Mitigating wind noise with a spherical microphone array.

The area of sound field synthesis has significantly advanced in the past decade, facilitated by the development of high-quality sound-field capturing and re-synthesis systems. Spherical microphone arrays are among the most recently developed systems for sound field capturing, enabling processing and analysis of three-dimensional sound fields in the spherical harmonics domain. In spite of these developments, a clear relation between sound fields recorded by spherical microphone arrays and their perception with a re-synthesis system has not yet been established, although some relation to scalar measures of spatial perception was recently presented. This paper presents an experimental study of spatial sound perception with the use of a spherical microphone array for sound recording and headphone-based binaural sound synthesis. Sound field analysis and processing is performed in the spherical harmonics domain with the use of head-related transfer functions and simulated enclosed sound fields. The effect of several factors, such as spherical harmonics order, frequency bandwidth, and spatial sampling, are investigated by applying the repertory grid technique to the results of the experiment, forming a clearer relation between sound-field capture with a spherical microphone array and its perception using binaural synthesis regarding space, frequency, and additional artifacts. The experimental study clearly shows that a source will be perceived more spatially sharp and more externalized when represented by a binaural stimuli reconstructed with a higher spherical harmonics order. This effect is apparent from low spherical harmonics orders. Spatial aliasing, as a result of sound field capturing with a finite number of microphones, introduces unpleasant artifacts which increased with the degree of aliasing error.

This paper concerns the problem of convolutive sound source separation from mutlichannel recordings made with a spherical microphone array. In particular, we formulate two state-of-the-art separation techniques based on Expectation Maximization (EM) and Nonnegative Tensor Factorization (NTF) in the spherical harmonic domain (SHD). Furthermore, we adjust and incorporate the Gaussian Localization Prior (GLP) to the proposed algorithms, which yields two variants of the derived methods. For the source signal reconstruction, a Minimum Variance Distortionless Response (MVDR) beamformer with a single-channel Wiener post-filter is employed. The performance comparison is based on experimental evaluation using micro-phone signals simulated with the image-source method in several scenarios, including diverse geometrical setup, different number of sources and various types of source signals, namely the recordings of speech utterances and musical instruments. The experimental results for the first-order ambisonic signals show that the proposed methods enable high-quality sound source separation in the spherical harmonic domain. In particular, we show that incorporation of the Gaussian Localization Prior to the proposed algorithms leads to a substantial improvement in separation performance.

Fast direction-of-arrival estimation algorithm for multiple wideband acoustic sources using multiple open spherical arrays

Source localization has been studied in the spatial domain using differential geometry in earlier work. However, parameters of the sensor array manifold have hitherto not been investigated for source localization in spherical harmonics domain. The objective of this work is to represent and model the manifold surface using differential geometry. The system model for source localization over a spherical harmonic manifold is first formulated. Subsequently, the manifold parameters are modeled in the spherical harmonics domain. Source localization methods using MUSIC and MVDR over the spherical harmonics manifold are developed. Experiments on source localization using a spherical microphone array indicate high resolution in noise.

Estimation of the direction of arrival (DOA) of reflections in a room using spherical microphone arrays can be performed using several methods such as plane-wave decomposition, and delay-and-sum beamforming, for which the resolution of the estimation process is usually bounded by the array order, or the number of microphones. In this paper a method for DOA estimation of the reflections in a room using room impulse-response data and spherical microphone arrays is presented, in which only a first order microphone array is required. The proposed method uses signal processing in the spherical harmonics domain, utilizing an analytical dependence of the spherical harmonic coefficients on different acoustics parameters. It is shown that the zero order spherical harmonics coefficients is a function of the time delays and amplitudes of the planes waves composing the sound field, while the first order coefficients are a function of the time delays, amplitudes and the elevation and azimuth components of the wave arrival direction. Using this property leads to the development of an estimator based on the least-squares approach, in which each order is used to estimate a single sound-field parameter.

Spherical array plays important role in 3D targets localization. In this paper, we develop a novel DOA estimation method for the spherical array with super resolution approach. The proposed method operates in spherical harmonic domain. Based on the atomic norm minimization, we develop a gridless L1-SVD algorithm in spherical harmonic domain and then we adopt the spherical ESPRIT method to two-dimensional DOA estimation. Compared to the previous work, the proposed method acquires better estimation performance. Numerical simulation results verify the performance of the proposed method.

Radial filter design for processing near field speech sources over a spherical microphone array is challenging. Polynomial based radial filter design procedures have been proposed in earlier work. In this paper we address the issue of radial filter design using a family of orthogonal polynomials called the Gegenbauer polynomials. The radial filters designed using this approach indicate an improved radial response and greater efficiency in attenuating distant sources. Improved white noise gain and directivity index are also noted from experimental evaluations. The radial filters hence designed are used to separate directionally co-incident near field speech sources. Subjective evaluation is conducted on separated sources using measures like LSD, PESQ, and SDR. The subjective evaluation scores are motivating enough to be considered for practical speech and audio applications.

High-order spherical microphone arrays offer many practical benefits including relatively fine spatial resolution in all directions and rotation invariant processing using eigenbeams. Spatial filtering can reduce interference from noise and reverberation but in even moderately reverberant environments the beam pattern fails to suppress reverberation to a level adequate for typical applications. In this paper we investigate the feasibility of applying dereverberation by considering multiple beamformer outputs as channels to be dereverberated. In one realisation we process directly in the spherical harmonic domain where the beampatterns are mutually orthogonal. In a second realisation, which is not limited to spherical microphone arrays, beams are pointed in the direction of dominant reflections. Simulations demonstrate that in both cases reverberation is significantly reduced and, in the best case, clarity index is improved by 15 dB.

