• 한국정밀공학회:학술대회논문집
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• 한국정밀공학회 2001년도 춘계학술대회 논문집
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• pp.235-238
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• 2001

Deformation phase can be obtained by using Least-Square Fitting. In extraction of phase values, Least-Square Fitting is superior to usual method like as 2, 3, 4-Bucket Algorithm. That can extract almost noise-free phase and retain 2$\pi$discontinuities. But more fringe in phase map, 2$\pi$ discontinuities is destroyed when that is filtered and reconstruction of deformation is not reliable. So, we adapted Least-Square Fitting using an isotropic window in dense fringe. using Sine-Cosine filter give us perfect 2$\pi$discontinuities information. We showed the process and result of extraction of phase map and filtering in this paper.

• 한국정밀공학회지
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• 제19권12호
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• pp.20-26
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• 2002

Deformation phase can be obtained by using Least-Square fitting. In extraction of phase values, Least-Square Fitting is superior to usual method such as 2, 3, 4-Bucket Algorithm. That can extract almost noise-free phase and retain 2 $\pi$ discontinuities. But more fringes in phase map, 2 $\pi$ discontinuities are destroyed when that are filtered and reconstruction of deformation is not reliable. So, we adapted Least-Square fitting using an isotropic window in dense fringe. Using Sine/cosine filter give us perfect 2 $\pi$ discontinuities information. We showed the process and result of extraction of phase map and filtering in this paper.

• 한국산업정보학회논문지
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• 제4권3호
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• pp.79-88
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• 1999

적응 신호처리 분야에서 LMS(Least Mean Square)알고리즘은 그 식의 간편함과 구현의 용이함으로 가장 널리 이용되고 있다. 대부분의 LMS 알고리즘은 수렴비를 조절하는 적응계수를 일정한 값으로 정하는데, 이는 안전성과 속도사이에서 트레이드오프가 존재한다. 이러한 단점을 해결하고 성능을 개선하기 위하여 가변 LMS(VLMS: Variable LMS)가 발표되었다. 그러나 기존에 발표된 알고리즘들은 급격한 환경변화에 적응하지 못하고 발산하는 경우도 있으며 수렴속도에 문제가 있다. 본 논문에서는 기존의 적응계수의 특성을 일부 변형시킴으로서 계산량을 줄이고, 급격한 환경변화에서도 수렴하도록 하는 새로운 알고리즘을 제안하였다. 제안한 적응계수의 성능 확인을 위하여 시스템 식별 및 잡음 제거 시스템에 적용하여 기존의 알고리즘들과 비교하였다.

• 한국정보전자통신기술학회논문지
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• 제8권1호
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• pp.45-50
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• 2015

본 논문은 능동 소음 제어를 위한 적응 알고리즘들의 성능을 비교함으로써 효과적인 적응 알고리즘을 보인다. 일반적인 적응 알고리즘으로는 normalized least mean square (NLMS) 알고리즘이 있다. NLMS는 구조가 간단하고 수렴 속도가 빠르다는 장점이 있어서 널리 사용되고 있다. 하지만 상관도가 높은 신호에 대해서는 수렴 성능이 떨어지는 문제가 발생한다. 이에 수렴 성능을 개선하기 위해 affine projection (AP) 알고리즘을 사용하고 있다. 하지만 연산량의 문제로 AP 알고리즘의 사용이 제한적이다. 이러한 사실을 바탕으로 협대역 소음 제어를 위한 능동 소음 제어 시스템에서 NLMS와 AP 알고리즘을 연산량과 수렴 성능을 비교함으로써 효과적인 알고리즘을 도출하였다. 실험을 통해 NLMS와 AP 알고리즘의 소음 제어 성능이 차이가 크게 발생하지 않는 것을 확인함으로써 NLMS가 AP 알고리즘에 비해 소음 제어에 효과적임을 확인하였다.

