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Ultrasonic phased array technology has been widely used in the field of industrial nondestructive examination (NDE) but how to obtain the ideal results is still lacking. Based on Huygens’ principle, a one-dimensional ultrasonic phased array model was established to study the resultant sound field. Important parameters to the acoustic field characteristics of ultrasonic phased array were systematically investigated, such as the number of array elements, the working frequency, the focal length and the steering angle. Based on the simulation results, the optimal parameters can be obtained which can provide a theoretical basis for the actual application of ultrasonic phased array.

Recent advancements of spatial audio technologies to enhance human’s emotional and immersive experiences are gathering attention. Many studies are clarifying the neural mechanisms of acoustic spatial perception; however, they are limited to the evaluation of mechanisms using basic sound stimuli. Therefore, it remains challenging to evaluate the experience of actual music contents and to verify the effects of higher-order neurophysiological responses including a sense of immersive and realistic experience. To investigate the effects of spatial audio experience, we verified the psychophysiological responses of immersive spatial audio experience using sound field synthesis (SFS) technology. Specifically, we evaluated alpha power as the central nervous system activity, heart rate/heart rate variability and skin conductance as the autonomic nervous system activity during an acoustic experience of an actual music content by comparing stereo and SFS conditions. As a result, statistically significant differences (p < 0.05) were detected in the changes in alpha wave power, high frequency wave power of heart rate variability (HF), and skin conductance level (SCL) among the conditions. The results of the SFS condition showed enhanced the changes in alpha power in the frontal and parietal regions, suggesting enhancement of emotional experience. The results of the SFS condition also suggested that close objects are grouped and perceived on the basis of the spatial proximity of sounds in the presence of multiple sound sources. It is demonstrating that the potential use of SFS technology can enhance emotional and immersive experiences by spatial acoustic expression.

Previous studies about micro non-uniform linear arrays show that the simple non-uniform element distribution has some potential in minimizing sidelobe and improving imaging quality. However, the numerical relationship between the specific element distribution and sidelobe has not been explored. So we propose a quantitative method as theoretical guidance for non-uniform linear array optimization. The sound field pattern comparison between uniform and nonuniform linear array is discussed. It indicates that the first sidelobe of the non-uniform linear array can be down about −15.44 dB from the main lobe, which is lower than the theoretical value of the uniform linear array (−13.5 dB).

With the frequent occurrence of large-scale indoor stampedes, people pay more and more attention to the problem of indoor crowd density, and indoor crowd density estimation has become a very important security work. This paper proposes an indoor crowd density estimation method based on acoustic means. This method makes use of the reverberation effect and spatial distribution of sound wave propagation in indoor environment, and realizes the estimation of crowd density by analyzing the disturbance of indoor crowd to sound field. It can be applied to intelligent office, indoor security and other scenes where there is a high demand for crowd density estimation and there are many obstacles, that traditional computer vision methods cannot be applied.

The simulation of external sound field is one of the key problems in the radiated noise calculation of underwater structure. The influence of external acoustic environmental factors, such as limited water depth and free surface, is also lack of enough research at present. Firstly, the calculation method of underwater radiated noise is verified by a stiffened cylindrical shell, and then the area size, shape and element size of the external sound field is discussed. Finally, the effect of finite water depth and free surface on the radiated sound field is analyzed. The results can provide a reference for the calculation of radiated noise of underwater structure.

Internal solitary wave is a universal phenomenon in ocean which can bring enormously significant impact to sound propagation. To our knowledge, most previous acoustic-based methods on the problems of the effect of internal solitary waves on acoustic propagation were done in the frequency domain. It is hard to make inconsistent spatial variation of sound speed with these methods. Therefore, a time domain model based on the lattice Boltzmann method is presented to simulate the sound propagation with and without internal solitary waves. The mechanisms of how the internal solitary waves influence the acoustic propagation involved is investigated. Through compare with and without solitary wave, the main result is this wave will changed the distribution of sound field and interfere with the structure of it.

In this paper, a long-range ocean experiment was carried out by underwater glider, and the sound velocity profile of the sea area was obtained by Kriging method. Using the sound speed profile, the propagation characteristics of long-range sound field were simulated and compared with experiments. On this basis, the characteristics of horizontal and vertical direction of sound field and its application direction are analyzed.

