Performance Of Microphone Arrays Research Articles

Acoustic source localization using the open spherical microphone array in the low-frequency range

Recently, spherical microphone arrays (SMA) have become increasingly significant for source localization and identification in three dimension due to its spherical symmetry. However, conventional Spherical Harmonic Beamforming (SHB) based on SMA has limitations, such as poor resolution and high side-lobe levels in image maps. To overcome these limitations, this paper employs the iterative generalized inverse beamforming algorithm with a virtual extrapolated open spherical microphone array. The sidelobes can be suppressed and the main-lobe can be narrowed by introducing the two iteration processes into the generalized inverse beamforming (GIB) algorithm. The instability caused by uncertainties in actual measurements, such as measurement noise and configuration problems in the process of GIB, can be minimized by iteratively redefining the form of regularization matrix and the corresponding GIB localization results. In addition, the poor performance of microphone arrays in the low-frequency range due to the array aperture can be improved by using a virtual extrapolated open spherical array (EA), which has a larger array aperture. The virtual array is obtained by a kind of data preprocessing method through the regularization matrix algorithm. Both results from simulations and experiments show the feasibility and accuracy of the method.

A mixed-order ambisonic scheme to improve performance for sound sources on the horizontal plane

This paper proposes a mixed-order ambisonic scheme called #P#L that improves performance in terms of estimation of the coefficient of spherical harmonics expansion, reconstruction of sound fields, and spatial resolution for sound sources located on the horizontal plane. Mixed-order ambisonics (MOA) selects appropriate components of spherical harmonics expansion instead of truncating up to a certain order as in higher-order ambisonics (HOA). Several schemes of MOA including the proposed scheme and HOA are compared each other in several ways. The results show that the proposed scheme has several advantages compared with the other MOA schemes. This scheme can be useful to further improve performance of microphone arrays if the number of the microphones is greater than what is required for HOA schemes or a non-uniform layout of microphones is used.

Comparison of the measured and theoretical performance of a broadband circular microphone array

The design and construction of a circular microphone array (CMA) that has a wide frequency range suitable for human hearing is presented. The design of the CMA was achieved using a technique based on simulated directivity index (DI) curves. The simulated DI curves encapsulate the critical microphone array performance limitations: spatial aliasing, measurement noise, and microphone placement errors. This paper demonstrates how the non-regularized DI curves for a given beamforming order clearly define the bandwidth of operation, in other words, the frequency band for which the beamformer has relatively constant and maximum directivity. Detailed and comprehensive experimental data that characterizes the CMA beamformer are also presented.

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.

Spatial Aliasing in Spherical Microphone Arrays

Performance of microphone arrays at the high-frequency range is typically limited by aliasing, which is a result of the spatial sampling process. This paper presents analysis of aliasing for spherical microphone arrays, which have been recently studied for a range of applications. The paper presents theoretical analysis of spatial aliasing for various sphere sampling configurations, showing how high-order spherical harmonic coefficients are aliased into the lower orders. Spatial antialiasing filters on the sphere are then introduced, and the performance of spatially constrained filters is compared to that of the ideal antialiasing filter. A simulation example shows how the effect of aliasing on the beam pattern can be reduced by the use of the antialiasing filters

Acoustic array tracking performance under moderately complex environmental conditions

The direction-of-arrival (DOA) tracking performance of microphone arrays having aperture sizes ranging from 0.3 to 34 m is examined for an experiment involving a vehicle traversing a moderately complex terrain. A segment of the vehicle’s path was obscured behind a small, 6.7-m high, vegetated hill. The combination of the hill and upwind propagation created an acoustic shadow during this segment. DOA tracks were estimated with a minimum-variance distortionless response (MVDR) beamformer operating in two frequency bands: 25–60 Hz and 60–105 Hz. In the lower frequency band, array sizes between 1 and 8 m gave the best results, with DOA errors between 2° and 5°. Furthermore, in this band shadowing from the hill and wind refraction had a minimal affect on DOA error. In the higher frequency band, the acoustic shadow zone produced a distinct interval of high DOA error, with the 8-m array giving the best overall performance. Modeling of the beamforming process shows that high DOA errors corresponded to MVDR wavenumber patterns that are degraded by distortions to the propagating wavefronts. Our experimental results indicate that small acoustic arrays with apertures less than 0.3 m, operating at frequencies above 100 Hz, should be considered line of sight sensors. Given the moderate complexity of the test conditions, it is anticipated that the observed effects are likely to be present in most attempts to localize outdoor sound sources.

