Sound Field Analysis Research Articles

Flow and sound field analysis of agricultural ultrasonic atomizing nozzle

To solve the problem of large size of fog droplets generated in plant protection, which are not conducive to absorption by target plants and result in pollution due to excessive application, an ultrasonic atomizing nozzle suitable for agricultural plant protection was designed. First, a geometric model of the agricultural ultrasonic atomizing nozzle was established using the Design Modeler module in ANSYS FLUENT. The FLUENT simulation software program was then employed to simulate the internal flow field of the nozzle, and the internal flow field cloud image and sound pressure for various cavity depths and cavity diameters were investigated. Finally, the vapor holdup of the flow field inside the nozzle were simulated. The results indicate that the internal cavity depth and diameter of the agricultural ultrasonic atomizing nozzle affect the generation of a cavitation vortex inside the nozzle and the magnitude of the sound pressure. As the cavity depth and diameter are increased, the amplitude of sound pressure first increases and then gradually decreases. The cavity diameter has a stronger influence on the amplitude of sound pressure than the cavity depth does. The sound pressure amplitude changes marginally with the cavity depth. Simulation revealed that the ultrasonic intensity is highest and the corresponding atomization effect is strongest when the depth and diameter of the of the resonant cavity are 4 and3 mm, respectively. When the inlet pressure is 2MPa, the percentage of the flow field of the ultrasonic atomizing nozzle with vapor content higher than 80% is approximately 33.94% higher than that achieved before parameter optimization. The effective space utilization rate inside the nozzle is improved. Keywords: ultrasonic atomization, resonant cavity, sound pressure amplitude, vapor content; computational fluid dynamics DOI:  10.33440/j.ijpaa.20190202.36.  Citation: Gong J L, Wang M X, Zhang Y F, Lan Y B, Mostafa K. Flow and sound field analysis of agricultural ultrasonic atomizing nozzle.  Int J Precis Agric Aviat, 2019; 2(2): 32–37.

Introduction of generalized curvilinear coordinates to spectral nodal Galerkin methods for irregular-shaped two-dimensional sound field analysis

Introduction of generalized curvilinear coordinates to spectral nodal Galerkin methods for irregular-shaped two-dimensional sound field analysis

Diffusion Equation-Based Finite Element Modeling of a Monumental Worship Space

In this work, a diffusion equation model (DEM) is applied to a room acoustics case for in-depth sound field analysis. Background of the theory, the governing and boundary equations specifically applicable to this study are presented. A three-dimensional geometric model of a monumental worship space is composed. The DEM is solved over this model in a finite element framework to obtain sound energy densities. The sound field within the monument is numerically assessed; spatial sound energy distributions and flow vector analysis are conducted through the time-dependent DEM solutions.

Modeling of reverberant room responses for two-dimensional spatial sound field analysis and synthesis.

A unified framework is proposed for analysis and synthesis of two-dimensional spatial sound field in reverberant environments. In the sound field analysis (SFA) phase, an unbaffled 24-element circular microphone array is utilized to encode the sound field based on the plane-wave decomposition. Depending on the sparsity of the sound sources, the SFA stage can be implemented in two manners. For sparse-source scenarios, a one-stage algorithm based on compressive sensing algorithm is utilized. Alternatively, a two-stage algorithm can be used, where the minimum power distortionless response beamformer is used to localize the sources and Tikhonov regularization algorithm is used to extract the source amplitudes. In the sound field synthesis (SFS), a 32-element rectangular loudspeaker array is employed to decode the target sound field using pressure matching technique. To establish the room response model, as required in the pressure matching step of the SFS phase, an SFA technique for nonsparse-source scenarios is utilized. Choice of regularization parameters is vital to the reproduced sound field. In the SFS phase, three SFS approaches are compared in terms of localization performance and voice reproduction quality. Experimental results obtained in a reverberant room are presented and reveal that an accurate room response model is vital to immersive rendering of the reproduced sound field.

Solution Strategies for Linear Inverse Problems in Spatial Audio Signal Processing

The aim of this study was to compare algorithms for solving inverse problems generally encountered in spatial audio signal processing. Tikhonov regularization is typically utilized to solve overdetermined linear systems in which the regularization parameter is selected by the golden section search (GSS) algorithm. For underdetermined problems with sparse solutions, several iterative compressive sampling (CS) methods are suggested as alternatives to traditional convex optimization (CVX) methods that are computationally expensive. The focal underdetermined system solver (FOCUSS), the steepest descent (SD) method, Newton’s (NT) method, and the conjugate gradient (CG) method were developed to solve CS problems more efficiently in this study. These algorithms were compared in terms of problems, including source localization and separation, noise source identification, and analysis and synthesis of sound fields, by using a uniform linear array (ULA), a uniform circular array (UCA), and a random array. The derived results are discussed herein and guidelines for the application of these algorithms are summarized.

