Array Of Omnidirectional Microphones Research Articles

AeroSense: Sensing Aerosol Emissions from Indoor Human Activities

The types of human activities occupants are engaged in within indoor spaces significantly contribute to the spread of airborne diseases through emitting aerosol particles. Today, ubiquitous computing technologies can inform users of common atmosphere pollutants for indoor air quality. However, they remain uninformed of the rate of aerosol generated directly from human respiratory activities, a fundamental parameter impacting the risk of airborne transmission. In this paper, we present AeroSense, a novel privacy-preserving approach using audio sensing to accurately predict the rate of aerosol generated from detecting the kinds of human respiratory activities and determining the loudness of these activities. Our system adopts a privacy-first as a key design choice; thus, it only extracts audio features that cannot be reconstructed into human audible signals using two omnidirectional microphone arrays. We employ a combination of binary classifiers using the Random Forest algorithm to detect simultaneous occurrences of activities with an average recall of 85%. It determines the level of all detected activities by estimating the distance between the microphone and the activity source. This level estimation technique yields an average of 7.74% error. Additionally, we developed a lightweight mask detection classifier to detect mask-wearing, which yields a recall score of 75%. These intermediary outputs are critical predictors needed for AeroSense to estimate the amounts of aerosol generated from an active human source. Our model to predict aerosol is a Random Forest regression model, which yields 2.34 MSE and 0.73 r2 value. We demonstrate the accuracy of AeroSense by validating our results in a cleanroom setup and using advanced microbiological technology. We present results on the efficacy of AeroSense in natural settings through controlled and in-the-wild experiments. The ability to estimate aerosol emissions from detected human activities is part of a more extensive indoor air system integration, which can capture the rate of aerosol dissipation and inform users of airborne transmission risks in real time.

Spherical-sector harmonics domain processing for wideband source localization using spherical-sector array of directional microphones

The spherical microphone array can be uneconomical for applications where the sources arrive only from a known section of the sphere. For this reason, the spherical-sector harmonics was developed for processing spherical sector array. The orthonormal spherical sector harmonics (SSH) basis functions which accounts for the discontinuity arising from sectioning the sphere have been developed and shown to work for the array of omnidirectional microphones. In this work, the SSH basis functions are applied to far-field wideband sound source localization using spherical-sector array of first-order directional microphones (cardioid microphones). The array manifold interpolation method is used to produce the steered covariance matrix and the MUSIC algorithm applied for the direction of arrival estimation. The root-mean-square error performance of this spherical-sector array of the first-order cardioid microphones is compared against that of the omnidirectional microphones for different directions and signal-to-noise ratio.

Listeners Sensitivity to Different Locations of Diffusive Surfaces in Performance Spaces: The Case of a Shoebox Concert Hall

Diffusive surfaces are considered as one of the most challenging aspects to deal with in the acoustic design of concert halls. However, the acoustic effects that these surface locations have on the objective acoustic parameters and on sound perception have not yet been fully understood. Therefore, the effects of these surfaces on the acoustic design parameters have been investigated in a real shoebox concert hall with variable acoustics (Espace de Projection, IRCAM, Paris, France). Acoustic measurements have been carried out in six hall configurations by varying the location of the diffusive surfaces over the front, mid, and rear part of the lateral walls, while the other surfaces have been maintained absorptive or reflective. Moreover, two reference conditions, that is, fully absorptive and reflective boundaries of the hall have been tested. Measurements have been carried out at different positions in the hall, using an artificial head and an array of omnidirectional microphones. Conventional ISO 3382 objective acoustic parameters have been evaluated in all conditions. The results showed that the values of these parameters do not vary significantly with the diffusive surface location. Moreover, a subjective investigation performed by using the ABX method with auralizations at two listening positions revealed that listeners are not sensitive to the diffusive surface location variations even when front-rear asymmetric conditions are compared. However, some of them reported perceived differences relying on reverberance, coloration, and spaciousness.

FPGA implementation of adaptive beamforming in hearing aids.

