Acoustic Sound Field Research Articles

Efficient calculation of the three-dimensional sound pressure field around a slab track

The wavenumber domain Boundary Element (2.5D BE) method is well suited to calculate the acoustic sound field around structures with a constant cross-section along one dimension, such as noise barriers or railway track. By expressing the sound field along this dimension in wavenumber domain, the numerical model is reduced from a 3D model to 2D model at each wavenumber. A consequence of the required discrete Fourier domain representation is that the sound field is represented by periodically repeating sections, of which only one section is physically meaningful. The resolution and the number of required wavenumbers increases with the desired length and spatial discretisation of this section. Describing the sound field adequately to auralise the sound without disturbing artefacts requires a large number of wavenumbers (and thus 2D BE computations), which is not feasible for large geometries. Here, a method is introduced that allows the calculation of the 3D sound field by solving a single 2D BE problem for a dense frequency spectrum and interpolating at higher wavenumbers. The calculation efficiency is further increased by precalculating the acoustic transfer functions between each BE surface element and receiver positions. Combining these two methods allows for efficient calculation of the 3D sound field around acoustically rigid structures such as slab tracks. The numerical approach is validated by comparison with a standard 2.5D BEM calculation and an analytical solution. Precalculated transfer functions to calculate the sound radiation from railway track, which are made available online, are illustrated. An example application is presented.

Experimental research on global active rotor noise control using near-field acoustic holography and sound field reproduction

This study employs a 1.6 m diameter rotor to validate the global active noise control method, developed from our previously work, which is based on near-field acoustic holography and sound field reproduction. The validation is conducted through experiments in an anechoic chamber. Additionally, we achieved control over nearly all acoustic modal components of multiple blade passing frequency (BPF) noises of the rotor by extending the time-domain least mean square (LMS) to the acoustic modal domain. The experimental results from near-field acoustic holography revealed that each BPF noise contained 3–4 main acoustic modal components, accounting for almost 90 % of the sound power. The global active noise control experiments demonstrated that the adaptive algorithm in the acoustic modal domain effectively mitigates the influence of rotation speed fluctuations, resulting in steady-state global noise reduction through sound field reproduction. Furthermore, for the first 3 BPF noise, our method achieved noise reduction of 9, 11, and 6 dB by controlling the first 3–4 main acoustic modal components, respectively. The concurrent reduction of various harmonic noises at all observation points validates the method's capability to globally control multiple harmonic noises.

Influence of noise masking on leak pinpointing: Experimental analysis on a laboratory test rig for leak noise correlation.

The efficient management of water distribution networks requires the detection and mitigation of leaks that result in significant water losses, economic costs, and potential health hazards. This experimental study focuses on the optimal frequency range for accurately locating a water leak in a distribution network using the noise correlation approach while mitigating the influence of background noise associated with water consumption from users. Through laboratory experimental tests, a digital bandpass filter was defined in order to pre-process signals and to improve cross-correlation analysis within the pertinent frequency range. This was done by characterising the frequency content of leak-induced noise, noise masking resulting from water consumption, and background noise. The analysed signals were recorded using hydrophones installed within the pipes of a laboratory test rig under different conditions of water flow rate. The spectra analysis assessed the influence of varying water flow rates on the acoustic sound field within the pipeline system. Additionally, the results of the analysis made it possible to discriminate clearly between noise generated by leaks and noise attributed to water consumption. This distinction is useful for implementing suitable filters, and also to establish a threshold ratio of leak flow rate to water consumption flow rate for effective pre-filtering. This study aims to establish a methodological framework that can effectively differentiate between acoustic signals stemming from leaks and noise generated by other sources within the system, to improve leak location accuracy through the noise correlation approach.

Active control of acoustic scattering from a passively optimised spherical shell

At low frequencies, the sound power scattered from a spherical shell can be minimised by designing its material properties and thickness so that the mass and compressibility are the same as that of the displaced fluid. The scattered power is then dominated at higher frequencies by that due to the resonances of the structural modes of the shell, particularly the ovalling mode. The peaks in the scattered power due to structural resonances can be reduced somewhat by material damping but are more effectively attenuated with active control using structural actuators as secondary sources. Simulations are presented of the scattered sound power of such a shell when subject to feedforward control, which assumes knowledge of both the incident and scattered acoustic sound fields, and structural feedback control, which only assumes that the velocity on the surface of the sphere can be measured. The performance of the feedback controller is also examined if the structural actuators and sensors are distributed over the surface of the sphere, rather than just acting at single points.