A single focus is formed to maintain a high detection resolution through controlling the element’s excitation time in ultrasonic phased array testing. However, the testing resolution and sensitivity outside of focus area may be decreased. Therefore, it is necessary to develop new focusing strategies for multiple focusing in a specific application. Firstly, three kinds of new focusing strategies were proposed for controlling focusing sound field. Secondly, the Finite Element Method (FEM) was adopted to analyze the characteristics of sound field. Finally, the testing capabilities of focusing strategies for the near-surface and the far-surface testing were investigated by experiments. The results indicate that the experimental results are consistency with finite element simulation, which can be used to assist in the design of the focusing strategies. The new focusing strategies make full use of the array resources can flexibly control the focused sound field to satisfy some special testing requirements.

The acquisition of the spatio-temporal characteristics of a sound field over a large volume of space is experimentally challenging, as a large number of transducers is required to sample the sound field. Sound field reconstruction methods are a resourceful approach, as they enable the interpolation and extrapolation of the sound field from a limited number of observed data. In this study we examine the spatio-temporal and spatio-spectral reconstruction of the sound field in a room from distributed measurements of the sound pressure. Specifically, a variational Gaussian process regression model is formulated, using time-domain anisotropic kernels to reconstruct the direct sound and early reflections, and frequency-domain isotropic kernels for reconstructing the late reverberant field. The proposed methodology is compared experimentally to classical regression models based on plane wave decompositions, which are widely used in sound field reconstruction in enclosures due to their simplicity and accuracy.

Aiming at the additive manufacturing part by Electron Beam Rapid Manufacturing of TC4 (Ti-6Al-4V) titanium alloy, the research of total focusing detecting and imaging of internal small defects was carried out. Firstly, the CIVA software was used to simulate the sound field distribution with different transducer parameters, and the effect of the element center distance on the sound field was analyzed. Then, the total focusing method was used to detect the Φ0.4mm flat-bottom holes with buried depths between 3mm and 50mm. A high signal-to-noise ratio detecting method for this kind of structure was analyzed and summarized. The results show that, in a certain range, with the increase of the center distance of the array, the width and depth of the focal zone gradually decrease, and the resolution gradually increases. But at the same time, the energy of the sound field gradually disperses and the grating lobe phenomenon occurs; Olympus 10L64 linear array probe can be used to effectively detect the defects with a buried depth of 20mm-50mm, but the signal-to-noise ratio is relatively poor for the two cases with a buried depth of 3mm and 10mm, respectively. To overcome the disadvantage, a new transducer with smaller center distance is proved to be effective.

Due to the energy conservation law, traditional active noise control system cannot realize the suppression of omnidirectional sound field and will bring fluctuation effect in the far field. This paper presents a method of sound field control using beam deflection to overcome the shortcomings of traditional active noise control system. The principle of beam deflection is firstly derived based on the principle of beam forming. Two methods including conventional beam deflection and optimal beam deflection are proposed. Then technical route of the sound field control using beam deflection is given. Finally, simulations are performed and the results show that proposed method can effectively suppress the energy of a controlled sound source in the predetermined direction.

This paper introduces one method for room sound field characterization and clustering and it presents part of the research carried out for the purpose of the doctoral dissertation. Room impulse response is the basic source of information in room acoustics. The algorithm provides impulse response characterization and clustering using global multifractal parameters. Large Deviation method is used in calculating the multifractal spectrum. The correlation between different multifractal parameters and room acoustic characteristics is determined. Clusters of impulse responses with identical acoustic characteristics are formed. The test results verify the usability of the proposed algorithm.

Aiming at the influence of anti-acoustic baffle on the acoustic characteristics of hydrophone, the sound field model of hydrophone under the influence of anti-acoustic baffle is established by using finite element software, and the sensitivity variation law is analyzed. The simulation results show that the anti-acoustic baffle and the hydrophone can be placed at 45°, and the former can increase the sensitivity of the latter by 6dB in a certain frequency range. This frequency range varies with the change in the anti-sound coefficient of the baffle and the distance between the hydrophone and the baffle. The research results have guiding significance for designing high-sensitivity hydrophones in the future.

