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Damage identification of wind turbine blades using the microphone array under different parametric and measuring conditions: A prototype study with laboratory-scale models

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STRUCTURAL HEALTH MONITORING-AN INTERNATIONAL JOURNAL 2023年第1期22卷 201-215页

作者： Sun, Shilin Wang, Tianyang Yang, Hongxing Chu, Fulei Tsinghua Univ Dept Mech Engn Beijing 100084 Peoples R China Hong Kong Polytech Univ Res Inst Sustainable Urban Dev Renewable Energy Res Grp RERG Hong Kong Peoples R China

Structural health monitoring (SHM) of wind turbine blades is significant to the reliability and efficiency of wind energy generation, and it is a challenging issue due to the complicated structures and variational operating conditions. In this investigation, a SHM method for wind turbine blades based on the microphone array and acoustic source identification is proposed. With the equipment of loudspeakers in blade cavities, damage-related information is excited to be captured by the array. To generate accurate acoustic maps with high spatial resolutions, a novel algorithm for sparsity-based sound field reconstruction is developed based on the generalized minimax-concave penalty function. With a laboratory-scale wind turbine model, damage identification performance of the proposed method is evaluated under different parametric and measuring conditions, and experiments are conducted under diverse blade health conditions. Results reveal that and both internal and external damage in operating blades can be recognized as acoustic sources, and satisfactory performance of the proposed method can be guaranteed with appropriate parameters. Furthermore, determination criteria for parameters are concluded with respect to the variation of measuring conditions. This prototype study provides useful insights into the development of effective SHM systems.

关键词： structural health monitoring damage identification wind turbine blade microphone array acoustic source identification

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microphone array based trajectory reconstruction for small UAV fly-by measurements

Microphone array based trajectory reconstruction for small U...

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AIAA Aviation Forum

作者： Herold, Gert Sarradj, Ennes Tech Univ Berlin Dept Engn Acoust Einsteinufer 25 D-10587 Berlin Germany

ISBN: (数字)9781624107047

ISBN: (纸本)9781624107047

The accurate in-flight characterization of noise emissions from unmanned aerial vehicles requires knowledge of their flight path, i.e. the position and velocity at any given time. These parameters can be obtained from microphone array measurements with the help of suitable algorithms. However, the reconstructed path may deviate from the ground truth depending on the chosen array geometry, the relative position and emission characteristics of the object, and the selected evaluation algorithm parameters. In this contribution, these deviations and their extent depending on several parameters are investigated by employing an exemplary array measurement setup in an anechoic environment. Based on measurements from a small quadcopter drone, simulated data of a dipole point source with precisely known trajectory characteristics are generated and further processed to locate the moving source at each instant in time. The achievable accuracy of the calculated flight paths is quantified using appropriate error metrics derived from positional deviations of the reconstruction, and recommendations for similar test setups are given.

关键词： microphone array Unmanned Aerial Vehicle Quadcopter Frequency Domain Beamforming White Noise Acoustic Measurement Spline Interpolation Sound Propagation Cruise Flight Python

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microphone array POST-FILTER BASED ON AUDITORY FILTERING

MICROPHONE ARRAY POST-FILTER BASED ON AUDITORY FILTERING

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6th International Symposium on Chinese Spoken Language Processing

作者： Li, Peng Liao, Fengchai Cheng, Ning Xu, Bo Liu, Wenju Chinese Acad Sci Inst Automat Digital Content Tech Res Ctr Beijing 100864 Peoples R China Sanming Univ Dept Math & Comp Sci Sanming Fujian Peoples R China Chinese Acad Sci Inst Automat Natl Lab Pattern Recognit Beijing 100864 Peoples R China

ISBN: (纸本)9781424429424

In this paper, an auditory filtering based microphone array post-filter is proposed to enhance the quality of the output signal. By using a gammatone filterbank to band pass each input of the array, the input signals are decomposed into a two-dimensional T-F representation. Then, for each auditory filter channel, the post-filter's coefficients are estimated in each frame using the decomposed multi-channel input signals. Followed by the post-filtering and synthesis processing, the enhanced speech with better quality is acquired. Systematical evaluations on the CMU microphone array database prove that the proposed method could improve not only the noise reduction measure but also the speech quality measures.

关键词： microphone array speech enhancement auditory filter

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microphone array Post-filter based on Spatially-Correlated Noise Measurements for Distant Speech Recognition

Microphone Array Post-filter based on Spatially-Correlated N...

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13th Annual Conference of the International-Speech-Communication-Association

作者： Kumatani, Kenichi Raj, Bhiksha Singh, Rita McDonough, John Carnegie Mellon Univ Sch Comp Sci Pittsburgh PA 15213 USA

ISBN: (纸本)9781622767595

This paper presents a new microphone-array post-filtering algorithm for distant speech recognition (DSR). Conventionally, post-filtering methods assume static noise field models, and using this assumption, employ a Wiener filter mechanism for estimating the noise parameters. In contrast to this, we show how we can build the Wiener post-filter based on actual noise observations without any noise-field assumption. The algorithm is framed within a state-of-the-art beamforming technique, namely maximum negentropy (MN) beamforming with super directivity. We investigate the effectiveness of the proposed post-filter on DSR through experiments on noisy data collected in a car under different acoustic conditions. Experiments show that the new post-filtering mechanism is able to achieve up to 20% relative reduction of word error rates (WER) under the represented noise conditions, as compared to a single distant microphone. In contrast, super-directive (SD) beamforming followed by Zelinski post-filtering achieves a relative WER reduction of only up to 11%. Other post-filters evaluated perform similarly in comparison to the proposed post-filter.

