Applied AcousticsPub Date : 2025-04-03DOI: 10.1016/j.apacoust.2025.110706
Shuming Luan , Yukoh Wakabayashi , Tomoki Toda
{"title":"Generalized sound field interpolation for freely spaced microphone arrays in rotation-robust beamforming","authors":"Shuming Luan , Yukoh Wakabayashi , Tomoki Toda","doi":"10.1016/j.apacoust.2025.110706","DOIUrl":"10.1016/j.apacoust.2025.110706","url":null,"abstract":"<div><div>In this paper, we present a novel method for <figure><img></figure> aimed at achieving rotation-robust beamforming with circular microphone arrays (CMAs), where the microphone distribution is unknown. Previous methods require a known microphone distribution, limiting their applicability in real-world scenarios where such information is often unavailable. Our proposed method addresses this limitation by utilizing an unequally spaced circular microphone array (unes-CMA) with unknown microphone positions. The method comprises two key components: unsupervised calibration and unequally spaced <figure><img></figure>. Unsupervised calibration employs an innovative iterative optimization technique to estimate the positional errors of the microphones without any pre-existing location information, thereby determining the microphone distribution on the unes-CMA in an unsupervised manner. Once these positional errors are estimated, the unequally spaced <figure><img></figure> method enables the reconstruction of the target signal of the unes-CMA before rotation. This process of unsupervised calibration enables accurate <figure><img></figure> even with unknown microphone distributions. Additionally, we further improve the method by modifying the cost function used during unsupervised calibration and extending its applicability to nearly circular microphone arrays, which are not strictly a circle anymore. Simulation experiments were conducted to evaluate the performance of our method. The results demonstrated that our method effectively mitigates the negative impact of unknown microphone placements. It yielded significant improvements in estimating the signal before rotation and the performance of beamforming compared with previous methods.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"236 ","pages":"Article 110706"},"PeriodicalIF":3.4,"publicationDate":"2025-04-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143760529","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"OA","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Applied AcousticsPub Date : 2025-04-03DOI: 10.1016/j.apacoust.2025.110709
Haoxin Ruan , Shang Zhao , Zhibin Lin , Jing Lu
{"title":"Computationally-efficient blind source extraction under extremely low SNR conditions based on principal component analysis","authors":"Haoxin Ruan , Shang Zhao , Zhibin Lin , Jing Lu","doi":"10.1016/j.apacoust.2025.110709","DOIUrl":"10.1016/j.apacoust.2025.110709","url":null,"abstract":"<div><div>Signal extraction under extremely low signal-to-noise ratio (SNR) conditions remains a challenging task in signal processing. Conventional separation and extraction algorithms can achieve satisfactory performance but the computational burden is heavy since they usually involve complicated update rules and require a sufficient number of iterations. In this paper, we prove that, under extremely low SNR conditions, principal component analysis (PCA) can be used as an engineering approximation of the optimal solution for independent vector extraction (IVE). Based on this conclusion, two computationally-efficient methods are proposed to obtain the demixing vector. Both methods involve only a few simple operations and thus require much less computational resources. Numerical experiments across various scenarios demonstrate that the proposed algorithms can achieve comparable performance to conventional algorithms with light computational burden.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"236 ","pages":"Article 110709"},"PeriodicalIF":3.4,"publicationDate":"2025-04-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143760530","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}
Applied AcousticsPub Date : 2025-04-02DOI: 10.1016/j.apacoust.2025.110697
Min Qiu , Tian Ran Lin , Xinying Wang
{"title":"Sound insulation of a lightweight lattice sandwich panel","authors":"Min Qiu , Tian Ran Lin , Xinying Wang","doi":"10.1016/j.apacoust.2025.110697","DOIUrl":"10.1016/j.apacoust.2025.110697","url":null,"abstract":"<div><div>In this study, an analytical approach based on the Reissner sandwich panel theory is presented for the analysis of sound insulation properties of a lightweight sandwich panel with lattice cores. A comparison of the sound insulation of the sandwich panel predicted using the analytical solution and finite element analysis shows that by ignoring the shear stiffness of face plates of the sandwich panel can lead to a deviation in the predicted sound insulation result, exemplified by the frequency difference of the valleys in the STL of the panel. An equivalent shear energy principle is then adopted to render a revised equivalent shear stiffness of the sandwich panel by including the contribution of the face plate shear stiffness. It is shown that the analytical STL prediction of the panel using the equivalent shear stiffness matches quite well with that using finite element analysis under a normal incident sound wave excitation. There is a deviation in the results when the sandwich panel is under an oblique sound wave excitation due to the isotropic assumption in the evaluation of the equivalent shear stiffness of the lattice core in the analytical solution. A sound insulation experiment is also carried out in the study to verify the theoretical prediction result of the sandwich panel.