• 한국해양공학회:학술대회논문집
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• 한국해양공학회 2004년도 학술대회지
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• pp.386-389
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• 2004

Beamforming for underwater acoustic communication is affected by the broadband feature of underwater acoustic communication signal, which has the low center frequency compared to the signal bandwidth. In this paper, the baseband equivalent array signal model is derived and we present computer simulation results for the broadband finite impulse response (FIR) beamformer performance according to the FIR filter order and the tap spacing. If the FIR filter order is increased above the optimum value, the beamformer performance is degraded. Also the tap spacing is related to the optimum FIR filter order.

• 한국정보통신학회논문지
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• 제10권12호
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• pp.2151-2156
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• 2006

수중음향통신을 위한 빔형성 기법은 대역폭이 반송주파수에 비해서 큰 광대역 신호 특성을 고려해야한다. 수중 음향통신에서는 협대역 신호가정이 성립하지 않는다. 본 논문에서는 기저 대역 배열신호 모델을 이용한 수중음향통신 광대역 FIR 빔형성기에 대해서 논한다. 반송주파수 25Hz, 심볼 속도 5kHz인 QPSK 방식의 수중음향통신에 있어서 광대역 FIR 빔형성기를 고려했다. 배열 센서는 8개의 등방형 센서로 구성된 선형등간격 구조이고, 센서간 간격은 반송주파수 파장의 절반이다 컴퓨터 모의실험을 통하여 각 센서에 길이가 2인 FIR 필터를 채택하고 탭간 간격이 심볼 주기의 1/4일 때 광대역 FIR 빔형성기는 최적 신호 대 간섭잡음비에 근접하였으며, 기존의 통상적인 협대역 빔형성기보다 신호 대 간섭 잡음비가 0.5dB 향상된 결과를 보였다. 광대역 FIR 빔형성기의 성능은 FIR 필터 길이가 특정 값 이상으로 커지면 더 나빠지고, 탭간 간격이 심볼 주기의 절반보다 작으면 탭간 간격은 성능에 영향을 주지 않았다. 탭간 간격 이 심볼 주기와 같은 경우에, 훈련 신호열이 통상적인 경우보다 더 많이 필요하였다.

• 한국음향학회지
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• 제28권6호
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• pp.520-525
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• 2009

적응 빔형성기에 대한 기존의 연구들은 간섭신호가 비교적 정상상태인 것을 가정하여 시간-평균을 취하거나 비교적 간단한 최소자승법 (Least Mean Squares)등을 이용하여 구현하고 있으나, 비정상 (Non-stationary) 상태의 간섭잡음의 제거 성능은 상대적으로 떨어진다. 따라서 본 논문에서는, 이러한 비정상 간섭 신호의 제거를 위해 MAFF-RLS 알고리즘을 도출하고, 이를 수정, 적용한 광대역 GSC의 구조를 제안하였다. 그리고 이를 시뮬레이션을 통하여 검증한 결과, 기존의 SMI, RLS 방법을 이용한 빔형성기보다 성능이 우수함을 확인하였다.

• 한국음향학회지
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• 제28권6호
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• pp.548-556
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• 2009

본 논문에서는 반향이 있는 환경에서 스피커 에레이를 이용하여 소리를 집중 시키는 기술을 제안한다. 제안 방법은 각 스피커에서 원하는 지점까지의 시스템 응답 함수의 에너지 합을 최대화함으로써 반향이 있는 환경에서도 원하는 지점에서의 음압을 최대화할수 있다. 기존에 제안된 광대역 빔 형성기 (Broadband beamformer)를 이용한 소리 집중 기술의 반향 환경 하에서의 문제점에 대하여 살펴본 후, 제안 방법이 갖는 장점을 서술한다. 16채널로 구성된 스피커 어레이 시스템을 이용한 실험 결과를 통해 반향이 있는 환경에서 제안 방법이 음압 집중도 측면에서 기존 방법보다 우수한 성능을 나타냄을 확인하였다.

• The Journal of the Acoustical Society of Korea
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• 제13권2E호
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• pp.14-19
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• 1994

본 논문에서는 벡터적 접근 방법을 이용하여 noncoherent 상황에서 선형 제한 조건을 갖는 광대역 빔 형성기의 출력 SINR식을 도출하였다. 여기서 입사되는 광대역 신호들은 평탄한 스펙트럼을 갖는다고 가정하였다.

• 대한의용생체공학회:의공학회지
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• 제32권3호
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• pp.198-206
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• 2011

Hearing devices like cochlear implant, vibrant soundbridge, etc. try to offer better sound for people. In hearing devices, several beamformers including conventional directional microphone are applicable to noise reduction. Each beamformer has different directional response and it could change sound intelligibility or quality for listeners. Therefore, we investigated the performance of three beamformers, which are first and second order directional microphone, and broadband beamformer(BBF) with a computer simulation assuming hearing device microphone configuration. We also calculated objective measurements which have been used to evaluate speech enhancement algorithms. In the simulation, a single speech and a single babble noisewere propagated from the front and $135^{\circ}$ azimuth degrees respectively. Microphones were configured in an end-fire array and the spacing was varied in comparison. With 3 cm spacing, BBF had about 3 dB higher enhanced SNR than that of directional microphones. However, enhancement of segmental SNR and frequency weighted segmental SNR were similar between the first order directional microphone and broadband beamformer. In addition when steady state noise was used, broadband beamformer showed the increased performance and had the highest enhanced SNR, and segmental SNR.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2004년도 학술대회지
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• pp.880-882
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• 2004

This paper presents the design and simulation results of 7 waveguide dielectric rod array antennas for 7 channel DBF(Digital beamformer) receivers for stratospheric system. A waveguide. dielectric antenna type, which has FTEP(Flat-toped element pattern) to manufacture easily. Also, the calculated element patterns according to the permittivity are compared. The designed antenna will be used for generating multibeam radiation pattern by means DBF.

• 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.

• 대한의용생체공학회:의공학회지
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• 제34권1호
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.