Microphone Array Technique Research Articles

SLAM-based Online Calibration for Asynchronous Microphone Array

This paper addresses online calibration of an asynchronous microphone array. Although microphone array techniques are effective for sound localization and separation, these techniques have two issues; geometry information on a microphone array or time-consuming measurements of transfer functions between a microphone array and a sound source is necessary, and a fully synchronous multichannel analog-to-digital converter should be used. To solve these issues, we proposed an online framework for microphone array calibration by combining simultaneous localization and mapping (SLAM), and beamforming. SLAM simultaneously calibrates locations of microphones and a sound source, and clocks differences between microphones every time a microphone array observes a sound event. Beamforming works as a cost function to decide the convergence of calibration by localizing the sound using the transfer functions calculated from the estimated microphone locations and clock differences. We implemented a prototype system based on the proposed framework using extended Kalman filter-based SLAM and delay-and-sum beamforming. The experimental results showed that the proposed framework successfully calibrated an eight-channel asynchronous microphone array both in a simulated and a real environment even when system parameters such as variances are set to be 10 times larger than the optimal values. Furthermore, the error of sound localization with the calibrated microphone array was as small as the desired one, that is, the grid size for beamforming.

A Normalized Beamforming Algorithm for Broadband Speech Using a Continuous Interleaved Sampling Strategy

The delay beamforming method, one of the most effective microphone array techniques, requires minimal computational complexity and enjoys wide usage in narrowband signal applications. However, the method suffers from high losses of low-frequency sound and is generally not suitable for broadband signals. In this paper, a novel normalized beamforming algorithm is introduced for broadband speech applications in multichannel cochlear implant (CI), which can be adjusted based on application conditions. The proposed algorithm can, with minimal computational complexity, accurately and effectively compensate for changes in signal beam pattern, in accordance with a preselected benchmark. In addition, this method can be combined with the widely used continuous interleaved sampling (CIS) strategy, adding to its practical value.

Transmission loss measurements using a scanned array nearfield acoustical holography system in an anechoic/reverberation facility. Part II. Comparison of transmission loss methods.

Transmission loss is an important metric used to assist in the selection of materials for noise control and design. In order to assess and improve the performance of transmission loss measurements, two different techniques were compared: traditional microphone and/or matched pair microphone intensity anechoic chamber‐side techniques, and nearfield acoustical holography techniques. Intensity and sound power measurements using traditional methods were made on a test panel installed in an anechoic‐reverb facility transmission loss window, then repeated using a conformal nearfield microphone array and nearfield acoustical holography processing techniques. The full vectored intensity components were used to assess the radiation characteristics of the test panel. This assessment quantifies the acoustics as panel radiation, flanking path contamination, and boundary effects. Data are presented to qualify the best estimate for transmission loss by rejecting contamination. Also, the dynamic range of the transmission loss measurement is compared between traditional and nearfield acoustical holography techniques with high transmission loss panel samples.

Transmission loss measurements using a scanned array nearfield acoustical holography technique in an anechoic/reverberation facility. Part I. Measurement methodology.

Transmission loss is an important metric when used to assist in the selection of materials for noise control and design. In order to assess and improve the performance of transmission loss measurements, two different techniques were compared: traditional microphone and/or matched pair microphone intensity anechoic chamber chamber‐side techniques, and nearfield acoustical holography (NAH) techniques. Intensity and sound power measurements using traditional methods were made on a test panel installed in an anechoic‐reverb facility transmission loss window, then repeated using a conformal nearfield microphone array and nearfield acoustical holography processing techniques. In order to process the data with the relative phase between microphones preserved, a set of stationary reference signals was used. A number of spectral preprocessing techniques and reference combinations are presented. The nearfield acoustical holography processing methodology is also presented with an emphasis on data quality and error reduction.

