• ETRI Journal
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• v.16 no.1
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• pp.1-15
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• 1994

A computationally-efficient approach to the calculation of the transient field of an acoustic radiator was developed. With this approach, a planar or curved source, radiating either continuous or pulsed waves, is divided into a finite number of shifted and/or rotated versions of an incremental source such that the Fraunhofer approximation holds at each field point. The acoustic field from the incremental source is given by a 2-D spatial Fourier transform. The diffraction transfer function of the entire source can be expressed as a sum of Fraunhofer diffraction pattern of the incremental sources with the appropriate coordinate transformations for the particular geometry of the radiator. For a given spectrum of radiator velocity, the transient field can be computed directly in the frequency domain using the diffraction transfer function. To determine the accuracy of the proposed approach, the impulse response was derived using the inverse Fourier transform. The results obtained agree well with published data obtained using the impulse response approach. The computational efficiency of the proposed method compares favorably to those of the point source method and the impulse response approach.

• Journal of the Computational Structural Engineering Institute of Korea
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• v.20 no.2
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• pp.105-111
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• 2007

A finite element analysis scheme is proposed to assess the dynamic response due to impulse excitation of ocean mining pipe system. Transient dynamic response analysis based on the proposed scheme was carried out for various types of impulses, and the magnitude of cable tension induced by impulse was discussed by using a model of 'Segero', a special purpose ship of KT-Submarine as well as ROV carrier 'Onnuri' A qualitative, prospective guideline for the relevant marine operation is obtained.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.2532-2537
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• 2005

FIR(Finite Impulse Response) filter is well known to be ideal for the finite time state-space model, but it requires much computation due to its inherent non-recursive structure especially when the measurement interval grows to a large extent. And often a fixed-lag smoother based on the finite time interval is needed to monitor the soundness of the system model and the measurement model, but the computation burden of FIR-type smoother imposes much restriction of its usage for real-time application. Conventional recursive forms of FIR estimator[1]-[4] could not be used for real time applications, since they are numerically unstable in their recursive equations. To cope with this problem, we suggest a stable recursive form FIR estimator(SRFIR) and its usefulness is demonstrated for designing the real-time fixed-lag smoother on the finite time window through an example of detection of rate bias in the anti-aircraft gun fire control system.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.1
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• pp.35-44
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• 2004

We Propose the methods to design the finite impulse response (FIR) and the infinite impulse response (IIR) lattice filters using Schur algorithm through the spectral factorization of the covariance matrix by circulant matrix factorization (CMF). Circulant matrix factorization is also very powerful tool used for spectral factorization of the covariance polynomial in matrix domain to obtain the minimum phase polynomial without the polynomial root finding problem. Schur algorithm is the method for a fast Cholesky factorization of Toeplitz matrix, which easily determines the lattice filter parameters. Examples for the case of the FIR filter and for the case of the In filter are included, and performance of our method check by comparing of our method and another methods (polynomial root finding and cepstral deconvolution).

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.1
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• pp.49-55
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• 2003

The use of digital filters in the signal process field is increasing rapidly with development of the modern industrial society. Especially, detail processors, Y/C separators, ghost removing filters, standard converters (NTSC to PAL or PAL to NTSC) and noise reducers, all of which use digital filters, tend to be used in digital video and audio processing, CATV and various communication fields. Generally, there are two different digital filters, the Rf (infinite impulse response) filter and the FIR (finite impulse response) filter in digital filter. In this paper, we have designed FIR filter which has the phase linearity and the easiness of creation. In the design of the FIR digital filter, the window function is used to alleviate the ripples caused by Gibbs Phenomenon around the cut off frequency of the band pass. But there're some problems to choose proper window function for the design destination due to its fixed values. Therefore, in this paper, we designed a modified Hanning window with new parameter which is adaptively chosen corresponding to design objectives. The digital filter was simulated to prove the validity of the model and it was compared with the Hamming, the Manning, the Blacknan and the Kaiser window function. And we have used peak side-lobe and transient characteristics as standard of judgement.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.443-446
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• 2005