Numerical analysis of two-dimensional sound fields with a spectral method in generalized curvilinear coordinates

Numerical analysis of two-dimensional sound fields with a spectral method in generalized curvilinear coordinates

Numerical analysis of sound field by BEM and application to design for noise reduction

Numerical analysis of sound field by BEM and application to design for noise reduction

Sound field analysis assuming a cuboid sound field and the same acoustic absorption medium.

Sound field analysis assuming a cuboid sound field and the same acoustic absorption medium.

Analysis of Onstage Acoustics Preference of Musicians of Traditional Performance of Javanese Gamelan Based on Normalized Autocorrelation Function

On-stage sound field analysis of a traditional performance of the Javanese gamelan at Pendopo ISI Surakarta, Indonesia was conducted by analyzing the effective decay time of a normalized autocorrelation function called tau-e, τ e , during a performance of the Gambyong Pare Anom dance. The parameter tau-e is used to describe the richness of the frequency content, tempo, and types of gamelan instruments being played at a certain time and position on stage. The tau-e parameter is important for musicians in order to maintain communication between each other such that they can keep the performance in harmony. In order to determine the acoustic parameters heard by gamelan musicians on stage, sound measurements were conducted at 4 points on stage during a performance. Each position represents a specific group of gamelan instruments, which have different characteristics of loudness and frequency, different functions and different ways the instruments are played. The analysis showed that each of the four positions had a different value of τ e , which fluctuated throughout the performance. Overall, the dominant τ e at position 1 was 20 ms; at position 2 it was 50 ms; at position 3 it was 20 ms; and at position 4 it was 40 ms. The distribution of τ e on the stage shows that positions 1 and 3 had more frequency richness compared to positions 2 and 4.

Analysis and synthesis of spatial sound fields with room response modeling using an integrated microphone and loudspeaker array system

A unified framework is proposed for analysis and synthesis of spatial sound fields. In the sound field analysis (SFA) phase, an unbaffled 24-element circular microphone array (CMA) is utilized to “encode” the sound field based on the plane-wave decomposition, whereas in the sound field synthesis (SFS) phase a 32-element rectangular loudspeaker array is employed to “decode” the target sound field using pressure matching technique. Depending on the sparsity of the sound sources, the SFA stage can be implemented in two ways. For the sparse-source scenario, a two-stage algorithm is utilized to estimate the source bearings using the minimum power distortionless response (MPDR) and the associated amplitude coefficients of plane waves using the Tikhonov regularization (TIKR) algorithm. Alternatively, a one-stage algorithm based on compressive sensing (CS) algorithm can be used. For the nonsparse-source scenario, a one-stage TIKR is utilized to solve for the amplitude coefficients for plane-wave components uniformly distributed in the angular domain. The SFA technique for the non-sparse source scenario is also useful in establishing the room response model, as required in the pressure matching step of the SFS phase. The integrated acoustic array system is validated with localization and listening tests.

A synthesized boundary condition for wave-based room acoustics simulations using ensemble averaged impedance measured in-situ

The importance of wave-based room acoustics simulation has been increasing in various fields of researches and design-stages. The problem remained, however, is how to model the rooms’ absorptive boundary conditions. It is known that traditional absorption indices like sound absorption coefficient and normal impedance are not always suit for such a fine method like finite element method. Then, the concept and measurement technique of ensemble averaged impedance were presented. In our previous studies, a series of in-situ measurement of ensemble averaged impedances were successfully performed to result excellent reproducibility and repeatability. Herein, the outline of the concept of ensemble averaged impedance including the measurement technique is summarized, first. Second, several example results of practical measurements are given to exhibit that their uncertainties stay within the range for wave-based simulations to keep their resulting uncertainties less than just noticeable difference. Then, a mathematical-physical model of synthesized boundary condition is given using the ensemble averaged impedance in the sound field analysis by finite element method. Finally, the results of sound field simulations of realistic rooms are shown to examine the plausibility of the model.

