Sensitivity is one of the most important properties of a microphone. The commonly used methods to analyze microphone sensitivity include lumped parameter method (LPM) and finite element method (FEM). ...The intricate structure of the capacitive dual-membrane microelectromechanical systems(MEMS) microphone poses challenges for the two methods. The lumped-parameter method is computationally efficient but lacks precision in describing the the complex structure, while the finite element method can effectively analyze structural influences albeit with lower computational efficiency. To address this, In this paper, the combination of LPM and FEM is used to analyze the sensitivity of MEMS microphones with dual-membrane structure, which can not only ensure the calculation accuracy of FEM but also greatly improve the calculation efficiency. In this way, several main parameters affecting the sensitivity of MEMS microphones with double membrane structure are analyzed, such as the gap distance between the membrane and the backplate, the diameter of the pillar connecting the two membranes, the volume of the front cavity, and the membrane tension.
Wind noise interference can adversely affect the performance of behind-the-ear hearing aids, particularly when used outdoors. In this study, we developed a novel strategy to mitigate this problem. ...Our methodology involved a modified microphone enclosure design, preprocessing stages of beamforming and wide dynamic range compression (WDRC) (which are inherent in conventional hearing aids), and advanced deep-learning-based noise reduction methods. We first explored the cavity aspect ratios and wall slanting of the microphone enclosure's design and determined that noise could be reduced by modifying the upper and lower enclosure widths from 0.8 mm (for both) to 1.1 and 0.5 mm, respectively. In terms of signal processing, a 1-D convolutional neural networks (CNNs) model achieved wind noise detection with an accuracy of 99.25% in various scenarios. The U-Net deep learning architecture was implemented for noise reduction and substantially improved short-time objective intelligibility (STOI) by 18.97%-209.09% compared with traditional high-pass filters (HPFs). Training with a voice database further improved the STOI. In terms of the mean opinion score-listening quality objective (MOS-LQO) and STOI metrics, all combinations of preprocessors with U-Net outperformed U-Net alone, and beamforming was the optimal preprocessing method. In conclusion, adaptive signal preprocessing based on wind classification, microphone enclosure optimization, and U-Net deep learning techniques effectively reduced wind noise interference, improving the outdoor usability, and listening experience provided by hearing aids.
An analytical method for 3-D sound source localization based on a five-element microphone array is proposed in this work. With four time-difference-of-arrival (TDOA) values, the proposed method ...realizes the sound source localization by analytical solution. A new five-element microphone array configuration (which is characterized in that four microphones are in the same plane and at four vertices of a concave quadrilateral respectively, while the fifth microphone is out of the plane) is analyzed and studied. It is proven that with the proposed configuration, the five-element microphone array can obtain a unique and analytical solution. To investigate the effectiveness and the localization performance of the proposed method, a practical five-element microphone array is developed and 3-D sound source localization experiments are carried out. The TDOA values are obtained by the generalized cross correlation (GCC) method based on the phase transform (PHAT). The experimental results show that the proposed method is an effective near-field localization method. The localization performance of the practical five-element microphone array is satisfactory. Compared with the conventional methods, the proposed method realizes the 3-D localization by analytical solution and has the advantages of simple construction and lower computational complexity.
A sound field recording method based on spherical or circular harmonic analysis for arbitrary array geometry and directivity of microphones is proposed. In current methods based on harmonic analysis, ...a sound field is decomposed into harmonic functions with a center given in advance, which is called a global origin, and their coefficients are obtained up to a certain truncation order using microphone measurements. However, the accuracy of the reconstructed sound field depends on the predefined position of the global origin and the truncation order, which makes it difficult to apply this technique to an asymmetric array since the criterion to determine the position of the global origin and the truncation order is not obvious. We formulate an estimate of the harmonic coefficients on the basis of infinite-order analysis. This formulation enables us to estimate the harmonic coefficients at an arbitrary desired position independently of the position of the global origin without truncation errors. Numerical simulation results indicated that the proposed method makes it possible to avoid performance degradation caused by inappropriate setting of the global origin.
Voice assistants are widely integrated into a variety of mobile devices, enabling users to easily complete daily tasks and even critical operations like online transactions with voice commands. Thus, ...once attackers replay a secretly-recorded voice command by loudspeakers to compromise users' voice assistants, this operation will cause serious consequences, such as information leakage and property loss. Unfortunately, most voice liveness detection approaches against replay attacks mainly rely on detecting lip motions or subtle physiological features in speech, which are limited within a very short range. In this paper, we propose VoShield to check whether a voice command is from a genuine user or a loudspeaker imposter. VoShield measures sound field dynamics, a feature that changes fast as the human mouths dynamically open and close. In contrast, it would remain rather stable for loudspeakers due to the fixed size. This feature enables VoShield to largely extend the working distance and remain resilient to user locations. Besides, sound field dynamics are extracted from the difference between multiple microphone channels, making this feature robust to voice volume. To evaluate VoShield, we conducted comprehensive experiments with various settings in different working scenarios. The results show that VoShield can achieve a detection accuracy of 98.2% and an Equal Error Rate of 2.0%, which serves as a promising complement to current voice authentication systems for smart mobile devices.
