• Title/Summary/Keyword: Time Domain Beamforming

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Study on Be-Dopplerization Technique for Rotating Source Localization (마이크로폰 어레이를 이용한 회전하는 소음원 가시화에 관한 연구)

  • Park, Sung;Lee, Ja-Hyung;Choi, Jong-Soo;Kim, Jai-Moo;Rhee, Wook
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2005.11a
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    • pp.200-204
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    • 2005
  • The use of beamforming method and de-Dopplerization technique was applied in studying the rotating sound sources. Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. Two main issues of the signal reconstruction in time domain are addressed herein: First, to remove Doppler effect from the measured data and to restore the original emission data of the moving source. The difference of the time domain beamforming from the frequency domain beamforming was mentioned. Also, the time domain beamforming method is deployed in the test and the comparisons were made to the frequency domain results. The time domain signal reconstruction was numerically simulated prior to the application. To validate the de-Dopplerization Performance, the rotating Point sources were examined and localized by the use of a phased array of microphone. The application of prop-rotor was conducted in a hovering condition. The results of reconstructing time signals of rotating sources and its locations were shown in the power distribution maps. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies of interest.

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Time-Domain Model of Surface Clutter for Airborne Phase-Array Radar (항공기 위상 배열 레이더에서 시간 영역의 지상클러터 생성 모델)

  • Kim, Tae-Hyung;Kim, Eun-Hee;Kim, Seon-Joo
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.24 no.7
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    • pp.730-736
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    • 2013
  • Time-domain clutter generation model for airborne pulse doppler phase-array radar is presented. Time-domain surface clutter signal is generated assuming earth of a sphere and considering geometry of a clutter patch, and generation of sub-array clutter signal is presented. The generated sub-array clutter signal can be used by simulation input signal in various radar applications of DBF(Digital Beamforming), ABF(Adaptive Beamforming), Stap(Space-Time Adaptive Processing) and etc.

Impulsive sound localization using crest factor of the time-domain beamformer output (빔형성기 출력의 파고율을 이용한 충격음의 방향 추정)

  • Seo, Dae-Hoon;Choi, Jung-Woo;Kim, Yang-Hann
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2014.10a
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    • pp.713-717
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    • 2014
  • This paper presents a beamforming technique for locating impulsive sound source. The conventional frequency-domain beamformer is advantageous for localizing noise sources for a certain frequency band of concern, but the existence of many frequency components in the wide-band spectrum of impulsive noise makes the beamforming image less clear. In contrast to a frequency-domain beamformer, it has been reported that a time-domain beamformer can be better suited for transient signals. Although both frequency- and time-domain beamformers produce the same result for the beamforming power, which is defined as the RMS value of its output, we can use alternative directional estimators such as the peak value and crest factor to enhance the performance of a time-domain beamformer. In this study, the performance of three different directional estimators, the peak, crest factor and RMS output values, are investigated and compared with the incoherent interfering noise embedded in multiple microphone signals. The proposed formula is verified via experiments in an anechoic chamber using a uniformly spaced linear array. The results show that the peak estimation of beamformer output determines the location with better spatial resolution and a lower side lobe level than crest factor and RMS estimation in noise free condition, but it is possible to accurately estimate the direction of the impulsive sound source using crest factor estimation in noisy environment with stationary interfering noise.

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Localization of Moving Sound Source Using Various Beamforming Methods (이동하는 소음원 위치 추정을 위한 다양한 빔형성 기법 적용)

  • Go, Yeong-Ju;Lee, Jaehyung;Choi, Jong-Soo;Ha, Jae-Hyoun
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.26 no.5
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    • pp.501-510
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    • 2016
  • Capabilities of several beamforming techniques are compared for estimating the position of a moving source. Beamforming has enabled to widen our perspective of aeroacoustics in wind tunnel experiments and has provided useful approach in array measurements. Meanwhile beamforming techniques have been developed in a way to improve estimation accuracy and to save ing effort at the same time. In order to achieve reasonable outcome from aeroacoustic measurement, it is important to identify the spectral characteristics of source and to select an appropriate beamformer. Though aeroacoustic sources normally generates broadband noises, many array signal processing have been focused on narrowband processing which makes calculation numerically efficient. However, calculation in frequency-domain requires selection of single frequency of interest which affects spatial resolution and sidelobe level as a consequence. To be able to localize broadband noise source, it is proposed to use broadband beamforming. The formulas implements the deletion of diagonal term from cross spectral matrix. In this study, trajectory of flying source emitting broadband noise was simulated and several beamformers are applied.

Acoustic Noise Measurement for the Wind Turbine Blade by Using Time-domain Beamforming (시간영역 빔포밍을 사용한 풍력터빈 축소모델 소음원 측정)

  • Cho, Tae-Hwan;Kim, Cheol-Wan
    • New & Renewable Energy
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    • v.5 no.2
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    • pp.25-30
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    • 2009
  • The wind tunnel test to identify the acoustic noise source position of the wind turbine blade was conducted in KARI low speed wind tunnel. Microphone array and time-domain beamforming methodology was applied to this study. To reduce the data processing time, a modified beamforming method with a criteria between calculation time step and grid size for rotating angle in the cylinderical coordinate system was proposed. The test results shows that the data processing time to identify the noise source position was reduced to 20% compared with conventional method. And the dominant noise source of the blade moves from inboard to blade tip as the frequency increases.

