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Acoustic Characterization of Hemadpanti-Style Hindu Temples: A Case Study of the Markanda and Mrikunda Temples Hemadpanti 式印度教寺庙的声学特征:马尔坎达庙和姆里昆达庙案例研究
IF 1.7 4区 物理与天体物理 Pub Date : 2024-05-08 DOI: 10.1007/s40857-024-00320-4
Apoorva A. Dandge, Akshay P. Patil

This article presents the acoustic characterization of two well-known Hemadpanti-style Indian Hindu temples in Maharashtra, India, built during the twelfth century. The studies of architectural acoustics in Indian Hindu temples are sparse. Therefore, characterizing the acoustic nature of such historical Hindu temples is vital. This study may provide insight into the role of architectural characteristics that support the desired sound field, ensuring that the music ritual, singing of devotional songs, and Vedic chanting are suitable in Hemadpanti-style Hindu temples. The research aimed to report and investigate the acoustic behavior of the Hindu temples through in-situ measurements in an unoccupied condition. Virtual acoustic models were developed and validated using the in-situ measurements under the same conditions. Objective room acoustic indicators considered are reverberation time (T30), clarity of music (C80), and Speech Transmission Index (STI), which are later simulated and analyzed for two sound source positions in occupied conditions. The results report that the spatially and spectrally unoccupied averaged values for reverberation time (T30) and clarity of music (C80) of the Markanda temple are 0.98 s and 3.98 dB, and the Mrikunda temple (T30) and (C80) values are 0.73 s and 5.62 dB respectively. The values obtained for both temples are within the optimum range adopted for this study. The average subjective rating for speech intelligibility of the Markanda and the Mrikunda temples is “good”. After analyzing indicators, the results emphasize the influence of architectural features on the acoustic characteristics of the Hemadpanti style of Hindu temples.

本文介绍了印度马哈拉施特拉邦两座著名的 Hemadpanti 风格印度教寺庙的声学特征,这两座寺庙建于 12 世纪。对印度教寺庙建筑声学的研究很少。因此,确定这些历史悠久的印度教寺庙的声学特性至关重要。这项研究可以让人们深入了解建筑特点在支持理想声场方面的作用,从而确保在 Hemadpanti 风格的印度教寺庙中适合音乐仪式、虔诚歌曲的演唱和吠陀吟唱。这项研究旨在通过在无人居住的情况下进行现场测量,报告和调查印度教寺庙的声学行为。利用相同条件下的现场测量结果,开发并验证了虚拟声学模型。客观的室内声学指标包括混响时间(T30)、音乐清晰度(C80)和语音传输指数(STI)。结果表明,Markanda 寺庙的混响时间(T30)和音乐清晰度(C80)的空间和频谱非占用平均值分别为 0.98 秒和 3.98 分贝,而 Mrikunda 寺庙的混响时间(T30)和音乐清晰度(C80)值分别为 0.73 秒和 5.62 分贝。两座寺庙的数值都在本研究采用的最佳范围之内。马尔康达寺和姆里坤达寺的语音清晰度平均主观评分为 "良好"。在对各项指标进行分析后,结果强调了建筑特征对 Hemadpanti 风格印度教寺庙声学特征的影响。
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引用次数: 0
Virtual Augmentation of the Beamforming Array Based on a Sub-cross-spectral Matrix Computation for Localizing Stationary Signal Noise Sources 基于子跨谱矩阵计算的波束成形阵列虚拟增强,用于定位固定信号噪声源
IF 1.7 4区 物理与天体物理 Pub Date : 2024-05-07 DOI: 10.1007/s40857-024-00322-2
R. Singh, A. Mimani