Toward a Real-World Technical Test Battery for Remote Microphone Systems Used with Hearing Prostheses.

Remote microphones (RMs) enable clearer reception of speech than would be normally achievable when relying on the acoustic sound field at the listener's ear (Hawkins, J Sp Hear Disord 49, 409-418, 1984). They are used in a wide range of environments, with one example being for children in educational settings. The international standards defining the assessment methods of the technical performance of RMs rely on free-field (anechoic) delivery, a rarely met acoustic scenario. Although some work has been offered on more real-world testing (Husstedt et al., Int J Audiol 61, 34-45. 2022), the area remains under-investigated. The electroacoustic performance of five RMs in a low-reverberation room was compared in order to assess just the RM link, rather than measurements at the end of the signal chain, for example, speech intelligibility in human observers. It pilots physical- and electro-acoustic measures to characterize the performance of RMs. The measures are based on those found in the IEC 60118 standards relating to hearing aids, but modified for diffuse-field delivery, as well as adaptive signal processing. Speech intelligibility and quality are assessed by computer models. Noise bands were often processed into irrelevance by adaptive systems that could not be deactivated. Speech-related signals were more successful. The five RMs achieved similar levels of good predicted intelligibility, for each of two background noise levels. The main difference observed was in the transmission delay between microphone and ear. This ranged between 40 and 50 ms in two of the systems, on the upper edge of acceptability necessary for audio-visual synchrony.

A global rotor noise control method based on near-field acoustic holography and sound field reproduction

Aiming at the severe aerodynamic noise generated by the rotor of rotorcrafts, a global active noise control method is proposed based on near-field acoustic holography (NAH) and sound field reproduction (SFR). The microphone array here is not only used to evaluate errors as in existing methods, but also to predict the rotor noise by extracting acoustic modal information. The loudspeaker array reproduces a secondary sound field with the same acoustic modal components but in opposite phase to achieve global noise reduction. Specifically, by arranging microphones on the fuselage, the rotor noise prediction problem is transformed into an external domain problem of the NAH. Then, by extending the SFR to the external domain, the loudspeaker array is used to reproduce a target sound field which just achieves sound-sound cancellation with the rotor noise globally. Further, the application of the acoustic modal analysis shows that the acoustic modal distribution of the rotor noise is determined by the number of blades, which greatly reduces the scale of loudspeaker/microphone array and is conducive to on-line noise control. The acoustic modal decomposition of the rotor noise shows that the amplitude of the main acoustic mode decays the slowest with distance compared with other acoustic modes, and it is the main component of the far-field noise. The comprehensive evaluation of the model rotor noise reduction performance at different scales shows that the global noise reduction effect for controlling all acoustic modes exceeds 40 dB, while for controlling the main acoustic mode reaches 18 dB, and the required power is reduced by 84%. Finally, the global active noise control experiments for the model rotor are conducted. The results show that the global noise reduction can reach 17 dB through controlling the main acoustic mode. The GANC method shows great advantage of global noise reduction and high engineering application potential.

Spatio-temporal Gaussian process regression for room impulse response interpolation with acoustically informed priors

The reconstruction of sound fields in a room from a limited set of measurements is a central problem in acoustics, with relevant applications in e.g. acoustic analysis, audio, or sound field control. Conventional approaches rely on measuring the room impulse response (RIR) at several locations in the room, and fitting a wave model that enables to estimate the field at other locations—via solving an inverse problem. Previous studies have shown that the reconstruction of RIRs strongly depends on the regularization method used to solve the problem, and that enforcing sparsity is beneficial for the reconstruction of the early part of the RIR. This work studies Gaussian processes with time-dependent regularization in order to exploit the temporal properties of RIRs. The inverse problem is solved in the time domain, where hierarchical Bayes is applied in order to explicitly promote solutions where waves are more likely to propagate as time evolves. The model aims at reconstructing both the early part, and the late part of the RIRs. The performance of the proposed model is studied with experimental measurements, and compared to classical reconstruction methods with l1 and l2-norm regularization schemes

Identification of a new acoustic sound field trend in modern catholic churches

This work describes the sound field transformation in catholic churches up today, focusing on a new possible acoustic trend in modern buildings.In the past, the sacred sense inside a church was greatly reinforced by the surrounding atmosphere of a very reverberant space, with possible echoes, that infatuated the perception of God and his presence. On the contrary, nowadays, concentration and intimacy are preferred.Modern Church buildings have become acoustically complex places. They are complex from the geometric point of view, because their dimensions and shapes may be extremely varied, passing from simple auditorium-like churches to complicated ones characterized by curved walls, side chapels, vaults which often act as partially coupled volumes. The range of the possible sound messages is also extremely varied passing from the spoken word to congregational singing and sometimes to organ music during the same liturgical service.In this paper some main recommendations to follow in the acoustic design of new catholic churches are presented, explaining two case studies in detail: the Santa Gianna Beretta Molla Church in Trezzano, Milan (IT) and the Massimiliano Kolbe Church in Bergamo (IT).Room Acoustics procedures and implementations of computer simulation techniques are included. A new approach for the acoustic design is proposed in which the acoustic response is suitable for both speech and music as recent room acoustics’ studies for other typologies of building such as theatres and multipurpose auditoria suggest.

