• The Journal of the Acoustical Society of Korea
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• v.35 no.1
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• pp.55-62
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• 2016

The least-mean fourth (LMF) algorithm is well known for its fast convergence and low steady-state error especially in non-Gaussian noise environments. Recently, there has been increasing interest in the least mean square (LMS) algorithms with variable step size. It is because the variable step-size LMS algorithms have shown to outperform the conventional fixed step-size LMS in the various situations. In this paper, a variable step-size LMF algorithm is proposed, which adopts an algebraic optimal step size as a variable step size. It is expected that the proposed algorithm also outperforms the conventional fixed step-size LMF. The superiority of the proposed algorithm is confirmed by the simulations in the time invariant and time variant channels.

• The KIPS Transactions:PartB
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• v.9B no.2
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• pp.223-230
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• 2002

In a speech coding system using excitation source of voiced and unvoiced, it would be involves a distortion of speech waveform in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new method of TSIUVC (Transition Segment Including Unvoiced Consonant) approximate-synthesis by using Least Mean Square. The TSIUVC extraction is based on a zero crossing rate and IPP (Individual Pitch Pulses) extraction algorithm using residual signal of FIR-STREAK Digital Filter. As a result, This method obtain a high Quality approximation-synthesis waveform by using Least Mean Square. The important thing is that the frequency signals in a maximum error signal can be made with low distortion approximation-synthesis waveform. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.

• Journal of Communications and Networks
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• v.6 no.1
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• pp.19-25
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• 2004

In this paper, we present a trellis-based blind channel estimation and equalization technique coupling two kinds of adaptive Viterbi algorithms. First, the initial blind channel estimation is accomplished by incorporating the list parallel Viterbi algorithm with the least mean square (LMS) updating approach. In this operation, multiple trellis mappings are preserved simultaneously and ranked in terms of path metrics. Equivalently, multiple channel estimates are maintained and updated once a single symbol is received. Second, the best channel estimate from the above operation will be adopted to set up the whole trellis. The conventional adaptive Viterbi algorithm is then applied to detect the signal and further update the channel estimate alternately. A small delay is introduced for the symbol detection and the decision feedback to smooth the noise impact. An automatic switch between the above two operations is also proposed by exploiting the evolution of path metrics and the linear constraint inherent in the trellis mapping. Simulation has shown an overall excellent performance of the proposed scheme in terms of mean square error (MSE) for channel estimation, robustness to the initial channel guess, computational complexity, and channel equalization.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.7 no.6
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• pp.1175-1179
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• 2006

In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of exist a voiced and an unvoiced consonants in a frame. In this paper, I propose a new method of TSIUVC(Transition Segment Including Unvoiced Consonant) approximate-synthesis by using Least Mean Square. As a result, a method by using Least Mean Square was obtained a high quality approximation-synthesis waveform . The important thing is that the frequency signals in a maximum error signal can be made with low distortion approximation-synthesis waveform. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and synthesis.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.10C
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• pp.1007-1012
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• 2003

In order to accelerate the convergence speed of the conventional BP algorithm, constrained optimization techniques are applied to the BP algorithm. First, the noise-constrained least mean square algorithm and the zero noise-constrained LMS algorithm are applied (designated the NCBP and ZNCBP algorithms, respectively). These methods involve an important assumption: the filter or the receiver in the NCBP algorithm must know the noise variance. By means of extension and generalization of these algorithms, the authors derive an adaptive error-constrained BP algorithm, in which the error variance is estimated. This is achieved by modifying the error function of the conventional BP algorithm using Lagrangian multipliers. The convergence speeds of the proposed algorithms are 20 to 30 times faster than those of the conventional BP algorithm, and are faster than or almost the same as that achieved with a conventional linear adaptive filter using an LMS algorithm.

• Journal of the Korean Statistical Society
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• v.27 no.4
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• pp.485-494
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• 1998

In the white noise model we construct an adaptive estimate for f(0) for a decreasing function f. We also show that the maximum mean square error of this estimate attains the same rate as the minimax risk simultaneously over a range of Lipschitz classes of order less than or equal to one.

• 제어로봇시스템학회:학술대회논문집
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• 1989.10a
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• pp.109-112
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• 1989

Gun system operation is represented as a first-order Markov process, and an optimum linear filter is derived for closed-loop control of mean square error. Potential improvement is then estimated by contrasting the variance in performance and the auto correlation for open-loop system with that for the optimum linearly corrected process.

• ETRI Journal
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• v.8 no.1
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• pp.3-5
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• 1986

본 고는 image의 pattern을 identify하기 위해 그 image data의 FFT(Fast Fourier Transform)를 취한 후 에너지 스펙트럼의 크기를 폐적분한 값으로 부터 original input object와 비교대상의 object에 대한 mean square error 값의 차이를 시뮬레이션한 결과 얻은 threshold value와 비교함으로써 matching 을 구현하기 위함이다. Vax11-780/vms와 Fortran77 Language를 사용하여 시뮬레이션을 수행하였으며 Tektronix graphic terminal이 digitized된 이미지의 모니터용으로 사용되었다.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2013.04a
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• pp.793-794
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• 2013

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8C
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• pp.613-616
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• 2008

In this paper, a fine residual frequency offset estimation scheme is proposed for multiband orthogonal frequency division multiplexing ultra-wideband (MB-OFDM UWB) systems. The basic idea of our approach is based on the fact that two adjacent OFDM symbols carry the identical information in the MB-OFDM UWB system, thus removing the need of pilot symbols. The mean square error of the synchronization scheme is evaluated and simulation results are used to verify the effectiveness of the proposed estimator. When compared to the pilot-aided conventional estimator, the proposed estimator has a lower estimation error.