This paper presents a generalized algorithm called the sub-cross-spectral matrix (SCSM) beamforming technique for the virtual augmentation of an N-channel beamforming array based on sequential computation of the cross-spectral matrix (CSM) terms for localizing stationary signal sources. To this end, first, the diagonal sub-cross-spectral matrices (SCSMs) of the N-channel array pertaining to M different spatial locations were obtained. Next, the off-diagonal SCSMs were systematically computed by directly evaluating the cross-spectral terms between some microphones placed in the array at (i{text{th}}) location ((1 le i le M)) and the remaining microphones placed in the array at (j{text{th}}) location ((j ne i, , 1 le j le M)). As a proof of concept, the SCSM beamforming was used to virtually construct a 32-channel planar Underbrink spiral array by sequentially measuring data using (left( {begin{array}{*{20}c} {32} 2 end{array} } right)) microphone pairs. The resultant 2-D beamforming map of a loudspeaker source was found to be nearly identical to the counterpart result produced when data from 32-channel simultaneous measurements were used. The SCSM technique was then extended to increase the density and aperture of a planar array by constructing a virtual 64-channel planar array from 32-channel simultaneous measurements. For the former case, the source maps were found to be identical to the counterpart results obtained from the existing geometric mean and combined CSM algorithms. However, for the latter case, the SCSM beamforming delivered a noticeably improved focal-resolution along the direction in which there was a virtual increase in aperture. For localizing loudspeaker source(s) in a 3-D domain, the SCSM beamforming implemented using two orthogonal Underbrink arrays was shown to deliver a significantly improved resolution (focal lobe) and unambiguous localization because it considers the complete CSM unlike the multiplicative beamforming and combined CSM algorithms which do not account for the phase-information between the two orthogonal arrays.

本文提出了一种名为 "子交叉谱矩阵(SCSM)波束成形技术 "的通用算法,该算法基于用于定位静止信号源的交叉谱矩阵(CSM)项的连续计算,用于对 N 信道波束成形阵列进行虚拟增强。为此,首先要获得 N 信道阵列与 M 个不同空间位置相关的对角子交叉谱矩阵(SCSM)。接下来,通过直接评估阵列中位于 (i{text{th}}) 位置 ((1le i le M))的部分麦克风与阵列中位于 (j{text{th}}) 位置 ((j ne i, , 1 le j le M))的其余麦克风之间的交叉谱项,系统地计算了非对角 SCSMs。作为概念验证,SCSM 波束成形被用于通过使用 (left( {begin{array}{*{20}c} {32} 2 end{array} } right)) 麦克风对顺序测量数据来虚拟构建一个 32 通道平面 Underbrink 螺旋阵列。结果发现,扬声器声源的二维波束成形图与使用 32 通道同步测量数据时产生的对应结果几乎相同。随后,SCSM 技术被扩展用于增加平面阵列的密度和孔径,即通过 32 通道同步测量构建一个虚拟的 64 通道平面阵列。在前一种情况下,源图与现有几何平均算法和 CSM 组合算法得到的对应结果相同。然而,在后一种情况下,SCSM 波束成形在虚拟孔径增加的方向上明显提高了焦点分辨率。在三维域中定位扬声器声源时,使用两个正交 Underbrink 阵列实现的 SCSM 波束成形可显著提高分辨率(焦叶)和定位的准确性,因为它考虑了完整的 CSM,而乘法波束成形和组合 CSM 算法则不同,它们不考虑两个正交阵列之间的相位信息。
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引用次数: 0
Multi-channel ANC System with Online Secondary Path Modeling for Turboprop Aircraft Cabin 用于涡轮螺旋桨飞机客舱的带在线次级路径建模的多通道 ANC 系统
IF 1.7 4区 物理与天体物理 Pub Date : 2024-04-23 DOI: 10.1007/s40857-024-00317-z
Hao Shen, Qing Xue, Ningjuan Dong, Yixiao Chen, Xing Shen

The engineering implementation of the multi-channel active noise control (MCANC) system for turboprop aircraft cabin is seriously hampered by its enormous computational complexity. This paper proposes the variable-P-sequential-partial-update filtered-x least mean square (VP-SPUFxLMS) algorithm, which achieves noise reduction performance comparable to that of the multi-channel FxLMS (MCFxLMS) algorithm while significantly reducing the computational complexity. Additionally, considering the time-varying nature of the secondary paths in practical applications, the Eriksson online secondary path modeling (OSPM) method is extended from single-channel to multi-channel, the problems that may be faced when the method is applied to MCANC systems are analyzed, and an improved alternative online secondary path modeling (AOSPM) method is proposed to address the above problems, which exhibits great online modeling capabilities without introducing excessive computational load. Simulation and experiment results validate the noise control performance of the proposed method, and the ANC experiment has achieved an average reduction of more than 15 dB in the sound pressure level (SPL) of the four channels, which fully demonstrates its broad engineering application prospects.