关键词： microphone array Post-filter Distant speech recognition Automotive speech application

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microphone array for 2D sound localization and capture 1

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9th International Symposium on Experimental Robotics (ISER)

作者： Kagami, S Mizoguchi, H Tamai, Y Kanade, T Natl Inst Adv Sci & Technol Digital Human Res Ctr Tokyo Japan JST Jap Sci & Technol Corp CREST Program Tokyo Japan Tokyo Univ Sci Tokyo Japan Carnegie Mellon Univ Robot Inst Pittsburgh PA 15213 USA

ISBN: (数字)9783540330141

ISBN: (纸本)3540288163

This paper describes two circular microphone arrays and a square microphone array which can be used for sound localization and sound capture. Sound capture by microphone array is achieved by Sum and Delay Beam Former (SDBF). Simulation of sound pressure distribution of 32 & 128ch circular microphone array and 128ch square microphone array are shown. According to simulation results, dedicated PCI 128-channel simultaneous input board and Firewire (IEEE1394) 32-channel board are developed with maximum sampling rate of 44.1 kHz and 11.025 kHz sample respectively. Then a 32ch circular microphone array and a 128ch square microphone array have been developed. The 32ch circular microphone array can capture sound from an arbitrary direction. The 128ch square microphone array can capture sound from a specific point. Both systems are evaluated by using frequency components of the sound. The circular type system will be used on a mobile robot including humanoid robot, and square type will be extend towards room coverage type application.

关键词： microphone array

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microphone array beamforming approach to blind speech separation

Microphone array beamforming approach to blind speech separa...

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4th International Workshop on Machine Learning for Multimodal Interaction

作者： Himawan, Ivan McCowan, Iain Lincoln, Mike Queensland Univ Technol Brisbane Qld Australia CSIRO E Hlth Res Ctr Brisbane Qld Australia Ctr Speech Technol Res Edinburgh Midlothian Scotland

ISBN: (纸本)9783540781547

In this paper, we present a microphone array beamforming approach to blind speech separation. Unlike previous beamforming approaches, our system does not require a-priori knowledge of the microphone placement and speaker location, making the system directly comparable other blind source separation methods which require no prior knowledge of recording conditions. microphone location is automatically estimated using an assumed noise field model, and speaker locations are estimated using cross correlation based methods. The system is evaluated on the data provided for the PASCAL Speech Separation Challenge 2 (SSC2), achieving a word error rate of 58% on the evaluation set.

关键词： speech separation microphone array speech recognition

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MIRACLE-a microphone array impulse response dataset for acoustic learning

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EURASIP JOURNAL ON AUDIO SPEECH AND MUSIC PROCESSING 2024年第1期2024卷 1-16页

作者： Kujawski, Adam Pelling, Art J. R. Sarradj, Ennes TU Berlin Dept Engn Acoust Einsteinufer 25 D-10587 Berlin Germany

This work introduces a large dataset comprising impulse responses of spatially distributed sources within a plane parallel to a planar microphone array. The dataset, named MIRACLE, encompasses 856,128 single-channel impulse responses and includes four different measurement scenarios. Three measurement scenarios were conducted under anechoic conditions. The fourth scenario includes an additional specular reflection from a reflective panel. The source positions were obtained by uniformly discretizing a rectangular source plane parallel to the microphone for each scenario. The dataset contains three scenarios with a spatial resolution of 23 mm \documentclass[12pt]{minimal} \usepackage{amsmath} \usepackage{wasysym} \usepackage{amsfonts} \usepackage{amssymb} \usepackage{amsbsy} \usepackage{mathrsfs} \usepackage{upgreek} \setlength{\oddsidemargin}{-69pt} \begin{document}$$23\,\textrm{mm}$$\end{document} at two different source-plane-to-array distances, as well as a scenario with a resolution of 5 mm \documentclass[12pt]{minimal} \usepackage{amsmath} \usepackage{wasysym} \usepackage{amsfonts} \usepackage{amssymb} \usepackage{amsbsy} \usepackage{mathrsfs} \usepackage{upgreek} \setlength{\oddsidemargin}{-69pt} \begin{document}$$5\,\textrm{mm}$$\end{document} for the shorter distance. In contrast to existing room impulse response datasets, the accuracy of the provided source location labels is assessed and additional metadata, such as the directivity of the loudspeaker used for excitation, is provided. The MIRACLE dataset can be used as a benchmark for data-driven modelling and interpolation methods as well as for various acoustic machine learning tasks, such as source separation, localization, and characterization. Two timely applications of the dataset are presented in this work: the generation of microphone array data for data-driven source localization and characterization tasks and data-driven model order reduction.

