• The Journal of the Acoustical Society of Korea
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• v.17 no.3
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• pp.90-95
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• 1998

본 논문에서는 CHMM인 CDHMM과 ARHMM을 이용하여 화자적응화 하는 방법을 각각 연구하였다. CDHMM에서는 최대사후화확률 추정법에 의하여 각 상태마다 하나의 가 지를 이용하여 화자에 적응시킨다. 본 논문에서는 음성의 다양한 음향학적 특징을 표현하기 위하여 상태마다 여러 개의 가지를 갖는 방법을 제안하였다. 상태마다의 적절한 가지 수를 결정하기 위하여 각 상태에 속하는 프레임 수와 특징 벡터들의 분산행렬의 행렬식값을 이용 하였다. ARHMM에서는 특징벡터로 선형예측계수를 사용하기 때문에 최대사후화확률 추정 법을 사용할 수 없게 된다. 따라서 화자독립모델을 이용하여 적응화자에 대한 음성을 Viterbi 알고리듬으로 상태별로 분할한 후 k-means 알고리듬을 이용하여 각 상태마다 하나 의 가지를 갖는 모델로 적응시키는 방법을 제안하였다.

• Proceedings of the KIEE Conference
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• 2000.07d
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• pp.3115-3117
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• 2000

기존의 LMS 알고리듬을 이용한 적응필터에 비해 연산횟수를 줄이고 입력신호의 통계적 특성에 덜 민감한 적응필터를 제안한다. 입력 신호와 기준신호에 대한 고속 하다마드 변환을 수행한 후 하다마드 변환 영역에서 LMS 알고리듬을 적용한다 기존의 적응필터와 비교하여 필터의 입력신호 추정 성능은 유지하면서 고속 하다마드 변환으로 인해 적응과정에서의 곱셈연산이 크게 줄어드며 잡음의 분산값 변화와 같은 입력신호의 변화에 대한 필터의 안정도와 강인성이 크게 향상됨을 보인다.

• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.45-55
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• 1997

In this paper, we propose a fast algorithm with adaptive thresholding for the wavelet transform (WT) based blocking artifact reduction. In the fast algorithm, all processings that are equivalent to the processing in WT domain of the first and second scale are performed in spatial domain. In the adaptive thresholding, the threshold values used to classify the block boundary are selected adaptively according to each input image by using the statistical properties of the WT of the coded signal at block boundary and at block center, which can be obtained in spatial domain. Experimental results showed that the proposed fast algorithm is about 10 times faster than the WT-based algorithm. It also was found that the postprocessing with proposed adaptive thresholding yields some PSNR improvement and better subjective quality over that with nonadaptive thresholding which has best performance at high compression ratios of a certain .image, even at low compression ratios.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2
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• pp.40-49
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• 1995

Application of the filtered-x LMS adaptive filter to active noise cancellation requires to estimate the transfer charactersitics between the output and the error signal of the adaptive canceler. In this paper, we derive the filtered-x adaptive noise cancellation algorithm and analyze its convergence behavior when the acoustic noise consists of multiple sinusoids. The results of the convergence analysis of the filtered-x LMS algorithm indicate that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct components : Phase estimation error and estimated gain. In particular, the convergence is shown to strongly affected by the accuracy of the phase response estimate. Simulation results are presented to support the theoretical convergence analysis.

• Journal of Biomedical Engineering Research
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• v.17 no.1
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• pp.121-128
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• 1996

This paper presents the ECG data compression method referred the adaptive fractal interpolation algorithm. In the previous piecewise fractal interpolation(PFI) algorithm, the size of range is fixed So, the reconstruction error of the PFI algorithm is nonuniformly distributed in the part of the original ECG signal. In order to improve this problem, the adaptive fractal interpolation(AEI) algorithm uses the variable range. If the predetermined tolerance was not satisfied, the range would be subdivided into two equal size blocks. large ranges are used for encoding the smooth waveform to yield high compression efficiency, and the smaller ranges are U for encoding rapidly varying parts of the signal to preserve the signal quality. The suggested algorithm was evaluated using MIT/BIH arrhythmia database. The AEI algorithm was found to yield a relatively low reconstruction error for a given compression ratio than the PFI algorithm. In applications where a PRD of about 7.13% was acceptable, the ASI algorithm yielded compression ratio as high as 10.51, without any entropy coding of the parameters of the fractal code.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.4B
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• pp.695-703
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• 2010

