• 대한전기학회논문지
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• 제39권10호
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• pp.1057-1065
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• 1990

A parallel processing algorithm for the discrete Kalman filter, which is one of the most commonly used filtering techniques in modern control, signal processing, and communication, is proposed. To decrease the number of computations critical in the Kalman filter, previously proposed parallel algorithms are of the hierarchical structure by distributed processing of measurements, or of the systolic structure to disperse the computational burden. In this paper, a new parallel Kalman filter employing a structure similar to recursive doubling is proposed. Estimated valuse of state variables by the new algorithm converge faster to the true values because the new algorithm can process data twice faster than the conventional Kalman filter. Moreover, it maintains the optimality of the conventional Kalman filter.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1990년도 하계학술대회 논문집
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• pp.64-67
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• 1990

A parallel processing algorithm for discrete Kalman filter, which is one of the most commonly used filtering technique in modern control, signal processing, and communication. is proposed. Previously proposed parallel algorithms to decrease the number of computations needed in the Kalman filter are the hierachical structures by distributed processing of measurements, or the systolic structures to disperse the computational burden. In this paper, a new parallel Kalman filter employing a structure similar to recursive doubling is proposed. Estimated values of state variables by the new algorithm converge with two times faster data processing speed than that of the conventional Kalman filter. Moreover it maintains the optimality of the conventional Kalman filter.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1996년도 Proceedings of the Korea Automatic Control Conference, 11th (KACC); Pohang, Korea; 24-26 Oct. 1996
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• pp.157-160
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• 1996

Physical systems axe generally continuous-time in nature. However as the data measured from these systems is generally in the form of discrete samples, and most modern signal processing is performed in the discrete-time domain, discrete-time models are employed. This paper describes methods for estimating the coefficients of continuous-time system within a closed loop control system. The method employs a recursive estimation algorithm to identify the coefficients of a discrete-time bilinear-operator model. The coefficients of the discrete-time bilinear-operator model closely approximate those of the corresponding continuous-time Laplace transform transfer function.

• Journal of Communications and Networks
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• 제1권2호
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• pp.86-88
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• 1999

The Discrete Fourier Transform (DFT) and the Inverse Discrete Fourier Transform (IDFT) are commonly used in signal processing applications, in particular in digital communication sys-tems using the multi-carrier modulation principle. In such systems an IDFT is computed at the transmitter end, and a DFT at the re-ceiver end. This paper examines a technique of computations, for which only negligible differences appear between the DFT and the IDFT calculations while the number of arithmetic operations re-quired is substantially reduced. This offers significant advantages for the design of an IDFT/DFT processor for Discrete Multitone(DMT) systems.

• 한국인터넷방송통신학회논문지
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• 제19권1호
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• pp.9-14
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• 2019

소형 밀리미터파 추적 레이다는 저속으로 기동 중인 큰 RCS를 갖는 바다위의 함정 표적에 대하여 TWS(Track While Scan) 방식을 통하여 실시간으로 표적을 탐색, 탐지 하여 추적하는 펄스 방식의 레이더이다. 본 논문에서는 저속으로 기동을 하지만 채프, 디코이 등 다양한 기만체를 운영하는 함정 표적에 대하여 LPRF와 DBS, 및 HRR 신호처리 기법을 통하여 표적 정보를 획득하고 추적하기 위하여 고속의 CPU가 탑재된 보드 개발과 표적정보를 실시간 처리하기 위하여 FPGA(Field Programmable Gate Array)를 활용하여 실시간 FFT 연산이 가능한 DFT(Discrete Fourier Transform) 모듈 설계를 포함한 신호처리기 구조를 설계하고 성능시험을 통해 구현한 신호처리기를 검증하였다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2002년도 춘계학술대회논문집
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• pp.71-77
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• 2002

The spectrum of impulse response signal which is obtained from an impulse hammer testing is used for frequency response function, nevertheless it has serious faults when the record length for the signal processing is not very long. The faults cannot be avoided with the conventional signal analyzer that is processing all the signals as if they are always periodic. The signals generated by the impact hammer are undoubtedly non-periodic because of the damping, and are acquired for limited recording time due to the memory as well as the computation performance of the signal analyzer. This paper will make clear the relation between the faults and the length of recording time, and propose the way for solving the faults.

