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Moving sound source localization and tracking for an autonomous robot equipped with a self-rotating bi-microphone array

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JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA 2023年第2期154卷 1261-1273页

作者： Gala, Deepak Sun, Liang New Mexico State Univ Klipsch Sch Elect & Comp Engn Elect & Comp Engn Dept Las Cruces NM 88001 USA New Mexico State Univ Dept Mech & Aerosp Engn Las Cruces NM 88001 USA

In this paper, we present two approaches to localizing and tracking a sound source that moves in a three-dimensional (3D) space. The sound signal was captured by a unique bi-microphone system that rotates at a constant angular velocity. The motion of the sound source along with the rotation of the bi-microphone array produces a sinusoidal inter-channel distance difference (ICDD) signal with time-varying amplitude and phase. Four state-space models were developed and employed to design extended Kalman filters (EKFs) that identify instantaneous amplitude and phase of the ICDD signal. Both theoretical and numerical observability analyses of the four state-space models were performed to reveal singularities of the proposed EKFs in the domain of interest. We also developed a Hilbert-transform based method that localizes the sound source by comparing the true analytic ICDD signal to a virtual reference signal with zero elevation and azimuth angles. A moving average filter is then applied to reduce the noise and the effect of the artifacts at the beginning and the ending portions of the estimates. The effectiveness of the proposed methods was evaluated using comparison studies in simulation.

关键词： microphone array

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On microphone array beamforming and insights into the underlying signal models in the short-time-Fourier-transform domain

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JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA 2021年第1期149卷 660-672页

作者： Pan, Chao Chen, Jingdong Shi, Guangming Benesty, Jacob Northwestern Polytech Univ Ctr Intelligent Acoust & Immers Commun 127 Youyi West Rd Xian 710072 Shaanxi Peoples R China Xidian Univ Sch Artificial Intelligence Optoelect Imaging & Brain Inspired Percept Lab 2 Taibai South Rd Xian 710071 Shaanxi Peoples R China Univ Quebec Inst Natl Rech Sci Energie Mat & Telecommun 800 Gauchetiere Ouest Montreal PQ H5A 1K6 Canada

This paper studies signal models for microphone array beamforming in the short-time-Fourier-transform (STFT) domain with long acoustic impulse responses. The major contributions are as follows. First, the signal modeling problem is investigated in the STFT domain and a general decomposition is proposed for the convolved source signal. Second, new insights into the array manifold are presented: the STFT of the windowed acoustic impulse response from the source to the sensors. Third, the structure of the reference signal is analyzed: it can be viewed as the output of a beamformer without considering the noise in the observation signal. Fourth, based on the new perspectives and decomposition, a signal model is derived based on the use of the superdirective beamformer. Finally, three performance measures are defined, based on which three optimal/suboptimal signal models are derived and their performance is assessed under different acoustic environments and analysis window lengths. The performance of the well-known minimum variance distortionless response (MVDR) beamformer is evaluated, which justifies the properties of the developed signal models.

关键词： microphone array

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microphone array TECHNIQUES FOR TRAILING EDGE NOISE BASED ON CONVOLUTIONAL NEUTRAL NETWORK 32

MICROPHONE ARRAY TECHNIQUES FOR TRAILING EDGE NOISE BASED ON...

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32nd Congress of the International Council of the Aeronautical Sciences, ICAS 2021

作者： Song, Zhangchen Liu, Peiqing Guo, Hao Ministry of Industry and Information technology Beijing100191 China

ISBN: (纸本)9783932182914

Trailing edge noise is a major contribution to the airframe noise and has been studied in various processing methodologies such as wavelet analysis and velocity field comparison. To further understand the noise mechanism and improve the result quality in wind tunnel, microphone array technique has been applied to obtain the sound source distribution by different traditional processing algorithms such as beamforming algorithms and deconvolution algorithms. Rather than solving the sound transfer function, a convolutional neutral network (CNN) method has been proposed based on machine learning for complex objects, and proven efficient in limited number of incoherent monopole sources. However, trailing edge noise is an emission of distributed sound sources and most phased array methods including the CNN method assume the monopole sources. In this study, CNN method is investigated from monopole sources to simple line sources in order to find out an applicable condition for trailing edge noise. By using different definitions of accuracy and loss, the network is trained, tested, and compared with other researchers' results to validate on monopole sources. Then a new test strategy has been proposed and proven helpful in understanding the wrong prediction of the CNN model. The confidence level of the applied region and the method to use the prediction results are proposed to standardize and normalize the applicable range of the phased array. Then, a new training strategy is proposed by adding statistical parameters of the line sources as training data to get a direct source strength distribution of line sources. An experiment of the trailing edge noise of NACA0012 airfoil is conducted in Beihang D5 aeroacoutics wind tunnel to validate the methods. These results indicate that CNN method might be able to detect a more general feature of the sound source distributions, which might be helpful in constructing a network with a statistical output. © 2021 32nd Congress of the Intern