Beamforming is a spatial filtering technique used in hearing aids to improve target sound reception by reducing interference from other directions. In this paper we propose improvements in an existing architecture present for two omnidirectional microphone array based adaptive beamforming for hearing aid applications and implement the same on Xilinx Artix 7 FPGA using VHDL coding and Xilinx Vivado® 2015.2. The nulls are introduced in particular directions by combination of two fixed polar patterns. This combination can be adaptively controlled to steer the null in the direction of noise. The beamform patterns and improvements in SNR values obtained from experiments in a conference room environment are analyzed.

Acoustic Monitoring of the Great Reed Warbler Using Multiple Microphone Arrays and Robot Audition

[abstFig src='/00290001/21.jpg' width='300' text='Spatial distribution pattern of the observed birds and localized sounds' ] This paper reports the results of our field test of HARKBird, a portable system that consists of robot audition, a laptop PC, and omnidirectional microphone arrays. We assessed its localization accuracy to monitor songs of the great reed warbler (Acrocephalus arundinaceus) in time and two-dimensional space by comparing locational and temporal data collected by human observers and HARKBird. Our analysis revealed that stationarity of the singing individual affected the spatial accuracy. Temporally, HARKBird successfully captured the exact song duration in seconds, which cannot be easily achieved by human observers. The data derived from HARKBird suggest that one of the warbler males dominated the sound space. Given the assumption that the cost of the singing activity is represented by song duration in relation to the total recording session, this particular male paid a higher cost of singing, possibly to win the territory of best quality. Overall, this study demonstrated the high potential of HARKBird as an effective alternative to the point count method to survey bird songs in the field.

Objective and perceptual evaluation of distance-dependent scattered sound effects in a small variable-acoustics hall.

Performance spaces are characterized by a complex sound field, due to the presence of absorptive and diffusive surfaces. In situ evaluations of the acoustic effects that these surfaces have on the objective acoustic parameters and on sound perception have not yet been fully understood. To this aim, acoustic measurements have been performed in a variable-acoustic concert hall, the Espace de Projection, at the Institut de Recherche et Coordination Acoustique/Musique. These measurements have allowed the effects of one single wall to be determined. A diffusive and a reflective condition of one of the long lateral walls of the shoebox-like hall have been considered, while the other surfaces have been fixed in absorptive mode. Measurements have been carried out at different distances from the test wall, using an artificial head and an array of omnidirectional microphones. Objective acoustic parameters, such as early decay time, reverberation time (T30), clarity (C80), definition (D50), and interaural cross correlation, have been compared between both conditions. In addition to the objective indexes, a perceptual evaluation has been performed using listening tests that had the purpose of determining the maximum distance from a diffusive surface at which acoustic scattering effects are still audible.

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.

Individual Variability in Benefit from Fixed and Adaptive Directional Microphones

This study aimed to document improvements in speech recognition when listening through fixed or adaptive directional microphone arrays relative to listening through omnidirectional microphone arrays in a variety of near-field noise conditions. Twelve participants with mild to severe hearing loss were recruited. Data were analyzed in a manner to facilitate discussion of both group and individual performance. The activation of directional microphones significantly improved speech recognition in all listening conditions when compared to omnidirectional listening conditions. Significant differences in speech recognition performance were not observed between fixed and adaptive directional microphone conditions. Analysis of individual outcomes shows a wide range of both absolute speech recognition scores and directional benefit across treatment conditions.

32-Channel Omni-Directional Microphone Array Design and Implementation

This paper presents the design and evaluation of a microphone array. The proposed evaluation index is the directional characteristic of delay and sum beamforming, which is used to optimize the microphone array design. Using beamforming simulation, a microphone arrangement that minimizes sidelobes and improves the basic performance of beamforming is selected. The new hardware has omni-directional directivity and high tolerance for exogenous noise. It has 32 microphones on a 335-mm diameter disk designed to be mounted on a mobile robot. The microphone array performance is verified in different real environments. Experimental results in indoor/outdoor sound localization show the effectiveness of the array in reverberant environments and its robustness against different pressure sound sources for covering larger areas.

2A2-B02 32ch低サイドローブ設計マイクロホンアレイの開発と屋外での音源定位評価

2A2-B02 32ch低サイドローブ設計マイクロホンアレイの開発と屋外での音源定位評価

Microphone array beamforming with near-field correlated sources.

