Directional Microphone Arrays Research Articles

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.

An mm-sized biomimetic directional microphone array for sound source localization in three dimensions

Fly Ormia ochracea ears have been well-studied and mimicked to achieve subwavelength directional sensing, but their efficacy in sound source localization in three dimensions, utilizing sound from the X-, Y-, and Z-axes, has been less explored. This paper focuses on a mm-sized array of three Ormia ochracea ear-inspired piezoelectric MEMS directional microphones, where their in-plane directionality is considered a cue to demonstrate sound source localization in three dimensions. In the array, biomimetic MEMS directional microphones are positioned in a 120° angular rotation; as a result, six diaphragms out of three directional microphones keep a normal-axis relative to the sound source at six different angles in the azimuth plane starting from 0° to 360° in intervals of ±30°. In addition, the cosine-dependent horizontal component of the applied sound gives cues for Z-axis directional sensing. The whole array is first analytically simulated and then experimentally measured in an anechoic chamber. Both results are found to be compliant, and the angular resolution of sound source localization in three dimensions is found to be ±2° at the normal axis. The resolution at the azimuth plane is found to be ±1.28°, and the same array shows a ± 4.28° resolution when sound is varied from the elevation plane. Looking at the scope within this area combined with the presented results, this work provides a clear understanding of sound source localization in three dimensions.

Automatic environmental soundscape classification of continuous field recordings around Lake George, NY

The amount of audio data required for long-term bioacoustics monitoring is often too large to be manually sorted. Automatic environmental sound recognition techniques are therefore applied to extract relevant acoustic stimuli and classify these stimuli after a training period. In this work, a continuous 24-h, remote audiovisual recording system was developed and deployed near a tributary stream of Lake George, NY for the automatic collection of environmental and animal sounds. Besides monitoring natural environments, this system can also be used to establish an automatic protocol for collaborative business meetings. In conjunction with the efforts of the RPI/IBM Jefferson Project to create a network of sensors continuously monitoring the lake, the goal of this research is to automatically transcribe the soundscape of the lake using a network of directional microphone arrays positioned throughout the watershed. After establishing a training database of relevant acoustic stimuli, a convolutional neural network is used to classify unprocessed audio data. Pilot studies show good results when learning takes place directly from spectrograms, since the variability of non-speech acoustic events often requires more rich detail than can be provided by MFCCs or other compressed feature sets. [Work supported by NSF #1631674, CISL, and the Jefferson Project.]

A nested beamforming approach to sound source separation using an array-of-arrays recording system

Beamforming is a technique used with microphone arrays to selectively isolate sounds according to their direction of incidence. However, conventional methods struggle to separate sound sources if the spatial resolution of the recording system is not enough to define a recording zone containing only the target. In this presentation, a sound source separation method using multiple beamforming arrays is introduced. The proposal consists of three stages. The first stage applies modal beamforming methods to individual spherical microphone arrays; this results in a virtual array of directional microphones. The second stage distributes the recordings from the virtual microphones onto a new angular coordinate system. This stage transforms the radial coordinates for the different sound sources into pairs of parallax angles between them and the spherical microphone arrays. Finally, the third stage uses a least-squares optimized beamforming method defined on the parallax-angle coordinate system. The combination of t...

Analytical approach to transforming filter design for sound field recording and reproduction using circular arrays with a spherical baffle.

A sound field recording and reproduction method using circular arrays of microphones and loudspeakers with a spherical baffle is proposed. The spherical baffle is an acoustically rigid object on which the microphone array is mounted. The driving signals of the loudspeakers must be obtained from the signals received by the microphones. A transform filter for this signal conversion is analytically derived, which is referred to as the wave field reconstruction filter. The proposed method using a spherical baffle is compared with methods using an array of directional microphones and a microphone array mounted on a cylindrical baffle. Numerical simulations indicated that the proposed method is advantageous for sound field recording and reproduction compared with the other two methods. The results of measurement experiments in a real environment are also demonstrated.

