• Journal of Korea Multimedia Society
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• v.6 no.1
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• pp.114-120
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• 2003

Blind channel estimation of communication channels is a problem of important current theoretical concerns. Recently proposed solutions for this problem exploit the diversity induced by antenna array or time oversampling, leading to the so-called, second order statistics techniques. This paper proposes the blind adaptive channel estimation using multichannel linear prediction method. Computer simulations are presented to compare the proposed algorithm with the existing ones.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.8 no.2
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• pp.100-107
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• 2009

Velocity estimation is one of important issues for efficient system management in mobile cellular systems. In this paper, a modified velocity estimation scheme which works in Adaptive Antenna System (AAS) is proposed. The proposed scheme estimates user velocity based on moving distance information and sojourn time information. From numerical results, it is shown that the proposed scheme can estimate user velocity accurately with low cost.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.9B
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• pp.1215-1225
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• 2001

통신 채널에서 블라인드 채널 인식은 매우 중요한 문제이다. 블라인드 채널 인식은 고차 통계를 이용하면 구할 수 있으나 최근에는 오버샘플링한 수신신호를 이용하거나 수신측의 안테나 어레이를 이용한 신호의 2차 통계값을 이용한 방법에 관한 많은 연구가 진행되고 있다. 기존의 알고리듬은 잡음이 없는 환경에서 LS 방법에 기반을 두고 있기 때문에 잡음이 강한 채널에서는 원하는 성능을 얻을 수 없는 단점이 있다. 수신신호의 상관행렬의 최소 고유값에 대응하는 고유벡터는 채널의 임펄스 응답에 관한 정보를 포함하고 있다. 본 논문에서는 이러한 고유벡터를 매 시간마다 갱신시키면서 구하는 적응 알고리듬을 제안하고 이를 이용하여 블라인드 채널 인식 알고리듬을 제안한다. 제안한 알고리듬은 잡음에 강인한 특성을 보일 뿐만 아니라 기존의 알고리듬들 보다 우수한 채널 추정 성능을 보임을 모의실험을 통하여 검증하였다.

• Proceedings of the Korean Information Science Society Conference
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• 1999.10c
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• pp.42-44
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• 1999

주문형 비디오(Video On Demand) 서버를 구현하는데, 현재까지도 많은 부분에서 한계에 부딪치고 있으며, 이를 해결하기 위한 연구들이 활발히 진행되고 있다. 주문형 비디오 서비스에서의 한계는 대부분이 자원(디스크, 버퍼)의 제한 때문에 발생하고 있으며, 디스크어레이나, 메모리 뱅크와 같은 특수한 하드웨어를 사용하여 해결하고 있다. 또한, 이에 맞추어 여러 가지 소프트웨어적인 방법도 제안되었다. 본 논문에서는 주문형 비디오 서버의 한계 중 디스크대역폭과 버퍼 공간의 한계점에 의한 문제를 해결하기 위하여 기존의 연구분야인 버퍼 공유 및 관리 방법을 제안한다. 제안하는 버퍼 공유 및 관리 방법은 버퍼의 크기 및 디스크의 대역폭 가용량에 따른 적응적 병합, 분할 방식을 사용한다. 제안한는 병합, 분할방식은 피기백킹(piggybacking) 개념을 도입한 방법을 사용한다. 이러한 적응적 버퍼관리의 효율성을 보기 위해 주워진 환경을 설정한 시뮬레이션을 통해 제안방법의 성능을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.38 no.2
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• pp.145-153
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• 2019

Adaptive beamforming methods are known to suppress sidelobes and improve detection performance of weak signal by constructing weight vectors depending on the received signal itself. A standard adaptive beamforming like the MVDR (Minimum Variance Distortionless Response) is very sensitive to mismatches between weight vectors and actual signal steering vectors. Also, a large computational complexity for estimating a stable covariance matrix is required when wideband beamforming for a large-scale array is used. In this paper, we exploit the WBRCB (Wideband Robust Capon Beamforming) method for stable and robust wideband adaptive beamforming of a passive large uniform line array sonar. To improve robustness of adaptive beamforming performance in the presence of mismatches, we extract a optimum mismatch parameter. WBRCB with extracted mismatch parameter shows performance improvement in beamforming using synthetic and experimental passive sonar signals.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.10a
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• pp.347-350
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• 2007