Spatial Wiener filter to reduce spatial aliasing with spherical microphone arrays.

Three dimensional acoustic soundfields can be represented by a set of spherical harmonic coefficients that are extracted from pressure signals recorded by a spherical microphone array. The extraction method used and truncation order chosen introduce errors of spatial aliasing in the coefficients and truncation error in a reconstructed signal. A spatial Wiener filter (SWF) extraction method is proposed in this paper, using second order statistics of typical soundfield characteristics (signal power, estimated source locations, and internal microphone noise) and accounts for the presence of coefficients beyond the truncation order to reduce spatial aliasing in the extracted coefficients. The SWF can also distinguish between "wanted" and "unwanted" sources, reducing the contributions of unwanted sources to the extracted coefficients. The SWF is compared against the state of the art methods; regularized inverse (or generalized inverse), and orthonormal extraction methods, which are explored under a similar framework to the SWF. The authors compare these methods and show the benefit of the SWF for plane waves, with varying assumptions about the source characteristics. The SWF can also extract coefficients beyond the traditional truncation limit of a given array, unlike other methods.

Design issues for eigenbeamforming spherical microphone arrays

Approximating an analytic function by orthonormal bases functions has its origins in the theory of least squares developed by Gauss and Legendre. The use of orthonormal bases functions has been a standard tool in soundfield analysis since Rayleigh. One common set of orthonormal bases functions that naturally fit the solution of the spherical wave equation are the spherical harmonics. In this talk we describe the concept of building a microphone array on a rigid spherical baffle. We process the output signals from the spherical array such that we spatially decompose the acoustic soundfield into spherical harmonic Eigenbeam signals. Computationally efficient Eigenbeamforming, or modal beamforming, with these processed Eigenbeam signals will be discussed. Two main design issues are the growth in sensitivity to self-noise for higher-order spherical modes at lower values of k*r where, k is the wavenumber and r is the radius, and spatial aliasing due to the finite number of microphones covering the spherical surface at higher values of kr. We will discuss how to practically handle these design issues and how these design solutions impact the overall performance of the Eigenbeams and the use of these components in the design of modal beamformers.

A methodology to objectively assess the performance of sound field amplification systems demonstrated using 50 physical simulations of classroom conditions.

Introduction:The effect of a sound reinforcement system, in terms of speech intelligibility, has been systematically determined under realistic conditions. Different combinations of ambient and reverberant conditions representative of a classroom environment have been investigated.Materials and Methods:By comparing the measured speech transmission index metric with and without the system in the same space under different room acoustics conditions, it was possible to determine when the system was most effective. A new simple criterion, equivalent noise reduction (ENR), was introduced to determine the effectiveness of the sound reinforcement system which can be used to predict the speech transmission index based on the ambient sound pressure and reverberation time with and without amplification.Results:This criterion had a correlation, R2 > 0.97. It was found that sound reinforcement provided no benefit if the competing noise level was less than 40 dBA. However, the maximum benefit of such a system was equivalent to a 7.7 dBA noise reduction.Conclusion:Using the ENR model, it would be possible to determine the suitability of implementing sound reinforcement systems in any room, thus providing a tool to determine if natural acoustic treatment or sound field amplification would be of most benefit to the occupants of any particular room.