As the price of omnidirectional microphone arrays has fallen, new applications have emerged for large room audio capture. Further, large room acoustics use loudspeakers, which create correlated sources and severely limit the use of optimum beamformers. Most sources are in the near field due to the large wavelength of speech. Widrow and Kailath’s work on correlation in sonar and radar assumed far field, but it cannot be directly applied to near-field acoustics. A new beamforming method has been devel-oped, which incorporates near field and correlation. The source bearings are estimated and an uncorrelated model is formed. A better DOA estimator was developed for the near-field correlated case limited by the width of the mainlobe. The new beamformer is able to null strong, perfectly and nearly perfectly correlated signals using the uncorrelated model and minimum variance distortionless response beamforming.

Introduction to dual sensory loss issue.

Introduction to dual sensory loss issue.

Improved BTE hearing-aid directivity using a directional microphone array

Extraction of speech in noise is of great importance to hearing-impaired listeners. Directional microphones are incorporated in some hearing aids to improve noise rejection through increased directivity. A symmetric cardioid response is created either with a single directional microphone, or using beamforming with two omnidirectional microphones. The head-related transfer function (HRTF), however, introduces an asymmetry that cannot be exploited by a linear array of omnidirectional microphones. A new BTE array consisting of a gradient directional microphone with nulls in the front–back vertical plane and two omnidirectional microphones exploits the asymmetry of the HRTF to obtain almost 2 dB better directivity than the best cardioid. HRTFs measured on KEMAR with this array were transformed to the frequency domain, where directivity-maximizing coefficients in each band were derived. The Articulation-Index (AI) weighted directivity gain of this optimal three-element directional array was 6.4 dB greater than a single omnidirectional microphone on the BTE, whereas the directivity gain of the HRTF-optimized two-omni beamformer was 4.6 dB, and the optimal free-field cardioid placed on the head yielded 4.4 dB. [Work partially supported by NIH (NIDCD) under Grant No. 1 R01 DC005762-01A1.]

Active stereo sound localization.

Estimating the direction of arrival of sound in three-dimensional space is typically performed by generalized time-delay processing on a set of signals from a fixed array of omnidirectional microphones. This requires specialized multichannel A/D hardware, and careful arrangement of the microphones into an array. This work is motivated by the desire to instead only use standard two-channel audio A/D hardware and portable equipment. To estimate direction of arrival of persistent sound, the position of the microphones is made variable by mounting them on one or more computer-controlled pan-and-tilt units. In this paper, we describe the signal processing and control algorithm of a device with two omnidirectional microphones on a fixed baseline and two rotational degrees of freedom. Experimental results with real data are reported with both impulsive and speech sounds in an untreated, normally reverberant indoor environment.

Comparison of binaural beamformers for speech extraction in complex auditory scenes

The goal of this study was to assess and compare the capacities of several array-processing algorithms to extract on-axis signals in the presence of one or more off-axis interferers. Performance comparisons are made between the localization/extraction (L/E) algorithm [Liu et al., J. Acoust. Soc. Am. 108, 1888–1905 (2000)], the frequency-banded minimum-variance beamformer (FBMVB) [Lockwood et al., J. Acoust. Soc. Am. 106, 2278 (1999)], the Frost beamformer, the Griffiths–Jim generalized sidelobe canceller (GSC), and the Peissig–Kollmeier (P-K) algorithm [Kollmeier et al., J. Rehab. Res. Dev. 30, 82–94 (1993)]. The two metrics used to evaluate algorithm performance were the signal-to-noise ratio gain (SNRG) and the intelligibility-weighted signal-to-noise ratio gain (IWSNRG). The signals used simulated an omnidirectional microphone array in free field and were composed of sentence-length speech by male and female speakers against a background of one to four interfering sound sources. Overall, the IWSNRGs for the L/E and FBMVB algorithms were similar to those for the Frost, GSC, and P-K algorithms when there was a single interferer, but 3–4 dB higher when there were two or more interferers. This study shows that FBMVB and L/E algorithms hold promise for hearing aids to be used in noisy environments. [Work supported by NIH-NIDCD Grant R21DC04840.]