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Numerical investigation of the impeller blockage effect on centrifugal fan noise using a hybrid prediction approach with rotating system 叶轮堵塞对离心风机噪声影响的混合预测方法与旋转系统
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-22 DOI: 10.1016/j.apacoust.2025.111154
Ye Yuan , Youhong Xiao , Qiannan Xu , Fujian Zhao
In this study, a hybrid aeroacoustic prediction approach incorporating the effects of a rotating system is proposed to calculate the aerodynamic noise generated by centrifugal fan operation. Additionally, the blockage effect caused by the impeller on the propagation of broadband noise is investigated. The relative motion between the rotating impeller and the volute is modeled using a sliding grid technique. The Improved Delayed Detached-Eddy Simulation (IDDES) is employed to resolve the flow field and compute the acoustic source term, which is subsequently coupled with the Acoustic Perturbation Equations (APEs) to simulate the generation and propagation of the fan noise. The flow and acoustic fields are solved simultaneously on a single grid, eliminating the need to store large datasets and avoiding interpolation errors between grids. The accuracy of the approach is validated using benchmark acoustic problems. Comparison of the numerical results of the centrifugal fan with the experimental data shows that the deviation in Overall Sound Pressure Level (OASPL) at the four microphone locations ranges from 1.9 dB to 3.4 dB, demonstrating the validity of the approach in predicting aeroacoustic behavior in complex rotating systems. Furthermore, the blockage effect of the impeller on noise propagation is investigated in detail. For acoustic sources located in the volute tongue region, this effect not only influences the spatial distribution of acoustic energy but also causes a shift and redistribution of acoustic energy across different frequency bands.
本文提出了一种考虑旋转系统影响的混合气声预测方法来计算离心风机运行时产生的气动噪声。此外,还研究了叶轮对宽带噪声传播的阻塞效应。采用滑动网格技术对旋转叶轮与蜗壳之间的相对运动进行了建模。采用改进的延迟分离涡模拟(IDDES)对流场进行解析,计算声源项,并结合声扰动方程(ape)模拟风机噪声的产生和传播。在单个网格上同时求解流场和声场,消除了存储大型数据集的需要,避免了网格之间的插值误差。利用基准声学问题验证了该方法的准确性。离心风机的数值计算结果与实验数据的比较表明,4个传声器位置的总声压级(OASPL)偏差在1.9 ~ 3.4 dB之间,证明了该方法预测复杂旋转系统气动声学行为的有效性。此外,还详细研究了叶轮的堵塞对噪声传播的影响。对于位于蜗壳舌区的声源,这种效应不仅会影响声能的空间分布,还会引起声能在不同频段间的移位和再分布。
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引用次数: 0
Distributed particle filtering based on stein variational gradient descent for acoustic source tracking 基于stein变分梯度下降的分布式粒子滤波声源跟踪
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-20 DOI: 10.1016/j.apacoust.2025.111168
Yuan Jing , Hao Song
This paper proposes a novel distributed particle filtering algorithm for acoustic source tracking in distributed microphone networks within the Bayesian filtering framework. The core innovation leverages multiple tempering Stein variational gradient descent to efficiently fuse probability density functions of the local posteriors across the network, minimizing Kullback-Leibler divergence. At each sensor node, the probability density functions of local posteriors are first approximated using particles via sequential Monte Carlo sampling. To enable communication-efficient distributed consensus, these particles model local posteriors as Gaussian mixture distributions. Neighboring nodes exchange these local Gaussian mixture posteriors. Within each average consensus iteration, a multiple tempering Stein variational gradient descent sampler generates samples approximating the fused posterior, defined as the product of fractional powers of the received Gaussian mixture posteriors. This process yields a consistent approximation of the global posterior probability density on every node from the fused samples. The algorithm’s effectiveness is validated through both distributed acoustic source tracking simulations and a real-world recording experiment.