• Journal of Power Electronics
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• 제15권6호
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• pp.1619-1627
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• 2015

This paper presents a power system harmonic elimination using the mixed adaptive linear neural network and variable step-size leaky least mean square (ADALINE-VSSLLMS) control algorithm based active power filter (APF). The weight vector of ADALINE along with the variable step-size parameter and leakage coefficient of the VSSLLMS algorithm are automatically adjusted to eliminate harmonics from the distorted load current. For all iteration, the VSSLLMS algorithm selects a new rate of convergence for searching and runs the computations. The adopted shunt-hybrid APF (SHAPF) consists of an APF and a series of 7^th tuned passive filter connected to each phase. The performance of the proposed ADALINE-VSSLLMS control algorithm employed for SHAPF is analyzed through a simulation in a MATLAB/Simulink environment. Experimental results of a real-time prototype validate the efficacy of the proposed control algorithm.

• 한국산업정보학회논문지
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• 제7권2호
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• pp.68-73
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• 2002

빠른 수렴속도를 얻기 위해서 LS(Least Square)에 기초한 적응 알고리즘에 대한 연구가 많이 이루어지고 있다. 본 논문에서는 수렴속도의 향상 그리고 계산량의 감소를 위하여 웨이브렛 기반 적응알고리즘을 제안하고, 음성신호의 특성에 따라서 두 가지 구조의 형태로 적응잡음 제거기에 적용시켰다. 컴퓨터 시뮬레이션을 통하여 기존의 시간영역 적응알고리즘, 주파수영역 적응알고리즘 그리고 제안한 알고리즘을 적응잡음제거기에 적용하여 비교하였다. 그 결과 제안한 알고리즘은 음성을 사용하는 적응신호처리 분야에 적합하다는 것을 확인하였다.

• 디지털산업정보학회논문지
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• 제10권4호
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• pp.133-142
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• 2014

The least mean square(LMS) algorithm has been popular owing to its simplicity, stability, and availability to implement. But it inherently has a problem of slow convergence speed, and the presence of a transfer function in the secondary path following the adaptive controller and the error path has been shown to generally degrade the stability and the performance of the LMS algorithm in applications of acoustical noise control. In general, in order to solve these problems, the filtered-x LMS (FX-LMS) type algorithms can be used and the bi-directional Filtered-x LMS(BFXLMS) algorithm is very attractive among them, which increase the convergence speed and the performance of the controller with nearly equivalent computation complexity. In this paper, a mathematical analysis for the BFXLMS algorithm is presented. In terms of view points of time domain, frequency domain, and stochastic domain, the characteristics and stabilities of algorithm is accurately analyzed.

• ETRI Journal
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• 제33권5호
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• pp.806-809
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• 2011

This letter considers the channel estimation for two-way relay MIMO OFDM systems. A least square (LS) channel estimation algorithm under block-based training is proposed. The mean square error (MSE) of the LS channel estimate is computed, and the optimal training sequences with respect to this MSE are derived. Some numerical examples are presented to evaluate the performance of the proposed channel estimation method.

• ETRI Journal
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• 제45권3호
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• pp.379-393
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• 2023

Underwater communication (UWC) is widely used in coastal surveillance and early warning systems. Precise channel estimation is vital for efficient and reliable UWC. The sparse direct-adaptive filtering algorithms have become popular in UWC. Herein, we present an improved adaptive convex-combination method for the identification of sparse structures using a reweighted normalized leastmean-square (RNLMS) algorithm. Moreover, to make RNLMS algorithm independent of the reweighted l₁-norm parameter, a modified sparsity-aware adaptive zero-attracting RNLMS (AZA-RNLMS) algorithm is introduced to ensure accurate modeling. In addition, we present a quantitative analysis of this algorithm to evaluate the convergence speed and accuracy. Furthermore, we derive an excess mean-square-error expression that proves that the AZA-RNLMS algorithm performs better for the harsh underwater channel. The measured data from the experimental channel of SPACE08 is used for simulation, and results are presented to verify the performance of the proposed algorithm. The simulation results confirm that the proposed algorithm for underwater channel estimation performs better than the earlier schemes.

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