Implementation issues of the nearfield equivalent source imaging microphone array

This paper revisits a nearfield microphone array technique termed nearfield equivalent source imaging (NESI) proposed previously. In particular, various issues concerning the implementation of the NESI algorithm are examined. The NESI can be implemented in both the time domain and the frequency domain. Acoustical variables including sound pressure, particle velocity, active intensity and sound power are calculated by using multichannel inverse filters. Issues concerning sensor deployment are also investigated for the nearfield array. The uniform array outperformed a random array previously optimized for far-field imaging, which contradicts the conventional wisdom in far-field arrays. For applications in which only a patch array with scarce sensors is available, a virtual microphone approach is employed to ameliorate edge effects using extrapolation and to improve imaging resolution using interpolation. To enhance the processing efficiency of the time-domain NESI, an eigensystem realization algorithm (ERA) is developed. Several filtering methods are compared in terms of computational complexity. Significant saving on computations can be achieved using ERA and the frequency-domain NESI, as compared to the traditional method. The NESI technique was also experimentally validated using practical sources including a 125 cc scooter and a wooden box model with a loudspeaker fitted inside. The NESI technique proved effective in identifying broadband and non-stationary sources produced by the sources.

Speech enhancement using an equivalent source inverse filtering-based microphone array

This paper presents a microphone array technique aimed at enhancing speech quality in a reverberant environment. This technique is based on the central idea of single-input-multiple-output equivalent source inverse filtering (SIMO-ESIF). The inverse filters required by the time-domain processing in the technique serve two purposes: de-reverberation and noise reduction. The proposed approach could be useful in telecommunication applications such as automotive hands-free systems, where noise-corrupted speech signal generally needs to be enhanced. SIMO-ESIF can be further enhanced against uncertainties and perturbations by including an adaptive generalized side-lobe canceller. The system is implemented and validated experimentally in a car. As indicated by numerous performance measures, the proposed system proved effective in reducing noise in human speech without significantly compromising the speech quality. In addition, listening tests were conducted to assess the subjective performance of the proposed system, with results processed by using the analysis of variance and a post hoc Fisher's least significant difference (LSD) test to assess the pairwise difference between the noise reduction (NR) algorithms.

Source identification and reduction of engine noise

Engine acoustic behavior is inherently complicated because it depends on many design and manufacturing factors. Therefore, identifying engine noise sources is a challenging task. For an engine model, a further challenge is to define engine acoustic refinement targets for the noise level for the car models equipped with the engine of interest. Based on the automaker's feedback of the vehicle acoustic assessments, a significant engine-induced interior noise occurred at 400 to 410 Hz. This paper addresses the critical engine noise problem�from source identification to countermeasure validation�to resolve the noise at 400 to 410 Hz. To identify the dominant engine noise sources, conventional sound pressure measurements and advanced microphone array technique were performed in a hemi-anechoic chamber. Spectrum and order analyses were used to identify the air cleaner and the oil pan as the major sources at frequencies from 400 to 410 Hz. Furthermore, a microphone array system visually confirmed secondary noise sources as the alternator and oil pump. A resonator for the air cleaner and enhanced structural ribs for the oil pan were used to successfully reduce the engine noise. In turn, the related vehicle interior booming noise was also abated

Blind location and separation of callers in a natural chorus using a microphone array

Male frogs and toads call in dense choruses to attract females. Determining the vocal interactions and spatial distribution of the callers is important for understanding acoustic communication in such assemblies. It has so far proved difficult to simultaneously locate and recover the vocalizations of individual callers. Here a microphone-array technique is developed for blindly locating callers using arrival-time delays at the microphones, estimating their steering-vectors, and recovering the calls with a frequency-domain adaptive beamformer. The technique exploits the time-frequency sparseness of the signal space to recover sources even when there are more sources than sensors. The method is tested with data collected from a natural chorus of Gulf Coast toads (Bufo valliceps) and Northern cricket frogs (Acris crepitans). A spatial map of locations accurate to within a few centimeters is constructed, and the individual call waveforms are recovered for nine individual animals within a 9 x 9 m(2). These methods work well in low reverberation when there are no reflectors other than the ground. They will require modifications to incorporate multi-path propagation, particularly for the estimation of time-delays.