本文提出了一种基于贝叶斯滤波框架的分布式粒子滤波算法,用于分布式传声器网络中的声源跟踪。核心创新利用多重回火Stein变分梯度下降,有效融合整个网络局部后验的概率密度函数,最大限度地减少Kullback-Leibler散度。在每个传感器节点上,首先使用粒子通过顺序蒙特卡罗采样来近似局部后验的概率密度函数。为了实现高效通信的分布式共识,这些粒子将局部后验建模为高斯混合分布。相邻节点交换这些局部高斯混合后验。在每次平均共识迭代中,多重回火Stein变分梯度下降采样器生成近似于融合后验的样本,该样本被定义为接收到的高斯混合后验的分数次幂的乘积。这一过程产生了融合样本中每个节点的全局后验概率密度的一致近似值。通过分布式声源跟踪仿真和实际录音实验验证了该算法的有效性。
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引用次数: 0
Comparison of Zwicker and Moore–Glasberg loudness levels with common level-based measures in the assessment of environmental noises in daily life Zwicker和Moore-Glasberg响度等级与常用的基于等级的方法在日常生活环境噪声评价中的比较
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-19 DOI: 10.1016/j.apacoust.2025.111164
Akiko Sugahara , Miki Yonemura , Shinichi Sakamoto
A-weighting is commonly used in environmental noise regulations and standards across many countries due to its simplicity and practicality. However, it does not always align with subjective loudness perception, particularly for characteristic sounds. In contrast, the loudness levels defined in ISO 532–1 (Zwicker method) and ISO 532–2 (Moore–Glasberg method) more accurately reflect human auditory response. Despite this advantage, their practical application has been limited by computational complexity and limited field data. This study focuses on a systematic comparison of the two standardized loudness metrics (Zwicker and Moore–Glasberg) against one another and with A-weighted sound pressure levels, in the context of environmental noise evaluation. To support this comparison, an efficient and practical system for real-time loudness level measurement was developed and implemented on a tablet-based, multifunctional measuring device. The system enables simultaneous calculation and direct comparison of both ISO 532–1 and ISO 532–2 loudness levels during on-site environmental noise monitoring. First, the frequency and level dependencies of each method were analyzed through numerical experiments. Subsequently, field measurements were conducted to capture instantaneous loudness levels and energy-averaged equivalent loudness levels over the entire measurement period for various environmental noise sources. The results consistently demonstrated method-dependent differences in loudness indices across both numerical experiments and field measurements. Notably, characteristic phenomena such as reversals in loudness levels between methods were observed, depending on the frequency content, temporal variations, and level ranges of the noise sources. These findings provide novel insights into the behavior and interpretability of standardized loudness metrics under real-world acoustic conditions and may support their broader application in perceptually relevant environmental noise assessment.
a加权法因其简单实用,在许多国家的环境噪声法规和标准中得到普遍应用。然而,它并不总是与主观的响度感知一致,特别是对于特征声音。相比之下,ISO 532-1 (Zwicker方法)和ISO 532-2 (Moore-Glasberg方法)中定义的响度水平更准确地反映了人类的听觉反应。尽管有这样的优势,但它们的实际应用受到计算复杂性和有限的现场数据的限制。本研究的重点是在环境噪声评价的背景下,对两种标准化响度指标(Zwicker和Moore-Glasberg)进行系统比较,并与a加权声压级进行比较。为了支持这种比较,开发了一种高效实用的实时响度测量系统,并在基于平板电脑的多功能测量设备上实现了该系统。该系统可以在现场环境噪声监测期间同时计算和直接比较ISO 532-1和ISO 532-2的响度水平。首先,通过数值实验分析了每种方法的频率依赖性和水平依赖性。随后,进行了现场测量,以捕获各种环境噪声源在整个测量期间的瞬时响度水平和能量平均等效响度水平。结果一致地证明了数值实验和现场测量中响度指数的方法依赖差异。值得注意的是,根据噪声源的频率含量、时间变化和电平范围,观察到不同方法之间响度水平的逆转等特征现象。这些发现为标准响度指标在真实声学条件下的行为和可解释性提供了新的见解,并可能支持其在感知相关环境噪声评估中的更广泛应用。
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引用次数: 0
Ultra-broadband dual-mode nonlocal sound absorber based on phase gradient Fabry-Perot slit channels 基于相位梯度法布里-珀罗狭缝通道的超宽带双模非局域吸声器
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-17 DOI: 10.1016/j.apacoust.2025.111166
Jin He, Qingxuan Liang, Dichen Li
Acoustic artificial structures have promoted the high-performance of sound absorbing devices. However, the local coupling and inherent structural losses limit the development of ultra-0broadband sound absorbing devices. Here, we propose a dual-mode nonlocal design method to improve the performance of the structure-based absorbers. By utilizing the nonlocal coupling effect caused by multiple reflections of the period phases gradient structures, the losses in the units can be amplified without extra resonance. A transverse nonlocal coupling is then excited by introducing pores in the wall of Fabry-Perot channels to enhance and transfer the internal loss. The loss power density distribution uncovers the physical mechanism of the coupling effect. A dual-mode nonlocal absorber incorporating two types of walls is designed and experimentally validated, achieving an ultra-broadband sound absorption performance from 970 Hz to 10,000 Hz with an average absorption coefficient exceeding 0.9. Our design provides a new route to improve the absorption performance of the phase gradient absorbers without considering the strong resonance of the units.