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"235 ","pages":"Article 110697"},"PeriodicalIF":3.4,"publicationDate":"2025-04-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143746891","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}
Applied AcousticsPub Date : 2025-04-01DOI: 10.1016/j.apacoust.2025.110711
Xianhui Li
{"title":"Predicting the diffuse sound absorption coefficient of a baffled finite absorber","authors":"Xianhui Li","doi":"10.1016/j.apacoust.2025.110711","DOIUrl":"10.1016/j.apacoust.2025.110711","url":null,"abstract":"<div><div>Due to the edge effect, the diffuse sound absorption coefficient of a finite absorber can be quite different from that of the corresponding infinite one. In this paper, a matrix form formula is derived for the diffuse sound absorption coefficient of a baffled finite planar absorber. The diffuse sound field is modeled as the reverberant loading on the absorber, which is shown to be proportional to the time-averaged mean squared pressure in the far field and the imaginary part of the half space Green's function. By employing the direct boundary integral formulation for the scattering problem, the diffuse sound absorption coefficient is obtained as ratio of the mean value of the power dissipated by the absorber to the intensity of the diffuse sound field and the area of the absorber. Sound absorption in non-homogeneous diffuse field is also considered. The proposed method is validated both analytically and experimentally with earlier works.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"235 ","pages":"Article 110711"},"PeriodicalIF":3.4,"publicationDate":"2025-04-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143739256","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}
Applied AcousticsPub Date : 2025-03-31DOI: 10.1016/j.apacoust.2025.110682
Wenxia Wang, Min Wang, Ping Yang, Longbiao He, Guangzhen Xing, Ke Wang
{"title":"Three-parameter signal modeling for amplitude estimation of direct sound in semi-anechoic environments","authors":"Wenxia Wang, Min Wang, Ping Yang, Longbiao He, Guangzhen Xing, Ke Wang","doi":"10.1016/j.apacoust.2025.110682","DOIUrl":"10.1016/j.apacoust.2025.110682","url":null,"abstract":"<div><div>Amplitude measurements of the direct sound under semi-anechoic environments are investigated. This paper proposes using high-frequency acoustic pulses to accurately measure the time-delay difference between the first reflected sound and the direct sound, so as to establish a signal model containing three unknown parameters, namely, the direct sound amplitude, the reflected sound amplitude, and the initial phase. Based on the three-parameter signal model, a moving phase least squares algorithm is proposed to estimate the direct sound amplitude by combining the direct and superimposed steady-state sound data. The Fisher information matrices and Cramer-Rao lower bounds of the signal model are derived, and then the effects of time-delay difference, initial phase and data length on the parameter estimation performance are investigated. Laboratory water tank experiments are carried out, and the theoretical analysis and experimental results show that the proposed three-parameter signal model improves the accuracy and stability of the amplitude estimation of direct sound by effectively utilizing the superimposed sound data.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"235 ","pages":"Article 110682"},"PeriodicalIF":3.4,"publicationDate":"2025-03-31","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143739254","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}
Applied AcousticsPub Date : 2025-03-31DOI: 10.1016/j.apacoust.2025.110680
Zubin Liu, Ze Peng Xu, Dapeng Tan