Speech Enhancement Based on the General Transfer Function GSC and Postfiltering

In speech enhancement applications microphone array postfiltering allows additional reduction of noise components at a beamformer output. Among microphone array structures the recently proposed general transfer function generalized sidelobe canceller (TF-GSC) has shown impressive noise reduction abilities in a directional noise field, while still maintaining low speech distortion. However, in a diffused noise field less significant noise reduction is obtainable. The performance is even further degraded when the noise signal is nonstationary. In this contribution we propose three postfiltering methods for improving the performance of microphone arrays. Two of which are based on single-channel speech enhancers and making use of recently proposed algorithms concatenated to the beamformer output. The third is a multichannel speech enhancer which exploits noise-only components constructed within the TF-GSC structure. This work concentrates on the assessment of the proposed postfiltering structures. An extensive experimental study, which consists of both objective and subjective evaluation in various noise fields, demonstrates the advantage of the multichannel postfiltering compared to the single-channel techniques.

Developing portable acoustic arrays on a large‐scale e‐textile substrate

This research deals with the production of electronic textiles (e‐textiles) demonstrators. Initially, the research dealt with the creation of 4×5 microphone array on a large area conformal textile substrate. Once the interface electronics were connected to the 4×5 microphone array, this system became an effective acoustic array. Here, a new acoustic eight microphone array design has been designed, fabricated and tested. Changes were made to improve microphone array performance, and to optimize the associated software for data capture and analysis. This new design was based on UC‐Berkeley mote microcomputer technology. The mote‐based system addresses the issue of scaling acoustic arrays, to allow for distributing microphones over large‐areas, and to allow performance comparisons to be made with the original 4×5 microphone acoustic array.

Performance of microphone array geometries in a reverberant room

AbstractThe choice of sensor positions plays a critical role in determining the performance of a nearfield microphone array used for hands‐free speech acquisition. For a single source in a reverberant room, we propose a performance measure based on the expected reverberant power at the array output, and use the proposed measure to choose the microphone locations. One result of applying the proposed method is that it may be better to distribute the microphones throughout the room rather than placing them together in a single array (as conventional farfield array theory would dictate).

Relating change in signal-to-noise ratio to array gain for microphone arrays used in rooms

Microphone arrays have proven effective in improving speech intelligibility in noise for the hearing impaired. Two different acoustic performance metrics have been proposed for evaluating the performance of microphone arrays in reverberant rooms. The signal-to-noise ratio (SNR) metric uses the improvement in SNR at the microphone output relative to the microphone input, and includes the reverberant speech as part of the desired speech signal. The array-gain metric, on the other hand, ignores the reverberant speech and measures the degree to which the noise in the room has been suppressed. An approximate relationship between these two metrics is derived so that results using one metric can be compared to results using the other even when the signal and noise correlation matrices are not available.

A flexible experimental microphone array.

Sound pickup for hands-free communications often requires the microphone to be located at a large distance from the talker. However, in typical environments, the resulting speech signal is corrupted by reverberation and noise which degrade perceived voice quality and impair the operation of automatic speech recognizers. Microphone arrays have been shown to provide substantial improvements in predicted voice quality in computer simulations. To measure microphone array performance in more realistic conditions, a flexible, experimental microphone array has been developed based on digital signal processing technology. This array is flexible enough to permit the implementation of a wide variety of signal processing strategies and array geometries. In this paper the array design will be described and the performance of this array in the variable-acoustics listening room at the National Research Council will be discussed.