Super-resolution in sound field recording and reproduction based on sparse representation

Sound field recording and reproduction enables us to construct more realistic audio systems. In practical systems, sound pressure is obtained with microphones in a recording area, and then the sound field is reproduced with loudspeakers in a target area. Therefore, a signal conversion algorithm for obtaining the driving signals of the loudspeakers from the signals received by the microphones is necessary. Most of the current methods are based on the sound field analysis and synthesis in the spatial frequency domain. Although these methods make it possible for stable and efficient signal conversion, artifacts originating from spatial aliasing notably occur, which depends on the intervals of microphones and loudspeakers. We have proposed a signal conversion method based on sparse sound field decomposition, which enables sound field recording and reproduction above the spatial Nyquist frequency. By using sparse decomposition algorithms, the sound pressure distribution can be represented using a small number of fundamental solutions of the Helmholtz equation, such as Green's functions. Group sparse signal models are also required for accurate and robust decomposition. In this presentation, we reports on comparisons between several group sparse signal models and decomposition algorithms as well as their relation to reproduction performances.

Finite element sound field analysis for investigation on measurement mechanism of reverberation time in non-diffuse sound field

As well known, the reverberation time is essential to evaluate room acoustics; however, it is not so easy to predict in a practical room where the assumption of diffuse sound field is not satisfied. On the other hand, the numerical analyses based on the wave equation have been intensively used to explore many kinds of acoustic problems. In this study, 15 kinds of rooms with different absorption surfaces are analyzed by time domain finite element method. This study shows effectiveness of the analysis for investigation on measurement mechanism of the reverberation time of sound field in rooms where the assumption of diffuse sound field is not satisfied. To evaluate a sound field in a room for the measurement, the ratio of incident sound energy to a part of boundary in those to all boundary of a sound field in a room is calculated from results of the time domain finite element analysis. At first, the calculation method of the ratio of incident sound energy to a part of boundary in those to all boundary of a sound field in a room is described, and the ratios are calculated in the 15 kinds of rooms at each frequency. Finally, relationship among the ratios, mean absorption coefficients, reverberation times calculated from the results of TDFE analysis and those from Sabin’s formula is investigated.

Time-space analysis, synthesis, and room response modeling of virtual-reality sound fields using microphone and loudspeaker arrays

This paper proposes a unified framework to analyze and synthesize spatial sound fields. In the analysis phase, an unbaffled 24-element circular microphone array (CMA) is utilized to “encode” the sound field based on plane-wave decomposition, whereas in the synthesis phase, a 32-element rectangular loudspeaker array is employed to “decode” the above-encoded sound field based on pressure matching. Plane-wave decomposition can be implemented in two ways. First, a two-stage algorithm uses the minimum variance distortionless response (MVDR) beamformer to estimate the directions of significant plane-wave components. Next, the amplitude coefficients of plane waves are calculated using the Tikhonov regularization (TIKR) algorithm. Alternatively, a one-stage algorithm that treats the sound field as a great number of plane-wave components uniformly distributed at all angles can be used. In this case, either TIKR or convex (CVX) optimization can be utilized to solve the underdetermined problem for a minimum-norm solution or a minimum-cardinality solution. The same technique is also useful in establishing the response model for a reverberation room with all direct sounds and reflections coded into plane-wave components. In the synthesis phase, pressure matching is achieved by the inverse of the preceding room response model for rendering loudspeakers in the reproduction room.

Compressive sensing with a spherical microphone array

A wave expansion method is proposed in this work, based on measurements with a spherical microphone array, and formulated in the framework provided by Compressive Sensing. The method promotes sparse solutions via ℓ1-norm minimization, so that the measured data are represented by few basis functions. This results in fine spatial resolution and accuracy. This publication covers the theoretical background of the method, including experimental results that illustrate some of the fundamental differences with the "conventional" least-squares approach. The proposed methodology is relevant for source localization, sound field reconstruction, and sound field analysis.

Beamforming with Optimal Aliasing Cancellation in Spherical Microphone Arrays

Spherical microphone arrays facilitate three-dimensional processing and analysis of sound fields in applications such as music recording, beamforming and room acoustics. The frequency bandwidth of operation is constrained by the array configuration. At high frequencies, spatial aliasing leads to side-lobes in the array beam pattern, which limits array performance. Previous studies proposed increasing the number of microphones or changing other characteristics of the array configuration to reduce the effect of aliasing. In this paper we present a method to design beamformers that overcome the effect of spatial aliasing by suppressing the undesired side-lobes through signal processing without physically modifying the configuration of the array. This is achieved by modeling the expected aliasing pattern in a maximum-directivity beamformer design, leading to a higher directivity index at frequencies previously considered to be out of the operating bandwidth, thereby extending the microphone array frequency range of operation. Aliasing cancellation is then extended to other beamformers. A simulation example with a 32-element spherical microphone array illustrates the performance of the proposed method. An experimental example validates the theoretical results in practice.

