution in Historical Urban Centers

In Memoriam Professor Eugeniusz Kozaczka
type: item
description: Professor Eugeniusz Tadeusz Kozaczka, member of the Polish Academy of Sciences, passed away on May 27, 2025, in Gdynia. We write these words with profound sorrow and deep regret, mourning the loss of such an exceptional person as Professor Eugeniusz Kozaczka.
subject: In Memoriam
  topic: section
creator: Bogumił LINDE
creator: Iwona KOCHAŃSKA
rights: Copyright (c) 2025 https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 419
ending page: 420
doi: 10.24425/aoa.2025.157220

Subwavelength Underwater Imaging of a Wire Array Metamaterial Based on Fabry–Perot Resonance
type: item
description: Metamaterials with Fabry–P´erot (FP) resonance have proven effective for underwater ultrasound imaging. The propagation phenomenon can be understood as a spatial filter with linear dispersion over a finite bandwidth. However, conventional imaging techniques are constrained by the diffraction limit or rely on a strong impedance mismatch between the metamaterial and water. In this paper, we propose a columnar array metamaterial designed for underwater imaging based on FP resonances and validate the proposed design through numerical simulations. The acoustic pressure transmission coefficient, together with the normalized acoustic pressure distribution, is analyzed to quantitatively evaluate imaging quality and verify the physical effectiveness of the model. This novel structure enables deep subwavelength imaging underwater, maintaining excellent and stable imaging performance within a 0.4 kHz bandwidth centered around the operating frequency. We use air-filled metamaterials to create strong acoustic coupling and establish effective sound isolation. This approach significantly enhances imaging resolution, while optimizing energy loss at multiple interfaces, an issue in previous studies. Additionally, in contrast to resonance- or refraction-based approaches such as Helmholtz resonators or hyperlens designs, the proposed FP-resonant metamaterial offers an alternative mechanism for achieving near-field subwavelength imaging through controlled wave transmission and confinement. We also examine the influence of various parameters, such as imaging distance, incidence distance, and array periodicity, on imaging performance. The results demonstrate that the columnar array metamaterial holds great potential for underwater ultrasound imaging applications.
subject: Research Papers
  topic: section
subject: Fabry–P´erot (FP) resonance
  topic: keywords
subject: metamaterial underwater imaging
  topic: keywords
subject: air-filled metamaterials
  topic: keywords
subject: finite element simulation
  topic: keywords
subject: underwater acoustics
  topic: disciplines
creator: Guo LI
creator: FeiLong LI
creator: LiQing HU
creator: QunFeng LI
creator: GuanJun YIN
rights: Copyright (c) 2025 Guo LI, FeiLong LI, LiQing HU, QunFeng LI, GuanJun YIN https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 477
ending page: 490
doi: 10.24425/aoa.2025.156930

Single-Sensor Passive Ranging of Underwater Monopoles Using Near-Field/Far-Field Energy Contrasts
type: item
description: While acoustic vector sensors (AVS) are well-established for detection and direction-of-arrival (DOA) estimation using co-located pressure and particle motion (PM) measurements, their potential for passive range estimation remains largely unexplored. This paper introduces a novel single-AVS method for passive range estimation to an acoustic monopole source by exploiting the fundamental near-field dominance of PM energy. We derive the frequency and the distance dependent ratio (ξ) of kinetic to potential acoustic energy density – a key near-field signature inaccessible to conventional hydrophones. By leveraging simultaneous AVS pressure and PM velocity measurements, our method estimates ξ, inverts the monopole near-field model to obtain the Helmholtz number, and directly computes the range. Crucially, we demonstrate that PM sensors offer a potential signal-to-noise ratio (SNR) advantage over pressure sensors within the near-field (>7.8 dB). Validation under simulated noise conditions shows accurate range estimation (RMSE <10 %) for low-frequency sources (<100 Hz) within 8 m–25m ranges at 0 dB SNRs, with performance degrading as frequency increases or SNR decreases. Critically, robustness is confirmed using recorded basin noise profiles, overcoming the isotropic Gaussian noise assumption. This technique extends AVS functionality beyond DOA, enabling single-sensor passive ranging without arrays, environmental priors, or reference signals where conventional methods fail.
subject: Research Papers
  topic: section
subject: monopole source
  topic: keywords
subject: passive ranging
  topic: keywords
subject: acoustic vector sensor (AVS)
  topic: keywords
subject: particle motion (PM)
  topic: keywords
subject: near-field acoustics
  topic: keywords
subject: underwater acoustics
  topic: keywords
subject: energy ratio
  topic: keywords
subject: single-sensor localization
  topic: keywords
creator: Saier MAHMOUD
creator: Louay SALEH
creator: Ibrahim CHOUAIB
rights: Copyright (c) 2025 Saier MAHMOUD, Louay SALEH, Ibrahim CHOUAIB https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 491
ending page: 499
doi: 10.24425/aoa.2025.154829

Analysis of Decision Fusion in Speech Detection
type: item
description: This article addresses the issue of detecting speech signal segments in an acoustic signal and analyzes potential decision fusion for a group of voice activity detectors (VADs). We designed ten new VADs using three different types of neural network architectures and three time-frequency signal representations. One of the proposed models has higher classification efficiency than competitive solutions. We used our VAD models to analyse data fusion and improve the final classification decision. For this purpose, we used gradient-free and gradient-based optimizers with different objective functions. The analysis revealed the impact of individual classifiers on the final decisions and the potential gains or losses resulting from VAD fusion. Compared with existing models, the models we proposed achieved higher classification accuracy at the cost of increased memory requirements. The final choice of a specific model depends on the platform constraints on which the VAD system will be deployed.
subject: Research Papers
  topic: section
subject: voice activity detection (VAD)
  topic: keywords
subject: deep neural networks
  topic: keywords
subject: data fusion
  topic: keywords
subject: other
  topic: disciplines
creator: Tomasz MAKA
creator: Lukasz SMIETANKA
rights: Copyright (c) 2025 Tomasz MAKA, Lukasz SMIETANKA https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 445
ending page: 454
doi: 10.24425/aoa.2025.156933