Adaptive arrays have been proposed as an effective means of mitigating Multiple Access Interference (MAI) and improving the performance of existing and future wireless communication systems. In this paper, we apply the analytical method proposed to analyse the theoretical Mean Bit Error Rate (BER) of an uncoded IS-95 based CDMA system with an array antenna at the BS in a Ricean fading environment. We present a modified expression for the Signal to Interference plus Noise (SINR) ratio as a function of the number of users, number of antennas and noise levels. We also verify the analytical results by means of simulations by considering different user and channel scenarios.

• Journal of Advanced Navigation Technology
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• v.24 no.5
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• pp.393-407
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• 2020

In order to reduce the threat by illegal unmanned aerial vehicles, a tracking system based on sound was implemented. There are three main points to the drone acoustic tracking method. First, it scans the space through variable beam formation to find a sound source and records the sound using a microphone array. Second, it classifies it into a hidden Markov model (HMM) to find out whether the sound source exists or not, and finally, the sound source is In the case of a drone, a sound source recorded and stored as a tracking reference signal based on an adaptive beam pattern is used. The simulation was performed in both the ideal condition without background noise and interference sound and the non-ideal condition with background noise and interference sound, and evaluated the tracking performance of illegal drones. The drone tracking system designed the criteria for determining the presence or absence of a drone according to the improvement of the search distance performance according to the microphone array performance and the degree of sound pattern matching, and reflected in the design of the speech reading circuit.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.8 no.2
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• pp.116-125
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• 1997

This paper presents a novel method, entitled Orthogonal Weights Searching(OWS), for the Direction-Of-Arrival(DOA) estimation. Utilizing the modified Conjugate Gradient Method(MCGM), the weight vector which is orthogonal to the signal subspace is directly computed from the signal matrix. The proposed method does not require the computation of the eigenvalues and eigenvectors. In addition, the new technique excludes the procedure for the detection of the number of signals under the assumption that the number of weights in the array is greater than the number of input signals. Since the proposed technique can be performed independently of the detection procedure, it shows a good performance in adverse signal environments in which the detection of the number of array inputs cannot be obtained successfully. The performance of the proposed technique is compared with that of the convectional eigen-decomposition method in terms of angle resolution for a given signal-to-noise ratio(SNR) and a required amount of computations.

• Journal of the Korean Society for Nondestructive Testing
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• v.32 no.5
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• pp.551-563
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• 2012

Medical ultrasound systems have been used since 1950s, and are now widely used in most hospitals as indispensable diagnostic imaging systems. Since array probe was introduced in 1970s, beamforming technology using electronic signal processing has been adopted to the medical ultrasound system, and has been improved. Beamforming is a important technology which defines the resolution of the ultrasound system. In this paper, the technologies are introduced from basic beamforming principles to current trend. They include principles of beamforming using array probe, basic theory, and practical implementation, and recent topics of synthetic aperture imaging, adaptive beamforming, 2-dimensional beamforming using 2-dimensional array are also introduced. These various technologies will improve system performances continuously by merging innovatively with various technologies in other fields.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9A
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• pp.1013-1021
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• 2004

In the WCDMA mobile multimedia communication system, the adaptive array antenna IS adopted to improve the performance of the system by reducing inter-user interference using antenna beam control. Usually, the interference resulted from the higher data rate users is much more significant to the lower data rate users than the other way around, so the overall performance can be enhanced by reducing the interference from higher data rate users. In order to maximize the efficiency of adaptive antenna operation, an optimal call admission control, especially during handoff, adaptive to the data rates is a critical problem. In this paper, We propose a call admission control algorithm based on the Soft QoS concept for the efficient processing of the handoff of higher data rate calls, and an adaptive handoff control mechanism according to the data rates. The proposed algorithm has been evaluated by computer simulation that it accommodates high data rate users among many lower data rate users much better, and the average call blocking probability for lower rate users becomes much lower than the conventional call admission control algorithm.