605 直方体吸音音場のモード解析と統計的解析手法の比較研究(GS10 音場評価・防音,研究討論セッション)

605 直方体吸音音場のモード解析と統計的解析手法の比較研究(GS10 音場評価・防音,研究討論セッション)

考虑磁致收缩的750 kV电抗器的噪声计算

电抗器噪声作为变电站噪声的主要来源之一，对于变电站外部声环境有重大影响。研究电抗器噪声对于电抗器的降噪分析具有重要意义。本文通过对750 kV甘肃永登变电站并联电抗器噪声声场分布的现场测量，得到了电抗器周围声场的干涉与衰减特性。并基于有限元法建立了并联电抗器的电磁、结构、声场仿真计算模型，通过时域电磁场分析得到了电抗器铁心与线圈中的磁场分布，将磁场所引起的铁心饼和线圈中的电磁力以及铁心中产生的磁致伸缩效应作为激励施加给铁心和线圈，通过振动分析得到铁心和线圈表面各节点的振动加速度，并将其作为声场分析的边界条件，最终得到电抗器噪声声场分布情况。将仿真结果与实测值进行对比分析，最大误差不超过15%，验证了有限元模型的可行性，为电抗器噪声预测及降噪措施提供了理论依据及控制方法。 Reactor noise, as one of the main sources of substation noise, has a significant impact on the ex-ternal acoustic environment of the substation. It is very important to study reactor noise for the noise reduction analysis of the reactor. In this paper, the interference and attenuation character-istics of the acoustic field around the reactor are obtained by field measurement of the noise field distribution of the shunt reactors in the 750 kV Gansu Yongdeng substation. The simulation calculation model of electromagnetism, structure and acoustic field of the reactor is established based on the finite element method. The magnetic field distribution of the core and the coil of the reactor is obtained by the time-domain electromagnetic field analysis. The electromagnetic force between the core cakes and coil and the magnetostrictive effect in the iron core caused by the magnetic field are applied as excitation to the core and the coil. The vibration acceleration of each node on the core and coil surface is obtained by vibration analysis, which is used as the boundary condition of the sound field analysis, and finally the noise field distribution of the reactor is obtained. The simulation results are compared with the measured values, and the maximum error is less than 15%. The feasibility of the finite element model is verified, which provides the theoretical basis and control method for the reactor noise prediction and noise reduction.

Theory and design of compact hybrid microphone arrays on two-dimensional planes for three-dimensional soundfield analysis.

Soundfield analysis based on spherical harmonic decomposition has been widely used in various applications; however, a drawback is the three-dimensional geometry of the microphone arrays. In this paper, a method to design two-dimensional planar microphone arrays that are capable of capturing three-dimensional (3D) spatial soundfields is proposed. Through the utilization of both omni-directional and first order microphones, the proposed microphone array is capable of measuring soundfield components that are undetectable to conventional planar omni-directional microphone arrays, thus providing the same functionality as 3D arrays designed for the same purpose. Simulations show that the accuracy of the planar microphone array is comparable to traditional spherical microphone arrays. Due to its compact shape, the proposed microphone array greatly increases the feasibility of 3D soundfield analysis techniques in real-world applications.

Acoustic Source Localization and Deconvolution-Based Separation

This paper examines two fundamental issues in sound field analysis: acoustic sources localization and separation. Algorithms are developed to locate and separate acoustic signals on the basis of plane-wave decomposition. In the localization stage, directions of plane waves are determined using either minimum variance distortionless response (MVDR) method or multiple signal classification (MUSIC) method. For broadband scenarios, coherent and incoherent techniques are utilized in the localization procedure. In the separation stage, two approaches with overdetermined and underdetermined settings can be employed. In the overdetermined approach, Tikhonov regularization (TIKR) is utilized to recover the source signals. In the underdetermined approach, the steering matrix is augmented by including the directions that have been determined in the localization stage. Hence, the separation problem is formulated into a compressive sensing (CS) problem which can be effectively solved by using convex (CVX) optimization. Simulation and experiments are conducted for a 24-element circular array. Objective tests using perceptual evaluation of speech quality (PESQ) tests and subjective listening tests demonstrate that the proposed methods yield speech signals with well separated and improved quality, as compared to the mixed signals.