As modern society needs to process of acquisition, storage and transmission of much information, the importance of signal processing is increasing and various digital filters are used in the two-dimensional signal such as image. And kinds of these digital filters are IIR(infinite impulse response) filter and FIR(finite impulse response) filter. And FIR filter which has the phase linearity, the easiness of creation and stability is applied to many fields. In design of this FIR filter, flatness property is a important factor in pass-band and stop-band. In this paper, we designed a 2-D Circular FIR filter using the Bernstein polynomial, it is presented flatness property in pass-band and stop-band. And we simulated the designed filter with noisy test image and compared the results with existing methods.

• Journal of the Korean Institute of Telematics and Electronics
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• v.20 no.1
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• pp.27-36
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• 1983

The 12-Channe1 TDM/FDM translator is proposed which uses a periodically varying digital filter and the multiplexing weaver modulators. The general 12-Channel TDM/FDM translator using the Weaver modulators requires 24 interpolating FIR(finite impulse response) filters and 24 sinusoidal modulators, however the TDM/FDM translator proposed in this paper consists of one interpolating periodically varying digital filter and 12 sinusoidal modulators. The results obtained in this paper show that the system is simplified and the computation time is reduced. These facts are verified by the computer simulation.

• Journal of Korea Society of Digital Industry and Information Management
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• v.13 no.3
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• pp.35-42
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• 2017

This paper presents hardware implementation of background timing-skew calibration technique for time-interleaved analog-to-digital converters (TI ADCs). The timing skew between any two adjacent analog-digital (A/D) channels is detected by using pure digital Finite Impulse Response (FIR) delay filter. This paper includes hardware architecture of the system, main units and small sub-blocks along with control logic circuits. Moreover, timing diagrams of logic simulations using ModelSim are provided and discussed for further understanding about simulations. Simulation process in MATLAB and Verilog is also included and provided with basic settings need to be done. For hardware implementation it not practical to work with all samples. Hence, the simulation is conducted on 512 TI ADC output samples which are stored in the buffer simultaneously and the correction arithmetic is done on those samples according to the time skew algorithm. Through the simulated results, we verified the implemented hardware is working well.

• JSTS:Journal of Semiconductor Technology and Science
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• v.16 no.3
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• pp.261-273
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• 2016

Reconfigurable finite impulse response (FIR) filters whose filter coefficients and filter order change dynamically during run-time play an important role in the software defined radio (SDR) systems, multi-channel filters, and digital up/down converters. However, there are not many reports on such reconfigurable designs which can support dynamic variation of filter order and filter coefficients. The purpose of this paper is to provide an architectural solution for the FIR filters to support run-time variation of the filter order and filter coefficients. First, two straightforward designs, namely, (i) single-MAC based design and (ii) full-parallel design are presented. For large variation of the filter order, two designs based on (iii) folded structure and (iv) fast FIR algorithm are presented. Finally, we propose (v) high throughput design which provides significant advantage in terms of hardware and/or time complexities over the other designs. We compare complexities of all the five structures, and provide the synthesis results for verification.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.04a
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• pp.714-720
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• 2008

This research presents a laboratory study about an subjective response of impulsive sound caused by leisure shooting. The sources are sampled from outdoor noise and their levels range from 40 to 75 dB at the interval of 5dB. The noise unit is based on A-weighted sound exposure level (ASEL; $L_{AE}$). To make equal ASEL of outdoor noise, finite impulse response (FIR) filter is applied to the originally sampled source to include the effect of distance attenuation. The evaluation method of the jury test adopted a Semantic Difference(SD) Method. In the result of the jury test for impulsive noise, the mean response rating expressed a linear relation and %HA(percent highly annoyed) displayed a exponential growth relation.