Adaptive beamforming for acoustic echo cancellation using measured array models and subband filtering

Acoustic echo that can substantially undermine speech quality is one of the key issues must be addressed in practical telecommunication systems. In this paper, an evolutionary exposition is given in regard to the enhancing strategies for acoustic echo cancelers (AEC). On the basis basic adaptive filtering, a highly directional microphone array is designed as a fixed beamformer (FBF) to focus on the near-end speaker and suppress the echo from the far-end as well as noise and interference from the background. Design of the beamformer is based on array models interpolated from measured frequency responses. Subband (SB) filtering with polyphase decomposition is exploited to accelerate the cancellation process with multiple choice of step size in the subbands. To further enhance the canceler, adaptive generalized sidelobe canceler (GSC) can be utilized, which is comprised of a FBF for the near-end speech and an adaptive blocking module for the echo and noise. Objective tests in terms of echo return loss enhancement (ERLE) and perceptual evaluation of speech quality (PESQ) are conducted to compare four algorithms with various combinations of enhancement. The results show that the GSC-SB-AEC approach has attained the highest ERLE and best speech quality, even in a double-talk scenario.

Small directional microelectromechanical systems microphone arrays

Directional microphone arrays that are physically small compared to the acoustic wavelength are of great interest for hand-held communication devices. Spatially directive microphones can reduce the impact of background acoustic noise without adding distortion to the signal. This talk will present some design topologies and requirements as well as a new physical design that could enable directional microphone responses while being small in size.

Performance of a highly directional microphone array in a multi-talker reverberant environment

A visual guided hearing aid (VGHA) has recently been developed, which uses an eye tracker to steer the “acoustic look direction” (ALD) of a beamforming microphone array. The current study evaluates the performance of this highly directional microphone in providing spatial release from masking (SRM) under acoustically dry and reverberant conditions. Four normal-hearing subjects participated in a speech intelligibility test with collocated and spatially separated speech maskers when listening either through the microphone array or through KEMAR to simulate “natural” binaural conditions. The results indicated that near-normal SRM was achieved by listening through the VGHA in both environments. In the acoustically dry condition, SRM was similar to the measured signal-to-noise ratio (SNR) gain from the microphone array. However, in the reverberant condition, subjects showed significantly greater SRM than predicted from the measured SNR gain from the array. This is consistent with the measured improvement in SNR for the early part of the room impulse response for the target but not for the spatially separated maskers. This indicates that in some reverberant conditions the microphone array provides substantial source selection benefits.

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.

On the measurement of directivity index for adaptive directional hearing aids

Adaptive directional microphone arrays have been widely utilized in hearing aids to help wearers understand speech in noisy environments. However, a practical and objective way of measuring the benefit of this technique is not obvious. One commonly used measure for directional microphone arrays is directivity index (DI), which evaluate the arrays' capability to attenuate a diffuse noise field. However, the DI has rarely been considered for the Adaptive directional microphone arrays because a truly diffuse noise field is required for the measurement. This paper investigates the feasibility of measuring the DI of adaptive directional hearing aids in a diffuse noise field generated by a loudspeaker array playing uncorrelated noise signals simultaneously. The requirement for the degree of diffuseness of the noise field is studied based on numerical simulations for first-order and second-order adaptive microphone arrays. The possibility of measuring the DI in both anechoic and non-anechoic conditions is investigated. The accuracy of the proposed approach is further discussed.

Response surface optimization of a directional endfire microphone array for hearing aids

The optimal operating parameters for a directional microphone array worn in situ are not necessarily equivalent to the optimal parameters while operating in the absence of head and torso related scattering. These parameters include the relative magnitude and phase of the microphones and their positional placement on the head, characterized as factors, operating over a range of levels, characterized by their production spread and susceptibility to drift. The goal is to understand how these factors operating over their levels contribute to the in-situ directional responses on a measurement manikin, characterized by the directivity index and the unidirectional index. Using 614 impulse responses acquired in ten deg resolution on the manikin, a simple central composite design of experiments was conducted to fit a quadratic polynomial and generate a response surface to the aforementioned directional indices, thereby leading to the critical first-order and two factor interactions of the system. The interactions, statistical validity of the predictive polynomials, and the sweet spot of operation will be described for some common hearing aid microphone arrays.