We propose an analytical method of 2.5-dimensional exterior sound field reproduction by using a multipole loudspeaker array. The method reproduces the sound field modeled by expansion coefficients of spherical harmonics based on multipole superposition. We also present an analytical method for converting the expansion coefficients of spherical harmonics to weighting coefficients for multipole superposition. In contrast to pressure-matching methods, which tend to be ill-conditioned problems, the proposed method gives stable solutions based on an analytical conversion from the expansion coefficients of spherical harmonics. We derive the analytical method by converting the sound field modeled by expansion coefficients of spherical harmonics to a linear combination of ’basic multipoles’ for obtaining the weighting coefficients of each multipole. Computer simulation results indicate that the proposed method reproduces the sound field more accurately than existing pressure-matching based method.

The process of capturing, analyzing and predicting sound fields is finding novel areas of applications in AR/VR. One of the key processes in such applications is to estimate the sound field at locations that differ from the actual measurement points—i.e., the sound field reconstruction. However, it's a difficult spatial audio processing problem. Though theoretical solutions exist to reconstruct sound fields, they are practically infeasible due to hardware and computational requirements. This paper discusses the implementation of a system for large area sound field recording and reconstruction and proposes an improved sound field reconstruction algorithm. The proposed algorithm introduces a practical improvement in order to overcome implementation issues. In addition, we present a preliminary real-world results on an innovative but highly challenging application.

Many physical sound field synthesis techniques have been studied. To implement these techniques, a large-scale reproduction system is usually required with a large number of loudspeakers, D/A converters, and amplifiers. Thus, the techniques are not suitable for use in a rooms of an ordinary house. In our recent studies, a physical sound field reproduction system was developed with a digital loudspeaker array and suitable for an ordinary, small room. To more accurately reproduce the sound field with the implemented loudspeakers, an effective method is local sound field synthesis (LSFS). Because the area of the reproduced sound field is limited in LSFS, the reproduction area must track the listener's head movement. To track the listener's head movement, the driving functions of loudspeakers must be changed in real time. In this study, the effect of switching the reproduction area is investigated to determine the appropriate switching rate and size of the local area.

Virtual-reality reproduction of real-world acoustic environments often fix the listener position to that of the microphone. In this paper, we propose a method for listener translation in a virtual reproduction that incorporates a mix of near-field and far-field sources. Compared to conventional plane-wave techniques, the mixed-source method offers stronger near-field reproduction and translation capabilities in the case of a sparse virtualization.

The bunching sound field model of a rotating ellipsoidal reflector in different structure of discharge electrode based on ANSYS/LS-DYNA is established in this paper. The bunching process of shock wave is simulated by LS-DYNA's multi-material ALE method, and the bunching characteristics of shock wave is analyzed and validated through relevant experiments. The results show that the actual focus location is deviated from its geometrical focus location due to the nonlinearity of shock wave propagation and bunching process, and the shielding effect of the vertical discharge electrode to shock wave is stronger than the transverse discharge electrode, which cause less bunching gain.

The authors of the article reviewed the methods of researching the formation of the sound field of a room using examples from audience halls for various purposes. The task of forming the optimal acoustic conditions in the rooms of the audience halls was formulated, objective and, related to them, subjective indicators of the room acoustic quality were described, the characteristics of the indicators were indicated. The results of the implementation of all stages of acoustic expertise obtained by the authors allow to conduct a comprehensive assessment of the sound field generated in the premises, taking into account the peculiarities of musical and speech materials in the audience halls.

The waveguide invariants are stable in shallow water when receiver is far away from source, and they are related to the ocean environment parameters. In this paper, through theoretical research and simulation analysis, it is found that the change of sound source depth will affect the simulating results of waveguide invariants, when using single-element receiving hydrophone and broadband frequencies sound source, and the distance of source and receiver is close. At the same time, it is found that at a long distance, the waveguide invariants are correlated with the sound source depth under the condition of shallow sea negative thermocline waveguide.

The perceived quality of a sound played in a closed room is often degraded by the added reverberation. In order to combat these effects, the methods of room impulse response equalization may be used. Typically, properties of the human auditory system, such as temporal masking, are used for a better control of the late echoes. There, a prefilter is used to modify the played signal, which renders the echoes inaudible for a given position. When targeting a human listener who is always performing small movement, the method fails due to the spatial mismatch. In such a case, a whole volume needs to be equalized, which requires a huge amount of measurements. In this work, we propose to reduce this burden by measuring only a random subset of the required room impulse responses and reconstruct a grid employing sparse methods. By further interpolation an equalization of the whole target area is achieved.