涡轮螺旋桨飞机客舱多通道有源噪声控制系统(MCANC)的工程实施因其巨大的计算复杂性而受到严重阻碍。本文提出了可变-P-序列-部分更新滤波-x 最小均方差(VP-SPUFxLMS)算法,在显著降低计算复杂度的同时,实现了与多通道 FxLMS(MCFxLMS)算法相当的降噪性能。此外,考虑到实际应用中二次路径的时变性,将 Eriksson 在线二次路径建模(OSPM)方法从单通道扩展到多通道,分析了该方法应用于 MCANC 系统时可能面临的问题,并针对上述问题提出了一种改进的替代在线二次路径建模(AOSPM)方法,该方法在不引入过多计算负荷的情况下展现了强大的在线建模能力。仿真和实验结果验证了所提方法的噪声控制性能,ANC 实验实现了四个通道声压级 (SPL) 平均降低 15 dB 以上,充分展示了其广阔的工程应用前景。
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引用次数: 0
Assessing the Acoustic Noise in Intensive Care Units via Deep Learning Technique 通过深度学习技术评估重症监护室的声学噪声
IF 1.7 4区 物理与天体物理 Pub Date : 2024-04-22 DOI: 10.1007/s40857-024-00321-3
Awwab Qasim Jumaah Althahab, Branislav Vuksanovic, Mohamed Al-Mosawi, Hongjie Ma

Intensive care unit (ICU) noise is a critical and often overlooked issue, impacting patient recovery and healthcare staff well-being. Existing research primarily relies on costly sound level meters for monitoring noise levels, where the characteristics of noise sources cannot be determined and discriminated. This study employs deep neural networks to detect and classify ICU noise events, enhancing source identification. A cost-effective internet of things-based audio recording and monitoring system has been designed and deployed in three ICUs for data collection. The acoustic event classification system described in the paper integrates convolutional neural networks for event detection, followed by clustering to isolate noise sources. Results demonstrate precise classification, with speech identified as a major contributor in all ICUs. This model offers valuable insights for characterising acoustic sources in typical ICUs, which could be the first step towards tackling the problem of excessive noise in ICUs as well as a starting point for further research in this area.

重症监护室(ICU)的噪声是一个关键问题,但往往被忽视,影响着病人的康复和医护人员的健康。现有的研究主要依赖昂贵的声级计来监测噪声水平,而声级计无法确定和区分噪声源的特征。本研究采用深度神经网络来检测 ICU 噪音事件并对其进行分类,从而提高了声源识别能力。我们设计了一套经济高效的基于物联网的音频记录和监测系统,并将其部署在三间重症监护室进行数据收集。论文中描述的声学事件分类系统集成了用于事件检测的卷积神经网络,然后通过聚类来隔离噪声源。结果表明,分类准确,在所有重症监护室中,语音都是主要的噪声源。该模型为描述典型重症监护室的声源特征提供了宝贵的见解,这可能是解决重症监护室噪音过大问题的第一步,也是该领域进一步研究的起点。
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引用次数: 0
Combining Statistical Noise Levels and Application to Wind Farm Guidelines 综合统计噪声水平并应用于风电场指南
IF 1.7 4区 物理与天体物理 Pub Date : 2024-04-13 DOI: 10.1007/s40857-024-00318-y
Renzo Tonin

It is stated in wind farm standards that logarithmic addition and subtraction of LAF90,T sound pressure levels is “not strictly mathematically correct”. An analytical and experimental study reported in Tonin 2024 (a related article) examines the underlying accuracy of combining statistical noise levels as a general proposition, particularly the LAF90,T and the LAF10,T. The objective of that study was to explore the accuracy of combining statistical noise levels and what might influence that accuracy. The objective of this study, as foreshadowed in Tonin 2024, is to apply the results to wind farms. It was concluded in Tonin 2024 that values of D90 (being the difference between the logarithmic sum and actual values of LAF90,T) are negative in the range 0 to − 3 dB (for the cases in the study), meaning that the logarithmic sum of LAF90,T for the ambient and source sound pressure level distributions is less than the actual value of LAF90,T for the combined distribution. As a result, in deriving the wind farm noise level (as a contribution), the actual value of LAF90,T will be less than that determined by logarithmic subtraction of the individual components. In respect of the question of the underlying accuracy of combining statistical noise levels for wind farms, it is concluded that the difference between the logarithmic addition of the LAF90,T and the true value is less than 1 dB (for the cases in the study). The results are applied herein to a typical wind farm concluding that the simple energy subtraction method adopted in wind farm guidelines is conservative even allowing for the hypothesis that the fluctuation strength of wind farm noise is not invariant but increases with distance. It is also concluded that if wind farm guidelines were to assess wind farm noise on the basis of LAeq,T rather than LAF90,T then adding a value of 2.5 dB to the derived wind farm noise level LAF90,T as currently specified in the guidelines (i.e., with D90 = 0 dB) would be conservative even allowing for the hypothesis that the fluctuation strength of wind farm noise is not invariant but increases with distance.