Potential Applications of Ultrasonic Parametric Array Loudspeakers (PALs) in Room Acoustic Measurements
type: item
description: In this paper, the potential use of parametric array loudspeakers (PALs) in acoustic measurements of the room is analysed, especially in the assessment of the effectiveness of reflective panels and intentionally angled surfaces. PALs are sound sources capable of emitting highly directional acoustic beams within the audible frequency range. Their operation is based on the emission of a high-frequency (ultrasonic) carrier modulated so that, through nonlinear demodulation in air, audible sound is generated. This process results in a narrow, focused sound beam, enabling precise acoustic emission. To explore PALs potential for acoustic measurement applications, the propagation behaviour of PAL-generated signals is first investigated under free-field conditions, focusing on how different surface types influence sound reflection. Subsequent experiments are carried out in a controlled indoor space, where impulse responses are recorded for various beam incidence angles and receiver positions. The collected data are used to generate sound-level distribution maps, allowing for the visualization and quantification of reflected sound coverage areas. The results show that PALs produce beams with substantially reduced lateral dispersion compared to conventional loudspeakers, enabling precise identification of reflection points and incidence angles. This directional precision makes it possible to accurately assess how effectively the reflective acoustic elements and structures shape the sound field within the room. Overall, these findings may contribute to optimising sound design in acoustically complex environments.
subject: OSA 2025
  topic: section
subject: parametric array loudspeakers (PALs)
  topic: keywords
subject: room acoustics
  topic: keywords
subject: ultrasonic
  topic: keywords
subject: reflection
  topic: keywords
subject: directivity
  topic: keywords
creator: Filip WĘGRZYN
creator: Adam PILCH
rights: Copyright (c) 2025 Filip WĘGRZYN, Adam PLICH https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 513
ending page: 524
doi: 10.24425/aoa.2025.156934

Simulation Analysis of Beam Intensity Attenuation Patterns and Source Depth Estimation Using a Vertical Long Line Array
type: item
description: In the deep-water reliable acoustic path (RAP), when estimating target depth using a vertical array, a large-aperture array can enhance the extraction of the acoustic field interference structure under low signal-to-noise ratio (SNR). However, this operation introduces slow envelope modulation (the envelope amplitude of peak beam intensity decreases with frequency) to the broadband acoustic field interference pattern, significantly degrading the performance of estimating the source depth. The Kraken normal-mode model can accurately calculate low-frequency sound fields in deep-water environments. This paper uses this tool to find that, in the deep-water direct arrival zone (DAZ), the peak beam intensity output of a vertical linear array varies across a broadband frequency range, exhibiting a pattern combining periodic changes of Lloyd’s mirror interference and inherent envelope attenuation changes. The physical mechanism of envelope attenuation is explained through both theoretical derivation and simulation analysis, key factors affecting the envelope-attenuation pattern are clarified, and the impact of beam-intensity envelope attenuation on the depth-estimation method based on matched beam intensity processing (MBIP) is pointed out. Based on this, a modified target depth estimation method of matched beam intensity processing (M-MBIP) that contains an attenuation coefficient is proposed, and its effectiveness is verified through simulated data.
subject: Research Papers
  topic: section
subject: deep-water direct arrival zone (DAZ)
  topic: keywords
subject: Lloyd’s mirror interference
  topic: keywords
subject: broadband attenuation pattern
  topic: keywords
subject: underwater acoustics
  topic: disciplines
creator: Hao WANG
creator: Guangying ZHENG
creator: Fangwei ZHU
creator: Xiaohong YANG
creator: Shuaishuai ZHANG
creator: Xiaowei GUO
rights: Copyright (c) 2025 Hao WANG, Guangying ZHENG, Fangwei ZHU, Xiaohong YANG, Shuaishuai ZHANG, Xiaowei GUO https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 501
ending page: 512
doi: 10.24425/aoa.2025.156931

Comparison of Methods for Determining the Airflow Resistivity of Porous and Covering Materials
type: item
description: This article compares two methods for determining the airflow resistivity of porous and coating materials – a key parameter in sound absorption modelling. The analysis involves a modified static airflow measurement procedure in accordance with International Organization for Standardization [ISO] (2018), using a linear approximation algorithm (PLA), and a reverse method consisting of matching the measured absorption coefficient in an impedance tube to the Miki model. The analysis was conducted on both porous materials utilised in acoustic panel fillings and thin coverings. It is evident that both methods yield analogous outcomes for materials exhibiting low resistivity. However, for materials characterised by higher resistivity, discrepancies of up to 50 % were observed. Nevertheless, a high degree of agreement was obtained between the calculated and measured absorption coefficients. For thin coating materials, an air gap of at least 70 mm is required. For materials with a thickness of up to approximately 30 mm, differences in resistivity do not significantly affect the absorption coefficient. It is evident that both methods can be used to determine the airflow resistivity of porous materials and layered structures, supporting the effective selection of materials according to requirements.
subject: Research Papers
  topic: section
subject: airflow resistivity
  topic: keywords
subject: specific airflow resistance
  topic: keywords
subject: sound absorption coefficient
  topic: keywords
subject: impedance tube
  topic: keywords
subject: porous materials
  topic: keywords
subject: other
  topic: disciplines
creator: Mykhaylo MELNYK
creator: Jarosław RUBACHA
creator: Artur FLACH
creator: Aleksandra CHOJAK
creator: Tadeusz KAMISIŃSKI
creator: Wojciech ZABIEROWSKI
creator: Marek IWANIEC
creator: Andryi KERNYTSKYY
rights: Copyright (c) 2025 Mykhaylo MELNYK, Jarosław RUBACHA, Artur FLACH, Aleksandra CHOJAK, Tadeusz KAMISIŃSKI, Wojciech ZABIEROWSKI, Marek IWANIEC, Andryi KERNYTSKYY https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 421
ending page: 433
doi: 10.24425/aoa.2025.154831

The Impact of Training Strategies on Overfitting in Vowel Classification Using PS-HFCC Parametrization for Automatic Speech Recognition
type: item
description: This paper investigates the overfitting problem in vowel classification task for automatic speech recognition (ASR). It utilizes a pitch synchronized human factor cepstral coefficients (PS-HFCC) as the parametrization method, which outperforms traditional methods like HFCC and mel-frequency cepstral coefficients (MFCC) in frame-level classification accuracy. While deep learning models are prevalent in contemporary ASR systems, they often lack explainability, a characteristic of classical classifiers. Therefore, this study examines overfitting phenomenon using a range of classifiers with well-understood properties. Specifically, it analyzes the impact of different training strategies on classifier performance, comparing the susceptibility to overfitting of several widely used classifiers, including the Gaussian mixture model (GMM), a standard approach in speech recognition. The analysis of training strategies considers various data splitting methods: random, speaker-based, and cluster-based. Our analysis of training strategies highlights the crucial role of data splitting methods: while random splitting is commonly used, it can lead to inflated accuracy due to overfitting. We demonstrate that speaker-independent splitting, where the classifier is trained on one set of speakers and tested on a separate, unseen set, is essential for robust evaluation and for accurately assessing generalization to new speakers. Potentially, the resulting insights may inform the future development and training of more reliable ASR systems.
subject: Research Papers
  topic: section
subject: automatic speech recognition
  topic: keywords
subject: vowel classification
  topic: keywords
subject: classifier training strategy
  topic: keywords
subject: pitch synchronized human factor cepstral coefficient
  topic: keywords
subject: overfitting
  topic: keywords
subject: robust parametrization
  topic: keywords
subject: speaker grouping
  topic: keywords
subject: speech processing and communication systems
  topic: disciplines
creator: Stanislaw GMYREK
creator: Urszula LIBAL
creator: Robert HOSSA
rights: Copyright (c) 2025 Stanislaw GMYREK, Urszula LIBAL, Robert HOSSA https://creativecommons.org/licenses/by/4.0
date: 2025-09-26
publication date: 2025-09-26
starting page: 371
ending page: 382
doi: 10.24425/aoa.2025.154823

