• 대한전기학회:학술대회논문집
• /
• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
• /
• pp.689-692
• /
• 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.

• 한국전산구조공학회:학술대회논문집
• /
• 한국전산구조공학회 2000년도 봄 학술발표회논문집
• /
• pp.365-370
• /
• 2000

A sliding mode fuzzy control (SMFC) algorithm is applied to design a controller for a benchmark problem on a wind- excited building. The structure is a 76-story concrete office tower with a height of 306 meters, hence the wind resistance characteristics are very important for the serviceability as well as the safety. A control system with an active tuned mass damper is assumed to be installed on the top floor. Since the structural acceleration is measured only at ,limited number of locations without measurement of the wind force, the structure of the conventional continuous sliding mode control may have the feed-back loop only. So, an adaptive least mean squares (LMS) filter is employed in the SMFC algorithm to generate a fictitious feed-forward loop. The adaptive LMS filter is designed based on the information of the stochastic characteristics of the wind velocity along the structure. A numerical study is carried out. and the performance of the present SMFC with the ,adaptive LMS filter is investigated in comparison with those of' other control, of algorithms such as linear quadratic Gaussian control, frequency domain optimal control, quadratic stability control, continuous sliding mode control, and H/sub ∞///sub μ/, control, which were reported by other researchers. The effectiveness of the adaptive LMS filter is also examined. The results indicate that the present algorithm is very efficient .

• 한국음향학회지
• /
• 제24권5호
• /
• pp.238-247
• /
• 2005

본 논문에서는 디지털 보청기의 보상필터와 적응 궤환제거기의 부밴드 설계기법을 다룬다 감음 신경성 난청은 주파수 대역에서 비선형 특성을 보이는 가청 한계값 (hearing thresholds)을 가지며 시변하는 궤환경로에 의해 보상이 어렵다. 그러므로 디지틸 보청기는 주파수 대역별 비선형적 이득조정이 가능하며 반향을 빠르게 제거할 수 있는 보상기를 필요로 한다. 제안한 디지털 보청기에서 보상필터는 부밴드 구조에서 적응 시스템 식별기법을 이용하여 설계되며 적응 궤환제거기는 부밴드 인접투사 알고리즘을 이용하여 설계된다. 설계된 보상필터는 비선형 이득을 각 부밴드에서 조정할 수 있으므로 보다 정확한 이득보상이 가능하다. 그리고 부밴드 적응필터를 사용하는 궤환제거기는 빠른 수렴속도를 가진다. 설계된 보상필터는 주파수 대역별로 비선형적 이득보상이 가능하다. 그리고 적응 궤환제거기는 원치 않은 반향을 빠르게 제거할 수 있다. 제안한 보상필터의 성능을 컴퓨터 시뮬레이션을 통해 이전 방법의 성능과 비교하여 입증한다.

• 한국음향학회지
• /
• 제23권6호
• /
• pp.446-453
• /
• 2004

본 논문에서는 잡음에 의해 열화된 오디오 신호를 가변 임계값을 이용한 적응 지각 필터를 사용하여 음질을 개선하는 알고리즘을 제안한다. 제안된 적응 지각 필터는 신호 구간마다 달라지는 신호의 세기와 잡음의 영향 정도를 고려하여 임계값을 가변적으로 조정함으로써 잔여 잡음을 효과적으로 제어하는 방식으로 지각적으로 개선된 음질의 신호를 얻을 수 있다 제안한 방식은 잡음에 의해 열화된 오디오 신호를 주파수 영역으로 변환한 후 임계 대역 기반의 임계 대역 에너지 (Critical intensity energy)와 마스킹 영향이 고려된 청각 자극 에너지 (Excitation energy)를 계산한 다음, 지각 필터를 기반으로 한 적응 지각 필터 알고리즘으로 각 단계별 지각 필터 응답을 임계값으로 이용하여 가변 임계값이 재조정되는 단계를 결정하게 된다. 신호의 구간별 에너지 크기에 의한 잡음에 의해 열화된 정도의 차이를 가변 임계값을 이용하여 고려함으로써 잔여 잡음의 효과적인 제어가 가능하게 된다. 제안한 방법은 다양한 신호대 잡음비에서 열화된 오디오 신호를 입력으로 사용하였다. 입력 신호대 잡음비가 15dB, 20dB, 25dB와 30dB의 각각의 경우에 대하여 잡음대 마스킹비 (Noise-to-mask ratio, NMR)와 청감 테스트 (Mean opinion score, MOS Test)를 시행하였다. 그 결과, 잡음대 마스킹비의 개선 측면에서 각각의 경우에 대해 17.4dB, 15.3dB, 12.8dB, 9.8dB의 개선을 확인할 수 있었고, 청감 테스트의 개선 측면에서는 각각 2.9, 2.5, 2.3, 1.7의 개선된 음질을 확인할 수 있었다.