• 디지털산업정보학회논문지
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• 제7권4호
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• pp.119-131
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• 2011

In this paper, we explore the application of 2-D dual-tree discrete wavelet transform (DDWT), which is a directional and redundant transform, for image coding. DDWT main property is a more computationally efficient approach to shift invariance. Also, the DDWT gives much better directional selectivity when filtering multidimensional signals. The dual-tree DWT of a signal is implemented using two critically-sampled DWTs in parallel on the same data. The transform is 2-times expansive because for an N-point signal it gives 2N DWT coefficients. If the filters are designed is a specific way, then the sub-band signals of the upper DWT can be interpreted as the real part of a complex wavelet transform, and sub-band signals of the lower DWT can be interpreted as the imaginary part. The quincunx lattice is a sampling method in image processing. It treats the different directions more homogeneously than the separable two dimensional schemes. Quincunx lattice yields a non separable 2D-wavelet transform, which is also symmetric in both horizontal and vertical direction. And non-separable wavelet transformation can generate sub-images of multiple degrees rotated versions. Therefore, non-separable image processing using DDWT services good performance.

• 한국통신학회논문지
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• 제35권1A호
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• pp.71-79
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• 2010

다중채널 신호처리 알고리즘은 사용 주파수 대역의 가변성, 효율적인 전송전력 할당, 서로 다른 전송률과 대역을 요구하는 서비스 형태를 충족시키기 위한 가변 주파수 대역 변환을 요구한다. 본 논문에서는 다중채널 반송파 신호의 가변 주파수 대역 변환을 위해 시간 영역의 윈도우 함수와 DFT(Discrete Fourier Transform)를 이용한 다중채널 신호처리 알고리즘을 제안한다. 제안한 알고리즘은 기존의 주파수 영역에서 대역통과 신호처리를 하는 다중채널 신호처리 알고리즘과 달리, 시간 영역에서 윈도우 함수를 사용한 블록 신호처리를 하기 때문에 기존의 주파수 영역에서 신호처리 방식보다 연산이 간단하며 효율적인 주파수 변환을 할 수 있다. 전산모의 실험을 통해 제안한 알고리즘의 출력신호 복원과 가변 주파수 대역 변환이 효율적으로 이루어지는 것을 보였다.

• 한국안전학회지
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• 제6권2호
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• pp.21-30
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• 1991

In this paper, we study effect of waveness at the optical surface roughness measurement. In generally, waveness components cause errors in calculation of the roughness value of metal surface. We study about surface roughness signals In the frequency domain for separate two signal component of real roughness and waveness by digital signal processing methods. Thereafter, determine low and high Component of frequency spectrum. By this separating frequency value we design liner low and high pass filter which cutoff frequency is 1 Hz. After this process, converted each filtered spectrum by inverse discrete fourier transformation to time domain waveness and real roughness signals We calculate surface roughness value from filtered roughness signals. For evaluate this method, we use five specimens roughness signal which obtained from optical surface roughness measuring system in 3mm/s moving speed with 0.1 mm laser beam spot size As a result, we obtain more linerized roughness value than that of unfiltered roughness signals.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅲ
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• pp.1685-1688
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• 2003

The modem multimedia applications which are video Processor, video conference or video phone and so forth require real time processing. Because of a large amount of image data, those require high compression performance. In this paper, the proposed image processing encoder was designed by using wavelet transform encoding. The proposed filter block can process image data on tile high speed because of composing individual function blocks by parallel and compute both highpass and lowpass coefficient in the same clock cycle. When image data is decomposed into multiresolution, the proposed scheme needs external memory and controller to save intermediate results and it can operate within 33㎒.