• 전자공학회논문지S
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• 제35S권7호
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• pp.46-52
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• 1998

This paper presents an experimenal proof for criteria of selecting an optimum adaptive beamforming (ABF) algorithm for a large distorted phased array. A single point target embedded in clutter model is suggested to compare four well-known ABF algorithms. These algorithms are tested to low variance and high variance real data for self-calibrating a large distored phased array. It is shown that these algorithms require at least one dominant scatterer with large radar cross section (RCS) or multiple scatterers with moderate RCS in the field of view. Experimental results are provided to demonstrate the comparisons of the four algorithems in terms of gain loss and image correlaion coefficient, along with corresponding reconstructed cross-range images and range-azimuth images.

• Journal of information and communication convergence engineering
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• 제3권3호
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• pp.119-121
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• 2005

This paper aims to estimate the delay time of multiple signals in a multi-path environment. It also seeks to carry out a comparative analysis with the existing delay time under the proposed algorithm to develop a new algorithm that applies the space average method in a MUSIC algorithm. Unlike the existing delay time estimation algorithm, the developed algorithm was able to estimate the delay time in 5ns low. Therefore, the algorithm proposed in this paper improved the existing delay time estimated algorithm.

• 한국전자통신학회논문지
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• 제13권6호
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• pp.1207-1212
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• 2018

신호정보 수집(: Signal Intelligence, SIGINT) 기술은 군수산업을 비롯한 여러 분야에서 다양한 데이터 수집을 목적으로 활발하게 사용되고 있다. 효율적으로 신호정보 및 데이터를 수집하고 수집된 데이터를 송/수신하기 위해서는 신호의 정확한 도래각 정보가 필요하고, 간섭 또는 재밍 신호로부터의 통신 방해가 최소화되어야 한다. 본 논문에서는 효율적으로 신호정보 및 데이터를 수집하고 송/수신하기 위한, 평면 배열 안테나 기반의 적응 빔형성 위성 시스템 구조를 소개한다. 제시된 적응 빔형성 시스템은 폄면 배열 형태의 안테나, MUSIC(: Multiple Signal Classification) 알고리즘 기반의 도래각 추정기, MVDR(: Minimum Variance Distortionless Response) 간섭 제거기, 신호처리 및 D/B 유닛, MMSE(: Minimum Mean Square Error) 기반의 송신 빔형성기 등으로 구성되어 있다. 또한, 컴퓨터 시뮬레이션을 통해, 제시된 시스템의 성능을 평가하고 분석한다.

• 한국통신학회논문지
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• 제32권4C호
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• pp.379-388
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• 2007

In this paper, we propose an adaptive turbo decoding algorithm for high order modulation scheme combined with originally design for a standard rate-1/2 turbo decoder for B/QPSK modulation. A transformation applied to the incoming I-channel and Q-channel symbols allows the use of an off-the-shelf B/QPSK turbo decoder without any modifications. Adaptive turbo decoder process the received symbols recursively to improve the performance. As the number of iterations increase, the execution time and power consumption also increase as well. The source of the latency and power consumption reduction is from the combination of the radix-4, dual-path processing, parallel decoding, and early-stop algorithms. We implemented the proposed scheme on a field-programmable gate array (FPGA) and compared its decoding speed with that of a conventional decoder. From the result of implementation, we confirm that the decoding speed of proposed adaptive decoding is faster than conventional scheme by 6.4 times.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1998년도 추계종합학술대회 논문집
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• pp.1191-1194
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• 1998

This paper presents a new computationally efficient adaptive algorithm for blind signal separation, which is able to recover the narrowband source signals in the presence of cochannel interference without a prior knowledge of array manifold. We derive a new blind signal separation algorithm using the Natural gradient 〔1〕from an information-theoretic approach. The resulting algorithm has the Bussgang property which has been widely used in blind equalization 〔12〕. Extensive computer simulation results comfirm the validity and high performance of the proposed algorithm.

• 디지털산업정보학회논문지
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• 제5권3호
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• pp.99-105
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• 2009

This paper suggests novel signal processing methods for optimal beamforming of smart antenna system in CDMA2000 mobile communication environments. This method utilize characteristics of the reverse pilot channel of CDMA2000 mobile communication systems, and applies them to improve the performance of an adaptive algorithm, which is used to a smart antenna system for beamforming. To perform the best beamforming, it is important to get an exact beamforming algorithm. This paper proposed an algorithm based on Laglange multiplier which has such an adaptive process, and also proposed the method to demodulate the received signal through array antenna using pilot channel in CDMA2000 environment. This paper analysed the enhanced performance of proposed algorithm in various signal environment through signal modeling of physical layer in CDMA2000 reverse link.

