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: 1.6 (2023: 1.8) | SJR 2024: 0.233 (2023: 0.239) | SNIP 2024: 0.597 (2023: 0.570)
Institute of Fundamental Technological Research (IPPT) of the Polish Academy of Sciences
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
Indian Sign Language Alphabet Recognition and Speech Synthesis Using a Hybrid Deep Learning Approach
Indian Sign Language (ISL) is vital for communication among India's hearing-impaired community. However, the lack of standardised datasets and reliable identification frameworks has hampered the use of ISL in modern assistive technology. This paper presents a deep learning-based solution to robust ISL alphabet identification, with an emphasis on both accuracy and practical use. A curated static ISL alphabet collection was created by combining authoritative visual references from the official Indian Sign Language website and the Ramakrishna Mission Vivekananda Educational and Research Institute (RKMVERI). Multiple deep learning models were trained and assessed, including CNN, ResNet-50, DenseNet-121, VGG16, MobileNetV2, and EfficientNet-B0, with a new hybrid CNN-ResNet architecture outperforming the others. 98% classification accuracy is achieved by the suggested approach, outperforming individual baseline models. Furthermore, the framework is expanded to support real-time applications, combining webcam-based capture with immediate conversion of recognized signs to textual and synthesized vocal output. Comprehensive performance evaluation, including confusion matrix analysis and ROC curves, demonstrates the solution's durability and practical applicability. This research enhances accessibility, promotes inclusive education, and prepares the path for scalable sign language translation systems in real-world human-machine interaction scenarios by enabling accurate and real-time ISL recognition with voice feedback.
Copyright (c) 2025 Aswani Sivan, Chandra Eswaran https://creativecommons.org/licenses/by/4.0
Flanking Sound Transmission in Massive-Lightweight Connections
Assessing the impact of flanking sound transmission is one of the most significant challenges in the process of designing building partitions. Acoustic parameters declared by manufacturers of lightweight systems are subject to errors of up to several decibels – and in the case of inaccurate construction on site, these differences can reach even higher values. One factor contributing to this is the phenomenon called flanking sound transmission, which involves the transmission of acoustic energy through partitions connected to a partition directly dividing two adjacent rooms. For this reason, estimating the resultant acoustic insulation of a partition, taking into account the flanking paths, is crucial early in the design process to ensure compliance with the requirements outlined in standard recommendations and literature. Currently, there are regulations and studies that provide guidance on calculating the estimated reduction in acoustic insulation due to flanking transmission. However, in practice, situations arise that have not yet been addressed in standards or literature. Examples include partitions made of plasterboard, which are among the most common types of partition walls in Poland, yet are not covered by current normative procedures, as well as glass systems. This study aims to explore this topic further by analysing the impact of combining a massive partition with flanking lightweight partitions for selected structures (glass, plasterboard with single or double panelling, with full or partial sound-absorbing material infill, and without infill) and connection types.
Copyright (c) 2025 Agnieszka Wójtowicz, Tadeusz Kamisiński, Jarosław Rubacha https://creativecommons.org/licenses/by/4.0
Study on Dual-Stage Amplification Cascaded Piezoelectric Transducer for High-Power Applications
To reduce the size and enhance the efficiency of cascaded sandwich transducers with conical horns, a novel structural configuration of such transducers was investigated. This transducer incorporates two sets of piezoelectric stacks, enabling two-stage amplification for improved efficiency. An equivalent circuit model for the cascaded sandwich transducer with a conical horn was established, systematically deriving analytical expressions for core performance parameters including input impedance, velocity amplification ratio, and resonant characteristics. Through theoretical and simulation analyses, the dynamic influence of key structural parameters on electromechanical energy conversion efficiency was determined, specifically the output radius of the second-stage, the relative position of the variable cross-section of two sets of piezoelectric ceramic sandwich structures, and the spacing between the two sets of piezoelectric ceramics. Furthermore, a performance optimization strategy based on piezoelectric single-crystal materials was proposed. Numerical simulations validated against theoretical models revealed the governing principles of piezoelectric materials on transducer performance. Experimental results demonstrate excellent agreement between the operational characteristics of the optimized transducer and predictions from both theoretical models and finite element simulations. This work holds guiding significance for optimizing multi-mode transducers and demonstrates promising application potential in high-power ultrasonic fields.