Comparative Study of the Acoustic Efficiency of Prototype Sound Absorbing Panels Used in the Railway Track
type: item
description: The article presents the results of comparative studies concerning the efficiency of systems aimed at minimising the acoustic nuisance of noise generated by the railway vehicle movement. The issue of noise in railway traffic is a significant challenge, affecting both human health and the quality of life in the vicinity of railway lines. Prototype sound absorbing panels with varied surface geometry, a rubber slab, and ballast layer (stone aggregate, grain size 31.5/50 mm) were examined. Experiments were conducted in a reverberation chamber, analysing the response to broadband noise excitation. The reverberation chamber allows for obtaining repeatable results, eliminating the influence of external sound sources. It enables the assessment of the sound absorption properties of various materials which makes it possible to determine their effectiveness in noise reduction. The research methodology included measurements of reverberation time in different frequency bands for an empty chamber and a chamber containing the tested materials. The obtained differences in reverberation times provide information on the influence of the tested material on the distribution of acoustic energy in individual frequency bands. The research results allow for a preliminary assessment of the effectiveness of the tested materials in the task of reducing railway line noise.
subject: Research Papers
  topic: section
subject: railway traffic noise
  topic: keywords
subject: noise reduction
  topic: keywords
subject: acoustic insulation
  topic: keywords
subject: sound-absorbing materials
  topic: keywords
subject: vibration damping
  topic: keywords
subject: reverberation chamber
  topic: keywords
subject: noise and vibration
  topic: disciplines
creator: Grzegorz KLEKOT
creator: Mariusz WĄDOŁOWSKI
creator: Cezary KRAŚKIEWICZ
creator: Artur ZBICIAK
rights: Copyright (c) 2025 Grzegorz KLEKOT, Mariusz WĄDOŁOWSKI, Cezary KRAŚKIEWICZ, Artur ZBICIAK https://creativecommons.org/licenses/by/4.0
date: 2025-09-26
publication date: 2025-09-26
starting page: 331
ending page: 341
doi: 10.24425/aoa.2025.154817

Review of Microphone-Based Contactless Vital Signs Monitoring Systems
type: item
description: Microphones are sensors common to a variety of the Internet of Things (IoT) and healthcare applications. Many examples have proved that microphones can be useful in detecting, e.g., abnormal breathing rates. There are already applications that serve this purpose, e.g., respiratory acoustic monitoring, ResApp, etc. Breath signal was studied using a range of technologies and sensors, including the most common: radar, accelerometer, wearables, and so on. The majority of these sensors are attached to the body of a monitored person. However, the emergence of COVID-19 has drawn particular attention to the importance of using non-contact technologies for monitoring breath signals and other vital signs. This paper presents a comprehensive review of microphone-based non-contact vital sign monitoring, including the methodologies and concepts, while identifying new research gaps and opportunities for the future studies.
subject: Review Papers
  topic: section
subject: beamforming
  topic: keywords
subject: microphone
  topic: keywords
subject: machine learning
  topic: keywords
subject: vital signs
  topic: keywords
subject: acoustical measurements and instrumentation
  topic: disciplines
creator: Abiodun Ernest AMORAN
creator: Dariusz BISMOR
rights: Copyright (c) 2025 Abiodun Ernest AMORAN, Dariusz BISMOR https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 525
ending page: 541
doi: 10.24425/aoa.2025.154830

Experimental and Computational Analysis on the Acoustic Performance of Side Outlet Muffler with Semi-Circular Baffles
type: item
description: In this study, the impacts of the inclusion of two semi-circular baffles and their orientations on the acoustic performance of the side outlet muffler have been investigated. The side outlet muffler has a circular simple expansion chamber with an axial inlet and a side outlet and two semi-circular baffles that have been placed inside the expansion chamber at different orientations. The axis of the outlet is at the right angle to the axis of the inlet. The acoustical investigation of the side outlet muffler with two semi-circular baffles is done using the plane wave analysis, the finite element method (FEM), and the two-load technique. Based on the orientations of the two semi-circular baffles, three different models of side outlet muffler with semi-circular baffles have been investigated. The plane wave analysis, FEM, and two-load method are applied to all models and it is found that analytical, computational, and experimental transmission loss (TL) are in good agreement. The analytical modelling successfully predicts the presence of semi-circular baffles in the form of peaks and troughs in the TL of side outlet muffler with semi-circular baffles before the cut-off frequency and thus proves its effectiveness. Among all the models, model_2 gives 42 % higher TL than model_1 and model_3 shows 16.20 % higher TL than model_2. Hence, model_3 proves to be the best design for the side outlet muffler with semi-circular baffles in the attenuation of noise. The model_3 is effective for 1030 Hz–1480 Hz, 1500 Hz–1570 Hz, and 1640 Hz–2400 Hz frequency sound waves. The TL curve, sound pressure contours for model_3, and the band power variation in the 1/3 octave band for all the models have also been presented.
subject: Research Papers
  topic: section
subject: band power
  topic: keywords
subject: finite element method
  topic: keywords
subject: semi-circular baffles
  topic: keywords
subject: side outlet muffler
  topic: keywords
subject: sound pressure level
  topic: keywords
subject: transmission loss
  topic: keywords
subject: two-load method
  topic: keywords
subject: other
  topic: disciplines
creator: Sandeep Kumar VISHWAKARMA
creator: Suryappa Jayappa PAWAR
rights: Copyright (c) 2022 Sandeep Kumar VISHWAKARMA, Suryappa Jayappa PAWAR https://creativecommons.org/licenses/by/4.0
date: 2025-09-26
publication date: 2025-09-26
starting page: 355
ending page: 369
doi: 10.24425/aoa.2025.154815

