Pub Date : 2026-01-16DOI: 10.1016/j.apacoust.2026.111234
Ilaria Fichera, Cédric Van hoorickx, Maarten Hornikx
In the diffusion equation model, the diffusion coefficient quantifies the relationship between the sound intensity and the gradient of the sound energy density. It is a crucial factor that affects the diffusion equation model, mostly for non-proportionate rooms. Previous work has indicated that this parameter varies spatially based on the rooms’ dimensions, the absorption coefficient, and the distance between source and receiver positions. In this paper, the spatially dependent diffusion coefficient is obtained by optimizing a distance-dependent function for the diffusion coefficient using reference results obtained from the radiosity method. The estimated diffusion coefficient inside long rooms is also shown to depend on the source position. The empirical function for the diffusion coefficient for long rooms is established as a quadratic polynomial function and is applicable for elongated rooms with a constant absorption coefficient lower than 0.3 and a minimum cross section of 4 m2.
{"title":"An empirical diffusion coefficient function for the acoustic diffusion equation model in long rooms","authors":"Ilaria Fichera, Cédric Van hoorickx, Maarten Hornikx","doi":"10.1016/j.apacoust.2026.111234","DOIUrl":"10.1016/j.apacoust.2026.111234","url":null,"abstract":"<div><div>In the diffusion equation model, the diffusion coefficient quantifies the relationship between the sound intensity and the gradient of the sound energy density. It is a crucial factor that affects the diffusion equation model, mostly for non-proportionate rooms. Previous work has indicated that this parameter varies spatially based on the rooms’ dimensions, the absorption coefficient, and the distance between source and receiver positions. In this paper, the spatially dependent diffusion coefficient is obtained by optimizing a distance-dependent function for the diffusion coefficient using reference results obtained from the radiosity method. The estimated diffusion coefficient inside long rooms is also shown to depend on the source position. The empirical function for the diffusion coefficient for long rooms is established as a quadratic polynomial function and is applicable for elongated rooms with a constant absorption coefficient lower than 0.3 and a minimum cross section of 4 m<sup>2</sup>.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"246 ","pages":"Article 111234"},"PeriodicalIF":3.4,"publicationDate":"2026-01-16","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145979808","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2026-01-14DOI: 10.1016/j.apacoust.2026.111232
Chuanyang Liu , Xiaoyu Wang , Guangyu Zhang , Zhiliang Hong , Xiaofeng Sun
To investigate the noise reduction characteristics of a liner located at the blade tip on incident sound waves, this study combines experimental and numerical methods to explore the acoustic interaction mechanism between incident sound waves, the over-the-stator liner (OTSL), and the blade row under flow-free conditions. Experimental comparisons among the OTSL (with blade row), the liner only (no blade row), and the blade row only (with a hard wall casing) configurations reveal that the blade row significantly affects the liner’s noise reduction performance, and the OTSL exhibits effective sound absorption for both transmitted and scattered sound waves. Overall, the noise reduction effectiveness of the OTSL is superior to that of the liner only. Numerical analyses of acoustic particle velocity on the liner surface and pressure fluctuations on the blade surface confirm that the interaction between the OTSL and the blade row influences the liner’s dissipative characteristics and the blade row’s scattering behavior. Further investigations indicate that this interaction is correlated with the sound wave propagation direction, frequency, blade angle, and the sound absorption performance of the liner itself.