声学人工结构的发展促进了吸声装置的高性能。然而,局部耦合和固有结构损耗限制了超宽带吸声器件的发展。在此,我们提出了一种双模非局部设计方法来改善基于结构的吸波器的性能。利用周期相位梯度结构的多次反射引起的非局部耦合效应,可以在不产生额外共振的情况下放大单元中的损耗。然后通过在法布里-珀罗通道壁上引入孔隙来激发横向非局部耦合,以增强和转移内部损失。损耗功率密度分布揭示了耦合效应的物理机制。设计并实验验证了一种包含两种壁的双模非局部吸声器,实现了970 ~ 10000 Hz的超宽带吸声性能,平均吸收系数超过0.9。我们的设计为在不考虑单元强共振的情况下提高相位梯度吸收器的吸收性能提供了一条新的途径。
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引用次数: 0
Impact of varying basalt fibre size on the acoustical properties of low carbon concrete 不同玄武岩纤维尺寸对低碳混凝土声学性能的影响
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-18 DOI: 10.1016/j.apacoust.2025.111167
Haydar Aygün, Amer Ali, Ali Bayyati
The impact of varying basalt fibre size and perforating concrete samples was investigated to improve the acoustical properties of low-carbon concrete. The research was focused on how the length and quantity of basalt fibres and perforation of samples could influence sound absorption and transmission, with an intention to apply them as sound reflectors and sound absorbers. Concrete samples both with and without perforations were designed and created. Their sound absorption coefficient and transmission loss values of the samples were measured applying transfer function method in a circular metal impedance tube. Two-Dimensional (2D) computational simulations were carried out on periodic perforated material that are mimicking perforated concrete samples to determine their sound transmission loss values and absorption coefficients applying the thermo-viscous acoustics, pressure acoustics, and solid mechanics models in COMSOL Multiphysics software. Computational results matched up well with the experimental results for perforated materials in one-third octave bands. The results reveal that varying basalt fibre size and quantity of them within materials significantly influences the sound absorption mechanism of the concrete causing the samples to behave like resonators around their natural frequency. Introducing perforations into samples enhanced sound absorption performance and moved the peak absorption to higher frequencies. However, the increase in sound absorption came with a trade-off, decreased sound transmission loss, meaning more sound passed through the perforated samples. Finally, the average sound absorption coefficients and transmission loss values of the basalt fibre reinforced samples were compared to those of glass-bead samples and to the concrete samples without basalt fibre infusion. The results clearly showed that adding basalt fibres to concrete boosted sound absorption coefficient and the frequency at which the peak sound absorption occurs was shifted to lower frequencies.