Effect of Listener Head Position on Speech Intelligibility in an Automotive Cabin
type: item
description: When evaluating speech intelligibility (SI) in automotive cabins, binaural measurements typically employ a fixed dummy head. However, the impact of listener head positions on SI in nonuniform cabin sound fields remains unclear. This study analyzed SI under various listener head positions in an automotive cabin. An artificial mouth was regarded as the speaker, which was placed in three passenger positions. Binaural room impulse responses were measured using a dummy head in the driver’s seat with various head positions. The results show that head position significantly affects SI, with variations of up to 7 dB in octave band magnitudes, more than one just-noticeable difference in the speech transmission index, and shifts of up to 2.5 dB in the speech-reception threshold. SI variability depends on the speaker’s location. Directivity patterns play a crucial role in the front-passenger position, while seat occlusion affects SI at the back-right position, causing substantial decreases below a certain height threshold. At the back-left position, head positions close to the headrest enhance SI due to distance and reflections. Minor head displacements (4 cm apart) generally have insignificant effects on SI, except near seat obstructions or reach critical thresholds.
subject: Research Papers
  topic: section
subject: automotive cabin
  topic: keywords
subject: speech intelligibility
  topic: keywords
subject: head position
  topic: keywords
subject: speech reception threshold
  topic: keywords
subject: speech transmission index
  topic: keywords
subject: architectural
  topic: disciplines
subject: building and environmental acoustics
  topic: disciplines
creator: Linda LIANG
creator: Linghui LIAO
creator: Jiahui SUN
creator: Lingling LIU
creator: Liuying OU
creator: Xiaoyue HUANG
rights: Copyright (c) 2025 Linda LIANG, Linghui LIAO, Jiahui SUN, Lingling LIU, Liuying OU, Xiaoyue HUANG https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 455
ending page: 464
doi: 10.24425/aoa.2025.154828

Multi-label Bird Species Classification Using Transfer Learning Network
type: item
description: Bird sounds collected in the field usually include multiple birds of different species vocalizing at the same time, and the overlapping bird sounds pose challenges for species recognition. Extracting effective acoustic features is critical to multi-label bird species classification task. This work has extended an efficient transfer learning technique for labelling and classifying multiple bird species from audio recordings, further laying the foundation for conservation plans. A synthetic dataset was created by randomly mixing original single-species bird audio recordings from the Cornell Macaulay Library. The final dataset consists of 28 000 audio clips, each 5 s long, containing overlapping vocalizations of two or three bird species among 11 different species. Several pre-trained convolutional neural networks (CNNs), including InceptionV3, ResNet50, VGG16, and VGG19, were evaluated for extracting deep features from audio signals represented as mel spectrograms. The long short-term memory network (LSTM) was further employed to extract temporal features. A multi-label bird species classification was investigated. The absolute matching rate, accuracy, recall, precision, and F1-score of the InceptionV3+LSTM model for multi-label bird species classification are 98.25 %, 99.32 %, 99.41 %, 99.90 %, and 99.57 %, respectively, with the minimum Hamming loss of 0.0062. The results show that the proposed method has excellent performance and can be used for multi-label bird species classification.
subject: Research Papers
  topic: section
subject: transfer learning
  topic: keywords
subject: multi-label bird species classification
  topic: keywords
subject: InceptionV3
  topic: keywords
subject: LSTM
  topic: keywords
subject: bioacoustics
  topic: disciplines
creator: Xue HAN
creator: Jianxin PENG
rights: Copyright (c) 2025 Xue HAN, Jianxin PENG https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 223
ending page: 233
doi: 10.24425/aoa.2025.154812

Aeroacoustic Effect of Savonius Rotor Segmentation
type: item
description: Switching to renewable energy has been accelerated in recent decades due to the depleting fossil fuel reserves and the need to mitigate environmental and climate degradation. Wind power, especially in urban areas, has seen a significant growth. A critical consideration in the urban wind turbine installation is the noise impact on residents. This study investigates the noise generated by wind turbines under different operational conditions, comparing single-segment and five-segment rotor designs. Various acoustic analyses were conducted, including broadband analysis with weighting curves Z, A, C, and G, a narrowband analysis using 1/12 octave bands, and broadband calculations of sound quality indicators such as sharpness, roughness, and fluctuation strength (FS). The FS was also examined in the Bark scale frequency domain. The study linked the acoustic analysis with the rotor efficiency related to power production. The findings indicate that five-segment rotors generate less acoustic energy due to phase shifts, enhancing dissipation rates, and acoustic energy decreases with the increasing load, peaking when rotors are free at high revolutions per minute (RPM). While single-segment rotors show higher efficiency, they produce more noise. In contrast, five-segment rotors offer a better sound quality, making them preferable despite a lower efficiency. This research provides essential insights into designing urban wind turbines that balance efficiency and noise, crucial for sustainable energy solutions.
subject: Research Papers
  topic: section
subject: wind power
  topic: keywords
subject: wind turbines
  topic: keywords
subject: rotor efficiency
  topic: keywords
subject: Savonius rotor segmentation
  topic: keywords
subject: other
  topic: disciplines
creator: Shivangi SACHAR
creator: Piotr DOERFFER
creator: Paweł FLASZYŃSKI
creator: Józef KOTUS
creator: Krzysztof DOERFFER
creator: Joanna GRZELAK
creator: Michał PIOTROWICZ
rights: Copyright (c) 2025 Shivangi SACHAR, Piotr DOERFFER, Paweł FLASZYŃSKI, Józef KOTUS, Krzysztof DOERFFER, Joanna GRZELAK, Michał Piotrowicz https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 277
ending page: 288
doi: 10.24425/aoa.2025.153665

Structural Parametric Study of a Piezoelectric Energy Harvester for a Specific Excitation Frequency of an Electric Motor, Considering Fatigue Life
type: item
description: Over the past decade, extensive research has been conducted in the field of piezoelectric energy harvesting, which drives advancements in novel designs and techniques. In this study, the vibration of an electric motor is characterized, and a piezoelectric energy harvester (PEH) beam with a natural frequency of 50 Hz is designed and fabricated. Then, the electromechanical characteristics of the PEH are simulated and tested through both finite element simulation and experiment. The validated simulation model can accurately predict the vibration characteristics of the PEH, which can be utilized for design improvements. Based on this beam structure, three sets of PEHs with different sizes are designed. A study of the output voltage and fatigue life of these different PEH sizes is conducted, and the relationship between the electromechanical coupling effect and its varying values is discussed. Based on the results, design schemes 1∼6 demonstrate advantages in terms of output voltage efficiency and fatigue strength, making them suitable for various environments and application purposes. This study establishes an efficient method for analyzing the structural parametric performance of piezoelectric cantilever beams, which paves the way for future research on fatigue-based structural design guidelines for PEH in electric motors.
subject: Research Papers
  topic: section
subject: piezoelectric energy harvester (PEH)
  topic: keywords
subject: fatigue
  topic: keywords
subject: output voltage
  topic: keywords
subject: vibration
  topic: keywords
subject: noise and vibration
  topic: disciplines
creator: Ping YANG
creator: Nabil MOHAMAD USAMAH
creator: Abdullah Aziz SAAD
creator: Ahmad Zhafran AHMAD MAZLAN
rights: Copyright (c) 2025 Ping YANG, Nabil MOHAMAD USAMAH, Abdullah Aziz SAAD, Ahmad Zhafran AHMAD MAZLAN https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 265
ending page: 275
doi: 10.24425/aoa.2025.153664