{"title":"Investigation on the acoustic interaction between over-the-stator liner and stator blade row","authors":"Chuanyang Liu , Xiaoyu Wang , Guangyu Zhang , Zhiliang Hong , Xiaofeng Sun","doi":"10.1016/j.apacoust.2026.111232","DOIUrl":"10.1016/j.apacoust.2026.111232","url":null,"abstract":"<div><div>To investigate the noise reduction characteristics of a liner located at the blade tip on incident sound waves, this study combines experimental and numerical methods to explore the acoustic interaction mechanism between incident sound waves, the over-the-stator liner (OTSL), and the blade row under flow-free conditions. Experimental comparisons among the OTSL (with blade row), the liner only (no blade row), and the blade row only (with a hard wall casing) configurations reveal that the blade row significantly affects the liner’s noise reduction performance, and the OTSL exhibits effective sound absorption for both transmitted and scattered sound waves. Overall, the noise reduction effectiveness of the OTSL is superior to that of the liner only. Numerical analyses of acoustic particle velocity on the liner surface and pressure fluctuations on the blade surface confirm that the interaction between the OTSL and the blade row influences the liner’s dissipative characteristics and the blade row’s scattering behavior. Further investigations indicate that this interaction is correlated with the sound wave propagation direction, frequency, blade angle, and the sound absorption performance of the liner itself.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"246 ","pages":"Article 111232"},"PeriodicalIF":3.4,"publicationDate":"2026-01-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145979807","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2026-01-12DOI: 10.1016/j.apacoust.2026.111233
Xiao Liang , Zijing Yang , Jiangxia Luo , Nansha Gao , Guojian Zhou
The development of multi-band, wide-frequency, spectrum-tunable topological acoustic transmission is essential for the practical application of topological acoustic insulators. However, conventional approaches rely on complex structural reconfiguration or parameter modulation, which severely limits their flexibility. This paper addresses this issue by proposing a universal method of optimising multi-band structures based on stacked composite resonators. The key advantage of this strategy is that it provides comprehensive and flexible control over the number, width and position of operational frequency bands, simply by stacking and arranging resonators vertically. This approach neither alters the original scatterer geometry nor introduces additional parameters. Furthermore, it simplifies multiband control, allowing the operating bandwidth and position to be adjusted by merely altering the number and order of resonator layers. Research indicates that this method enables the on-demand introduction of multiple Dirac cones, as well as the flexible adjustment of existing frequency band widths and Dirac cone spectral positions. Each cone can open a bandgap independently and generate topologically protected one-way edge states. Superlattice simulations, acoustic field simulations and experimental measurements collectively confirm that all frequency bands exhibit the low transmission loss and strong defect immunity characteristic of topologically protected edge states. The proposed layered paradigm in this work revolutionises conventional band control approaches, offering a new way to develop high-performance, customisable, multiband acoustic topological devices.
{"title":"Multi-band controllable acoustic topological insulator based on stacked composite resonant cavities","authors":"Xiao Liang , Zijing Yang , Jiangxia Luo , Nansha Gao , Guojian Zhou","doi":"10.1016/j.apacoust.2026.111233","DOIUrl":"10.1016/j.apacoust.2026.111233","url":null,"abstract":"<div><div>The development of multi-band, wide-frequency, spectrum-tunable topological acoustic transmission is essential for the practical application of topological acoustic insulators. However, conventional approaches rely on complex structural reconfiguration or parameter modulation, which severely limits their flexibility. This paper addresses this issue by proposing a universal method of optimising multi-band structures based on stacked composite resonators. The key advantage of this strategy is that it provides comprehensive and flexible control over the number, width and position of operational frequency bands, simply by stacking and arranging resonators vertically. This approach neither alters the original scatterer geometry nor introduces additional parameters. Furthermore, it simplifies multiband control, allowing the operating bandwidth and position to be adjusted by merely altering the number and order of resonator layers. Research indicates that this method enables the on-demand introduction of multiple Dirac cones, as well as the flexible adjustment of existing frequency band widths and Dirac cone spectral positions. Each cone can open a bandgap independently and generate topologically protected one-way edge states. Superlattice simulations, acoustic field simulations and experimental measurements collectively confirm that all frequency bands exhibit the low transmission loss and strong defect immunity characteristic of topologically protected edge states. The proposed layered paradigm in this work revolutionises conventional band control approaches, offering a new way to develop high-performance, customisable, multiband acoustic topological devices.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"246 ","pages":"Article 111233"},"PeriodicalIF":3.4,"publicationDate":"2026-01-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145979806","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2026-01-10DOI: 10.1016/j.apacoust.2026.111221
Tian Zhang, Ning Han, Zhehua Duan
Remote microphone technique is designed to solve the problem of physical microphones being inconvenient to install in a zone of quiet requiring noise attenuation. However, a typical remote microphone technique based system necessitates a sufficient number of monitoring microphones, generally no fewer than the number of noise sources, to ensure optimal noise reduction performance. Such requirements may lead to increased equipment costs and significant computational demands. In this paper, we develop a methodological approach to simplify the system based on blind source separation and remote microphone technique, aiming to reduce the number of microphones within the system. The main approach operates in three steps. First, the correlation of the sound field is analyzed to determine the microphone configuration. Second, the observation filter is modeled with monitoring microphone signals de-correlated by the blind source separation method. Third, the virtual signal estimated using the observation filter is employed in the active noise control system. Experimental results verify the superiority of the proposed algorithm in computational complexity. Furthermore, results also show that under various noise conditions, the proposed algorithm can achieve similar noise reduction performance as traditional systems while reducing the number of microphones required.