A method for feedforward active noise control (ANC) over a spatial region is proposed. Conventional multipoint ANC aims to reduce the noise at multiple discrete positions; therefore, the noise reduction in the region between these points cannot be guaranteed. Recent studies revealed the possibility of spatial ANC, i.e., noise control in a continuous target region. These methods are essentially based on spherical/circular harmonic decomposition of the sound field by using spherical/circular arrays and have mainly been investigated for feedback control under the assumption of periodicity of the noise. We apply a sound field interpolation method based on kernel ridge regression to feedforward spatial ANC to control spatial nonstationary noise using distributed arrays. Numerical simulation results indicated that a large regional noise reduction is achieved by the proposed method compared with feedforward multipoint ANC.

This paper describes how to encode into a spherical-harmonic representation of a 3-D sound field using a cylindrical microphone array. The standard way to represent a sound field is to use higher order Ambisonics specified over a spherical region. If the sound field is instead measured using a cylindrical array, it is more naturally expressed in terms of cylindrical harmonics. In this paper, the transforms for conversion between cylindrical and spherical-harmonic representations are derived. Simulations then compare the performance of cylindrical and spherical array processing in estimating the spherical-harmonic coefficients of a sound field.

Array processing techniques are widely used in audio applications, especially in audio surveillance systems, where one of the tasks is the scan of the surroundings and the detection of acoustic sources. The array sensors sample the sound field in space, the same way an analog-to-digital converter samples sound in time. Correspondingly, the Nyquist criterion should be applied in the space domain in order to avoid aliasing artifacts. However, the audio spectrum is wide and reducing the sensor dimensions providing the correct reconstruction for all higher audio frequencies can be problematic. In this paper, we propose a comparative study on the usage of phased array solutions and the timed array counterpart.

Passive target classification is a crucial issue in underwater acoustic signal process domain. This paper starts from the difference of sound field caused by surface target and underwater target, a method of passive target classification based on the sound field coherence characteristic is proposed. According to the theory of normal mode, the mechanism of the interference pattern of sound field is expatiated, and the acoustic field coherence characteristics which can distinguish the surface target and underwater target are extracted, the numerical simulation of the proposed method is carried out, the result shows that, by filtering the normal mode and extracting the sound field coherence characteristic to discriminate the surface target and underwater target is valid.

In general, airborne sound power is considered to be independent of direction, distance and environment. Nevertheless, for single tone or low frequency sound sources, the sound power emitted into rooms depends also on the environment and on the exact position in the room. This effect is investigated using Monte-Carlo-Simulations. It is then proposed to describe a source by the sound power that would be emitted into a free sound field. The consequences of this approach for measuring sound insulation are discussed.

The paper describes a non-destructive diagnostic method primarily intended for large rotating machines and transformers. The method is based on the assumption that the emitted noise is characteristic of the mechanical condition of the machine concerned. On the theoretical side, subject to analysis have been the electric machine noise sources and their mathematical description. In practical situations, the noise signals are picked from sound field of the machine. A critical review of the strong points and weak spots of noise diagnostics is presented, as well as the prospects for practical utilization of the method based on noise measurement and analysis.

Kernel ridge regression with the constraint of the Helmholtz equation for three-dimensional sound field interpolation is proposed. Interpolation of a sound field is formulated as an optimization problem whose objective function to be minimized is a weighted sum of the squared residuals and the squared norm of the interpolating sound field. We show that this problem corresponds to kernel ridge regression and that the reproducing kernel for this problem is given by the zeroth-order spherical Bessel function of the first kind. In this kernel, the physical constraint on the sound fields, i.e., the Helmholtz equation, is taken into consideration, which is a major advantage over other commonly used kernels such as the Gaussian kernel. This advantage is also confirmed by numerical simulations.