Simulation and Experiment of a Medium-Distance Underwater Ultrasonic Wireless Power Transfer System
type: item
description: This paper focuses on ultrasonic wireless power transfer (UWPT), a form of underwater wireless power transfer (WPT) using acoustic (ultrasonic) waves. We propose an ultrasonic transducer designed based on the Langevin transducer model, along with its equivalent circuit model for underwater transmission, tailored for efficient medium-distance underwater WPT. The performance of this transducer is simulated and analyzed using the acoustic-piezoelectric structure module in COMSOL. The results demonstrate that the transducer exhibits excellent underwater transmission characteristics. Additionally, an equivalent circuit model of the underwater UWPT system is developed and analyzed to characterize its transmission properties. For validation, a prototype UWPT is fabricated. At a 35 cm separation between the transmitter and receiver, the transmitter power is 2.5 W. With a 1.2 kΩ load, the received root mean square (RMS) voltage is measured at 22.8 V, corresponding to a received power of 433mW and a transmission efficiency of 17 %. These results verify the proposed model, demonstrate a significant improvement in the transmission distance of underwater UWPT systems, and confirm the feasibility of medium-distance UWPT.
subject: Research Papers
  topic: section
subject: underwater
  topic: keywords
subject: wireless power transfer (WPT)
  topic: keywords
subject: ultrasonic
  topic: keywords
subject: ultrasonic transducer
  topic: keywords
subject: underwater acoustics
  topic: disciplines
creator: Chenyu LIAO
creator: Huaiqing ZHANG
creator: Zhi LI
creator: Qian ZHANG
rights: Copyright (c) 2025 Chenyu LIAO, Huaiqing ZHANG, Zhi LI, Qian ZHANG https://creativecommons.org/licenses/by/4.0
date: 2025-09-26
publication date: 2025-09-26
starting page: 391
ending page: 397
doi: 10.24425/aoa.2025.154821

Experimental Results of Diver Detection in Harbor Environments Using Single Acoustic Vector Sensor
type: item
description: This paper addresses the detection of divers with an open-circuit scuba. An acoustic vector sensor (AVS), which contains four channels, one for the pressure component, and three for orthogonal particle velocity components is proposed to be used. A novel covariance matrix analysis (CMA) method is presented for estimating the signal power using AVS in three-dimensional measurements. This method is based on solving a quartic equation that relates the determinant and trace of the AVS covariance matrix to the reciprocal of the signalto-noise ratio (SNR) in a three-dimensional isotropic acoustic field with spherical isotropic noise. This method is compared with two traditional methods: the AVS pressure channel power, and the minimum variance distortionless response (MVDR) beamformer, in estimating the acoustic power associated with the diver’s breathing. Experimental data from sea trials demonstrate the capability of all three methods to reconstruct the waveform of the acoustic diver signal and highlight the periodic breathing patterns. The diver’s breathing rate and corresponding power are estimated using the fast Fourier transform (FFT) of the power signal, therefore serving as a key signature for diver detection. The experiment demonstrates that the CMA method gives better diver detection index compared to the other two methods.
subject: Research Papers
  topic: section
subject: scuba diver
  topic: keywords
subject: acoustic vector sensor (AVS)
  topic: keywords
subject: detection
  topic: keywords
subject: minimum variance distortionless response (MVDR)
  topic: keywords
subject: signal-to-noise ratio (SNR)
  topic: keywords
subject: covariance matrix analysis (CMA)
  topic: keywords
subject: fast Fourier transform (FFT)
  topic: keywords
subject: underwater acoustics
  topic: disciplines
creator: Saier MAHMOUD
creator: Louay SALEH
creator: Ibrahim CHOUAIB
rights: Copyright (c) 2025 Saier MAHMOUD, Louay SALEH, Ibrahim CHOUAIB https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 173
ending page: 185
doi: 10.24425/aoa.2025.153663

CAPSE-ViT: A Lightweight Framework for Underwater Acoustic Vessel Classification Using Coherent Spectral Estimation and Modified Vision Transformer
type: item
description: Underwater acoustic target classification has become a key area of research for marine vessel classification, where machine learning (ML) models are leveraged to identify targets automatically. The major challenge is inserting area-specific understanding into ML frameworks to extract features that effectively distinguish between different vessel types. In this study, we propose a model that uses the coherently averaged power spectral estimation (CAPSE) algorithm. Vessel frequency spectra is first computed through the CAPSE analysis, capturing key machinery characteristics. Further, the features are processed via a vision transformer (ViT) network. This method enables the model to learn more complex relationships and patterns within the data, thereby improving the classification performance. This is accomplished by using self-attention mechanisms to capture global dependencies between features, enabling the model to focus on relationships throughout the entire input. The results, evaluated on standard DeepShip and ShipsEar datasets, show that the proposed model achieved a classification accuracy of 97.98 % and 99.19 % while utilizing just 1.90 million parameters, outperforming other models such as ResNet18 and UATR-Transformer in terms of both accuracy and computational efficiency. This work offers an improvement to the development of efficient marine vessel classification systems for underwater acoustics applications, demonstrating that high performance can be achieved with reduced computational complexity.
subject: Research Papers
  topic: section
subject: underwater acoustic targets
  topic: keywords
subject: CAPSE
  topic: keywords
subject: vision transformer
  topic: keywords
subject: CNN
  topic: keywords
subject: LOFAR gram
  topic: keywords
subject: underwater acoustics
  topic: disciplines
creator: Najamuddin NAJAMUDDIN
creator: Usman Ullah SHEIKH
creator: Ahmad Zuri SHA’AMERI
rights: Copyright (c) 2025 Najamuddin NAJAMUDDIN, Usman Ullah SHEIKH, Ahmad Zuri SHA’AMERI https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 161
ending page: 171
doi: 10.24425/aoa.2025.153662