Practical and mathematical aspects of pairing microphones for hearing aid directional arrays.

Directional microphone arrays in hearing aids are short baseline, end-fire arrays, which require tight matching of response characteristics between the microphones, as previously reported [J. Acoust. Soc. Am. 113(4,) 2219 (2003)]. Pairing tolerances for good directional performance are much tighter than normal manufacturing tolerances on individual microphones, so it is infeasible to produce entire production lots within these tolerances. Instead, microphones are selected in pairs based on their measured performance. In some cases, the pairing tolerance is on par with the measurement uncertainty of even a well-designed manufacturing test system. Standard gauge analysis is not well suited to this pairing measurement. An analysis of acoustical measurement error as it relates to the pairing problem, the statistics of successful pairing from a large quantity of microphones, and the underlying graph theory of optimum pairing will be discussed in this paper.

Three‐dimensional polar analysis of a directional endfire microphone array on a measurement manikin due to microphone drift.

A dual microphone endfire array is a directional system commonly used in hearing aids. The directional performance of such systems are sensitive to sensor mismatch and drift. In this study, a pair of matched omnidirectional microphones in a delay‐and‐sum configuration are mounted in a hearing aid and their directional response is measured in‐situ on KEMAR at 10 resolution in azimuth and elevation. The resulting 3‐D polar balloons, directivity indices, and unidirectional indices are computed as a function of frequency. The measured transfer functions are then perturbed with sensor mismatch responses acquired empirically from typical lots of hearing‐aid microphones. The resulting polar benchmarks are evaluated and compared to the original benchmarks. A detailed analysis, both visual and numerical, will be presented.

Higher order differential-integral microphone arrays

This paper develops theory to design higher order directional microphone arrays. The proposed higher order designs have similar inter sensor spacings as traditional first and second order differential arrays. The Jacobi-Anger expansion is used to exploit the underlying structure of microphone signals from pairs of closely spaced sensors. Specifically, the difference and sum of these microphone signals are processed to design the novel directional array.

Aeroacoustics of Drag-Generating Swirling Exhaust Flows

Aircraft on approach in high-drag and high-lift configuration create unsteady flow structures which inherently generate noise. For devices such as flaps, spoilers and the undercarriage there is a strong correlation between overall noise and drag such that, in the quest for quieter aircraft, one challenge is to generate drag at low noise levels. This paper presents a rigorous aero-acoustic assessment of a novel drag concept. The idea is that a swirling exhaust flow can yield a steady, and thus relatively quiet, streamwise vortex which is supported by a radial pressure gradient responsible for pressure drag. Flows with swirl are naturally limited by instabilities such as vortex breakdown. The paper presents a first aero-acoustic assessment of ram pressure driven swirling exhaust flows and their associated instabilities. The technical approach combines an in-depth aerodynamic analysis, plausibility arguments to qualitatively describe the nature of acoustic sources, and detailed, quantitative acoustic measurements using a medium aperture directional microphone array in combination with a previously established Deconvolution Approach for Mapping of Acoustic Sources (DAMAS). A model scale engine nacelle with stationary swirl vanes was designed and tested in the NASA Langley Quiet Flow Facility at a full-scale approach Mach number of 0.17. The analysis shows that the acoustic signature is comprised of quadrupole-type turbulent mixing noise of the swirling core flow and scattering noise from vane boundary layers and turbulent eddies of the burst vortex structure near sharp edges. The exposed edges are the nacelle and pylon trailing edge and the centerbody supporting the vanes. For the highest stable swirl angle setting a nacelle area based drag coefficient of 0.8 was achieved with a full-scale Overall Sound Pressure Level (OASPL) of about 40dBA at the ICAO approach certification point.