Real‐time guitar radiation sound synthesis of forced string and body eigenfrequency vibrations using microphone array techniques.

A guitar top plate vibration is investigated for different frequency regions taking string driving point dependencies into consideration using a microphone array of 128 microphones and back‐propagating the recording to the top plate positions. These radiation patterns are integrated in space at arbitrary virtual microphone positions. A guitar tone or piece is then synthesized using these radiation patterns where the microphone positions can be changed as in a real recording situation. The synthesis process takes the pressure and phase relations of more than 100 positions on the top plate and the sound hole into consideration for two cases of waves, the traveling waves of forced top plate vibration of the string frequencies and the eigenfrequencies of the guitar body appearing mostly during the initial transient phase. These eigenfrequencies are integrated using the much more simple mode shapes of the top plate eigenmodes. The amplitude behavior of these frequencies, appearing and decaying during the initial transient, is calculated by integrating the energy which is supplied by the string acting with a force on the top plate as an impulse train.

Microphone array techniques using cross-correlations.

Civilian noise complaints and damage claims have created a need to establish a detailed record of impulse noise generated at military training facilities. Wind noise is causing false positive impulse detections in the current noise monitoring systems. Multiple channel data methods were investigated in order to distinguish the characteristics of noise events. A microphone array was used to collect four simultaneous channels of military impulse and wind noise data. Cross‐correlation functions were then used to characterize the input waveforms. Three different analyses of microphone array data were developed. A new value, the min peak correlation coefficient, is defined from the minimum value of peaks of the cross‐correlation coefficient functions among the different channels. This value is a measure of the likelihood that a given waveform originated from a correlated noise source. The angle of incidence of the noise source is calculated using a sound source localization technique based on the geometry of th...

Tire radiation in vehicle environment: A review of some source identification methods

The REBECA research project aims at defining new architectural concepts and reduction strategies of vehicle external noise emission regarding the international pass‐by noise certification standard (ISO 362). The first task of the project is to qualify and quantify the major contributor of the external emission, the tires/road surface source. To this end, a 3D array composed of 164 microphones has been set up in the near and far field around vehicle. Measurements have been performed for several operational conditions with the use of a chassis dyno in a semi‐anechoic chamber. A review and a comparison of some source identification methods has been led including experimental FRFs based on inverse techniques (Nelson), conventional and optimized beamforming approaches (Elias, Dougherty, Brooks and Humphreys, Ravetta) and inverse BEM method (Hamdi, Omrani). This study permits to identify the more appropriate microphone array technique with respect to source identification robustness, power level and directivity radiation reconstruction in the far field.

The use of directional and binaural microphones for troubleshooting room acoustic problems

A common problem encountered when troubleshooting room acoustic issues (apart from the room exhibiting an overly long reverberation time or inadequate sound insulation) is the generation of late arriving reflections and echoes. The paper describes a number of techniques using both directional and binaural microphones as well as microphone array techniques for determining the direction, timing, and relative strength of such reflections. It is shown that in many situations, apparently highly directional microphones such as hypercardioid or rifle types, do not offer sufficient directional discrimination to be of much practical use. It is also shown that the inconsistent off axis frequency response of many such microphones can also lead to misleading analyses. Parabolic dish reflector microphones are shown to offer superior performance as do some microphone array techniques. It is shown that binaural measurements can also be a useful approach particularly with regard to assessing the subjective affect of reflections. The results of some early research into a novel extension to the Speech Transmission Index (STI) for assisting with acoustic trouble shooting will also be presented.

Experimental methods for multi-diagnostics of flow fields in wind tunnels

Advanced optical and acoustical measurement techniques as developed during the last decade allow new insight into complex unsteady flow fields and sound fields in aerodynamics. Parallel application of such experimental diagnostics methods allows a more complete description of complex flow phenomena and provides high quality data for validation of numerical calculations. In particular, three examples for multi-diagnostics of the flow field above a delta wing and within the slat of a high lift configuration by means of PSP and PIV, PIV and PDMS, and by means of PIV and the Acoustic Microphone Array Technique will be given during the presentation.