Benchmarking the First Realistic Dataset for Speech Separation
type: item
description: This paper presents a thorough benchmarking analysis of a recently introduced realistic dataset for speech separation tasks. The dataset contains audio mixtures that replicate real-life scenarios and is accompanied by ground truths, making it a valuable resource for researchers. Although the dataset construction methodology was recently disclosed, its benchmarking and detailed performance analysis have not yet been conducted. In this study, we evaluate the performance of four speech separation models using two distinct testing sets, ensuring a robust evaluation. Our findings underscore the dataset’s efficacy to advance speech separation research within authentic environments. Furthermore, we propose a novel approach for assessing metrics in real-world speech separation systems, where ground truths are unavailable. This method aims to improve accuracy evaluations and refine models for practical applications. We make the dataset publicly available to encourage innovation and collaboration in the field.
subject: Research Papers
  topic: section
subject: single-channel
  topic: keywords
subject: speech separation
  topic: keywords
subject: deep learning
  topic: keywords
subject: corpus
  topic: keywords
subject: datasets
  topic: keywords
subject: speech processing and communication systems
  topic: disciplines
creator: Rawad MELHEM
creator: Oumayma AL DAKKAK
creator: Assef JAFAR
rights: Copyright (c) 2022 Rawad MELHEM, Oumayma AL DAKKAK, Assef JAFAR https://creativecommons.org/licenses/by/4.0
date: 2025-09-26
publication date: 2025-09-26
starting page: 383
ending page: 390
doi: 10.24425/aoa.2025.154816

Assessment of Noise from Different Types of Rounds Fired from a Saluting Gun
type: item
description: The difference in sound pressure levels between two types of rounds fired from a saluting gun has been investigated; the rounds being identified as ‘current’ and ‘new’. A 3-pounder saluting gun mounted on a concrete floor based at HMNB Portsmouth, UK, was used in the survey. Sound pressure levels were measured at the two people responsible for operating the gun: the firer and the loader. Twelve current rounds and 24 new rounds were fired during the survey. The new rounds showed a greater variation in peak sound pressure levels between rounds (interquartile range of 2.1 dB, firer’s location) compared with the current rounds (interquartile range of 1.1 dB, firer’s location). The highest C-weighted peak sound pressure levels for the firer were 173.1 dB for the current round compared with 166.8 dB for the new round. The corresponding highest C-weighted peak sound pressure levels for the loader were 170.6 dB and 163.0 dB, respectively. The difference between median peak sound pressure levels was 8.8 dB for the firer and 9.8 dB for the loader. Similar differences were measured in sound exposure levels between the two types of rounds. Frequency data presented can be used for assessing the suitability of appropriate hearing protectors. Mitigation measures are proposed for further reducing noise exposure of the operators.
subject: Research Papers
  topic: section
subject: peak sound pressure level
  topic: keywords
subject: 3-pounder round
  topic: keywords
subject: saluting gun
  topic: keywords
subject: blank ammunition
  topic: keywords
subject: acoustical measurements and instrumentation
  topic: disciplines
creator: Gurmail S. PADDAN
creator: Matt J. HOWELL
rights: Copyright (c) 2025 Gurmail S. PADDAN, Matt J. HOWELL https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 243
ending page: 252
doi: 10.24425/aoa.2025.154813

Comparison of Plate Vibration and Structural Sound Reduction Using Square-Based Sensor–Actuator Piezoelectric Hybrids with Different Shapes of Sensor Part
type: item
description: This work is a continuation of the author’s previous research on modeling a piezoelectric sensor–actuator hybrid. It presents the results of vibration and structural sound reduction for a plate with attached piezoelectric elements. The models consist of a steel plate with two piezoelectric actuators attached on one side and a hemispherical air volume on the other side. One of the actuators is used to excite the plate’s vibration and has the same shape and size in all models. The second actuator is used for vibration and structural sound reduction and varies between a standard square-based full actuator and a sensor–actuator hybrid with different sizes and shapes of the sensor component (either square- or disc-based). Harmonic analyses were performed for the first four mode shapes (skipping the third mode since it is a square plate). Optimization was performed using internal ANSYS functions, with the objective of minimizing the sum of displacement vectors at a number of nodes corresponding to either the full plate or the sensor placed on the said plate.
subject: Research Papers
  topic: section
subject: AVC
  topic: keywords
subject: FEM
  topic: keywords
subject: plate vibration
  topic: keywords
subject: sensors
  topic: keywords
subject: noise and vibration
  topic: disciplines
creator: Roman TROJANOWSKI
rights: Copyright (c) 2025 Roman TROJANOWSKI https://creativecommons.org/licenses/by/4.0
date: 2025-09-26
publication date: 2025-09-26
starting page: 343
ending page: 353
doi: 10.24425/aoa.2025.154822

Sensitivity Analysis of Acoustic Parameters in a Theatre Hall: A Case Study of the Maria Zankovetska Theatre in Lviv
type: item
description: This article explores the potential for modifying acoustic parameters within a theatre hall, using the Maria Zankovetska Theatre in Lviv, as a case study. The study used sensitivity analysis to evaluate how changes in the sound absorption of specific surfaces affect selected acoustic parameters in the horseshoe-shaped theatre hall. A numerical model of the hall is developed and calibrated based on in-situ acoustic measurements to assess the sensitivity of parameters such as reverberation time (T20), early decay time (EDT), clarity (C50), and early sound strength (G80). The analysis reveals that surfaces such as the stage tower ceiling, stage walls with curtains, audience walls, and seating have the most significant impact on the acoustic parameters. Modifying the sound absorption of these surfaces can affect T20, EDT, C50, and G80. Notably, increasing the absorption of a single surface might not significantly alter C50 and G80 values, whereas reducing the absorption of surfaces such as seating can lead to noticeable changes in these parameters. These findings provide valuable insights for future renovations and acoustic adjustments, aiming to optimize the theatre’s acoustic performance while preserving its historical character.
subject: Research Papers
  topic: section
subject: room acoustics
  topic: keywords
subject: sensitivity analysis
  topic: keywords
subject: sound absorption coefficient
  topic: keywords
subject: reverberation time
  topic: keywords
subject: other
  topic: disciplines
creator: Jarosław RUBACHA
creator: Adam PILCH
creator: Roman KINASZ
creator: Wojciech BINEK
creator: Tadeusz KAMISIŃSKI
creator: Mykhaylo MELNYK
rights: Copyright (c) 2025 Jarosław RUBACHA, Adam PILCH, Roman KINASZ, Wojciech BINEK, Tadeusz KAMISIŃSKI, Mykhaylo MELNYK https://creativecommons.org/licenses/by/4.0
date: 2025-09-26
publication date: 2025-09-26
starting page: 307
ending page: 320
doi: 10.24425/aoa.2025.154826