• 한국음향학회:학술대회논문집
• /
• 한국음향학회 1985년도 학술발표회 논문집
• /
• pp.87-94
• /
• 1985

In this paper, enhancement of speech corrupted by additive white or colored noise is stuided. The nuconstrained frequency-domain block least-mean-square (UFBLMS) adaptation algorithm and its frequency-weighted version are newly applied to speech enhancement. For enhancement of speech degraded by white noise, the performance of the UFBLMS algorithm is superior to the spectral subtraction method or Wiener filtering technique by more than 3 dB in segmented frequency-weighted signal-to-noise ratio(FWSNERSEG) when SNR of speech is in the range of 0 to 10 dB. As for enhancement of noisy speech corrupted by colored noise, the UFBLMS algorithm is superior to that of the spectral subtraction method by about 3 to 5 dB in FWSNRSEG. Also, it yields better performance by about 2 dB in FWSNR and FWSNRSEG than that of time-domain least-mean-square (TLMS) adaptive prediction filter(APF). In view of the computational complexity and performance improvement in speech quality and intelligibility, the frequency-weighted UFBLMS algorithm appears to yield the best performance among various algorithms in enhancing noisy speech corrupted by white or colored noise.

• IEIE Transactions on Smart Processing and Computing
• /
• 제1권3호
• /
• pp.143-151
• /
• 2012

A new data-driven method for the design of a blind modulation frequency filter that suppresses the slow-varying noise components is proposed. The proposed method is based on the temporal local decorrelation of the feature vector sequence, and is done on an utterance-by-utterance basis. Although the conventional modulation frequency filtering approaches the same form regardless of the task and environment conditions, the proposed method can provide an adaptive modulation frequency filter that outperforms conventional methods for each utterance. In addition, the method ultimately performs channel normalization in a feature domain with applications to log-spectral parameters. The performance was evaluated by speaker-independent isolated-word recognition experiments under additive noise environments. The proposed method achieved outstanding improvement for speech recognition in environments with significant noise and was also effective in a range of feature representations.

• 한국통신학회논문지
• /
• 제24권10A호
• /
• pp.1471-1479
• /
• 1999

이 논문에서는 적응 배열 안테나를 갖는 MC-CDMA를 위한 적응 빔 형성 알고리듬을 제안한다. 다중경로 페이딩에 robust하고 간단히 단일 탭 등화기를 사용하여 고속 데이터 전송에 적합한 것으로 알려진 MC-CDMA 시스템의 수신단에 안테나 배열을 사용함으로서 그 성능을 크게 개선할 수 있음을 보인다. 본 논문에서 제안된 MC-CDMA 시스템의 적응 빔 형성 알고리듬은 원하는 사용자의 파이롯 심벌(기준신호)과 수신된 파이롯 신호의 오차를 주파수 영역에서 계산하고, 그 주파수 영역 오차신호를 시간영역 오차 신호로 변환한 후, MSE가 최소가 되는 방향으로 적응 빔 형성기의 계수를 갱신하므로서 유도된다. 모의 실험과 해석적 방법을 통하여 MC-CDMA 시스템에 제안된 적응 빔 형성기법을 적용할 경우 수렴특성과 성능개선 효과를 확인한다.

• 천문학회지
• /
• 제52권2호
• /
• pp.49-55
• /
• 2019

The New Vacuum Solar Telescope (NVST) is the largest solar telescope in China. When using CCDs for imaging, equal-thickness fringes caused by thin-film interference can occur. Such fringes reduce the quality of NVST data but cannot be removed using standard flat fielding. In this paper, a correction method based on multi-scale decomposition and adaptive partitioning is proposed. The original image is decomposed into several sub-scales by multi-scale decomposition. The region containing fringes is found and divided by an adaptive partitioning method. The interference fringes are then filtered by a frequency-domain Gaussian filter on every partitioned image. Our analysis shows that this method can effectively remove the interference fringes from a solar image while preserving useful information.

• 한국안전학회지
• /
• 제7권2호
• /
• pp.71-79
• /
• 1992

This paper presents a least-square algorithms of lattice structures and their use for adaptive prediction of time series generated from the dynamic system. As the view point of adaptive prediction, a new method of Identification of dynamic characteristics by means of estimating the parameters of linear auto regressive model is proposed. The fast convergence of adaptive lattice algorithms is seen to be due to the orthogonalization and decoupling properties of the lattice. The superiority of the least-square lattice is verified by computer simulation, then predictor coefficients are computed from the linear sequential time data. For the application to the dynamic characteristic analysis of unknown system, the transfer function of ideal system represented in frquency domain and the estimated one obtained by predicted coefficients are compared. Using the proposed method, the damping ratio and the natural frequency of a dynamic structure subjected to random excitations can be estimated. It is expected that this method will be widely applicable to other technical dynamic problem in which estimation of damping ratio and fundamental vibration modes are required.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2003년도 하계종합학술대회 논문집 I
• /
• pp.234-237
• /
• 2003

The accuracy of channel estimation significantly affects the performance of coherent OFDM receiver. It is desirable to employ a good channel estimator while requiring low implementation complexity. In this paper, we propose a channel estimator that employs a simple two-dimensional (2-D) moving average (MA) filter as the channel estimation filter. The optimum tap size of the 2-D MA FIR filter is analytically designed in the time and frequency domain in association with the channel condition and pilot signal to interference power ratio. The analytic results can be applied to the design of adaptive channel estimator. Finally, the performance of the proposed channel estimator is verified by computer simulation.