Directional room impulse responses (DRIR) measured with spherical microphone arrays (SMA) enable the reproduction of room reverberation effects on three-dimensional surround-sound systems (e.g., Higher-Order Ambisonics) through multichannel convolution. However, such measurements inevitably contain a nondecaying noise floor that may produce an audible “infinite reverberation effect” upon convolution. If the late reverberation tail can be considered a diffuse field before reaching the noise floor, the latter may be removed and replaced with an extension of the exponentially-decaying tail synthesized as a zero-mean Gaussian noise. This has previously been shown to preserve the diffuse-field properties of the late reverberation tail when performed in the spherical harmonic domain (SHD). In this paper, we show that in the case of highly anisotropic yet incoherent late fields, the spatial symmetry of the spherical harmonics is not conducive to preserving the energy distribution of the reverberation tail. To remedy this, we propose denoising in an optimized spatial domain obtained by plane-wave decomposition (PWD), and demonstrate that this method equally preserves the incoherence of the late reverberation field.

Vehicle detection is one of the fundamental tasks in the ITS (intelligent transport system). We are developing an acoustic vehicle detector that relies on two microphones at a sidewalk [1]. The vehicle detector successfully detected vehicles as well as their traveling directions. However, the detector has difficulties in vehicle detection in a high wind condition due to wind noise. This paper presents a wind noise suppressor for the acoustic vehicle detector. Our simple idea is to remove frequency components corresponding to wind noise. Our acoustic vehicle detector relies on TDOA (time difference of arrival) of sound signals on two microphones to detect vehicles, which can be derived from a part of frequency components of vehicle sound signals. We experimentally analyze frequency components of wind noise and design a filter to reduce wind noise. Initial experimental evaluations revel that our vehicle detector with a wind noise suppressor successfully detected vehicles with an F-measure of 0.77 in a normal wind condition.

It has been designed a sound pressure level detection for response of tympanic membrane based on optical fiber with extrinsic method. The fiber optic sensor is used to measure the sound pressure level by utilizing the output voltage change. The sound level pressure values obtained are displayed on OLED display. Testing tools with sound level meter comparison tool to test sound pressure level in poly ENT patients. The test results show that at sound pressure level measurement with frequency 200 Hz to 300 Hz, the average accuracy is 93.42% for right ear, at the same time for left ear the average accuracy is 94.58%. The test of compliance measurement the average accuracy is 77.97% for the right ear, at the same time for the left ear the average accuracy is 70.74%. The testing of the appliance to patient in Andalas University hospital poly ENT were detect the values of sound pressure level 60 dB to 65 dB.Â

The reliable estimation of fiber orientation distributions from diffusion-sensitized magnetic resonance imaging is an important processing step for revealing structural connectivity in the human brain. Fiber orientation distributions are usually represented in spherical harmonic (SH) domain, but many modern spatial regularization techniques were not yet formulated in SH-domain such that they cannot benefit from the well known advantages. In this article we propose a novel SH-formulation of the contour enhancement propagator (fiber continuity) which enables to implement a variety of processing steps in SH-domain.

Direction-of-arrival (DOA) estimation is an important step in acoustic scene analysis. Multiple signal classification in the eigenbeam domain (EB-MUSIC) is an accurate direction-of-arrival (DOA) estimation method for rigid spherical microphone arrays. Two important issues with this method are 1) the requirement of prior information about the number of coherent source components, and 2) its computational cost. In this paper, a computationally efficient two-stage method, which can alleviate these problems, is proposed. The proposed method is robust to reverberation, and can be used to efficiently localize multiple coherent sources. An evaluation of the method using simulated recordings under highly reverberant conditions is also presented.

A gridless sound field decomposition method based on the reciprocity gap functional (RGF) is proposed. An intuitive and powerful way of reconstructing a sound field inside a region including sound sources is to decompose the sound field into Green's functions. Current methods based on sparse representation require discretization of the reconstruction region into grid points to construct the dictionary matrix; however, this procedure causes an off-grid problem and has a high computational cost. We apply the RGF, which was first proposed in the field of inverse problems, to sound field decomposition in the spherical harmonic domain. The proposed method enables a sound field to be decomposed in a gridless manner with a computationally efficient algorithm. Numerical simulation results indicated that the reconstruction accuracy as well as the source localization accuracy can be improved by the proposed method compared with current methods, especially at low frequencies.

Excessive environmental noise and poor air quality can be adverse to human health, living comfort and the environment itself. Measurement of sound pressure levels and air quality in critical areas including libraries, campus areas, public parks and hospitals thus becomes necessary to monitor and mitigate existing noise levels. In a university environment, student activities will be less disrupted if the locations of the activities are sufficiently away from noise sources. The present study is intended to measure sound levels and air quality around the Engineering Complex, Universiti Teknologi Mara, Shah Alam. The measured data is compared with to acceptable sound pressure levels and air quality index specified by the Department of Environment (DOE), Malaysia. Sound pressure levels are measured using the Castle Sound Level Meter Type 6224 and air quality measurement was done by using the BW Gas Alert MicroClip XT device. Both measurements were conducted at five selected stations around the Engineering Complex for three times each weekday for five weeks. Results obtained indicated that sound levels at some locations and time zones are above the thresholds recommended by the DOE. The air quality is acceptable in most locations except the vicinity of a bus stop. With the growing number of students in the university and other factors like construction and redevelopment of existing roads, a continuously increasing noise situations and air pollution proportional to the traffic flow is inevitable.