Sound Source Localisation in Digital Hearing Aids: A Review of Critical Factors
type: item
description: Sound source localisation is a fundamental ability for a listener’s functional interaction with the environment, yet it remains a significant challenge for hearing aid users. In general, they perform worse on spatial hearing tests than individuals with normal hearing. This narrative literature review examines the critical factors affecting sound source localisation when hearing aids are worn, with a focus on direction identification. We analysed peer-reviewed articles published over the past three decades to evaluate the impacts of the type of fitting, form factor, acoustic coupling, processing delay, bandwidth, directional microphones, and dynamic range compression on localisation ability. As a general conclusion, there is a consensus in the literature that binaural and open fittings, with microphones at the ear entrance and with extended bandwidth, significantly improve localisation performance. In addition, this review is intended to provide valuable information that can guide future innovations in hearing aid technology to improve users’ hearing experiences and their integration into the environment.
subject: Review Papers
  topic: section
subject: Hearing aids
  topic: keywords
subject: sound source localisation
  topic: keywords
subject: binaural hearing
  topic: keywords
subject: head-related transfer functions
  topic: keywords
subject: electroacoustics
  topic: disciplines
creator: Fermín SCALITI
creator: Diego A. EVIN
creator: Fabián C. TOMMASINI
rights: Copyright (c) 2025 Fermín SCALITI, Diego A. EVIN, Fabián C. TOMMASINI https://creativecommons.org/licenses/by/4.0
date: 2025-09-26
publication date: 2025-09-26
starting page: 399
ending page: 415
doi: 10.24425/aoa.2025.154825

The Impact of HVAC Systems on Speech Intelligibility in University Classrooms
type: item
description: Good speech intelligibility in university classrooms is crucial to the learning process, ensuring that students can clearly hear all conversations taking place in the classroom. While it is well known that speech intelligibility depends on the geometrical characteristics of a space and the properties of its surfaces, other factors need also to be considered. Among the most important are: the heating, ventilation, and air conditioning (HVAC) systems used in classrooms. Fan noise from HVAC systems increases the background noise level (BNL), negatively affecting speech intelligibility. In addition, the movement of air caused by these systems alters room acoustic variables. Although this dynamic situation is often overlooked in the early design stages, HVAC systems are often active during lectures and influence acoustics variables, especially the speech transmission index (STI). In this study, the impact of HVAC systems on the STI was measured in five different unoccupied classrooms in the Rafet Kayıș Faculty of Engineering at Alanya Alaaddin Keykubat University. The results were evaluated according to relevant standards. The results of these evaluations offer insights for researchers, architects, and engineers working in the field of acoustics.
subject: Research Papers
  topic: section
subject: speech intelligibility
  topic: keywords
subject: speech transmission index (STI)
  topic: keywords
subject: room acoustic variables
  topic: keywords
subject: room impulse response (RIR)
  topic: keywords
subject: acoustic performance
  topic: keywords
subject: other
  topic: disciplines
creator: Akın OKTAV
creator: Arda ENER
creator: Faruk ÖZENÇ
creator: Sıla SARI
rights: Copyright (c) 2025 Akın OKTAV, Arda ENER, Faruk ÖZENÇ, Sıla SARI https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 253
ending page: 263
doi: 10.24425/aoa.2025.153661

A Hybrid Method for Predicting the Bistatic Target Strength Based on the Monostatic Scattering Sound Pressure Data
type: item
description: To predict the scattered acoustic field for underwater targets with separate transmission and reception points, a forecasting method based on limited scattered acoustic pressure data is proposed. This method represents the scattered acoustic field as the product of n acoustic scattering transfer function and a source density function. By performing numerical integration, the transfer function is obtained using the model surface grid information as input. An equation system concerning the unknown source density function is then derived using the computed scattering transfer matrix, the principle of acoustic reciprocity, and the geometric properties of the target. The unknown source density function is solved using the least squares method. The scattered field with separate transmission and reception points is then obtained by multiplying the calculated transfer matrix with the estimated source density function. This paper applies the finite element method (FEM) to solve the scattering field for a benchmark model with separate transmission and reception points. Using a subset of the elements as input, predictions of the omnidirectional scattered field were made. The predicted results were subsequently compared with those obtained from FEM simulations. The simulation results demonstrate that the proposed method maintains high computational accuracy and is applicable to the prediction of low-frequency scattered fields from underwater targets with spatially separated source and receiver. Further comparison with the FEM-calculated target strength patterns across varying incident–reception angles reveals a high level of agreement, indicating that accurate bistatic target strength predictions can be achieved with a limited amount of input data.
subject: Research Papers
  topic: section
subject: acoustic scattering properties
  topic: keywords
subject: bistatic scattering sound field
  topic: keywords
subject: scattering sound field prediction
  topic: keywords
subject: target strength
  topic: keywords
subject: underwater acoustics
  topic: disciplines
creator: Yuhang TANG
creator: Chenyu FAN
creator: Yinhao LI
creator: Changxiong CHEN
creator: Zilong PENG
rights: Copyright (c) 2022 Yuhang TANG, CHenyu FAN, Yinhao LI, Changxiong CHEN, Zilong PENG https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 199
ending page: 207
doi: 10.24425/aoa.2025.154814

Localization of Virtual Sound Source Reproduced by the Crosstalk Cancellation System Under Different Reflective Conditions
type: item
description: This study explores the localization of virtual sound source reproduced by the crosstalk cancellation system under different reflective conditions in virtual rooms and analyzes the localization results with binaural cues. Binaural room impulse responses are generated using the high-order image source method. By modifying the acoustic parameters of the virtual room to manipulate the intensity and temporal structure of the reflection, psychoacoustic experiments were conducted using headphone reproduction. The experimental results indicate that, changes in reflection intensity within a certain range by altering the room reverberation time (RT) do not cause noticeable variations in virtual source localization. Increasing the loudspeaker–listener distance (changing temporal structure of reflections) deteriorates localization performance. The primary distinction between variations in the loudspeaker–listener distance and RT lies in whether the temporal structure of the reflection component changes. Overall, the study highlights the importance of the reflection temporal structure in the virtual source localization. The analysis of binaural cues indicates that, even in reverberant environments, the interaural time difference exhibits greater consistency with localization than the interaural level difference.
subject: Research Papers
  topic: section
subject: sound localization
  topic: keywords
subject: crosstalk cancellation
  topic: keywords
subject: reflection environment
  topic: keywords
subject: binaural cues
  topic: keywords
subject: psychoacoustics
  topic: disciplines
creator: Wei TAN
creator: Guangzheng YU
creator: Jun ZHU
creator: Dan RAO
rights: Copyright (c) 2025 Wei TAN, Guangzheng YU, Jun ZHU, Dan RAO https://creativecommons.org/licenses/by/4.0
date: 2025-11-25
publication date: 2025-11-25
starting page: 465
ending page: 476
doi: 10.24425/aoa.2025.154827

