• Journal of the Korean Institute of Intelligent Systems
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• v.25 no.4
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• pp.349-354
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• 2015

This paper presents the stability conditions of the Takagi-Sugeno (T-S) fuzzy sampled-data filter. The error system between the T-S fuzzy system and fuzzy filter is presented. In the sense of the Lyapunov stability analysis, the stability conditions are given, which can be represented in terms of linear matrix inequalities (LMIs). The proposed stability conditions utilize the different approach from the conventional methods, and have better performance than that of the conventional ones. The simulation example is given to show the effectiveness of the proposed method.

• Journal of the Korea society of information convergence
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• v.7 no.2
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• pp.7-12
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• 2014

Many filters are made in order to dectect the accurate signal. The performance of the filter is evaluated as to remove the noise effectively. It is necessary to determine the threshold value for the accurate evaluation of the filter effectively. In this study, we investigated the optimization of the threshold value. It was used for the triangular wave signal and the power supply noise. SNR change according to the number of samples was studied. As the number of samples increases, SNR saw also stabilizde.

• The Journal of the Acoustical Society of Korea
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• v.18 no.8
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• pp.108-115
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• 1999

In this study we suggested a digital filter method to enhance car audio sound quality against the sound distortion due to cabin's acoustic characteristics and car driving noises. The digital filters designed were based on the characteristics on car driving noises and cabin acoustic characteristics. Car driving noises were analyzed by two ways; one is an objective method, octave band frequency analysis method. The other is a subjective method; sensory evaluation method, NCB method. On these results, seven sets of modified coefficients of eleven band digital filters were obtained. To find optimum audio sound quality among nine sound samples filtered by designing seven types of digital filters, which were mixed car driving noises at 100km/h, subjective evaluation method was used, paired comparison method; Scheffe' seven point method.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• v.2
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• pp.415-418
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• 2006

적응 횡단 필터는 상관 전 처리 기법으로서 실시간 간섭 신호 제거가 가능한 시간 영역 신호처리 기법을 사용한다. 적응 횡단 필터는 협대역 간섭 신호에 좋은 성능을 나타내며, 구현이 용이하고 높은 효율성을 갖는다. 적응 횡단 필터의 구성은 입력 샘플 신호의 지연 탭을 위한 FIR 필터와 전파 간섭 신호의 크기와 주파수를 결정하는 가중치 생성부로 나눌 수 있다. 본 논문에서는 가중치 생성부에 적용 되는 알고리즘 중 상대적으로 연산량이 적은 LMS와 NLMS를 적용한 적응 횡단 필터를 설계하고, GPS 수신기에 적용 하였다. 실제 측정치를 이용한 다양한 실험에 의한 항법 성능 평가를 통하여 NLMS가 LMS보다 좋은 성능을 나타냄을 확인하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.7
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• pp.956-962
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• 2018

As the study of the object feature detection from image, we explain methods to identify the species of the tree in forest using the picture taken from dron. Generally there are three kinds of methods, which are GLCM (Gray Level Co-occurrence Matrix) and Gabor filters, in order to extract the object features. We proposed the object extraction method using the statistical properties of trees in this research because of the similarity of the leaves. After we extract the sample images from the original images, we detect the objects using cross correlation techniques between the original image and sample images. Through this experiment, we realized the mean value and standard deviation of the sample images is very important factor to identify the object. The analysis of the color component of the RGB model and HSV model is also used to identify the object.

• Journal of Korea Game Society
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• v.15 no.2
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• pp.159-168
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• 2015

This paper introduces effective noise removal method for medical ultrasound volume data. Ultrasound volume data need to be filtered because it has a lot of noise. Conventional 2d filtering methods ignore information of adjacent layers and conventional 3d filtering methods are slow or have simple filter that are not efficient for removing noise and also don't equally operate filtering because that don't take into account ultrasound' sampling character. To solve this problem, we introduce method that fast perform in parallel bilateral filtering that is known as good for noise removal and adjust proportionally window size depending on that's position. Experiments compare noise removal and loss of original data among average filtered or biliteral filtered or adaptive biliteral filtered ultrasound volume rendering images. In this way, we can more efficiently and correctly remove noise of ultrasound volume data.