关键词： Room impulse response Dataset microphone array Acoustics Machine learning

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Characterisation of the acoustic field radiated by a rail with a microphone array: The SWEAM method

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JOURNAL OF SOUND AND VIBRATION 2015年第1期346卷 165-190页

作者： Faure, Saldrik Chiello, Olivier Pallas, Marie-Agnes Serviere, Christine IFSTTAR AME LAE F-69675 Bron France Univ Lyon Ctr Lyonnais Acoust F-69622 Lyon France Gipsa Lab F-38402 St Martin Dheres France

Beamforming methods are widely used for the identification of acoustic sources on rail bound vehicles with microphone airays, although they have limitations in case of spatially extended sources such as the rail. In this paper, an alternative method dedicated to the acoustic field radiated by the rail is presented. The method is called SWEAM for Structural Wavenumbers Estimation with an array of microphones. The main idea is to replace the elementary fields commonly used in beamforming (point sources or plane waves) by specific fields related to point forces applied on the rail. The vertical bending vibration of the rail is modelled using a simple beam assumption so that the rail vibration depends only on two parameters: the wavenurnber and the decay rate of the propagative wave. Together with a radiation model based on a line of coherent monopoles, the acoustic field emitted by the rail is easily derived. The method itself consists in using the signals measured on a microphone array to estimate both the structural parameters and the global amplitude of this specific source. The estimation is achieved by minimising a least squares criterion based on the measured and modelled spectral matrices. Simulations are performed to evaluate the performance of the method considering one or several sources at fixed positions. The comparison of the simulated and reconstructed fields are convincing at most frequencies. The method is finally validated in the case of a single vertical excitation using an original set up composed of a 30 m long experimental track excited by an electrodynamic shaker. The results show a great improvement of the wavenumber estimation in the whole frequency range compared with the plane wave beamforming method and a fair estimation of the decay rate. The underestimation of some low decay rates due to the poor selectivity of the criterion occurring in these cases requires further study. (C) 2015 Elsevier Ltd. All rights reserved.

关键词： microphone array

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Locating and tracking sound sources on a horizontal axis wind turbine using a compact microphone array based on beamforming

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APPLIED ACOUSTICS 2019年 146卷 295-309页

作者： Zhang, Cui Qing Gao, Zhi Ying Chen, Yong Yan Dai, Yuan Jun Wang, Jian Wen Zhang, Li Ru Ma, Jian Long Inner Mongolia Univ Technol Sch Energy & Power Engn Hohhot 010051 Peoples R China Inner Mongolia Tech Coll Mech & Elect Dept Elect Engn Hohhot 010070 Peoples R China Inner Mongolia Univ Technol Minist Educ Key Lab Wind Energy & Solar Energy Technol Hohhot 010051 Peoples R China Xinjiang Inst Engn Urumqi 830091 Peoples R China

As the number and capacity of wind-power installations increases rapidly worldwide, the noise problems generated by wind turbines, which are increasingly being installed near residential areas, become increasingly prominent. Wind turbines can be approached as aerodynamic noise generation mechanisms for noise reduction purposes, for which a precisely localized sound source is key. Sound source recognition algorithms capable of locating the radial sound source position and relative azimuth angle of a rotating wind turbine blade in real time are rare. This paper proposes an algorithm based on classical beamforming theory with instantaneous speed and delayed azimuth angle rotation correction. First, the sound identification and tracking accuracies of the algorithm are verified using a rotating simulation bench;next, a small horizontal axis wind turbine is tracked and identified according to the given step size in the wind tunnel. The results show that the algorithm can precisely tracked and identified sound source of horizontal axis wind turbine in real time. Some errors were identified between the difference in two consecutive azimuth angles and a set step size of 60 degrees, caused by fluctuations in rotating speed and characteristic vibrations of the wind turbine. (C) 2018 Elsevier Ltd. All rights reserved.

关键词： Horizontal axis wind turbine microphone array Beamforming Sound source location Sound source track

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The delay & sum algorithm applied to microphone array measurements: Numerical analysis and experimental validation

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MECHANICAL SYSTEMS AND SIGNAL PROCESSING 2007年第6期21卷 2645-2664页

作者： Cigada, Alfredo Ripamonti, Francesco Vanali, Marcello Politecn Milan Dept Mech I-20156 Milan Italy

The experimental validation of the beamforming technique applied to microphone array measurements is investigated in this paper. At first the method theoretical background is presented, highlighting the main parameters affecting its performances and studying their influence with the aim of producing an instrument useful in designing an efficient experimental set-up. Then, in order to prove the theory assessment and to give a first estimation of the method uncertainty, an experimental campaign is carried out to validate the exposed theory under controlled and repeatable conditions. The obtained results highlight a very good correspondence between the theoretical model and the "in field" tests, giving the possibility to correctly design microphones array for future experimental campaigns. (c) 2007 Elsevier Ltd. All rights reserved.

关键词： microphone array beamforming experimental validation

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