Listening Effort in Reverberant Rooms: A Comparative Study of Subjective Perception and Objective Acoustic Metrics
type: item
description: Modern room acoustics employs a variety of objective measures to characterize the acoustical properties of interiors. Despite these advancements, the relationship between these parameters and subjective assessments of room acoustics remains unclear. Subjective perception, particularly listening effort (LE), plays a critical role in how individuals experience acoustic environments, even when speech intelligibility (SI) is high. This study aims to bridge the gap between objective acoustic measures and subjective listening experiences. We conducted experiments in three rooms equipped with reverberation enhancement systems, resulting in nine different acoustic settings. Objective parameters, including reverberation time (RT), early decay time (EDT), clarity (C50), and the speech transmission index (STI), were measured. Additionally, subjective SI was assessed, and LE was rated on a 7-step Likert scale by 180 volunteers with normal hearing. The analysis revealed a nonlinear relationship between LE and both RT20 and EDT (R2 = 0.6), with an even weaker correlation for LE vs. C50 (R2 = 0.46). The Pearson correlation coefficient for STI was 0.74, compared to 0.55 for SI. These findings indicate that the relationship between LE and objective parameters, as well as SI, is complex and not straightforward. Our results suggest the importance of incorporating LE into room acoustic design and evaluation. The disparity between objective measures and subjective experiences suggests that LE may be a crucial factor in accurately assessing acoustic environments. This approach sheds the light on a more holistic understanding of acoustic quality that prioritizes human perception.
subject: Research Papers
  topic: section
subject: room acoustics
  topic: keywords
subject: listening effort
  topic: keywords
subject: speech intelligibility
  topic: keywords
subject: reverberation time
  topic: keywords
subject: early decay time
  topic: keywords
subject: clarity (C50)
  topic: keywords
subject: speech transmission index
  topic: keywords
subject: psychoacoustics
  topic: disciplines
creator: Anna PASTUSIAK
creator: Łukasz BŁASIŃSKI
creator: Jędrzej KOCIŃSKI
rights: Copyright (c) 2025 Anna Pastusiak, Łukasz BŁASIŃSKI, Jędrzej KOCIŃSKI https://creativecommons.org/licenses/by/4.0
date: 2025-09-26
publication date: 2025-09-26
starting page: 321
ending page: 329
doi: 10.24425/aoa.2025.154824

Determination of the Elastic Constant of the Top Plate of a Cello in the Interaction with the Bridge
type: item
description: This paper aims to determine the equivalent static elastic constant of a cello’s top plate in the interaction with the bridge. Experimental results detailing this constant are presented based on measuring the deformation and forces caused by a system of calibrated springs in similar conditions to that obtained when these forces are produced by the action of the strings. Subsequent tests are conducted following an intervention by a luthier to adjust the sound post, with the aim of assessing the impact on the elastic constants.
subject: Research Papers
  topic: section
subject: elasticity
  topic: keywords
subject: organology
  topic: keywords
subject: cello bridge
  topic: keywords
subject: musical acoustics
  topic: keywords
subject: boundary conditions
  topic: keywords
subject: interferometry
  topic: keywords
subject: other
  topic: disciplines
creator: Pablo PAUPY
creator: Pablo TABLA
creator: Dario HUGGENBERGER
creator: Federico ELFI
creator: Eneas N. MOREL
creator: Jorge R. TORGA
rights: Copyright (c) 2025 Pablo PAUPY, Pablo TABLA, Dario HUGGENBERGER, Federico ELFI, Eneas N. MOREL, Jorge R. TORGA https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 235
ending page: 242
doi: 10.24425/aoa.2025.153658

Single Vector Hydrophone DOA Estimation: Leveraging Deep Learning with CNN-CBAM
type: item
description: In recent years, single vector hydrophones have attracted widespread attention in target direction estimation due to their compact design and advantages in complex underwater acoustic environments. However, traditional direction of arrival (DOA) estimation algorithms often struggle to maintain high accuracy in nonstationary noise conditions. This study proposes the novel DOA estimation method based on a convolutional neural network (CNN) and the convolutional block attention module (CBAM). By inputting the covariance matrix of the received signal into the neural network and integrating the CBAM module, this method enhances the model’s sensitivity to critical features. The CBAM module leverages channel and spatial attention mechanisms to adaptively focus on essential information, effectively suppressing noise interference and improving directional accuracy. Specifically, CBAM improves the model’s focus on subtle directional cues in noisy environments, suppressing irrelevant interference while amplifying essential signal components, which is crucial for an accurate DOA estimation. Experimental results under various signal-to-noise ratio (SNR) conditions validate the method’s effectiveness, demonstrating superior noise resistance and estimation precision, providing a robust and efficient solution for underwater acoustic target localization.
subject: Research Papers
  topic: section
subject: single vector hydrophone
  topic: keywords
subject: direction of arrival (DOA)
  topic: keywords
subject: convolutional neural network (CNN)
  topic: keywords
subject: convolutional block attention module (CBAM)
  topic: keywords
subject: noise resistance
  topic: keywords
subject: underwater acoustics
  topic: disciplines
creator: Fanyu ZENG
creator: Yaning HAN
creator: Hongyuan YANG
creator: Dapeng YANG
creator: Fan ZHENG
rights: Copyright (c) 2025 Fanyu ZENG, Yaning HAN, Hongyuan YANG, Dapeng YANG, Fan ZHENG https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 187
ending page: 198
doi: 10.24425/aoa.2025.153659

An IT System for Assessing Noise Level Distribution in Historical Urban Centers
type: item
description: This article explores the challenge of identifying noise-generating factors in traffic flows (TFs) within the constrained spaces and imperfect transport networks of historical cities, using Lviv as a case study. Experimental studies were conducted to measure the equivalent noise levels at different times of the day on selected streets in Lviv. These streets are characterized by dense development, paved surfaces, and a high volume of vehicular and rail traffic. The study identified correlations between noise levels, traffic volumes, and vehicle speeds during daytime and nighttime periods. Notably, vehicle speed was found to have a more significant impact on noise levels than the number of vehicles. Through the analysis of these findings, empirical mathematical models were developed and validated using the Lagrange interpolation polynomial to predict noise pollution levels on selected streets at specific times. The developed computer system enables quick forecasting of noise levels for a given street while simultaneously provides data to manage TF as a factor affecting noise generation. Crucially, this tool can also assist in calculating the required specifications for acoustic insulation on building façades adjacent to these TFs.
subject: Research Papers
  topic: section
subject: IT system
  topic: keywords
subject: noise measurement
  topic: keywords
subject: road noise
  topic: keywords
subject: MATLAB
  topic: keywords
subject: equivalent noise level
  topic: keywords
subject: architectural
  topic: disciplines
subject: building and environmental acoustics
  topic: disciplines
creator: Mykhaylo MELNYK
creator: Tadeusz Marek KAMISIŃSKI
creator: Roman KINASZ
creator: Roman VYNAROVYCH
rights: Copyright (c) 2025 Mykhaylo MELNYK, Tadeusz Marek KAMISIŃSKI, Roman KINASZ, Roman VYNAROVYCH https://creativecommons.org/licenses/by/4.0
date: 2025-06-27
publication date: 2025-06-27
starting page: 215
ending page: 221
doi: 10.24425/aoa.2025.153660