风电场标准中规定,LAF90,T 声压级的对数加减法 "严格来说在数学上是不正确的"。Tonin 2024(相关文章)中报告的一项分析和实验研究考察了作为一般命题的统计噪声级组合的基本准确性,特别是 LAF90T 和 LAF10T。该研究的目的是探索统计噪声级组合的准确性以及可能影响准确性的因素。正如 Tonin 2024 所预示的,本研究的目的是将结果应用于风电场。Tonin 2024 中得出的结论是,D90 值(即 LAF90,T 的对数和与实际值之差)在 0 到 - 3 dB 范围内为负值(针对研究中的案例),这意味着环境声压级分布和声源声压级分布的 LAF90,T 对数和小于合并分布的 LAF90,T 实际值。因此,在得出风电场噪声级(作为贡献值)时,LAF90,T 的实际值将小于按对数减去各个组成部分后确定的值。关于风电场噪声水平统计组合的基本准确性问题,结论是 LAF90,T 的对数加法与真实值之间的差异小于 1 分贝(就研究中的案例而言)。本文将结果应用于一个典型的风电场,得出的结论是,即使考虑到风电场噪声的波动强度不是不变的,而是随着距离的增加而增加这一假设,风电场指南中采用的简单能量减法也是保守的。此外还得出结论,如果风电场指南是根据 LAeq,T 而不是 LAF90,T 来评估风电场噪声,那么即使考虑到风电场噪声的波动强度不是不变的,而是随距离增加而增加的假设,在指南中目前规定的风电场噪声级 LAF90,T 的基础上增加 2.5 dB 值(即 D90 = 0 dB)也是保守的。
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引用次数: 0
Influence of Glazing on Sound Quality in the Car: Validation of Auralizations Obtained from SEA Calculations 玻璃对车内音质的影响:验证通过 SEA 计算获得的听觉效果
IF 1.7 4区 物理与天体物理 Pub Date : 2024-04-09 DOI: 10.1007/s40857-024-00315-1
Valentin Miqueau, Etienne Parizet, Sylvain Germes

This paper explores the possibility to use statistical energy analysis (SEA)-based computations to synthesize sounds that can be used in a subjective evaluation of the unpleasantness of exterior noises transmitted in the car compartment through the glazing. A medium family car (C-segment car) was placed in a reverberation room. A sound source was placed outside the car. The resulting noise was measured at the driver’s position for nineteen different configurations of glazing. The transmission loss (TL) of each car window was computed using an in-house software and used in a SEA-based vibroacoustic synthesis model. The nineteen corresponding configurations were simulated. A listening test experiment was conducted to compare the signals synthesized from the measurements and from the simulations. The results showed a good agreement between the unpleasantness ratings of each glazing configuration. However, in the case of tempered glasses, a slight difference in the ratings was detected. Further analysis showed that this was due to an inaccurate prediction of the TL of the glazing, around its coincidence frequency. Additional measurements proved that this could be related to an underestimation of the damping. More precisely, because the intrinsic damping of a tempered glass is very low, the additional damping brought by the window seals must be taken into account. Further measurements were made to estimate the TL of a tempered glass mounted on a gasket. The use of these new values in the SEA calculation allowed for the correction of these difference in subjective ratings. The SEA computations can thus be used in the acoustic design process of cars.