研究了不同玄武岩纤维尺寸和不同穿孔混凝土试样对低碳混凝土声学性能的影响。研究的重点是玄武岩纤维的长度和数量以及样品的穿孔如何影响声音的吸收和透射,目的是将它们用作声音反射器和吸声器。设计并制作了带孔和不带孔的混凝土样品。采用传递函数法在圆形金属阻抗管中测量了样品的吸声系数和透射损耗值。利用COMSOL Multiphysics软件中的热粘性声学、压力声学和固体力学模型,对模拟穿孔混凝土样品的周期性穿孔材料进行二维(2D)计算模拟,以确定其声传输损失值和吸收系数。在三分之一倍频带穿孔材料中,计算结果与实验结果吻合良好。结果表明,材料中玄武岩纤维的大小和数量的变化会显著影响混凝土的吸声机制,导致样品在其固有频率附近表现为谐振器。在样品中打孔提高了吸声性能,并将吸声峰移动到更高的频率。然而,声吸收的增加是有代价的,声音传输损失减少,这意味着更多的声音通过穿孔的样品。最后,将玄武岩纤维增强试样的平均吸声系数和透射损失值与玻璃球试样和未注入玄武岩纤维的混凝土试样进行了比较。结果清楚地表明,在混凝土中加入玄武岩纤维提高了混凝土的吸声系数,并且吸声峰值发生的频率向低频偏移。
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引用次数: 0
Sound reductions of plenum windows installed with cylindrical scatterer arrays in the presence of a linear sound source 在存在线性声源的情况下,安装圆柱形散射体阵列的通风窗的减声效果
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-26 DOI: 10.1016/j.apacoust.2025.111172
Wai Kit Lam , S.K. Tang
A scale model experiment was carried out in the present study to understand the sound transmission losses across acoustic plenum windows installed with the ‘sonic crystal like’ cylindrical scatterer arrays with square and rhombic lattices in the presence of an incoherent linear sound source. The effects of the band structures of these arrays, the plenum window dimensions and the sound source orientation angle on the sound transmission losses were also examined. The results illustrated that the traffic noise transmission loss increases with increasing filling fraction, but meaningful noise reduction enhancement is only possible at filling fractions greater than 0.033. A larger window gap width results in improved angular uniformity of noise reduction. Stronger sound transmission loss is achieved with those scatterer array configurations having a half scatterer attached to the window panes. The arrays with square lattice and half scatterers on the window panes gives the best sound reduction. It is also evident that the band structures of the arrays have no effect on the spectral characteristics of the sound transmission when the sound source is nearly parallel to the window. Strong influences of the standing waves formed within the window inlet cavity are observed under such circumstance. The bandgaps of the scatterer arrays in general run over regions of high spectral sound transmission loss or of strongly increasing sound transmission loss. However, the sound transmission losses vary within each bandgap. Besides, it is found that the density of the bandgaps does not have direct relationship with the magnitude of the sound transmission loss. The length-wise acoustic modes of the window cavity only have limited effect of the sound transmission loss even at frequencies below the lowest bandgap eigenfrequency.
在本研究中进行了一项比例模型实验,以了解在非相干线性声源存在的情况下,安装了带有方形和菱形晶格的“声晶体”圆柱形散射体阵列的声学静压窗的声音传输损失。研究了这些阵列的能带结构、充气窗尺寸和声源取向角对声传输损失的影响。结果表明:交通噪声传输损失随填充分数的增加而增加,但只有在填充分数大于0.033时才有明显的降噪效果。较大的窗隙宽度可改善降噪的角度均匀性。具有半散射体附在窗玻璃上的散射体阵列配置实现了更强的声音传输损失。窗玻璃上的方形栅格和半散射体阵列具有最佳的降噪效果。当声源几乎平行于窗口时,阵列的波段结构对声传输的光谱特性没有影响。在这种情况下,观察到在窗口入口腔内形成的驻波的强烈影响。散射体阵列的带隙一般在高光谱声传输损耗或声传输损耗急剧增加的区域上运行。然而,声传输损失在每个带隙内是不同的。此外,还发现带隙的密度与声传输损失的大小没有直接关系。即使在低于最低带隙特征频率的频率下,窗腔的长度方向声模对声传输损失的影响也有限。
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引用次数: 0
Analog emulator designs for the auditory neurons and their performance comparison with the dynamic time warping algorithm 听觉神经元模拟模拟器的设计及其与动态时间规整算法的性能比较
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-28 DOI: 10.1016/j.apacoust.2025.111173
Nimet Korkmaz, Mustafa Molu
In auditory system, after the frequencies of sound signals are modulated by the basilar membrane, these sounds of different frequencies are processed by different auditory neurons. Design of the broadband electronic circuits is a critical issue to emulate the auditory neurons, because these emulator circuit processes the sound signal of various frequency without require any update. In this study, it is aimed to compare the response performance of the auditory neuron emulator circuits that are built by several analog devices. In this context, an auditory neuron model, which is defined by adapting the vibrations in a piezoelectric ceramic to the FitzHugh–Nagumo neuron model, is successfully implemented with voltage operational amplifier, operational transconductance amplifiers, current feedback operational amplifiers and field programmable analog arrays. After the notes of a musical composition are coded in a microcontroller, its electrical signal output is applied to the auditory neuron emulator circuits as an external excitation signal. The peaks of the excitation signal applying to the emulator circuits and the firing times of the dynamic responses measuring from the outputs of these analog circuits are recorded simultaneously. Then, time differences between stimulus peaks and neuronal firing responses are determined with the “Dynamic Time Warping” algorithm and the input/output synchronization performances of these analog emulators are compared according to the result of this quantitative measurement.