{"title":"The artificial rabbits optimized variational mode decomposition and its application on quality evaluation of machines","authors":"Zubin Liu, Ze Peng Xu, Dapeng Tan","doi":"10.1016/j.apacoust.2025.110680","DOIUrl":"10.1016/j.apacoust.2025.110680","url":null,"abstract":"<div><div>In order to distinguish the imperceptible differences of vibration and noise signals existing among a large amount of the eligible machinery of complex structures, a new feature extraction method, the artificial rabbits optimized variational mode decomposition (ARO-VMD), is proposed and applied to the quality evaluation of machines. Firstly, the ARO-VMD adaptively determinate the number of decomposition modes with three steps: preprocessing the original signal with generalized minimax-concave penalty function to remove the interference of irrelevant components, implementing adaptive extraction of penalty factors with a new fitness function obtained by ARO algorithm, and Tanimoto coefficient being applied as the judging condition to stop decomposition. Secondly, the integrated timbre parameter (ITP) is constructed based on the correlation analysis of the timbre parameters of the extracted signal components. Finally, the least squares support vector regression optimized by the ARO is used to evaluate the quality of the machine. The feasibility and superiority of the ARO-VMD method were verified by comparative analysis, and the accuracy of the quality evaluation model proposed was 95.65% on the verification over the industrial sewing machines, where the ITP have the highest correlation with quality evaluation results.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"235 ","pages":"Article 110680"},"PeriodicalIF":3.4,"publicationDate":"2025-03-31","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143739253","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}
Applied AcousticsPub Date : 2025-03-31DOI: 10.1016/j.apacoust.2025.110704
Wei Dai , Weirui Qin , Xiaoyu Wang , Guangyu Zhang , Xiaofeng Sun
{"title":"On the analytical acoustic propagation and mode decomposition method in C-duct","authors":"Wei Dai , Weirui Qin , Xiaoyu Wang , Guangyu Zhang , Xiaofeng Sun","doi":"10.1016/j.apacoust.2025.110704","DOIUrl":"10.1016/j.apacoust.2025.110704","url":null,"abstract":"<div><div>Aft-fan aero-engine nacelle noise is a significant component of overall aero-engine noise. The need for optimizing aft-fan liners necessitates a deeper understanding of acoustic propagation within the aft-fan duct. This study develops an analytical model for acoustic propagation in C-ducts to examine the effects of mode scattering in C-ducts due to incident modes from the annular duct. In a theoretical infinite duct system composed of both an annular duct and a C-duct, we illustrate the relationship between modes in the annular and C-ducts using analytical expression. Additionally, we identify and validate the methods and prerequisites for generating a single C-duct mode, which will aid in advancing the understanding of acoustic propagation mechanisms in C-ducts. Simulations and experiments are conducted to compare with the analytical results. Furthermore, mode decomposition methods for C-ducts are presented during experiments. To ensure the accuracy of decomposition results, we discuss methods for evaluating errors with different placement of measuring points.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"235 ","pages":"Article 110704"},"PeriodicalIF":3.4,"publicationDate":"2025-03-31","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143739255","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}
Applied AcousticsPub Date : 2025-03-30DOI: 10.1016/j.apacoust.2025.110705
Yibo Huang , Chong Li , Zhiyong Li , Qiuyu Zhang , Fanwang Yang
{"title":"Fractal matrix speech encryption algorithm based on fractional order robust chaos","authors":"Yibo Huang , Chong Li , Zhiyong Li , Qiuyu Zhang , Fanwang Yang","doi":"10.1016/j.apacoust.2025.110705","DOIUrl":"10.1016/j.apacoust.2025.110705","url":null,"abstract":"<div><div>The limitations of current speech encryption algorithms in planar dimensional transformation and encryption process, as well as the parameter discontinuity problem faced by fractional-order chaos, are optimized. In this paper, we propose a fractal matrix encryption algorithm for speech with high dimensionality based on fractional order robust chaos. The designed chaos equation utilizes the definition of differential fractional-order, the original integer-order robust chaos equation is transformed according to the definition of fractional-order differential and 3D fractional-FOHE exponential chaos is proposed. The fractional-order robust chaos effectively solves the problems of integer-order numerical aggregation and discontinuity of fractional-order chaos parameters. In the key proposed in the encryption algorithm, the nested and intermediate parameters are used to generate the key, which solves the problem of small key space at present. On this basis, an encryption scheme is proposed to change the one-dimensional speech matrix into cubic three-dimensional fractal matrix by using matrix transformation of space and to realize diffusion and obfuscation for the data in each three-dimensional fractal matrix. The proposed cubic 3D fractal matrix is irregular and self-similar. The core formulation in the encryption scheme utilizes the retrieval matrix of the chaotic sequence for the coordinate substitution operation of the cubic 3D fractal matrix and uses the algorithm to diffuse the values nested on the coordinates, breaking the current limitation of encryption only in low dimensions. Comparing the high-dimensional encryption algorithm proposed in this paper with other encryption literature, it is found that the designed algorithm has a larger key space, lower data correlation, and can resist a series of classical attacks. It provides a more practical encryption scheme for speech encryption.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"235 ","pages":"Article 110705"},"PeriodicalIF":3.4,"publicationDate":"2025-03-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143734563","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}