• 대한전자공학회논문지SP
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• 제45권3호
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• pp.46-54
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• 2008

최근 대부분의 보급형 CCD/CMOS 영상 센서는 크기와 비용을 줄이기 위해 한 가지 색상만 선택적으로 통과시키는 CFA(Color Filter Array)를 사용한다. 따라서 원래의 컬러 영상을 복원하기 위하여 패턴인식이나, 정규화 등을 이용한 많은 알고리즘이 제안되었으나, 지엽적인 색상오류, zipper 효과 등의 오류를 충분히 제거하지 못하고 있다. 본 논문에서는 전체 영상의 PSNR 뿐 아니라 주관적인 화질에 영향을 주는 에지 부분에서의 오류를 줄이기 위하여, 기존에 제시되었던 방법인 POCS(Projection Onto Convex Sets) 알고리즘을 기반으로 에지 가중치를 적응적으로 적용하였다. 그 결과 강한 에지 부분에서 보다 효율적인 컬러복원을 할 수 있었다.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2001년도 춘계종합학술대회
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• pp.749-752
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• 2001

최근 이동 통신 시스템에서 간섭 및 채널 왜곡, 잡음 둥에 의한 시스템의 성능 저하를 막고 통신 성능의 향상 및 시스템 용량 증가를 위해 사용자의 이동 상황에 파라 빔 추적 기능을 갖고 있는 스마트 안테나의 연구가 이루어져 왔다. 이에 본 논문에서는 실시간 처리를 위한 QR-RLS 기반 스마트 안테나 알고리즘을 설계하고, 이를 CPLD로 구현하였다. 구현된 알고리즘의 구조는 적응 필터링에 적합한 Systolic array 형태로 구성되어졌다. 연구된 방법은 컴퓨터 시뮬레이션과 아울러 Alters사의 Max+plus II를 사용하여 CPLD로 구현하였다.

• 전기학회논문지
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• 제58권10호
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• pp.1968-1975
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• 2009

The paper proposes a novel control algorithm for tracking maximum power of PV generation system.. The maximum power of PV array is determinated by a insolation and temperature. Prior considered the term in PV generation system is how maximum power point(MPP) is accurately tracked.. The paper proposes a fuzzy neural network(FNN) control algorithm so as to accurately track those maximum power points. The proposed control algorithm comprises the antecedence part of fuzzy rule and clustering method, multi-layer neural network in the consequent part. FNN has the advantages which are depicted both high performance and robustness in fuzzy control and high adaptive control in neural network.. Specially, it can show the outstanding control performance for parameter variations appling to non-linear character of PV array. In this paper, the tracking speed and the accuracy prove the validity through comparing a proposed algorithm with a conventional one.

• 전기학회논문지
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• 제58권4호
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• pp.857-866
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• 2009

In this paper, a noise cancellation-method using microphone array for digital hearing aids is proposed. The microphone array is located around the ear of a dummy. Speech sound is generated from the forward speaker positioned in the front of the dummy and noise sound is generated from the backward speaker. The speech and noise are mixed in the air space and entered into the microphones. VAD(voice activity detector) and ANC(adaptive noise cancellation) methods were used to eliminate noise in the sound of the microphones. 10 two-syllable words and 4 sentences were used for speech signals. Babble and car interior noise were used for noise signals. The performance of the proposed algorithm was evaluated by SNR(signal-to-noise ratio) and PESQ-MOS(perceptual evaluation of speech quality-mean opinion score). In babble noise condition, SNR was improved as much as $7.963{\pm}1.3620dB\;and\;3.968{\pm}0.6659dB$ for words and sentences respectively. In the case of car interior noise, SNR was improved as $10.512{\pm}2.0665dB\;and\;6.000{\pm}1.7642dB$ for words and sentences respectively. PESQ-MOS of the babble noise was improved as much as $0.1722{\pm}0.0861$ score for words and $0.083{\pm}0.0417$ score for sentences. And PESQ-MOS of the car interior noise was improved as $0.2661{\pm}0.0335$ score and $0.040{\pm}0.0201$ score for words and sentences respectively. It is verified that the proposed algorithm has a good performance in noise cancellation of microphone array for digital hearing aids.