在听觉系统中,声音信号的频率经过基底膜的调制后,这些不同频率的声音被不同的听觉神经元处理。宽带电子电路的设计是模拟听觉神经元的关键问题,因为这些仿真电路可以处理不同频率的声音信号而不需要任何更新。在本研究中,旨在比较由几种模拟设备构建的听觉神经元仿真电路的响应性能。在此背景下,通过将压电陶瓷中的振动适应FitzHugh-Nagumo神经元模型来定义听觉神经元模型,并成功地实现了电压运算放大器,运算跨导放大器,电流反馈运算放大器和现场可编程模拟阵列。乐曲的音符在微控制器中编码后,其电信号输出作为外部激励信号被应用到听觉神经元仿真电路中。同时记录应用于仿真电路的激励信号的峰值和从这些模拟电路的输出测量的动态响应的触发时间。然后,利用“动态时间扭曲”算法确定刺激峰值与神经元放电响应的时间差,并根据这一定量测量结果比较这些模拟模拟器的输入/输出同步性能。
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引用次数: 0
Identification of common excess-phase zeros for single-input multiple-output room response equalization via ringing quantification 通过振铃量化识别单输入多输出房间响应均衡的常见过相零
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-21 DOI: 10.1016/j.apacoust.2025.111153
Dongcai Wang, Ziyun Liu, Yiming Han, Keyu Pan, Yong Shen
Mixed-phase equalization filters are widely employed for enhancing multi-position audio quality in single-input multiple-output (SIMO) room response equalization systems. A spatially robust polynomial equalization method based on identification of common excess-phase zeros (CEZ) has been proposed to design mixed-phase filters that mitigate magnitude distortions and pre-ringing effects. However, existing CEZ identification methods are limited by clustering frameworks and fail to consider key factors such as filter design strategy, truncation length and mutual modulation effects, resulting in ineffective CEZ identification. To address this, a systematic three-stage CEZ identification method is proposed. First, excess-phase zeros are rapidly computed via sub-band decomposition. Next, relative CEZ clusters are extracted across frequency bands. Finally, CEZ are identified through sorting and screening of these candidates. The proposed method is evaluated using impulse responses from realistic environments to compare various CEZ identification approaches. Additionally, the effect of different parameter settings on equalization performance is examined. Objective and subjective evaluations indicate that filters designed with the CEZ identified by the proposed method achieve optimal filtering performance. The proposed method enhances the performance of SIMO mixed-phase filters, providing a systematic solution for achieving robust room response equalization in complex acoustic environments.
在单输入多输出(SIMO)房间响应均衡系统中,混合相位均衡滤波器被广泛用于提高多位置音频质量。提出了一种基于共过相零识别的空间鲁棒多项式均衡方法,用于设计抑制幅度失真和预振铃效应的混合相位滤波器。然而,现有的CEZ识别方法受聚类框架的限制,没有考虑滤波器设计策略、截断长度和互调制效应等关键因素,导致CEZ识别效果不佳。为了解决这一问题,提出了一种系统的三阶段CEZ识别方法。首先,通过子带分解快速计算多余的相位零。接下来,提取各频段的相对CEZ簇。最后,通过这些候选人的分类和筛选来确定CEZ。利用真实环境的脉冲响应对所提出的方法进行了评估,以比较各种CEZ识别方法。此外,还考察了不同参数设置对均衡性能的影响。客观和主观评价表明,采用该方法识别的CEZ设计的滤波器具有最优的滤波性能。该方法提高了SIMO混合相位滤波器的性能,为在复杂声环境中实现稳健的房间响应均衡提供了系统的解决方案。
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引用次数: 0
Spectral and tonality characterization of unwanted environmental sounds for automatic noise assessment 用于自动噪声评估的不需要的环境声音的频谱和调性特征
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-26 DOI: 10.1016/j.apacoust.2025.111171
Luca Fredianelli , Gino Iannace , Andac Akbaba
In environmental acoustics, especially during long-term outdoor measurements, unwanted sounds can compromise the reliability of noise assessments. Traditionally, their identification and removal require extensive manual work. Recent advances in machine learning for Sound Event Classification and Detection offer new perspectives for automating such tasks. This study, based on the publicly available DataSEC dataset, presents a spectral and tonal characterization of 40 classes of environmental sounds commonly encountered in urban and rural contexts. Third-octave spectra, spectrograms, and probabilistic tonality maps are analyzed to describe intra- and inter-class variability and to identify recurring spectral structures. The results provide a reference framework for understanding how different sources contribute to tonal and broadband features in environmental recordings. The analysis also discusses how these descriptors can support future automated workflows, including frequency-domain post-processing or spectral smoothing strategies, without claiming experimental validation of such methods. By offering comprehensive, normalized spectral and tonal features, this work provides an open reference for both acoustic professionals and researchers aiming to develop or train machine learning models for environmental sound assessment and event recognition.