The sound field interference structure is analyzed in the reliable acoustic path (RAP) duct based on ray theory. We give the interference striation, which is defined as the line predicted by the approximate formulae and Bellhop model used in this paper (see e.g., Fig. 3 for more information). We observe that interference striation is resulted from time delay difference between the direct (D) path and the surface-reflected (SR) path of the broadband noise radiated from the underwater moving source. And interference striations turn to be tight with the increase of the source depth. Based on these observations, we propose a source depth estimation method by using a single hydrophone in the RAP duct. We set up a cost function which measures the matching degree between the interference striations (predicted by the approximate formulae and Bellhop model) and the sound field range-frequency interference structure (extracted from the sampled data of a single hydrophone). And we take the peak location of the outputted matching degrees as the source depth estimate. Numerical simulations show the effectiveness of the proposed source depth estimation method.

Spatial audio may be performed in rooms using active-compensated sound field reproduction (AC-SFR), to cancel the reverberant reflections. Reproduction of a sound field may be performed using precalculated loudspeaker filters. However the robustness can be poor due to the temperature-induced perturbation of the room impulse responses (RIR). To compensate for reverberation, filters with significant non-causal components are designed. It is known that the requirements on the loudspeaker filters for room equalization may be reduced by applying the method of impulse response shaping (IRS). In this paper, we show that actively compensating paths with a long propagation distance are more heavily influenced by temperature than short paths. We hence impose a shaping penalty on the non-causal components of designed loudspeaker filters, to reduce the error caused by a change in temperature. We obtain AC-SFR designs that are more robust to changes in temperature.

Live performances are increasingly appreciated using popular commercial audiovisual transmission systems, such as live viewing systems. A reality enhancement method for live viewing had been proposed. This paper describes an applause sound extraction method for automation of applause sound transmission and a simulation experiment using the sound source recorded live at the venue to assess the applause sound extraction performance. We used an adaptive filter to emphasize applause sounds and extracted them using signal amplitude information. The experimental evaluation showed that the applause sound was extracted almost correctly from the performance sound source under various conditions.

The room impulse responses at multiple receiver positions can be measured efficiently with a continuously moving microphone. The acoustic system is periodically excited by a self-orthogonal signal, called perfect sequence, and the microphone captures the sound field on a pre-defined path. As shown in recent studies by the authors, the captured signal constitutes a spatio-temporal sampling of the sound field, and the impulse responses can be obtained by a spatial interpolation. So far, a uniformly moving microphone was mainly considered for the measurement of spatial room impulse responses. In this paper, the method is applied to non-uniformly moving microphones thereby addressing more general cases. The proposed method is evaluated by numerical simulations where the spatial room impulse responses on a circle are measured using a microphone with a fluctuating angular speed. The accuracy of the impulse responses are compared for varying interpolation orders.

The effect of traditional passive sound absorption material on low-frequency sound waves is limited. So, the active control method is put forward to eliminate the echo of low-frequency. But in the sound field, the incident wave and the reflected wave are superimposing together, we need to isolate the incident wave and the reflected wave in order to control the reflected wave. In this paper, we present two methods of isolating the incident wave and the reflected wave in the sound fields, the two methods are double-pressure sensor method and pressure-velocity sensor method. And the two kinds of acoustic separation methods are used to simulate and analyze the active control of reflected wave.

We describe a family of ambisonics rendering methods that is based on optimizing the soundfield component that lies in the null space of the operator that maps the loudspeaker signals onto the given ambisonics representation. In contrast to traditional rendering approaches, the new method avoids the coherent addition of loudspeaker contributions to the sound field. As a result, it provides a space-invariant timbre and good spatial directionality outside the spatial region where the ambisonics soundfield description is accurate. The new method is idempotent at all frequencies and has relatively low computational complexity. Our experimental results confirm the effectiveness of the method.

Recently, studies on reproduction of realistic sound field for communication with presence have been under-taken. In previous study, sound field reproduction using principle of boundary surface control did not consider acoustic echo. Because of acoustic echo, precision of soundfield reproduction is dropped. In this study, Modeling of acoustic tube and reflected wave rejection using the model are proposed because acoustic echo is caused by reflected wave. Validity of model was confirmed by servo analyzer and reflected wave rejection was verified by simulation.

We propose a method of estimating a sound field with two spherical microphone arrays (SMAs). This method estimates plane-wave expansion coefficients of the sound field by using sparse representation modeling in the frequency domain. The dictionary matrix we propose for this modeling achieves the integration of the measurements of two SMAs in a straightforward manner. The effectiveness of the proposed method was evaluated in simulations with computer-generated and measured impulse responses.