{"title":"A simplified active noise control system with remote microphone technique based on blind source separation","authors":"Tian Zhang, Ning Han, Zhehua Duan","doi":"10.1016/j.apacoust.2026.111221","DOIUrl":"10.1016/j.apacoust.2026.111221","url":null,"abstract":"<div><div>Remote microphone technique is designed to solve the problem of physical microphones being inconvenient to install in a zone of quiet requiring noise attenuation. However, a typical remote microphone technique based system necessitates a sufficient number of monitoring microphones, generally no fewer than the number of noise sources, to ensure optimal noise reduction performance. Such requirements may lead to increased equipment costs and significant computational demands. In this paper, we develop a methodological approach to simplify the system based on blind source separation and remote microphone technique, aiming to reduce the number of microphones within the system. The main approach operates in three steps. First, the correlation of the sound field is analyzed to determine the microphone configuration. Second, the observation filter is modeled with monitoring microphone signals de-correlated by the blind source separation method. Third, the virtual signal estimated using the observation filter is employed in the active noise control system. Experimental results verify the superiority of the proposed algorithm in computational complexity. Furthermore, results also show that under various noise conditions, the proposed algorithm can achieve similar noise reduction performance as traditional systems while reducing the number of microphones required.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"246 ","pages":"Article 111221"},"PeriodicalIF":3.4,"publicationDate":"2026-01-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145941254","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2026-01-10DOI: 10.1016/j.apacoust.2025.111213
Chang Liu , Fotis Georgiou , Maarten Hornikx
Urban vegetation, such as vegetated roofs, is effective in mitigating urban noise. In previous research, the acoustic impedance of vegetated roofs could be predicted by minimising the differences between the measured and pre-calculated level differences between two vertically placed microphones using the multiple-geometry technique. However, it was found that for some cases, the predicted sound pressure level differences deviated from the measured ones at the interference peaks. The sound pressure level produced by a single sound source above a ground surface is characterized by the interference of the direct and ground reflected sound waves, and the accuracy of the assumed locations of the transducers influences the prediction of the ground surface impedance. Therefore, the sensitivity of the transducers’ locations on the determination of the acoustic impedance of porous materials using the multiple-geometry technique was assessed in this research. It was found that small errors on the transducers’ locations lead to significant variations in predicted impedance and material properties. The tolerance of the extracted transducers’ locations is recommended to be within ± 0.005 m to achieve an accurate and unique prediction of the surface impedance of porous materials.