在环境声学中,特别是在长期的室外测量中,不必要的声音会损害噪声评估的可靠性。传统上,它们的识别和移除需要大量的手工工作。机器学习在声音事件分类和检测方面的最新进展为自动化这些任务提供了新的视角。本研究基于公开可用的DataSEC数据集,展示了城市和农村环境中常见的40类环境声音的频谱和音调特征。分析了三倍频谱、谱图和概率调性图,以描述类内和类间的变异性,并识别重复出现的谱结构。结果提供了一个参考框架,了解不同的来源如何有助于音调和宽带特征的环境记录。分析还讨论了这些描述符如何支持未来的自动化工作流程,包括频域后处理或频谱平滑策略,而没有要求对这些方法进行实验验证。通过提供全面的、标准化的光谱和音调特征,这项工作为声学专业人员和旨在开发或训练用于环境声音评估和事件识别的机器学习模型的研究人员提供了一个开放的参考。
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引用次数: 0
Acoustic reflection modeling and experimental study of rough fractal interfaces for dry-coupled ultrasonic testing 干耦合超声检测粗糙分形界面声反射建模及实验研究
IF 3.4 2区 物理与天体物理 Q1 ACOUSTICS Pub Date : 2026-02-15 Epub Date: 2025-11-28 DOI: 10.1016/j.apacoust.2025.111174
Yuanyao Hu , Donglin Tang , Sheng Rao , Chao Ding , Weiping Wu
This study investigates the enhancement of acoustic performance in ultrasonic dry coupling for solid rough interfaces, focusing on the coupling characteristics of ultrasound propagation through rough contact interfaces. A dry coupling acoustic model is developed based on the Distributed Spring Model (DSM), and a contact stiffness expression incorporating the fractal dimension is derived. Numerical simulations are conducted to analyze the relationship between interface coupling stiffness and load, and a piecewise mapping method is proposed to provide a quantitative basis for precise external load setting. To validate the theoretical model, an ultrasonic dry coupling experimental platform was constructed. The experimental results show that applying an appropriate external load improves the acoustic coupling effect, and the higher the fractal dimension, the lower the load required to achieve ideal coupling. Compared to traditional roughness parameters, the fractal dimension significantly improves measurement stability, reducing the standard deviation by 79.3%. Using the portable dry coupling wheel probe for thickness reduction measurement and applying the fractal dimension method, the minimum load required for optimal coupling was determined. The actual measurement error was maintained within 3%, confirming the feasibility of dry coupling technology for damage detection. Therefore, this study provides theoretical support and a quantitative reference for the performance optimization of dry coupling ultrasonic testing systems.
本文研究了固体粗糙界面下超声干耦合对声学性能的增强,重点研究了超声通过粗糙接触界面传播的耦合特性。基于分布式弹簧模型(DSM)建立了干耦合声学模型,推导了包含分形维数的接触刚度表达式。通过数值模拟分析了界面耦合刚度与载荷之间的关系,提出了分段映射法,为外载荷的精确设置提供了定量依据。为了验证理论模型,搭建了超声干耦合实验平台。实验结果表明,施加适当的外载荷可以改善声耦合效果,分形维数越高,实现理想耦合所需的载荷越低。与传统粗糙度参数相比,分形维数显著提高了测量稳定性,标准差降低79.3%。采用便携式干式耦合轮测厚仪进行减厚测量,并应用分形维数法确定了最优耦合所需的最小载荷。实际测量误差保持在3%以内,证实了干耦合技术用于损伤检测的可行性。因此,本研究为干耦合超声检测系统的性能优化提供了理论支持和定量参考。
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引用次数: 0
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