本文探讨了使用基于统计能量分析(SEA)的计算来合成声音的可能性,这些声音可用于主观评价通过玻璃在车厢内传播的外部噪音的难听程度。一辆中型家用轿车(C 级轿车)被放置在混响室中。声源置于车外。在驾驶员位置测量了 19 种不同配置的玻璃所产生的噪音。每个车窗的传输损耗(TL)都是通过内部软件计算得出的,并用于基于 SEA 的振动声学合成模型。模拟了 19 种相应的配置。为了比较测量和模拟合成的信号,进行了听力测试实验。结果表明,每种玻璃配置的不悦度评级都非常一致。不过,在钢化玻璃的情况下,检测到的评分略有不同。进一步的分析表明,这是由于围绕玻璃的重合频率,对玻璃的 TL 预测不准确造成的。额外的测量证明,这可能与低估了阻尼有关。更确切地说,由于钢化玻璃的固有阻尼非常低,因此必须考虑到窗户密封条带来的额外阻尼。通过进一步测量,我们估算出了安装在密封垫上的钢化玻璃的 TL 值。在 SEA 计算中使用这些新值可以修正这些主观评级的差异。因此,SEA 计算可用于汽车声学设计过程。
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引用次数: 0
News Item 新闻项目
IF 1.7 4区 物理与天体物理 Pub Date : 2024-04-09 DOI: 10.1007/s40857-024-00319-x
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引用次数: 0
Do Whales Sing to Their Own Tune? Comparing the Variability Within and Between Song Events of the Eastern Indian Ocean Pygmy Blue Whale 鲸鱼按照自己的曲调唱歌吗?比较东印度洋侏儒蓝鲸鸣唱事件内部和之间的可变性
IF 1.7 4区 物理与天体物理 Pub Date : 2024-03-10 DOI: 10.1007/s40857-024-00314-2
Capri D. Jolliffe, Robert D. McCauley, Alexander N. Gavrilov

Acoustic data from the Perth Canyon, Western Australia, were collected for the 2017 northern migration allowing for detailed acoustic analysis of eastern Indian Ocean pygmy blue (EIOPB) whale songs within a migratory season to explore fine-scale variability in song production. An algorithm was used to follow the unit II signal in time, tracking the change in frequency over the duration of the signal and enabling a comparison of song unit production within and between singing bouts. The results of this analysis indicate that units from within the same song event have relatively consistent characteristics but vary between song events, suggesting it is possible that individual whales may have distinct vocal characteristics. The presence of breaks within a unit was identified as a significant level of variability in song production within the 2017 data set and was seen to increase throughout the season. It is hypothesised that unit breaks may play a role in intra-species communication as well as represent a novel variation to song production that increases song complexity and thus may increase individual fitness through sexual selection.

从西澳大利亚珀斯峡谷收集了 2017 年北迁徙的声学数据,从而能够在一个迁徙季节内对东印度洋侏儒蓝鲸(EIOPB)的歌声进行详细的声学分析,以探索歌声产生的细微变化。使用一种算法对第二单元信号进行时间跟踪,追踪信号持续时间内的频率变化,从而能够比较各次鸣唱中和鸣唱之间的鸣唱单元产生情况。分析结果表明,同一歌声事件中的单元具有相对一致的特征,但不同歌声事件中的单元却各不相同,这表明鲸鱼个体可能具有不同的发声特征。在 2017 年的数据集中,一个单元内出现的间断被认为是鸣唱过程中的一个重要变异水平,并且在整个季节中呈上升趋势。据推测,单元断裂可能在种内交流中发挥作用,同时也代表了一种新的鸣声变异,它增加了鸣声的复杂性,从而可能通过性选择提高个体的适应性。
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引用次数: 0
Interior Noise Prediction of Metro Train in a Tunnel Caused by Wheel/Rail Rolling 由车轮/铁轨滚动引起的隧道内地铁列车内部噪声预测
IF 1.7 4区 物理与天体物理 Pub Date : 2024-03-05 DOI: 10.1007/s40857-024-00316-0
Yunfei Zhang, Li Li, Hongxiao Li

Metro running causes wheel/rail rolling radiation noise and reflects multiple times between the tunnel wall and the car body. Reverberation in a tunnel increases the interior noise and reduces riding comfort. A statistical energy analysis (SEA) model for a metro train in a tunnel is proposed to predict interior noise and improve ride comfort. The model considers the acoustic excitation caused by wheel/rail rolling, the damping/coupling loss factors, reverberation time in the tunnel/coach, and the equivalent panels. The results show that the error between the simulation and the measured is 3–6 dB; the SEA model is available. The mechanical wave of symmetrical loading may cancel out on the plane of symmetry. At low frequencies, the difference between the internal and external noise is slight (10 dBA), the transmission is robust, and the sound insulation of the car body is weak. In contrast, at high frequencies, the difference is significant (25 dBA). The tunnel reverberation effect increases the sound pressure inside the car by 8–12 dBA than the open-line, and the reverberation will reduce the spatial distribution gradient of the interior noise. Applying noise control treatment on the tunnel’s inner wall can reduce the noise by 5–10 dBA.