画像処理 会話シーンにおけるロボットと複数ユーザとの共同注意の形成

The ability to interact with multiple users as well as to recognize the ways in which people are interacting is essential for a robot participating in society. Forming shared attention is regarded as fundamental in improving the flexibility of human-robot interaction. This paper proposes an algorithm that enables shared attention between a robot and speakers in alternative conversational situations. We firstly present a microphone array technique and a method to combine auditory and visual information for estimating the physical location of the sound source. Secondly, the speaker is detected according to the linear combination of the result of sound source localization and the likelihood map representing the distribution of pixels having skin color. Finally, in the conversational situations, the speaker's focus of attention is alternatively detected and shared by the robot for upcoming communication based on the proposed algorithm. Several experimental results are presented that demonstrate the effectiveness of the proposed method.

Real-Time Automated Video and Audio Capture with Multiple Cameras and Microphones

This work presents the acoustic and visual-based tracking system functioning at the Harvard Intelligent Multi-Media Environments Laboratory (HIMMEL). The environment is populated with a number of microphones and steerable video cameras. Acoustic source localization, video-based face tracking and pose estimation, and multi-channel speech enhancement methods are applied in combination to detect and track individuals in a practical environment while also providing an improved audio signal to accompany the video stream. The video portion of the system tracks talkers by utilizing source motion, contour geometry, color data, and simple facial features. Decisions involving which camera to use are based on an estimate of the head's gazing angle. This head pose estimation is achieved using a very general head model which employs hairline features and a learned network classification procedure. Finally, a beamforming and postfiltering microphone array technique is used to create an enhanced speech waveform to accompany the recorded video signal. The system presented in this paper is robust to both visual clutter (e.g. ovals in the scene of interest which are not faces) and audible noise (e.g. reverberations and background noise).

A broadband microphone array technique for correlated interference rejection

A new method of broadband minimum variance beamforming is presented for use with microphone arrays in a reverberant environment. Conventional broadband algorithms (such as the well-known Frost algorithm) suffer from cancellation of the desired look-direction signal when employed in environments containing interference which is correlated with the desired signal. In this paper, a frequency invariant beamforming structure is used to perform spectral averaging. This reduces the correlation between the desired signal and the interference, thereby eliminating the signal cancellation problem. Such a technique is valuable for speech acquisition using a microphone array, since significant correlated interference will be present due to room reverberation.

Microphone arrays for reducing reverberation and noise in speech communication

Microphone pickup of sound in typical rooms is impaired by the combined effects of reverberation and noise. This degrades speech intelligibility and quality particularly in applications where the microphone is located far away from the talker. Recent advances in microphone array technology suggest a potential solution to such problems. This paper gives an overview of current microphone array techniques and discusses the potential benefits for speech communication. Various criteria for measuring the performance of a microphone array are described. A flexible, experimental microphone array intended for research in speech communications is under construction and will be described.

Adaptive Microphone Array Technique for Remote Monitoring of Components in Nuclear Power Plants

Adaptive Microphone Array Technique for Remote Monitoring of Components in Nuclear Power Plants

Adaptive Microphone Array Technique for Remote Monitoring of Components in Nuclear Power Plants.

Acoustic sensing technique provides an efficient measure to detect and diagnose incipient failures occurred in plant components. The study reported here investigates the feasibility of adaptive super-directive microphone array technique for detection and localization of anomaly in nuclear power plants. The technique extracts anomaly information of objective components from the acoustic signals obtained at geometrically arranged multiple microphones. A specific signal processing was used to obtain a super-directive sensitivity of the microphone array and to adapt the sensitivity following the acoustic environmental change. Two adaptive sensing algorithms have been developed and implemented in a personal computer, and their abilities to extract the objective acoustic signal have been tested through numerical simulation. For appropriate number of the microphones, the satisfactory performance of the sound extraction has been obtained within reasonable computational load. The practical applicability of this technique to component monitoring in nuclear power plants has been confirmed through the present study.