A novel systematic approach to the design of directivity patterns of higher order differential microphones is proposed. The directivity patterns are obtained by optimizing a cost function which is a ...convex combination of a front-back energy ratio and uniformity within a frontal sector of interest. Most of the standard directivity patterns - omnidirectional, cardioid, subcardioid, hypercardioid, supercardioid - are particular solutions of this optimization problem with specific values of two free parameters: the angular width of the frontal sector and the convex combination factor. More general solutions of practical use are obtained by varying these two parameters. Many of these optimal directivity patterns are trigonometric polynomials with complex roots. A new differential array structure that enables the implementation of general higher order directivity patterns, with complex or real roots, is then proposed. The effectiveness of the proposed design framework and the implementation structure are illustrated by design examples, simulations, and measurements.
•Applied the two-microphone method in an impedance tube based on two MEMS microphones. This simple setup maintains the low-cost nature of the MEMS microphone application.•Extended the upper-frequency ...range of impedance tube application to fully cover the upper audible range and the lower ultrasonic range. The high-frequency range is getting increasingly more attention in the industry.•Applied adaptive sine excitation and curve-fitting technique to handle the strongly nonlinear frequency response of the measurement system. The presence of nonlinear frequency response is common in the application of MEMS microphones and the high-frequency range.•Overcome the intrinsic drawback of the two-microphone method caused by the coincidence of microphone positions with pressure nodes.
The standardized two-microphone method is widely established for determining normal-incidence acoustic properties. Instead of numerous low-frequency usages, only a few high-frequency applications are reported. This paper provides insights into implementing the two-microphone method in an impedance tube based on two MEMS microphones for a frequency range of 5 kHz to 50 kHz. In doing so, we encountered technical issues associated with the MEMS microphone utilization and the high-frequency application. These issues can be addressed by implementing a modified method, in which an adaptive stepped sine excitation and a curve-fitting technique were applied. The modified method was validated by using three classical tube terminations, e.g., closed, open, and horn. The measured data of these terminations are in line with their analytic solutions. In conclusion, when proper concerns are taken, the acoustic properties up to the lower ultrasonic range can be determined using the modified two-microphone method in an impedance tube based on two MEMS microphones.
In this paper, we address the issue of near-field source localization using spherical microphone array. The spherical array has been widely used for far-field source localization due to ease of array ...processing in spherical harmonics domain. Various methods for far-field source localization has been reformulated in spherical harmonics domain. However, near-field source localization that involves joint estimation of range and bearing of the sources has hitherto not been investigated. In this paper, the near-field data model is developed in spherical harmonics domain. In particular, three methods that jointly estimate the range and bearing of multiple sources in the spherical array framework are proposed. Two subspace-based methods called the Spherical Harmonic MUltiple SIgnal Classification (SH-MUSIC) and the Spherical Harmonics MUSIC-Group Delay (SH-MGD) for near-field source localization are first presented. In addition, a method for near-field source localization and beamforming using Spherical Harmonic MVDR (SH-MVDR) is also formulated. Formulation and analysis of Cramér-Rao bound for near-field sources is presented in spherical harmonics domain. Various source localization experiments were conducted on simulated and signal acquired over spherical microphone array in an anechoic chamber. Root-mean-square error and probability of resolution are utilized as measures to evaluate the proposed methods. The significance and practical application of the proposed methods is discussed using experiment on interference suppression. The near-field SH-MVDR beamforming is utilized in this context.
Recent advances in neural engineering have enabled direct control of insect locomotion through neural and muscular stimulation. The resulting insect biobots, with a natural ability to crawl through ...small spaces, offer unique advantages over traditional synthetic robots. A cyberphysical network of such biobots could prove useful for search and rescue applications in uncertain disaster environments. We present a vision-based automated system for an objective assessment of biobotic navigation capability on Madagascar hissing cockroaches. We report the most precise control results obtained with insect biobots so far both manually and autonomously. We also demonstrate autonomous control capability where a low-power insect-mounted array of microphones was used to localize a sound source and guide the biobot toward it. Forming a wireless mobile sensor network with directional and omnidirectional microphones distributed within the structure of a rubble pile could be useful for both environmental mapping and localization of trapped survivors under the rubble.
In this paper, we propose a novel approach to noise suppression using multiple distributed recording devices with stereo microphones. In the proposed method, noise suppression based on phase ...information is applied to the synchronous stereo signals captured by each recording device and then output signals are utilized for transfer-function-gain nonnegative matrix factorization (NMF) as extra input signals. We intended to estimate the target signal more accurately by transfer-function-gain NMF. Experiments using impulse responses measured in a meeting room have shown that the proposed method outperformed conventional methods using transfer-function-gain NMF in terms of the signal-to-distortion ratio (SDR) and signal-to-interference ratio (SIR).