地铁运行会产生车轮/轨道滚动辐射噪声,并在隧道壁和车体之间多次反射。隧道内的混响会增加内部噪声,降低乘坐舒适度。本文提出了一种隧道内地铁列车的统计能量分析(SEA)模型,以预测车内噪声并改善乘坐舒适性。该模型考虑了车轮/轨道滚动引起的声激励、阻尼/耦合损失因子、隧道/车厢内的混响时间以及等效面板。结果表明,模拟与测量之间的误差为 3-6 dB;SEA 模型可用。对称加载的机械波可能会在对称面上抵消。在低频情况下,内部噪声和外部噪声之间的差异很小(10 dBA),传导性很强,车身的隔音效果较弱。相反,在高频情况下,两者的差异很大(25 dBA)。隧道混响效应使车内声压比开放线增加 8-12 dBA,混响将减小车内噪声的空间分布梯度。对隧道内壁进行噪声控制处理可使噪声降低 5-10 分贝。
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引用次数: 0
Vibrations of a Violin While the f-Holes were Sequentially Cut 依次切割 f 孔时小提琴的振动
IF 1.7 4区 物理与天体物理 Pub Date : 2024-02-22 DOI: 10.1007/s40857-024-00313-3
Jesús A. Torres, Liline-Daniel Canales-Hernández, Patricia Alejandra Acosta, José-Joel González-Barbosa, Juan-B. Hurtado-Ramos

The impact of cutting the f-holes in a completed violin was investigated based on two measurable characteristics of the instrument. One of these aspects focused on the signature modes, while the other involved monitoring the graph of the bridge mobility. A particular emphasis was placed on analyzing this graph due to existing literature suggesting the connection of the bridge hill to the presence of f-holes. In the case of two real violins, bridge mobility was measured using an ultra-miniature accelerometer and an impact hammer. Concurrently, signature modes were recorded using Electronic Speckle Pattern Interferometry, an optical technique that performs well on rough surfaces; hence, the violins employed were intentionally left unvarnished. One of the violins lacked f-holes but included the rest of the structural elements. It was measured in stages as the f-holes were cut. The second violin adhered to a standard design but featured slightly wider f-holes, serving as a reference. Additionally, bridge mobility and signature modes were simulated for a finite element violin soundbox both with and without f-holes. Contrary to expectations, both the experiments and simulations revealed the emergence of the Bridge Hill even in the absence of f-holes. However, the alteration of the middle and high-frequency response was evident during the f-hole cutting process, accompanied by the observation of CBR as well as (B_1^-) and (B_1^+) modes.

根据小提琴的两个可测量特征,研究了在一把已完成的小提琴上切割 f 孔的影响。其中一个方面侧重于签名模式,而另一个方面则涉及监测琴桥移动性的曲线图。由于现有的文献表明琴桥山丘与 f 孔的存在有关,因此特别强调了对这一图形的分析。在两把真琴的情况下,使用超小型加速度计和冲击锤测量了琴桥的流动性。与此同时,使用电子斑点模式干涉仪记录了签名模式,这种光学技术在粗糙表面上表现良好;因此,所使用的小提琴故意没有上漆。其中一把小提琴没有f孔,但包含其他结构元素。在切割 f 孔时分阶段对其进行了测量。第二把小提琴采用了标准设计,但f孔稍宽,作为参考。此外,还模拟了有孔和无孔有限元小提琴音箱的琴桥流动性和特征模式。与预期相反,实验和模拟结果均显示,即使没有孔,也会出现桥山。然而,在f孔切割过程中,中频和高频响应的改变是显而易见的,同时还观察到了CBR以及(B_1^-)和(B_1^+)模式。
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Acoustics Australia
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