Accurate interpolation of the array manifold is an important first step for the acoustic simulation of rapidly moving microphone ar- rays. Spherical harmonic domain interpolation has been proposed and well studied in the context of head-related transfer functions but has focussed on perceptual, rather than numerical, accuracy. In this paper we analyze the effect of measurement noise on spatial aliasing. Based on this analysis we propose a method for selecting the trunca- tion orders for the forward and reverse spherical Fourier transforms given only the noisy samples in such a way that the interpolation error is minimized. The proposed method achieves up to 1.7 dB im- provement over the baseline approach.

A multi-resolution technique is presented for sound field recovery based on measurements of one or multiple moving microphones. The interpolation of the spatial samples enables us to set up a system of linear equations that recovers room impulse responses on a virtual uniform grid in space. The spacing of the virtual grid must be very small when directly recovering for the entire bandwidth, thus, the system to be solved requires a large number of measurement positions. In this work, we propose two recovery schemes based on multiple virtual grids that represent distinct sub-bands of the spatiotemporal sound field. This allows for faster reconstruction of low frequencies at minor computational cost and improves the recovery quality when measurement noise is present.

Currently the controlled release of drugs in specific areas is of great importance for the treatment of diseases such as cancer, Parkinson, Alzheimer and others. This project works with focused ultrasound (HIFU) in rats, more specifically in the application of HIFU in a specific area of the brain called the substantia nigra, in order to generate opening in the membrane of this area called blood-brain barrier (BBB) with the purpose of introducing vectors containing the drug used to treat Parkinson, proposed by the Department of Physiology, Biophysics and Neuroscience Research Centro de Investigacion y de Estudios Avanzados del Instituto Politecnico Nacional. This work is the beginning of the investigation based on the modeling of a HIFU transducer in bone tissue of rats weighing approximately 400 gr. In order to obtain the sound field pattern, focus size, pressure and ultrasound path in a time dependent study. All of the above in order to get a clear idea of what you will get when it comes to experimentation.

There are several sources of noise and vibration in a contemporary ship, and propeller noise is the main one of them. The sound field radiated by rotating systems such as propellers have intricate structure, displaying a wide range of complex behaviour. This complexity makes the study of such systems both scientifically challenging and technologically important.

The hydrophone calibration system in a small water tank has broad application prospects. This paper studies some theory evidence of the practicability of the hydrophone calibration system in a small water tank. Simulation of the sound field in a small water tank shows the distribution of the acoustic field there in. An experiment is conducted to compare a vector hydrophone with a standard hydrophone, and, calibrate the sensitivity of the vector hydrophone, and thence calibrate the directivity pattern. The frequency is ranged from 50Hz to 1000Hz. Finally, the paper compared the result of calibration with the theoretical sensitivity and directivity pattern.

This paper presents a sound-field reproduction system able to equalize the acoustic channel impulse response and to cancel the crosstalk components between two loudspeakers and two microphones. The system is implemented over a network of two commodity mobile devices, and makes use of two loudspeakers connected to the devices via Bluetooth. The application connects each loudspeaker to one device, establishes a Wi-Fi connection between the two devices, and performs channel estimation, inverse filter calculation, delay compensation and sound reproduction in a seamless way, obtaining the desired sound signals at the devices' microphones. Considering the system as a two-node wireless acoustic network, distributed processing is proposed in order to balance the computational cost and the communication burden between the devices. A real experiment has been carried out using two tablets Samsung Galaxy TAB 3 and two Bluetooth Yamaha NXP100 loudspeakers obtaining over -10dB of average crosstalk attenuation.

In this work, a Head-Related Transfer Function (HRTF)-based Robust Least-Squares Frequency-Invariant (RLSFI) beamformer design is proposed. The HRTF-based RLSFI beamformer accounts for the influence of a robot's head on the sound field. The performance of the new HRTF-based RLSFI beamformer is evaluated using signal-based measures and word error rates for an off-the-shelf speech recognizer, and is compared to the performance of the original free-field RLSFI beamformer design. The experimental results confirm the efficacy of the proposed HRTF-based beamformer design for robot audition.