关键词： microphone array

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microphone array synthetic reconfiguration

Microphone array synthetic reconfiguration

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127th Audio Engineering Society Convention 2009

作者： Hur, Yoomi Abel, Jonathan S. Park, Young-Cheol Youn, Dae Hee CCRMA Music Department Stanford University Stanford CA 94305 United States DSP Lab. Dept. of Electrical and Electronic Eng. Yonsei University Seoul Korea Republic of Computer and Telecommunications Eng. Division Yonsei University Wonju Korea Republic of

ISBN: (纸本)9781615677122

This paper describes methods for processing signals recorded at a microphone array so as to estimate the signals that would have appeared at the elements of a different, collocated microphone array, i.e., " translating" measurements made at one microphone array to those hypothetically appearing at another array. Two approaches are proposed;a non-parametric method in which a fixed, low-sidelobe beamformer applied to the "source" array drives virtual sources rendered on the "target" array, and a parametric technique in which constrained beamformers are used to estimate source directions, with the sources extracted and rendered to the estimated directions. Finally, a hybrid method is proposed, which combines both approaches so that the extracted point sources and residual can be separately rendered. Experimental results using an array of 2mm-diameter microphones and human HRTFs are reported as a simple example.

关键词： microphone array

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A display-mounted high-quality stereo microphone array for high-definition videophone system

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 2008年第2期54卷 778-786页

作者： Hioka, Yusuke Okamoto, Manabu Kobayashi, Kazunori Haneda, Yoichi Kataoka, Akitoshi NTT Corp NTT Cyber Space Labs Tokyo 1808585 Japan

A stereo microphone array developed for a high-definition videophone system is presented. The array consists of a pair of fixed beamformers to collect sounds in stereo clearly while suppressing the far-end talk emitted from the loudspeakers to decrease the undesirable influence of acoustic echo. Both the objective and subjective experimental results demonstrate that the microphone array satisfies the required specifications, which have not been achieved by several conventional schemes.

关键词： microphone array stereo pick up filter-and-sum beamforming next-generation network speech localization

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A fast signal subspace approach for the determination of absolute levels from phased microphone array measurements

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JOURNAL OF SOUND AND VIBRATION 2010年第9期329卷 1553-1569页

作者： Sarradj, Ennes Brandenburg Tech Univ Cottbus Inst Verkehrstech D-03046 Cottbus Germany

Phased microphone arrays are used in a variety of applications for the estimation of acoustic source location and spectra. The popular conventional delay-and-sum beamforming methods used with such arrays suffer from inaccurate estimations of absolute source levels and in some cases also from low resolution Deconvolution. approaches such as DAMAS have better performance, but require high computational effort. A fast beamforming method is proposed that can be used in conjunction with a phased microphone array in applications with focus on the correct quantitative estimation of acoustic source spectra. This method bases on an eigenvalue decomposition of the cross spectral matrix of microphone signals and uses the eigenvalues from the signal subspace to estimate absolute source levels. The theoretical basis of the method is discussed together with an assessment of the quality of the estimation. Experimental tests using a loudspeaker setup and an airfoil trailing edge noise setup in an aeroacoustic wind tunnel show that the proposed method is robust and leads to reliable quantitative results. (C) 2009 Elsevier Ltd. All rights reserved.

关键词： microphone array

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A classification method for moving targets in the wild based on microphone array and linear sparse auto-encoder

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NEUROCOMPUTING 2017年 241卷 28-37页

作者： Guo, Feng Huang, Jingchang Zhang, Xin You, Xing Zu, Xingshui Zhao, Qin Ding, Yuanyuan Liu, Huawei Li, Baoqing Chinese Acad Sci Shanghai Inst Microsyst & Informat Technol Shanghai Peoples R China Univ Chinese Acad Sci Beijing Peoples R China IBM Res China Lab Beijing Peoples R China

Moving target classification is an important issue in wireless sensors. The wild environment makes it a difficult problem for the acoustic signals. In this paper, a new classification method for moving targets in the wild is proposed based on microphone array and linear sparse auto-encoder (LSAE). First, the acoustic signals of moving targets are enhanced by delay-and-sum (DS) beamformer in the narrowband way for the simplicity. The enhancing effects are given a detailed analysis. Then, a spatial feature named noise likelihood (NLH) is presented to further resist the interferences and noise widely existing in the wild. The NLH has a good ability to distinguish between the moving targets and noise. Moreover, to make full use of both the signals beamformed and the NLH, a classification network combining the LSAE layers to learn their representations by self-taught learning and the softmax layer for the classification is built. Experiments show that not only the representations learned by the LSAE layers are robust and much distinguishable but also the proposed method achieves a much better classification performance in comparison with the baseline classifiers for moving targets in the wild. (C) 2017 Elsevier B.V. All rights reserved.