In this paper, we propose an adaptive scalable encryption scheme for the layered architecture of SVC video. The proposed method determines an appropriate set of encryption algorithms to be applied for the layers of SVC by considering the importance and priority relationship among the SVC video layers. Unlike the conventional encryption method based on a fixed encryption algorithm for the whole video layers, the proposed method applies differentiated encryption algorithms with different encryption strength the importance of the video layers. Thereupon, higher security could be maintained for the lower video layer including more important data, while lower encryption strength could be applied for the higher video layer with relatively less important data. The effectiveness of the proposed adaptive scalable encryption method is proved by extensive simulations.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.3
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• pp.19-25
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• 2009

In this thesis, design effective elimination interference algorithm of ICS repeat system for repeater that improve frequency efficiency. Error convergence speed and accuracy of LMS Algorithm are influenced by reference signal. For improve LMS Algorithm, suggest Adaptive channel estimate algorithm. For using channel characteristic, adaptive channel estimate algorithm make reference signal similar interference signal by convolution operation and complement LMS algorithm demerit. For make channel similar piratical channel, apply Jake's Rayleigh multi-path model that random five path with 130Hz Doppler frequency. LMS algorithm and suggested adaptive channel estimate algorithm that have 16 taps apply to ICS repeat system under Rayleigh multi-path channel, so simulate with MATLAB. According to simulate, ICS repeat system with LMS algorithm show -40dB square error convergent after 150 datas iteration and ICS repeat system with adaptive channel estimate algorithm show -80dB square error convergent after 200 datas iteration. Analyze simulation result, suggested adaptive channel estimate algorithm show more three times iteration performance than LMS algorithm, and 40dB accuracy.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.829-832
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• 2001

본 논문에서는 2차 통계치를 이용하여 패널추징 및 등화 기법을 제안하였다. 기존의 채널 추정 알고리듬은 잡음이 없는 환경에서 LS방법을 이용하기 때문에 잡음이 강한 패널에서는 원하는 성능을 얻을 수 없는 단점이 있다. 수신신호의 상관행렬의 최소 고유값에 대응하는 고유벡터는 채널의 임펄스 응답에 관한 정보를 포함하고 있다. 이러한 고유 벡터를 매시간마다 갱신시키면서 구하는 적응 알고리듬을 제안하고 이를 이용하여 블라인드 채널 추정 및 등화기 파라미터를 추정하였다. 제안한 알고리듬은 잡음에 강인한 특성을 보일 뿐 아니라 기존의 알고리듬들 보다 우수한 채널 추정 및 등화 성능을 모의 실험을 통하여 검증하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2006.05a
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• pp.917-920
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• 2006

In mobile environments, their is more than one propagation path between each transceiver and Receiver have two or more delayed multipath signals. Delayed multipath signals can cause ISI and Receiver needs a adaptive algorithm to estimate a channel periodically. Also adaptive antenna using adaptive algorithm provide a significant increase in capacity, performance and coverage. This paper describes various LMS algorithm and evaluate the performance of array antenna Base station by using LMS algorithm in the presence of multipath signals and multiple users. As a result of simulation, Adaptive array antenna systems are able to adjust their antenna pattern to select desired signals, and reduce interference.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.4C
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• pp.217-225
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• 2011

In this paper, we propose an adaptive de-interlacing algorithm that is based on the degree of local complexity. The conventional intra field de-interlacing algorithms show the different performance according to the ways which find the edge direction. Furthermore, FDD (Fine Directional De-interlacing) algorithm has the better performance than other algorithms but the computational complexity of FDD algorithm is too high. In order to alleviate these problems, the proposed algorithm selects the most efficient de-interacing algorithm among LA (Line Average), MELA (Modified Edge-based Line Average), and LCID (Low-Complexity Interpolation Method for De-interlacing) algorithms which have low complexity and good performance. The proposed algorithm is trained by the DoLC (Degree of Local Complexity) for selection of the algorithms mentioned above. Simulation results show that the proposed algorithm not only has the low complexity but also performs better objective and subjective image quality performances compared with the conventional intra-field methods.