Copyright (c) 2025 Guo Li, Peiyu Tan, Ruihui Ma, Feilong Li, Xiaoli Zhang, Hua Tian https://creativecommons.org/licenses/by/4.0
In Memoriam Professor Eugeniusz Kozaczka
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.
Copyright (c) 2025 https://creativecommons.org/licenses/by/4.0
Analysis on Modal Distribution and Modal Density-Based Crossover Frequency in Cabin-Sized Enclosures
In the sound field simulation of cabin-sized enclosures, the Schroeder frequency (SF) is still employed to estimate the crossover frequency (CF) that determines the validity ranges of wave-based and geometrical acoustic methods. However, because cabin-sized enclosures exhibit distinct modal behaviors from typical medium- and large-scale rooms, the validity of SF in such enclosures has not been thoroughly tested. This study introduces the modal density-based crossover frequency (MDCF) to systematically evaluate the applicability of SF in cabin-sized enclosures. The MDCF employs the same dense modal criterion as SF. However, its modal parameters, are derived from the numerical eigenfrequency analysis. This contrasts with the SF formula, where these parameters are determined solely by the room volume and reverberation time. Ten models are constructed for evaluation, grouped into two volume sets: 8 m3 (cabin-sized) and 80 m3 (common-sized). Each set comprises five distinct geometrical shapes from rectangular models to simplified vehicle shapes. The results reveal that, for cabin-sized enclosures under low absorption boundary conditions, the MDCF is typically 70 Hz to 150 Hz lower than SF; the discrepancies decrease to 20 Hz to 50 Hz in 80 m3 rooms. Furthermore, the MDCF varies with room shapes at a constant volume, while the SF remains nearly unchanged. These findings demonstrate that MDCF provides a more reliable CF estimation for rooms with irregular shapes, highlighting the importance of considering accurate modal parameters in the acoustic analysis of cabin-sized models.
Copyright (c) 2026 Ziyu Wang, Liangfen Du, Guangzheng Yu https://creativecommons.org/licenses/by/4.0
TimeGAN and Coordinated Attention Prototype Network Based Prediction Model for Infrasound Signal
Due to the complexity of the infrasound environment and the high costs associated with data collection, frequent acquisition of infrasound data is often impractical, resulting in a limited amount of labeled data. To address the challenge of low classification prediction accuracy caused by data scarcity, this paper proposes an infrasound prediction model based on a time-series generative adversarial network (TimeGAN) and coordinated attention prototype network (CAPN) (TimeGAN-CAPN). The model begins by introducing TimeGAN, where the generative network is trained using a combination of unsupervised and supervised learning. This approach enables the network to operate within the latent space of temporal features and generate time-series data that closely aligns with the distribution of the original data. These generated samples are then combined with the original data to form an augmented dataset. Subsequently, the augmented data is input into the CAPN, which enhances the sample size per class, allowing for more precise class prototypes and improving the prediction accuracy of the model. Furthermore, the quality and diversity of the data generated by TimeGAN are quantitatively and qualitatively assessed using maximum mean discrepancy (MMD) and t-istributed stochastic neighbor embedding (t-SNE), facilitating a comparison and verification of the generated data’s performance. Experimental results show that TimeGAN-CAPN significantly outperforms the CAPN model in classification tasks with limited infrasound data, achieving an increase in accuracy of 7.15 %. This demonstrates that the proposed method is highly effective for predicting infrasound-related disasters, particularly in scenarios with small sample sizes.
time-series generative adversarial network
Copyright (c) 2026 Quanbo Lu, Xiaojuan Huang, Rao Li, Mei Li, Dong Zhu https://creativecommons.org/licenses/by/4.0
An Empirical Approach to Investigate Environmental Effects on Acoustic Signal Speed in Oceanic Layers
This paper investigates and demonstrates the effects of three significant environmental contributors: temperature, depth and salinity impact on the acoustic signal propagation across distinctive ocean layers: Mixed, Thermocline and Deep Layers. In the field of underwater wireless sensor networks (UWSN), exact and precise determination of coordinates for sensor localization is very crucial for data validation. Temperature dominates the upper layers; depth becomes prime factors for deeper domain with minimal thermal variations. Salinity while having a diminished effect, facilitates to finer alterations in propagation and deviation of acoustic signal speed. In our work we have analyzed these interdependences by using different empirical models (e.g. Mckenzie, Medwin) customized to each layer, accounting to their incomparable environmental parameters. In Mixed layers sound speed variation are mainly thermal driven, where depth is minimal important and salinity effect is negligible but as we go deeper the temperature starts to fall and depth (pressure) started getting importance and also salinity and temperature variation almost become corresponding. By evaluating ocean layer specified empirical formulas, we have calculated average speed of sound and measure the respective contribution of all parameters. Our work has provided a sub structure which will help to optimize UWSN nodes identification or localization. The results of this work underscored the essential to have an adaptive sound speed modeling for enhanced and precise acoustic signal communication systems.