Novel Bio-Logging Tool for Studying Fine-Scale Behaviors of Marine Turtles in Response to Sound

Increases in the spatial scale and intensity of activities that produce marine anthropogenic sound highlight the importance of understanding the impacts and effects of sound on threatened species such as marine turtles. Marine turtles detect and behaviorally respond to low-frequency sounds, however few studies have directly examined their behavioral responses to specific types or intensities of anthropogenic or natural sounds. Recent advances in the development of bio-logging tools, which combine acoustic and fine-scale movement measurements, have allowed for evaluations of animal responses to sound. Here, we describe these tools and present a case study demonstrating the potential application of a newly developed technology (ROTAG, Loggerhead Instruments, Inc.) to examine behavioral responses of freely swimming marine turtles to sound. The ROTAG incorporates a three-axis accelerometer, gyroscope, and magnetometer to record the turtle’s pitch, roll, and heading; a pressure sensor to record turtle depth; a hydrophone to record the turtle’s received underwater acoustic sound field; a temperature gauge; and two VHF radio telemetry transmitters and antennas for tag and turtle tracking. Tags can be programmed to automatically release via a timed corrodible link several hours or days after deployment. We describe an example of the data collected with these tags and present a case study of a successful ROTAG deployment on a juvenile green turtle (Chelonia mydas) in the Paranagua Estuary Complex, Brazil. The tag was deployed for 221 minutes, during which several vessels passed closely (<2km) by the turtle. The concurrent movement and acoustic data collected by the ROTAG were examined during these times to determine if the turtle responded to these anthropogenic sound sources. While fine-scale behavioral responses were not apparent (second-by-second), the turtle did appear to perform dives during which it remained still on or near the sea floor during several of the vessel passes. This case study provides proof of concept that ROTAGs can successfully be applied to free-ranging marine turtles to examine their behavioral response to sound. Finally, we discuss the broad applications that these tools have to study the fine-scale behaviors of marine turtles and highlight their use to aid in marine turtle conservation and management.

The implantation of Bonebridge in bilateral congenital malformation of external and middle ear

Objective: To evaluate the auditory efficacy of Bonebridge implantation in patients with bilateral congenital malformation of external and middle ear. Methods: Eleven cases (6 males and 5 females) had unilateral Bonebridge implantation. The age ranged from 8 to 26 and the average age was 16.9. Seven to ten days after operation, the first fitting was undergone. In acoustic sound field, the average auditory thresholds were respectively measured for unaided ears and Bonebridge implanted ears by pure tone auditory (PTA, 0.25, 0.5, 1, 2 and 4 kHz). For the group over 12-year-old, MSTM was applied to evaluate speech discrimination score (SDS). For the other cases, MLNT was used as the test material. The auditory efficacy post Bonebridge implantation would be analyzed and evaluated by comparing the differences between unaided ears and Bonebridge implanted ears. Results: The bone conduction audibility threshold after Bonebridge implantation was as well as the preoperative. The auditory threshold with Bonebridge aided was improved to 25-35 dB HL, when compared to that of the unaided ears in the sound field. The SDS in the group over 12-year-old was improved about 50%; the efficacy was slightly limited for the other two cases (both less than 12 years old). Statistical analysis showed that there were significant differences between unaided ears and Bonebridge implanted ears in the sound field and SDS(P<0.05). Conclusions: The auditory efficacy of Bonebridge is significant and noticeable in patients with bilateral congenital malformation of external and middle ear. Bonebridge provides a new and effective way for patient with congenital malformation of external and middle ear to reconstruct hearing.

統計的考えを用いた吸音音場の新たな解析手法に関する研究

統計的考えを用いた吸音音場の新たな解析手法に関する研究

Theoretical and experimental investigation of the pulse-echo nonlinearity acoustic sound fields of focused transducers

The pulse-echo nonlinear acoustic wave fields generated by focused transducers are calculated numerically and verified experimentally. Simulation results show that in the reflection direction, phase differences between newly generated second harmonic and reflected waves are about π/2, regardless of the rigid and stress-free boundaries; thus, the total second harmonic wave will neither be strongly enhanced nor be cancelled. Experiments are conducted with a focused transducer working in pulse-echo mode and the measured amplitudes of nonlinear waves reflected after water-air and water-aluminum/steel interfaces agree with the simulated results. This work helps advance applications using pulse-echo nonlinear acoustics.

Parametric Study of Sound Radiation Properties of Complex Vibration Patterns in Rectangular Plates using an Analytical Model

Complex vibration modes, which can be characterised by the superposition of standing and travelling bending waves, occur for instance in structures with inhomogeneous damping distribution. The existence of complex structural modes raises the question of their influence on sound radiation properties. In this paper an analytical model of a simply supported rectangular plate is introduced. It allows a quick calculation of complex vibration patterns with different properties, avoiding time consuming and sometimes unstable damped modal analysis in finite-element-software. Using this model it is possible to generate vibration patterns with varying level and spatial distribution of travelling waves as well as different symmetry of amplitude distributions, in order to reveal the important influence factors on sound radiation. Despite the fact that the analytical model generates idealised vibration patterns, the presented consideraton significantly improves phenomenological understanding of acoustic effects in complex vibration modes. The accuracy of the presented model is validated by results of finite element modal analysis and is evaluated in terms of resulting amplitude and phase distributions of the vibrating surface. The results of both suggested approaches are compared using modal radiation efficiencies, spatial distributions of far field acoustic intensity and sound field directivity. Some additional conclusions are made using the analytical model concerning important acoustic metrics of structural vibration patterns. It is shown, that vibration shapes of even order are very sensitive to the presence of travelling waves regarding their radiation capability and character of sound pressure field.