关键词： microphone array Delay-and-sum Noise likelihood Sparse auto-encoder Wireless sensors

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Echolocation and flight strategy of Japanese house bats during natural foraging, revealed by a microphone array system

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JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA 2011年第2期129卷 1081-1088页

作者： Fujioka, Emyo Mantani, Shigeki Hiryu, Shizuko Riquimaroux, Hiroshi Watanabe, Yoshiaki Doshisha Univ Fac Life & Med Sci Bionav Res Ctr Kyotanabe 6100321 Japan Doshisha Univ Fac Engn Kyotanabe 6100321 Japan

Using only a microphone array system, echolocation pulses and three-dimensional flight paths in the frequency-modulated bat, Pipistrellus abramus, during natural foraging, were simultaneously examined. During the search phase, the inter-pulse interval, pulse duration, and moving distance of the bat between successive emissions were relatively constant at around 89.5 +/- 18.7 ms, 6.90 +/- 1.31 ms, and 0.50 +/- 0.20 m, respectively. The bats started to decrease these acoustical parameters within 2-3 m of the prey capture point. For every emission along a flight path, the distance between a bat and its prey capture point was calculated as both direct distance to capture (DDC), which corresponded to the target distance, and flight distance to capture (FDC) along the flight path. The DDC matched the FDC after the start of the approach phase, indicating that foraging bats followed a straight-ahead path to the target. In addition, the duration of the quasi-constant frequency component of emitted pulses was slightly extended just before the convergence of the DDC with the FDC. These findings suggest that the bats confirm the presence of target prey by extending the duration of the pulse and then select a straight-ahead approach by forecasting the movement of the prey. (C) 2011 Acoustical Society of America.[DOI: 10.1121/1.3523300]

关键词： microphone array

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Musical-Noise Analysis in Methods of Integrating microphone array and Spectral Subtraction Based on Higher-Order Statistics

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EURASIP JOURNAL ON ADVANCES IN SIGNAL PROCESSING 2010年第1期2010卷 1-25页

作者： Takahashi, Yu Saruwatari, Hiroshi Shikano, Kiyohiro Kondo, Kazunobu Nara Inst Sci & Technol Grad Sch Informat Sci Nara 6300192 Japan Yamaha Corp Ctr Adv Sound Technol SP Grp Shizuoka 4380192 Japan

We conduct an objective analysis on musical noise generated by two methods of integrating microphone array signal processing and spectral subtraction. To obtain better noise reduction, methods of integrating microphone array signal processing and nonlinear signal processing have been researched. However, nonlinear signal processing often generates musical noise. Since such musical noise causes discomfort to users, it is desirable that musical noise is mitigated. Moreover, it has been recently reported that higher-order statistics are strongly related to the amount of musical noise generated. This implies that it is possible to optimize the integration method from the viewpoint of not only noise reduction performance but also the amount of musical noise generated. Thus, we analyze the simplest methods of integration, that is, the delay-and-sum beamformer and spectral subtraction, and fully clarify the features of musical noise generated by each method. As a result, it is clarified that a specific structure of integration is preferable from the viewpoint of the amount of generated musical noise. The validity of the analysis is shown via a computer simulation and a subjective evaluation.

关键词： microphone array

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In-situ impedance and absorption coefficient measurements using a double-layer microphone array

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APPLIED ACOUSTICS 2019年 143卷 74-83页

作者： Hald, Jorgen Song, Wookeun Haddad, Karim Jeong, Cheol-Ho Richard, Antoine Bruel & Kjaer SVM AS Skodsborgvej 307 DK-2850 Naerum Denmark Tech Univ Denmark Acoust Technol DK-2800 Kongens Denmark

Acoustic impedance is typically measured using an impedance tube, which requires a material sample physically fitted to the tube. However, the impedance can vary greatly between the material mounted in the tube and the material located in a real environment, where the mounting conditions are likely to be different. Also, oblique incidence cannot be measured in an impedance tube. In this paper, we investigate the use of a double-layer microphone array for in-situ measurement of surface impedance and absorption coefficient. With the array positioned near the material surface, a source emits broad-band sound towards the array and the material. A measurement is taken, and the sound pressure and the surface-normal particle velocity at the material surface are calculated using Statistically Optimized Near-field Acoustical Holography (SONAH). From the surface pressure and velocity, the impedance across a selected area is calculated, and finally the absorption coefficient is calculated from the impedance. A set of tests has been performed on porous material samples in an anechoic chamber as well as in a fitted room. Different sample sizes and different sound incidence angles have been considered. The results show consistency between the measurements in the anechoic room and the ordinary room as well as good agreement with Miki's model up to large oblique incidence angles. (C) 2018 Elsevier Ltd. All rights reserved.

关键词： Absorption coefficient Surface acoustic impedance microphone array Near-field acoustic holography In-situ measurement

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