Copyright (c) 2025 Khondoker Munim Salehin, Md. Khalid Mahbub Khan, Anisur Rahman https://creativecommons.org/licenses/by/4.0
Ultrasonic Estimation of Pressure Dependent Non-Linearity Index in Liver
This study introduces a proof-of-concept methodology for evaluating pressure-dependent non-linear acoustic properties of liver tissue. The proposed non-linearity index (NLI) is derived from echo amplitudes obtained at two substantially different acoustic pressures. Unlike previous harmonic-based approaches, the method relies solely on the fundamental frequency band, allowing clinical implementation without additional system modifications. The image acquired for the lower pressure is then amplified to correct for the pressure difference between the beams. Next, the NLI is estimated as a ratio of local amplitudes of the amplified low-pressure image (ALPI) to the high-pressure image (HPI). In the case of nonlinear media some energy of the wave is transferred from the pulse fundamental frequency to higher harmonics, which affects mainly the HPI. With the harmonics being filtered out from the signal, the HPI amplitude becomes lower than the ALPI amplitude. As a result, the NLI becomes higher than 1 and increases with the non-linearity of the imaged tissue. The hydrophone measurements were compared to the simulation (k-Wave) of the ultrasonic field in water and vegetable oil. Next, we performed NLI imaging of healthy and fatty livers using SonixTouch (Ultrasonix) systems and two acoustic pressures of 390 kPa and 1590 kPa. Preliminary studies – imaging healthy and fatty livers using SonixTouch (Ultrasonix) systems were performed on the 4 livers of the authors of the article showed that for ‘healthy’ livers the NLI was below 1.1, while in one of the authors with previously diagnosed steatosis falling between score 1 and 2, the NLI locally exceeded 1.3. These results show that the obtained NLI values increase with the degree of steatosis, which agrees with theoretical expectations based on tissue B/A coefficients. The work emphasizes methodological feasibility and physical consistency rather than clinical validation, given the limited number of volunteers and ethical restrictions on patient recruitment.
Copyright (c) 2025 Andrzej Nowicki, Jurij Tasinkiewicz, Piotr Karwat, Norbert Żołek, Ihor Trots, Ryszard Tymkiewicz https://creativecommons.org/licenses/by/4.0
Application of Acoustic Metamaterials in the Design of Muffling Unit of Internal Combustion Engine
Aiming at the problems of large volume, high exhaust resistance and difficulty in suppressing noise in the 500 Hz-100 Hz frequency band of traditional internal combustion engine exhaust mufflers, a noise reduction unit design based on acoustic metamaterials is proposed. Based on the equivalent medium theory, an acoustic model with a ring structure and multi-region variable refractive index was established. Phase control is achieved by helically winding the acoustic channel to change the refractive index, and the basic dimensions of the acoustic metamaterials muffling unit are calculated. The sound field distribution, transmission loss and flow field characteristics of the muffling unit are simulated and analyzed. This structure utilizes a multi-layer acoustic channel structure, effectively alleviating the problem of insufficient low-frequency noise elimination caused by the asymmetry of Fano interference. It achieved a transmission loss of over 10 dB within 85% of the 500 Hz to 1000 Hz frequency band, and still maintained excellent noise reduction performance under high-frequency conditions through multi-level phase control. By connecting multiple units in series, a transmission loss of 10 dB can be achieved within 85% of the 500 Hz to 1000 Hz frequency band. The exhaust flow field of the muffling unit was simulated and analyzed. Whether used alone or in series with the traditional muffling structure, the exhaust resistance remained within the range of 360 Pa to 370Pa. Experimental tests show that when the metamaterial muffler unit is used in combination with the traditional muffler, it effectively achieves targeted noise elimination in the 500 Hz-1000 Hz frequency band, and also demonstrates clear noise reduction capabilities in higher frequency ranges. The high noise suppression characteristics, high gas passage characteristics and compact volume characteristics of this structure provide more potential analysis methods and design schemes for the research and development of internal combustion engine mufflers and noise reduction accessories.