{"title":"Influence of transducer locations on acoustic impedance prediction in porous systems with application to vegetated roofs","authors":"Chang Liu , Fotis Georgiou , Maarten Hornikx","doi":"10.1016/j.apacoust.2025.111213","DOIUrl":"10.1016/j.apacoust.2025.111213","url":null,"abstract":"<div><div>Urban vegetation, such as vegetated roofs, is effective in mitigating urban noise. In previous research, the acoustic impedance of vegetated roofs could be predicted by minimising the differences between the measured and pre-calculated level differences between two vertically placed microphones using the multiple-geometry technique. However, it was found that for some cases, the predicted sound pressure level differences deviated from the measured ones at the interference peaks. The sound pressure level produced by a single sound source above a ground surface is characterized by the interference of the direct and ground reflected sound waves, and the accuracy of the assumed locations of the transducers influences the prediction of the ground surface impedance. Therefore, the sensitivity of the transducers’ locations on the determination of the acoustic impedance of porous materials using the multiple-geometry technique was assessed in this research. It was found that small errors on the transducers’ locations lead to significant variations in predicted impedance and material properties. The tolerance of the extracted transducers’ locations is recommended to be within ± 0.005 m to achieve an accurate and unique prediction of the surface impedance of porous materials.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"246 ","pages":"Article 111213"},"PeriodicalIF":3.4,"publicationDate":"2026-01-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145941255","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2026-01-09DOI: 10.1016/j.apacoust.2026.111223
Ruixin Nie , Kaiqi Zhao , Bin Wang , Jun Fan , Yehao Shi
Sound waves propagating through media with local fluctuations experience phase and amplitude perturbations that carry valuable information about the medium. This principle offers potential for indirectly detecting moving submerged objects, as flow-induced acoustic modulations can reveal their motion characteristics. This study experimentally investigates the Amplitude Modulation (AM) of acoustic waves induced by Ultra-Low-Frequency (ULF) oscillatory flows generated by moving submerged bodies. Controlled tank experiments were conducted to examine the influence of oscillation amplitude, frequency, and movement pattern on single-frequency acoustic signals. Experimental results confirmed the occurrence of AM, with the acoustic signal acting as the carrier and the oscillatory flow as the modulation signal. It was observed that the modulation depth systematically increases with higher oscillation amplitudes and frequencies, demonstrating the pronounced influence of oscillatory flows on acoustic wave propagation. To gain deeper insights into the modulation mechanism, a theoretical framework based on the adiabatic normal mode approximation and perturbation methods was developed to interpret the observed phenomena, treating flow-induced sound speed variations as the primary mechanism for the observed modulation. The agreement between theoretical predictions and experimental observations supports the model’s ability to predict modulation characteristics based on hydrodynamic parameters. This study suggests the potential of using flow-induced acoustic modulations from moving objects as an indirect detection method, providing a foundation for future applications in underwater target detection and tracking.
{"title":"Amplitude modulation of scattered acoustic waves by ULF oscillatory flow produced by submerged bodies: An experimental study","authors":"Ruixin Nie , Kaiqi Zhao , Bin Wang , Jun Fan , Yehao Shi","doi":"10.1016/j.apacoust.2026.111223","DOIUrl":"10.1016/j.apacoust.2026.111223","url":null,"abstract":"<div><div>Sound waves propagating through media with local fluctuations experience phase and amplitude perturbations that carry valuable information about the medium. This principle offers potential for indirectly detecting moving submerged objects, as flow-induced acoustic modulations can reveal their motion characteristics. This study experimentally investigates the Amplitude Modulation (AM) of acoustic waves induced by Ultra-Low-Frequency (ULF) oscillatory flows generated by moving submerged bodies. Controlled tank experiments were conducted to examine the influence of oscillation amplitude, frequency, and movement pattern on single-frequency acoustic signals. Experimental results confirmed the occurrence of AM, with the acoustic signal acting as the carrier and the oscillatory flow as the modulation signal. It was observed that the modulation depth systematically increases with higher oscillation amplitudes and frequencies, demonstrating the pronounced influence of oscillatory flows on acoustic wave propagation. To gain deeper insights into the modulation mechanism, a theoretical framework based on the adiabatic normal mode approximation and perturbation methods was developed to interpret the observed phenomena, treating flow-induced sound speed variations as the primary mechanism for the observed modulation. The agreement between theoretical predictions and experimental observations supports the model’s ability to predict modulation characteristics based on hydrodynamic parameters. This study suggests the potential of using flow-induced acoustic modulations from moving objects as an indirect detection method, providing a foundation for future applications in underwater target detection and tracking.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"246 ","pages":"Article 111223"},"PeriodicalIF":3.4,"publicationDate":"2026-01-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145941253","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2026-01-07DOI: 10.1016/j.apacoust.2026.111222