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Performance Improvement of MIMO MC-CDMA system with multibeamforming

  • Kim, Chan Kyu
    • International Journal of Internet, Broadcasting and Communication
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    • v.11 no.2
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    • pp.76-83
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    • 2019
  • In this paper, we propose the beamforming algorithm for the performance improvement of MIMO MC-CDMA system. The proposed multibeamforming of MIMO MC-CDMA structure having the same number of beamformer as the number of transmit antenna is derived by calculating the error signals between the coded pilot symbols and the corresponding received signals from the multiple transmitters of the desired user in the frequency domain, transforming the frequency-domain error signals into time-domain error signals, and updating the weights of the multibeamformer in the time-domain in the direction minimizing the mean squared error (MSE). The proposed approach can track each direction of arrival (DOA) of the signals from multi-antennas of a desired user. The performance improvement is investigated through computer simulation by applying the proposed approach to MIMO MC-CDMA system in a multipath fading channel with multiusers.

Block LMS-Based Adaptive Beamforming Algorithm for Smart Antenna (스마트 안테나를 위한 블록 LMS 기반 적응형 빔형성 알고리즘)

  • O, Jeong-Geun;Kim, Seong-Hun;Yu, Gwan-Ho
    • Proceedings of the KIEE Conference
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    • 2003.11c
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    • pp.689-692
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    • 2003
  • In this paper, we propose an adaptive beamforming algorithm for array antenna. The proposed beamforming algorithm, based on Block LMS (Block - Least Mean Squares) algorithm, has a variable step size from coefficient update. This method shows some advantages that the convergence speed is fast and the calculation time can reduced using a block LMS algorithm from frequency domain. As the adaptive parameter approaches a stationary state, it could reduce the number of filter coefficient update with the help of various step size. In this paper we compared the efficiency of the proposed algorithm with a standard LMS algorithm which is a representative method of adaptive beamforming.

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Study for Visualization of Rotating Sound Source Using Microphone Array (마이크로폰 어레이를 이용한 회전하는 소음원 가시화에 관한 연구)

  • Rhee, Wook;Park, Sung;Lee, Ja-Hyung;Kim, Jai-Moo;Choi, Jong-Soo
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.16 no.6 s.111
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    • pp.565-573
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    • 2006
  • Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. The purpose of this research is development of beamforming technique can be applied to the rotor noise source identification. For the do-Dopplerization and reconstruction of emitted sound wave, Forward Propagation Method is applied to the time domain beamforming technique. And validation test were performed using rotating sound source constructed by bended pipe and horn driver. In the validation test using sinusoidal sound wave, sufficient performance of signal processing can be seen, and the effect of measuring duration for accuracy was compared. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies and collective pitch angle, in hover condition.

Reception performance improvement of VSB in multipath channel using switched beamforming (선택 빔형성을 적용한 다중경로 환경 VSB 수신 성능 개선)

  • 배재휘;서재현;김주연;김승원
    • Proceedings of the IEEK Conference
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    • 2003.07a
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    • pp.35-38
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    • 2003
  • We propose a switched beamforming to improve the reception performance of VSB system in severe Rayleigh fading channel. The VSB system has only about 3% of known training signal for the receiver in a data field and the reception performance of VSB receiver is degraded significantly when there are near-0 dB ghosts in receiving signal. The switched beamforming forms several beams in different directions and selects only one beam among them. For the selection of a beam with best channel condition for VSB equalizer, we extracted the channel profiles in time domain for all the beams by correlating the PN511 sequence in VSB field sync and selected optimal beam by comparing the channel profiles. The simulation results show that the proposed method improves the reception performance of VSB system in severe Rayleigh channel.

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A method to find the position of fault in a moving vehicle using microphone arrays (마이크로폰 어레이를 이용하여 차량 하부에서 발생한 결함의 위치를 찾아내는 방법)

  • Kim, Yang-Hann;Jeon, Jong-Hoon
    • Proceedings of the KSR Conference
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    • 2006.11b
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    • pp.144-151
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    • 2006
  • Sound generated from a moving vehicle often carries information on the condition of vehicle, for example, whether it has faults or not, where the fault exists. The latter is possible especially by MFAH(moving frame acoustic holography) and beamforming method. MFAH is applicable to the sound source of pure tone or narrow band noise. For the beamforming method, we have to know what kind of wave the sound source radiates, for example, plane wave or spherical wave. That is, whether the above methods are applicable depends on the characteristics of sound source. To apply these methods to the fault detection, we have to know the characteristics of wave from faults. In this research, a machine diagnosis technique based on the above holographic approaches is introduced to find the position of faults. The signal due to faults is modeled based on the fact that the faults radiate impulsive noise, and analyzed in time and frequency domain. The way how MFAH and beamforming method can be used is introduced to find the position of source.

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