Copyright (c) 2025 Tianshuo Li, Lei Zheng, Zhengyang Bi, Hongchao Ji https://creativecommons.org/licenses/by/4.0
Subwavelength Underwater Imaging of a Wire Array Metamaterial Based on Fabry–Perot Resonance
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.
Copyright (c) 2025 Guo LI, FeiLong LI, LiQing HU, QunFeng LI, GuanJun YIN https://creativecommons.org/licenses/by/4.0
Application of ISO 12913 Standard to Assess Urban Soundscapes: A Case Study on Poznań
ISO 12913 standards provide a unified framework for describing and assessing soundscapes, yet the absence of a Polish translation has so far limited their practical use. This paper presents the first application of a validated Polish version of the ISO 12913-2 perceptual attributes, enabling full cross-language comparability of results. Whereas Polish research has traditionally focused on noise annoyance and broad judgements of acoustic comfort or discomfort, we outline the complete ISO-compliant assessment procedure, which combines: a soundwalk, questionnaires and audio-visual recording. The study was conducted at eight diverse urban locations in Poznań, Poland. Participants rated the soundscapes using eight attributes: przyjemne, tętniące życiem, bogate w wydarzenia, chaotyczne, dokuczliwe, monotonne, ubogie w wydarzenia, spokojne. Each rating set is mapped to a point in the two-dimensional pleasantness-eventfulness space defined in ISO 12913-3, facilitating visual comparison of locations and the identification of design needs. Results reveal pronounced perceptual differences between spatial typologies and demonstrate that the standardized approach provides richer, multidimensional information about the acoustic environment than conventional noise indicators. The proposed methodology establishes a reference framework for Polish soundscape studies and can support the creation of more people-friendly urban acoustic environments.
Copyright (c) 2025 Jakub Dumanowski, Anna Preis, Jan Felcyn https://creativecommons.org/licenses/by/4.0
Single-Sensor Passive Ranging of Underwater Monopoles Using Near-Field/Far-Field Energy Contrasts
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.
Copyright (c) 2025 Saier MAHMOUD, Louay SALEH, Ibrahim CHOUAIB https://creativecommons.org/licenses/by/4.0
Analysis of Decision Fusion in Speech Detection
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.
Copyright (c) 2025 Tomasz MAKA, Lukasz SMIETANKA https://creativecommons.org/licenses/by/4.0
Active Vibration Suppression of a Thin Circular Pipe
The article presents an active vibration damping system for a thin-walled cylindrical tube, which is a simplified model of a lightweight robot arm (LWR – Lightweight Robot). The proposed solution integrates control algorithms, piezoelectric materials and a hardware and software environment enabling real-time control. Macro Fiber Composite (MFC) elements were used for active vibration reduction, acting simultaneously as sensors and actuators. The object on which the research was conducted was a tube with an external diameter of 40 mm, this element was rigidly mounted at a distance of 1 meter from the free end, simulating cantilever conditions. The stimulation of the object to vibration was carried out using the MFC actuator, while the system response was recorded in the xPC Target environment. Based on the measurement data, the mathematical model of the object was identified in the discrete domain using the ARX method. The obtained model was used to design a controller based on the pole location method, which was implemented on a real test stand. The experimental results showed the effectiveness of the designed control system in reducing the amplitude of natural vibrations of the structure. The use of MFC elements as sensor elements and actuators enabled effective vibration damping in real time, confirming the usefulness of the proposed solution in the context of improving the precision of robotic systems.
Copyright (c) 2025 Marcin Pater, Lucyna Leniowska, Marcin Grochowina https://creativecommons.org/licenses/by/4.0
Potential Applications of Ultrasonic Parametric Array Loudspeakers (PALs) in Room Acoustic Measurements
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.