Qian Bai , Alireza Amiri-Simkooei , Sebastiaan Mestdagh , Dick G. Simons , Mirjam Snellen
Seabed backscatter data acquired by the multibeam echosounder (MBES) have been identified as a valuable indicator of sediment properties and benthic community characteristics. However, developing robust change detection models with MBES backscatter remains challenging due to the high costs and limited spatial coverage of seabed ground truth data. Lack of absolute backscatter calibration also hinders the comparison between repeated MBES measurements. To mitigate these issues, we propose an unsupervised method to detect seabed changes by fitting a Gaussian Mixture Model to the backscatter difference between two datasets. A relative calibration is conducted based on a stable reference area to eliminate the impact of possible drifts in echosounder characteristics on the backscatter difference. We then model the unchanged class as a zero-mean Gaussian distribution, with its variance constrained by the backscatter uncertainty estimated from the reference area. By processing each incident angle individually, the angular range with the greatest ability for seabed change detection can also be investigated. We demonstrate the effectiveness of the proposed method through two case studies in the Dutch North Sea. The detected changes reveal seasonal and temporal variations in benthic communities, such as sand mason worms, and are consistent with the sediment movement in one of the study areas. This research highlights the value of MBES backscatter data for seabed change detection and provides a cost-effective solution for seabed habitat monitoring with acoustic measurements.
{"title":"Unsupervised seabed habitat change detection with multibeam backscatter data using a constrained Gaussian mixture model","authors":"Qian Bai , Alireza Amiri-Simkooei , Sebastiaan Mestdagh , Dick G. Simons , Mirjam Snellen","doi":"10.1016/j.apacoust.2026.111222","DOIUrl":"10.1016/j.apacoust.2026.111222","url":null,"abstract":"<div><div>Seabed backscatter data acquired by the multibeam echosounder (MBES) have been identified as a valuable indicator of sediment properties and benthic community characteristics. However, developing robust change detection models with MBES backscatter remains challenging due to the high costs and limited spatial coverage of seabed ground truth data. Lack of absolute backscatter calibration also hinders the comparison between repeated MBES measurements. To mitigate these issues, we propose an unsupervised method to detect seabed changes by fitting a Gaussian Mixture Model to the backscatter difference between two datasets. A relative calibration is conducted based on a stable reference area to eliminate the impact of possible drifts in echosounder characteristics on the backscatter difference. We then model the unchanged class as a zero-mean Gaussian distribution, with its variance constrained by the backscatter uncertainty estimated from the reference area. By processing each incident angle individually, the angular range with the greatest ability for seabed change detection can also be investigated. We demonstrate the effectiveness of the proposed method through two case studies in the Dutch North Sea. The detected changes reveal seasonal and temporal variations in benthic communities, such as sand mason worms, and are consistent with the sediment movement in one of the study areas. This research highlights the value of MBES backscatter data for seabed change detection and provides a cost-effective solution for seabed habitat monitoring with acoustic measurements.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"246 ","pages":"Article 111222"},"PeriodicalIF":3.4,"publicationDate":"2026-01-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145904218","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2026-01-05DOI: 10.1016/j.apacoust.2025.111219
Alexandre Schiavini , Philippe Micheau , Pierre Grandjean , Gwénaël Gabard , Thomas Humbert
The Harmonic Acoustic Pneumatic Source (HAPS), a mechanical airflow modulator, is a promising alternative to loudspeakers as secondary sources for the active control of harmonic noise, thanks to its ability to generate high sound pressure levels and to precisely control signal frequency, phase, and amplitude. However, this source requires a specialized control strategy, as it operates through mechanical modulation at a predetermined frequency. Hence, conventional adaptive control techniques are not directly applicable. Furthermore, when dealing with time-varying harmonic noise, the dynamic performance of the control system, which is influenced by the dynamic properties of the HAPS, is challenged. The originality of this study lies in considering this source, not as a tonal source, but as a narrow-band mechanical modulator that generates a narrow-band anti-noise by driving its complex envelope, allowing the active control of time-varying primary noise. A model of the dynamic response of the Harmonic Acoustic Pneumatic Source and of its controller is presented, with an analysis of the theoretical dynamic performances and limitations of this control system. The controller uses either a far-field error microphone or a near-field error microphone with a previously established compensation strategy. The closed-loop model is validated with experimental results. Simulations and experiments of far-field and near-field active noise control on slow time-varying primary noise are conducted to characterize the dynamic performances and limitations of the controller. The simulations are in good agreement with the experimental results.