Maximum Likelihood Sound Source Localization and Beamforming for Directional Microphone Arrays in Distributed Meetings

In distributed meeting applications, microphone arrays have been widely used to capture superior speech sound and perform speaker localization through sound source localization (SSL) and beamforming. This paper presents a unified maximum likelihood framework of these two techniques, and demonstrates how such a framework can be adapted to create efficient SSL and beamforming algorithms for reverberant rooms and unknown directional patterns of microphones. The proposed method is closely related to steered response power-based algorithms, which are known to work extremely well in real-world environments. We demonstrate the effectiveness of the proposed method on challenging synthetic and real-world datasets, including over six hours of recorded meetings.

Measurements of sound leaks or ‘‘hot spots’’ and their effect on the architectural speech security of closed rooms

A new measurement procedure has been developed that accurately indicates the degree of speech security (speech privacy) of a closed office or meeting room. The procedure first determines the attenuation from an average sound-pressure field within a source room to single receiving points outside the room. These average-to-spot attenuations are used to predict transmitted speech levels outside of meeting rooms, which along with the background noise levels are used to derive a reliable indicator of the audibility or intelligibility of speech at the receiving points. A key aspect of the approach is that a localized weak spot in an otherwise highly insulating partition can be identified and assessed. Investigations were carried out as to the severity and detectability of different types of sound leaks intentionally introduced into an otherwise ‘‘good’’ wall separating two reverberation chambers. Steady-state and impulse response measurements of received pressure were made at a distance of 0.25 m from the wall. In addition, a highly directional beamforming microphone array was used in an effort to quickly locate the position of potential leaks. An overview of the measurement procedures will be presented, including discussion of the severity of several types of ‘‘hot spots,’’ such as holes, penetrations, and electrical boxes.

An electronic windscreen for microphones

Wind-noise pickup in microphones can be highly disturbing, especially for directional microphones that attain their directivity by using velocity or spatial derivatives of the acoustic pressure. Since small directional microphones are common in audio communications systems, the problem of wind noise is well known and the typical solution is to use a windscreen. Conventional windscreens that block the wind but do not significantly alter the sound field can be very effective, but need to be relatively large to attain a good reduction in wind-noise pick-up. Small directional microphone arrays are more commonly used in hearing aids since they have been clinically shown to improve hearing in noise for hearing-impaired users. Since hearing aids are required to be compact, an effective conventional windscreen is not practical. This talk describes a novel multi-microphone wind-noise reduction algorithm that we refer to as an ‘‘electronic windscreen.’’ This new algorithm and its variants have low computational cost and are therefore applicable to commercial digital hearing aids that use multi-microphone arrays to attain directionality.

Effects of Angle of Attack and Velocity on Trailing Edge Noise Determined Using Microphone Array Measurements

Trailing edge (TE) noise measurements for a NACA 63–215 airfoil model are presented, providing benchmark experimental data for a cambered airfoil. The effects of flow Mach number and angle of attack of the airfoil model with different TE bluntnesses are shown. Far-field noise spectra and directivity are obtained using a directional microphone array. Standard and diagonal removal beamforming techniques are evaluated employing tailored weighting functions for quantitatively accounting for the distributed line character of TE noise. Diagonal removal processing is used for the primary database as it successfully removes noise contaminants. Some TE noise predictions are reported to help interpret the data with respect to effects of flow speed, angle of attack, and TE bluntness on spectral shape and peak levels. Important findings include the validation of a TE noise directivity function for different airfoil angles of attack and the demonstration of the importance of the directivity function's convective amplification terms.

Microphone array having a second order directional pattern

A directional microphone system is disclosed, which comprises circuitry for low pass filtering a first order signal, and circuitry for high pass filtering a second order signal. The system further comprises circuitry for summing the low pass filtered first order signal and the high pass filtered second order signal. A method of determining whether a plurality of microphones have sufficiently matched frequency response characteristics to be used in a multi-order directional microphone array is also disclosed. For a microphone array having at least three microphones, wherein one of the microphones is disposed between the other of the microphones, a method of determining the arrangement of the microphones in the array is also disclosed.