Copyright (c) 2025 Filip WĘGRZYN, Adam PLICH https://creativecommons.org/licenses/by/4.0
Cube-Shaped Sound-Insulating Enclosures: Experimental Tests and Calculation Models
The research described in this article concerns sound-insulating enclosures used for sound sources imitating a noisy machine or device. It is a continuation of experimental tests and modelling studies previously conducted on a prototype test stand, in which the enclosure walls measured 0.7 m × 0.7 m. The main aim of the current research was to estimate the acoustic efficiency of the enclosures through experimental testing on a new stand with walls measuring 0.55 m × 0.55 m, conducted under conditions similar to those found in an industrial facility. Tests conducted on five wall types of varying thicknesses, made of materials such as steel, aluminium, and plexiglass, enabled the development of a calculation model for insertion loss (IL), which can be used based on the material data for the enclosure walls. The model was validated through further experimental tests covering four additional material variants, and a high correlation of the results was obtained. The influence of the calculation model used for the enclosure wall transmission loss on the IL result was also investigated. The results of the experimental tests and modelling studies were also compared with those obtained for a larger enclosure made of the same wall materials. The research described in the current article may have practical applications in the selection of walls of cube-shaped enclosures and in estimating their effectiveness in a cost-free manner, assuming that appropriate material data are used in the calculations.
Copyright (c) 2026 Krzysztof Kosała https://creativecommons.org/licenses/by/4.0
Laboratory Evaluation of Underwater Acoustic Attenuation Providedby Air Bubble Curtains
Air bubble curtains are widely used in Chilean salmon farming to protect ﬁsh from water pollutants. Due to new requirements from the Chilean Environmental Assessment Service, their use for underwater noise mitigation has become very important. This study aimed to evaluate the acoustic attenuation provided by a type of curtain widely used in salmon farming and seawater desalination under laboratory conditions, obtaining results independent of the conditions encountered in ﬁeld work. This allowed for comparative measurements of diﬀerent bubble curtain conﬁgurations. This study presents the results of an acoustic evaluation of underwater sound attenuation provided by diﬀerent bubble curtain conﬁgurations, conducted in the test channel of the Engineering Department at the Universidad Austral de Chile. The parameter evaluated was insertion loss IL. It was found that the bubble curtains evaluated provided a broadband insertion loss between 11 dB and 12 dB in the simplest conﬁguration and an Insertion Loss IL per third-octave band equal to or greater than 10 dB. A 10 dB attenuation in the source level represents a signiﬁcant reduction in the area of acoustic impact, reducing the physiological damage distances for the marine mammal species considered by at least 78.4%.
Copyright (c) 2025 Alfio Yori, Jose Luis Barros, Rodrigo Torres, Felipe Figueroa https://creativecommons.org/licenses/by/4.0
Simulation Analysis of Beam Intensity Attenuation Patterns and Source Depth Estimation Using a Vertical Long Line Array
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.
Copyright (c) 2025 Hao WANG, Guangying ZHENG, Fangwei ZHU, Xiaohong YANG, Shuaishuai ZHANG, Xiaowei GUO https://creativecommons.org/licenses/by/4.0
Comparison of Methods for Determining the Airflow Resistivity of Porous and Covering Materials
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.
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
An Improved Dynamic Time Warping Algorithm for Active Sonar Signal Matching
Active sonar signal matching is a critical technique for measuring inter-signal similarity and enhancing target detection and classification performance. However, in complex underwater environments, noise, reverberation, and prolonged signal durations often degrade matching accuracy and computational efficiency. To address these challenges, this paper proposes an adaptive extremum-aligned boundary-constrained dynamic time warping (AEB-DTW) algorithm, based on the classical dynamic time warping (DTW) framework. The algorithm extracts significant extrema from signal envelopes to suppress noise and reverberation while capturing salient features. By integrating the position and amplitude of extrema, an adaptive weighted matching strategy is introduced to enhance feature discrimination. In addition, spline fitting is applied to the residuals of the extremum matching path to dynamically generate upper and lower boundary constraints, thus restricting DTW computation to a meaningful region and achieving a balance between accuracy and efficiency. Experiments using lake-trial active sonar data under signal-to-reverberation ratios (SRRs) from 0 dB to 30 dB show that AEB-DTW outperforms Euclidean distance (ED), DTW, and its variants in matching accuracy, robustness, and angular resolution, while significantly improving computational efficiency, particularly for long-duration signals.