A Blind Dereverberation Method for Narrowband Source Localization

Narrowband source localization gets extremely challenging in strong reverberation. When the room is perfectly known, some dictionary-based methods have recently been proposed, allowing source localization with few measurements. In this paper, we first show that, for these methods, the choice of frequencies is important as they fail to localize sources that emit at a frequency near the modal frequencies of the room. A more difficult case, but also important in practice, is when the room geometry and boundary conditions are unknown. In this setup, we introduce a new model for the acoustic soundfield, based on the Vekua theory, that allows a separation of the field into its reverberant and direct source contributions, at the cost of more measurements. This can be used for the design of a dereverberation pre-processing step, suitable for a large variety of standard source localization techniques. We discuss the spatial sampling strategies for the sound field, in order to successfully recover acoustic sources, and the influence of parameters such as number of measurements and model order. This is validated in numerical and experimental tests, that show that this method significantly improves localization in strong reverberant conditions.

A method for predicting the performance of a steelband orchestra in arbitrary environments

One of the challenges in steelpan performance is in determining the best placement of the various instruments in an orchestra so as to achieve the level performance desired by the musical arranger. This research attempts to develop a method for analyzing the performance of the tenor steelpan in arbitrary acoustic spaces using the room acoustic modeling software, Enhanced Acoustic Simulation for Engineers (EASE). When provided with an acoustic characterization of a speaker system, particularly the magnitude and phase radiation data, and the room in which it is located, EASE can be used to predict the resulting acoustic sound-field. EASE also requires an electrical equivalent of the instrument's sound source and, through its auralization feature, allow the user to actually hear an actual performance. EASE uses a variety of reconstruction techniques, such as ray tracing and boundary element method, to provide the most accurate responses. The study will verify the approach by comparing simulation results on direct sound coverage, total SPL, Clarity (C80), Centre Time, Reverberation, and Early decay times of the instrument with real time measurements in the usual 1/3-band octave frequency resolution. Although the work focuses on the tenor steelpan, it can be easily extended to predict the performance of an entire ensemble of steelpans in any given space.

FDTD simulation of the acoustic interaction between a cavity and an adaptive Helmholtz resonator

Helmholtz resonators are used to manipulate the eigen behaviour of the acoustic fluid in a cavity. Using the analogy to a mass damper, it is possible to tune the resonators' eigen frequency by handling the resonator volume as spring and the resonator neck as mass of a vibration absorber. Simulating the transient interaction of the HR with the acoustic sound field in a cavity in terms of active sound control, a high simulation performance is demanded. This can be provided by the finite difference time domain method. The simulations are performed with respect to the LOEWE-Zentrum AdRIA acoustics demonstrator. It depicts the abstraction of a real room. Therefore, it consists of six sound reflecting walls that are implemented through acoustic impedances in the time domain. The semi-passive system is implemented using MATLAB/Simulink. In addition, the simulations are validated applying an adaptive Helmholtz resonator inside the acoustic fluid of the acoustics demonstrator.

Sound Field Classification in Small Microphone Arrays Using Spatial Coherences

The quality and performance of many multi-channel signal processing strategies in microphone arrays as well as mobile devices for the enhancement of speech intelligibility and audio quality depends to a large extent on the acoustic sound field that they are exposed to. As long as the assumption on the sound field is not met, the performance decreases significantly and may even yield worse results for the user than an unprocessed signal. Current hearing aids provide the user for instance with different programs to adapt the signal processing to the acoustic situation. Signal classification describes the signal content and not the type of sound field. Therefore, a further classification of the sound field, in addition to the signal classification, would increase the possibilities for an optimal adaption of the automatic program selection and the signal processing methods in mobile devices. To this end a sound field classification method is proposed that is based on the complex coherences between the input signals of distributed acoustic sensors. In addition to the general approach an exemplary setup of a hearing aid equipped with two microphone sensors is discussed. As only coherences are used, the method classifies the sound field regardless of the signal carried by it. This approach complements and extends the current signal classification approach used in common mobile devices. The method was successfully verified with simulated audio input signals and with real life examples.