{"title":"Dynamics of the harmonic acoustic pneumatic source for active control of time-varying tonal noise","authors":"Alexandre Schiavini , Philippe Micheau , Pierre Grandjean , Gwénaël Gabard , Thomas Humbert","doi":"10.1016/j.apacoust.2025.111219","DOIUrl":"10.1016/j.apacoust.2025.111219","url":null,"abstract":"<div><div>The Harmonic Acoustic Pneumatic Source (HAPS), a mechanical airflow modulator, is a promising alternative to loudspeakers as secondary sources for the active control of harmonic noise, thanks to its ability to generate high sound pressure levels and to precisely control signal frequency, phase, and amplitude. However, this source requires a specialized control strategy, as it operates through mechanical modulation at a predetermined frequency. Hence, conventional adaptive control techniques are not directly applicable. Furthermore, when dealing with time-varying harmonic noise, the dynamic performance of the control system, which is influenced by the dynamic properties of the HAPS, is challenged. The originality of this study lies in considering this source, not as a tonal source, but as a narrow-band mechanical modulator that generates a narrow-band anti-noise by driving its complex envelope, allowing the active control of time-varying primary noise. A model of the dynamic response of the Harmonic Acoustic Pneumatic Source and of its controller is presented, with an analysis of the theoretical dynamic performances and limitations of this control system. The controller uses either a far-field error microphone or a near-field error microphone with a previously established compensation strategy. The closed-loop model is validated with experimental results. Simulations and experiments of far-field and near-field active noise control on slow time-varying primary noise are conducted to characterize the dynamic performances and limitations of the controller. The simulations are in good agreement with the experimental results.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"245 ","pages":"Article 111219"},"PeriodicalIF":3.4,"publicationDate":"2026-01-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145938573","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2026-01-03DOI: 10.1016/j.apacoust.2025.111217
Sudhansu Sekhar Nayak , Anand D. Darji , Prashant K. Shah , Juan Rafael Orozco-Arroyave
There has been a growing interest in the development of automated methods to diagnose Parkinson’s disease from speech. These approaches can potentially be used in telemonitoring health applications; however, there is still much to be done in the process of developing accurate methods to perform the diagnosis. The purpose of this paper is to present a novel and efficient approach for detecting Parkinson’s disease from speech signals. Parkinson’s disease speech is modeled utilizing the Fourier–Bessel domain adaptive wavelet transform. The signal is decomposed by the Fourier–Bessel domain adaptive wavelet transform into several modes. Energy, entropy, increment entropy, and spectral entropy are extracted from each of the decomposed signals, and a combination of these features is evaluated using the isolated words and sustained vowels from the PC-GITA database. Support vector machine classifier achieves a maximum classification accuracy of 95 % for /drama/. Furthermore, with the aim of evaluating the generalization capability of the introduced approach, the model optimized with PC-GITA is used to perform the automatic classification of Parkinson’s disease vs. healthy control subjects in an independent dataset with a classification accuracy of 84 %. The results show that the proposed approach based on the Fourier–Bessel domain adaptive wavelet transform decomposition is accurate and efficient. Additionally, it showed robustness against unseen data collected under non-controlled acoustic conditions, making it a good candidate to develop computational systems that work properly in real-world clinical practice.
