• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2014년도 추계학술대회 논문집
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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.

• 대한전자공학회논문지SP
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• 제48권4호
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• pp.160-168
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• 2011

주파수 영역 빔형성기 (frequency domain beamformer)는 시간 영역 빔형성기에 비해 연산량 측면에서 효율성이 높아 다수의 빔과 센서가 요구되는 수동 소나 시스템에서 널리 사용되고 있다. 주파수 영역 빔형성기에서는 푸리에 변환을 통한 입력 신호의 스펙트럼 분석 과정이 필수적이며, 이 과정의 효율성에 따라 전체 시스템의 성능이 결정된다. 또한 기본적으로 실시간 신호처리가 요구되는 실제 전장 환경에서 이러한 스펙트럼 연산 과정 최적화의 중요성이 더욱 부각되고 있다. 따라서 본 논문에서는 실시간으로 수신되는 협대역 신호의 효율적인 국부 스펙트럼 분석 (partial spectrum analysis)을 위해, TD (transform decomposition) 기법과 GSFFT (generalized sliding fast Fourier transform) 기법의 장점을 모두 적용한 pruned-GSFFT 기법을 제안한다. 또한 제안된 기법의 구현상의 특성을 분석하고 요구되는 연산량을 기존 기법들과 비교한다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2010년도 추계학술대회 논문집
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• pp.543-543
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• 2010

The objective of this research is estimating the location of interested sound source by using the similarity between a beamformer output in time domain and the candidate signal. The waveform of beamformer output at the location of sound source is similar with the waveform emitted by that source. To estimate the location of sound source by using this feature, we define quantified similarity between candidate signal and beamformer output. The candidate signal describes the signal which is generated by interested source. In this paper, similarity is defined by four methods. The two methods use time vector comparison, and the other two methods use time-frequency map or linear prediction coefficients. To figure out the results and performance of localization by using similarities, we demonstrate two conditions. The one is when two pure tone sources exist and the other condition is when several bird sounds exist. As a consequence, inner product with two time-vectors and structural similarity with spectrograms can estimate the locations of interest sound source.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(4)
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• pp.199-202
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• 2002

This paper introduces the near field shading beamformer using widely known Chebyshev and Hanning window in the field of digital signal processing. The proposed shading beamformer improves the estimation of range as well as azimuth angle of targe residing in near field. A series of sensor weighting values are calculated from the FFT operation of given shading functions in time domain. This paper verifies the performance of the focused beamformer having the proposed shading sensor weights which are used to detect the range of target. Throughout computer simulations this paper exploits the performance improvement of the proposed shading beamformer as varying the frequency band of the received radiated signal along the non-uniform array.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2008년도 하계종합학술대회
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• pp.83-84
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• 2008

It is efficient to use the time-domain beamforming to operate the various pulse with the different pulse length, frequency, bandwidth in active sonar system. In this paper, we propose a time-domain beamformer with the cumulative processing in the decomposed channel using the polynomial interpolation to solve the problem of the computational cost, high transmission data rate, and the lack of internal memory.

• International Journal of Internet, Broadcasting and Communication
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• 제11권2호
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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.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2009년도 춘계학술대회 논문집
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• pp.490-495
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• 2009

An approach to 3D visualization of multiple sound sources has been developed with the application of a moving array technique. Frequency-domain beamforming algorithm is used to generate a beam power map and the sound source is modeled as a point source. When a conventional delay and sum beamformer is used, it is considered that 2D distribution of sensors leads to have deficiency in spatial resolution along a measurement distance. The goal of moving an array in this study is to form 3D array aperture surrounding multiple sound sources so that the improved spatial resolution in a virtual space can be expected. Numerical simulation was made to examine source localization capabilities of various shapes of array. The 3D beam power maps of hemispherical and spherical distribution are found to have very sharp resolution. For experiments, two sound sources were placed in the middle of defined virtual space and arc-shaped line array was rotated around the sources. It is observed that spherical array show the most accurate determination of multiple sources' positions.

• 한국소음진동공학회논문집
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• 제19권9호
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• pp.907-914
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• 2009

An approach to 3D visualization of multiple sound sources has been developed with the application of a moving array technique. Frequency domain beamforming algorithm is used to generate a beam power map and the sound source is modeled as a point source. When a conventional delay and sum beamformer is used, it is considered that 2D distribution of sensors leads to have deficiency in spatial resolution along a measurement distance. The goal of moving an array in this study is to form 3D array aperture surrounding multiple sound sources so that the improved spatial resolution in a virtual space can be expected. Numerical simulation was made to examine source localization capabilities of various shapes of array. The 3D beam power maps of hemispherical and spherical distribution are found to have very sharp resolution. For experiments, several sound sources were placed in the middle of defined virtual space and arc-shaped line array was rotated around the sources. It is observed that spherical array shows the most accurate determination of multiple sources' positions.

• 한국소음진동공학회논문집
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• 제26권5호
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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.

• 한국전자통신학회논문지
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• 제12권3호
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• pp.433-438
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• 2017

신호의 도래각(AOAm Angle-of-Arrival) 추정 및 간섭제거 기술 등이 기반이 되는 적응 빔형성기 (Beamformer)는 레이더, 위성 등을 포함한 각종 첨단장비를 활용하여 다양한 정보를 수집하는 신호정보수집(: Signal Intelligence, SIGINT)의 핵심기술 중의 하나이다. 빔형성 기술은 안테나 어레이를 이용하여 특정 방향으로 부터의 신호를 효율적으로 수신하도록 해당 방향으로 지향성(directivity)을 가질 수 있게 빔을 생성하는 기술이다. 본 논문에서는 도래각 추정기법 및 간섭제거 기술 등이 탑재된 신호정보 수집 시스템의 입력으로 사용되는 간섭과 잡음이 포함된 수신신호 모델을 제시하고, 이 수신신호에 포함될 수 있는 다양한 신호들에 대한 특성을 고찰하고 분석한다. 제시된 신호 모델은 다양한 빔형성 기술에 대한 성능평가에 직접적으로 적용될 수 있다. 또한, 컴퓨터 시뮬레이션을 통해 제시된 수신신호 모델에 대한 주파수 영역에서의 스펙트럼을 확인한다.