• KEPCO Journal on Electric Power and Energy
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• v.4 no.2
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• pp.55-61
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• 2018

본 연구는 시각 동기 위상 측정 정보를 이용하여 전력계통에 나타나는 여러 가지 동요 현상을 검출하기 위한 기초 연구로써, 시계열 데이터 분석 분야로 분류된다. 제시한 방법은 비선형 동특성에 해석 기반으로 접근하여 전력계통에 나타날 수 있는 여러 동요 현상을 범용적으로 검출해 낼 수 있다. 비선형 동요 현상의 신호적 패턴을 수학적으로 기본 순시치 파형으로부터 피크치 샘플링을 통해 전개하여 계통 요소간 간섭으로 인한 원하지 않는 진동 모드를 검출하고자 한다. 계통의 변화로 진동 모드가 나타날 때, 2차원 평면에 실효치로 환산한 시계열 전압 데이터와 선형화된 플로퀘트 상수(Floquet multiplier)를 맵핑하여 도시하고, 정상상태 지점으로부터 거리를 계산하여 최대 리아프노프 지수 계산을 통해 계통이 불안정하게 되는 시간을 시계열 데이터 분석으로 추정하는 것이 본 방법의 핵심이다. 이러한 접근으로 제시한 비선형 동요 검출 알고리즘을 적용하여 디지털 필터 적용 또는 주파수 영역 해석과 같은 오프라인 Study와 달리 온라인으로 신속하게 계통의 현재 상태를 알 수 있게 된다.

• Journal of Biomedical Engineering Research
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• v.18 no.4
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• pp.467-476
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• 1997

This paper presents a designing of a digital hearing aid processor (DHAP) chip being operated by a dedicated DSP core. The DHAP for hearing aid devices must be feasible within a size and power consumption required. Furthermore, it should be able to compensate for wide range of hearing losses and allow sufficient flexibility for the algorithm development. In this paper, a programmable 16-bit fixed-point DSP core is employed thor the designing of the DHAP. The designed DHAP performs a nonlinear loudness correction of 8 frequency bands based on audiometric measurements of impaired subjects. By employing a programmable DSP, the DHAP provides all the flexibility needed to implement audiological algorithms. In addition, the chip has low-power feature and $5, 500\times5000$$\mu$$m^2$ dimensions that fit for wearable hearing aids.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.4
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• pp.691-698
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• 2003

We extend the sue of the method of least square to develop a recursive algorithm for the design of adaptive transversal filters such that, given the least-square estimate of this vector of the filter at iteration n-l, we may compute the updated estimate of this vector at iteration n upon the arrival of new data. We begin the development of the RLS algorithm by reviewing some basic relations that pertain to the method of least squares. Then, by exploiting a relation in matrix algebra known as the matrix inversion lemma, we develop the RLS algorithm. An important feature of the RLS algorithm is that it utilizes information contained in the input data, extending back to the instant of time when the algorithm is initiated. In this paper, we propose new tap weight updated RLS algorithm in adaptive transversal filter with data-recycling buffer structure. We prove that convergence speed of learning curve of RLS algorithm with data-recycling buffer is faster than it of exiting RLS algorithm to mean square error versus iteration number. Also the resulting rate of convergence is typically an order of magnitude faster than the simple LMS algorithm. We show that the number of desired sample is portion to increase to converge the specified value from the three dimension simulation result of mean square error according to the degree of channel amplitude distortion and data-recycle buffer number. This improvement of convergence character in performance, is achieved at the B times of convergence speed of mean square error increase in data recycle buffer number with new proposed RLS algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7B
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• pp.1351-1360
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• 1999

In frequency selective channels, conventional PN code acquisition schemes are not ideal candidates. This is because they are primarily designed for the AWGN channel. In this paper, a direct-sequence spread-spectrum(DSSS) PN code acquisition system based on digital matched filtering (DMF) with automatic threshold control(ATC) algorithm is presented and analyzed with regards to probability of detection and probability of false alarm. These two important probabilities, the probability of detection ($P_D$) and the probability of false alarm ($P_{FA}$) are derived and analyzed in considering Doppler shift, sampling rate, mean acquisition time, and PN chip rate in frequency selective Rayleigh fading channel when using serial-search method as detection technique. From computer simulation results of a DMF acquisition system model, it is shown that the performance of the acquisition system using ATC algorithm is better than that of constant threshold system.