{"title":"Automatic detection of Parkinson’s disease from speech signals using the Fourier–Bessel domain adaptive wavelet transform","authors":"Sudhansu Sekhar Nayak , Anand D. Darji , Prashant K. Shah , Juan Rafael Orozco-Arroyave","doi":"10.1016/j.apacoust.2025.111217","DOIUrl":"10.1016/j.apacoust.2025.111217","url":null,"abstract":"<div><div>There has been a growing interest in the development of automated methods to diagnose Parkinson’s disease from speech. These approaches can potentially be used in telemonitoring health applications; however, there is still much to be done in the process of developing accurate methods to perform the diagnosis. The purpose of this paper is to present a novel and efficient approach for detecting Parkinson’s disease from speech signals. Parkinson’s disease speech is modeled utilizing the Fourier–Bessel domain adaptive wavelet transform. The signal is decomposed by the Fourier–Bessel domain adaptive wavelet transform into several modes. Energy, entropy, increment entropy, and spectral entropy are extracted from each of the decomposed signals, and a combination of these features is evaluated using the isolated words and sustained vowels from the PC-GITA database. Support vector machine classifier achieves a maximum classification accuracy of 95 % for /drama/. Furthermore, with the aim of evaluating the generalization capability of the introduced approach, the model optimized with PC-GITA is used to perform the automatic classification of Parkinson’s disease vs. healthy control subjects in an independent dataset with a classification accuracy of 84 %. The results show that the proposed approach based on the Fourier–Bessel domain adaptive wavelet transform decomposition is accurate and efficient. Additionally, it showed robustness against unseen data collected under non-controlled acoustic conditions, making it a good candidate to develop computational systems that work properly in real-world clinical practice.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"245 ","pages":"Article 111217"},"PeriodicalIF":3.4,"publicationDate":"2026-01-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145884082","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2026-01-03DOI: 10.1016/j.apacoust.2025.111220
P.Y. Chan , S.K. Tang , Chi-Chung Cheung
An investigation on the possible effects of reference microphone position on the improvement of sound reduction across a practical acoustic window by active noise control method was carried out in the present study. Error microphones were located over the indoor window opening, while the cancelling sound sources along the periphery of the window void started from the lower corner of the outdoor window opening to the overlapping region of the window. The analysis was done using experimentally measured transfer functions of all related sound transmission paths. In general, the active control performance is better when more error microphones are adopted. The best sound reduction improvement reaches 6.80 dB over the frequency range from 100 Hz to 1 kHz when the reference microphone was mounted midway on the external window vertical frame. The control performance is weakened at a reduced number of error microphones in general. However, it is found that a simple system with three error microphones and two cancelling sources can achieve an average 5.2 dB broadband sound reduction improvement, when the reference microphone was located at the top of the outdoor window opening. While the former reference microphone position performs better when more error microphones and cancelling sources are adopted in a control system, the latter works better with simpler systems. Also, the present results suggest that it is better to keep the number of microphones larger than that of the cancelling sources and indicate the possibility of a coherence-based approach for locating reference microphone.
{"title":"Effects of reference microphone position on active sound reduction across an acoustic plenum window","authors":"P.Y. Chan , S.K. Tang , Chi-Chung Cheung","doi":"10.1016/j.apacoust.2025.111220","DOIUrl":"10.1016/j.apacoust.2025.111220","url":null,"abstract":"<div><div>An investigation on the possible effects of reference microphone position on the improvement of sound reduction across a practical acoustic window by active noise control method was carried out in the present study. Error microphones were located over the indoor window opening, while the cancelling sound sources along the periphery of the window void started from the lower corner of the outdoor window opening to the overlapping region of the window. The analysis was done using experimentally measured transfer functions of all related sound transmission paths. In general, the active control performance is better when more error microphones are adopted. The best sound reduction improvement reaches 6.80 dB over the frequency range from 100 Hz to 1 kHz when the reference microphone was mounted midway on the external window vertical frame. The control performance is weakened at a reduced number of error microphones in general. However, it is found that a simple system with three error microphones and two cancelling sources can achieve an average 5.2 dB broadband sound reduction improvement, when the reference microphone was located at the top of the outdoor window opening. While the former reference microphone position performs better when more error microphones and cancelling sources are adopted in a control system, the latter works better with simpler systems. Also, the present results suggest that it is better to keep the number of microphones larger than that of the cancelling sources and indicate the possibility of a coherence-based approach for locating reference microphone.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"245 ","pages":"Article 111220"},"PeriodicalIF":3.4,"publicationDate":"2026-01-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"145884081","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
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