Copyright (c) 2026 Tong-jing Sun, Hun-yuan Wang, Lei Chen, Hao-ran Xu https://creativecommons.org/licenses/by/4.0
Masculinized or Feminized? Discriminant Analysis of Postmenopausal Women’s Voices
This study investigates the degree of vocal variation between men, pre- and postmenopausal women. The sample comprised 108 volunteers aged 18–66 y., divided into control and validation group. Each participant was subjected to voice recordings of five sustained vowels. Acoustic parameters were extracted using Praat software. The most significant parameters in intergroup correlation between canonical discriminant function and acoustic variables were: fundamental frequency (f0), shimmer, harmonics-to-noise ratio (HNR) and intensity. Premenopausal female voices were labeled with 97% correctness and male voices with 95.5% correctness. Interestingly, 65.5% of postmenopausal women were accurately classified as female voices and on average they had lower vocal pitches compared to premenopausal women. The differences in male and female voices are probably due to the difference in the size of the larynx and the length of the vocal cords. Hormonal changes during menopause may affect, but not significantly, the morphology of the laryngeal structures which develop during childhood and adolescence.
Copyright (c) 2025 Maja Pietras, Łukasz P. Pawelec, Monika Krzyżanowska, Anna M. Lipowicz https://creativecommons.org/licenses/by/4.0
The Impact of Training Strategies on Overfitting in Vowel Classification Using PS-HFCC Parametrization for Automatic Speech Recognition
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.
pitch synchronized human factor cepstral coefficient
speech processing and communication systems
Copyright (c) 2025 Stanislaw GMYREK, Urszula LIBAL, Robert HOSSA https://creativecommons.org/licenses/by/4.0
Comparative Study of the Acoustic Efficiency of Prototype Sound Absorbing Panels Used in the Railway Track
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.
Copyright (c) 2025 Grzegorz KLEKOT, Mariusz WĄDOŁOWSKI, Cezary KRAŚKIEWICZ, Artur ZBICIAK https://creativecommons.org/licenses/by/4.0
Comparative Perceptual Assessment of Sound Quality:Cone vs. Distributed Mode Loudspeakers
A conventional cone loudspeaker has a limited capacity for creating the impression of spatiality, while a distributed mode loudspeaker (DML) has an inherent ability to evoke it. DMLs have their specific drawbacks, but some of these can be compensated for. A key question arises – is it a cone loudspeaker or a compensated DML that is preferred by listeners? A listening experiment with carefully controlled conditions was carried out to answer this question; 30 subjects participated. The participants evaluated three stereo systems based on a DML speaker (with its power response equalized) and two conventional two-way active systems. Two perceptual attributes were evaluated: “overall preference,” and “spatial impression.” A graded pairwise comparison was used as an experimental paradigm; the results were analyzed according to the law of comparative judgment. The findings indicated that, even though the DMLs achieved slightly lower ratings than the conventional systems on average, the perceptual differences were very small. This was confirmed by the hypothesis testing that was performed on the raw results of the pairwise comparisons.
Copyright (c) 2025 Piotr Kleczkowski, Paweł Małecki, Dorota Młynaczyk https://creativecommons.org/licenses/by/4.0
Review of Microphone-Based Contactless Vital Signs Monitoring Systems
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.
acoustical measurements and instrumentation
Copyright (c) 2025 Abiodun Ernest AMORAN, Dariusz BISMOR https://creativecommons.org/licenses/by/4.0
Experimental and Computational Analysis on the Acoustic Performance of Side Outlet Muffler with Semi-Circular Baffles
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.
Copyright (c) 2022 Sandeep Kumar VISHWAKARMA, Suryappa Jayappa PAWAR https://creativecommons.org/licenses/by/4.0
Effect of Listener Head Position on Speech Intelligibility in an Automotive Cabin
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.
Copyright (c) 2025 Linda LIANG, Linghui LIAO, Jiahui SUN, Lingling LIU, Liuying OU, Xiaoyue HUANG https://creativecommons.org/licenses/by/4.0
Multi-label Bird Species Classification Using Transfer Learning Network
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.
Copyright (c) 2025 Xue HAN, Jianxin PENG https://creativecommons.org/licenses/by/4.0
Aeroacoustic Effect of Savonius Rotor Segmentation
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.
Copyright (c) 2025 Shivangi SACHAR, Piotr DOERFFER, Paweł FLASZYŃSKI, Józef KOTUS, Krzysztof DOERFFER, Joanna GRZELAK, Michał Piotrowicz https://creativecommons.org/licenses/by/4.0
Structural Parametric Study of a Piezoelectric Energy Harvester for a Specific Excitation Frequency of an Electric Motor, Considering Fatigue Life
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.
Copyright (c) 2025 Ping YANG, Nabil MOHAMAD USAMAH, Abdullah Aziz SAAD, Ahmad Zhafran AHMAD MAZLAN https://creativecommons.org/licenses/by/4.0