Applied AcousticsPub Date : 2025-03-30DOI: 10.1016/j.apacoust.2025.110685
Ting Wang , Huachang Cui , Wenkai Dong , Yuheng Wang , Kun Xie , Meixia Chen
{"title":"Vibroacoustic characteristics of a metamaterial plate cavity coupling system","authors":"Ting Wang , Huachang Cui , Wenkai Dong , Yuheng Wang , Kun Xie , Meixia Chen","doi":"10.1016/j.apacoust.2025.110685","DOIUrl":"10.1016/j.apacoust.2025.110685","url":null,"abstract":"<div><div>Metamaterials have garnered significant attention due to their remarkable ability to control low-frequency vibration and noise, particularly in applications involving underwater vehicles. This article specifically examines the vibration and acoustic properties of a metamaterial plate cavity coupling system (MPCCS). The mathematical model for the MPCCS is established using principles of energy functional and polynomial expansion. The study explores the bandgap characteristics, coupling modes, and mechanisms of vibration and sound reduction within the system. The key findings reveal that the MPCCS can generate two distinct low-frequency bandgaps, with the starting frequency remaining consistent across various cavity mediums, while the bandwidth varies. Within these band gaps, both vibration and cavity pressure experience significant reduction. To validate these findings, an underwater experiment on sound and vibration control using the metamaterial plate is conducted in an anechoic tank. Results demonstrate the effectiveness of the metamaterial plate in controlling vibration and acoustics compared to a bare plate, highlighting the dual bandgap characteristics of the lateral local resonance metamaterial plate. Furthermore, the experimental outcomes strongly confirm the metamaterial plate’s capability to achieve vibroacoustic control within specific frequency ranges, namely the bandgaps. These findings provide valuable insights and practical guidance for engineering applications, particularly in the design and optimization of metamaterial-based systems for controlling vibration and noise, especially in underwater environments.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"235 ","pages":"Article 110685"},"PeriodicalIF":3.4,"publicationDate":"2025-03-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143734646","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}
Applied AcousticsPub Date : 2025-03-30DOI: 10.1016/j.apacoust.2025.110688
Hong Yang, Chao Wang, Guohui Li
{"title":"Research on feature extraction of underwater acoustic signal based on hybrid entropy algorithms","authors":"Hong Yang, Chao Wang, Guohui Li","doi":"10.1016/j.apacoust.2025.110688","DOIUrl":"10.1016/j.apacoust.2025.110688","url":null,"abstract":"<div><div>The research on feature extraction of underwater acoustic signal (UAS) is of great significance in developing and protecting marine resources. Therefore, to effectively improve the feature extraction technology, a novel hybrid entropy feature extraction method based on enhanced singular spectrum decomposition (ZSSD), generalized phase-amplitude-aware permutation entropy and fractional singular value entropy (GAAPE&FSVE) is proposed. To better capture the dynamic fluctuation components in the UAS, an enhanced SSD algorithm based on the modified Cao algorithm and ensemble fluctuation-based dispersion entropy is proposed. To make AAPE better adapted to feature extraction of UAS, phase processing and entropy parameters transformation are introduced, and GAAPE is proposed. To solve the problem that the entropy value of SVE is unstable when the quantized signal changes dynamically, a new fractional-order processing is introduced, and FSVE is proposed. Firstly, UAS is decomposed into a series of singular spectrum components (SSCs) by ZSSD. Secondly, the useful information contained in SSC is calculated from the time and frequency domains, respectively. The best two SSCs are reconstructed as feature vectors. Then, 150 samples are randomly selected from the feature vectors, and the GAAPE and FSVE of each sample are calculated, respectively. Finally, compared with at least 15 other methods, the experimental results show that the proposed method exhibits stronger synergy in UAS feature extraction and outperforms all compared methods with a 99% recognition rate.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":"235 ","pages":"Article 110688"},"PeriodicalIF":3.4,"publicationDate":"2025-03-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"143734647","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}