• Journal of the Korea Institute of Military Science and Technology
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• 제24권5호
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• pp.475-481
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• 2021

The on-board processor of satellite Synthetic Aperture Radar(SAR) digitizes the back-scattered echoes and transmits them to the ground. As satellite SAR image of various operating conditions including broadband and high resolution is required, an enormous amount of SAR data is generated. Decimation filter is used for data compression to improve the transmission efficiency of these data. Decimation filter is implemented with the FIR(Finite Impulse Response) filter and here, the decimation ratio and tap length are constrained by resource requirements of FPGA used for implementation. This paper suggests to use a non-integer ratio decimation filter in order to optimize the data transmission efficiency. Also, it proposes a filter design method that remarkably reduces the resource constraints of the FPGA in-use via applying a polyphase filter structure. The required resources for implementing the proposed filter is analysed in this paper.

• Journal of the Institute of Electronics and Information Engineers
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• 제50권10호
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• pp.188-198
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• 2013

Maximally flat (MAXFLAT) half-band filters usually have wider transition band than other filters. This is due to the fact that the maximum possible number of zeros at $z={\pm}1$ is imposed, which leaves no degree of freedom, and thus no independent parameters for direct control of the frequency response. This paper describes a novel method for the design of FIR halfband filters with an explicit control of the transition-band width. The proposed method is based on a generalized Lagrange halfband polynomial (g-LHBP) with coefficients parametizing a 0-th coefficient $h_0$, and allows the frequency response of this filter type to be controllable by adjusting $h_0$. Then, $h_0$ is modeled as a steepness parameter of the transition band and this is accomplished through theoretically analyzing a polynomial recurrence relation of the g-LHBP. This method also provides explicit formulas for direct computation of design parameters related to choosing a desired filter characteristic (by trade-off between the transition-band sharpness and passband & stopband flatness). The examples are shown to provide a complete and accurate solution for the design of such filters with relatively sharper transition-band steepness than MAXFLAT half-band filters.

• Journal of the Institute of Electronics Engineers of Korea SP
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• 제39권5호
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• pp.556-563
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• 2002

In this paper, a novel method for designing an M-channel, causal, stable IIR cosine-modulated filter bank (CMFB) with near PR property is proposed. The IIR prototype filter is designed with a simple constraint using lattice stucture with 1st order allpass filter components. The IIR prototype filter which is designed by the proposed method has higher stopband attenuation and sharper roll-off characteristic than the one which is designed by the previously proposed method with similar complexity. The proposed M-channel IIR CMFB which is designed from this IIR prototype filter is applied to subband acoustic echo canceller (AEC). We obtained about 15dB higher ERLE using this subband AEC than when M-channel FIR subband AEC with similar complexity.

• Journal of the Korea Society of Computer and Information
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• 제17권2호
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• pp.113-118
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• 2012

In this paper, we discuss the design of the Digital Down Converters for the IEEE 802.11 wireless LAN receiver, which can be used for the customized receiver. The customized receiver can be used for special puropsed services which cannot be realized using the general custom chip. In the OFDM receiver, DDC receives the up sampled Inphase/Quadrature signal from the AD converter and process down sampling and filtering procedures using the Cascaded Intergrator Filter and FIR filters. We discuss the structure and design methodology of DDC's and analyze the simulation results.

• Journal of the Institute of Electronics Engineers of Korea SD
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• 제42권12호
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• pp.45-52
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• 2005

A systolic array formed by interconnecting a set of identical data-processing cells in a uniform manner is a combination of an algorithm and a circuit that implements it, and is closely related conceptually to arithmetic pipeline. High-performance computation on a large array of cells has been an important feature of systolic array. To achieve even higher degree of concurrency, it is desirable to make cells of systolic array themselves systolic array as well. The structure of systolic array with its cells consisting of another systolic array is to be called super-systolic array. This paper proposes a scalable bit-level super-systolic amy which can be adopted in the VLSI design including regular interconnection and functional primitives that are typical for a systolic architecture. This architecture is focused on highly regular computational structures that avoids the need for a large number of global interconnection required in general VLSI implementation. A bit-level super-systolic FIR filter is selected as an example of bit-level super-systolic array. The derived bit-level super-systolic FIR filter has been modeled and simulated in RT level using VHDL, then synthesized using Synopsys Design Compiler based on Hynix $0.35{\mu}m$ cell library. Compared conventional word-level systolic array, the newly proposed bit-level super-systolic arrays are efficient when it comes to area and throughput.

• Journal of the Korean Institute of Telematics and Electronics B
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• 제31B권12호
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• pp.11-19
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• 1994

In this paper, we propose a new method to desing IIR Nyquist filters. By thismethod, we can design IIR nyquist filters which prossess linear phase characteristics andare lower in order than FIR Nyquist filters. WE derive expressions for zero-phased IIR Nyquist filters and propose new desing methods for them. An iteration process is used to design the proposed IIR Nyquist filter, and in each iteration step a modified version of the Remez exchange algorithm is used. Finally, the propose desing method is demonstrated through design examples.

• Proceedings of the Korea Contents Association Conference
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• 한국콘텐츠학회 2005년도 춘계 종합학술대회 논문집
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• pp.341-345
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• 2005

In this paper, we propose an algorithm for the approximation of FIR filters by IIR filters. The algorithm is based on a concept of the balanced model reduction. Head-related transfer functions(HRTFs) of dummy-head are approximated by 32-order IIR filters. The binaural sounds using the approximated HRTFs are reproduced by headphone, and serves as a cue of sound image localization. Experiment of sound image are carried out for 10 participants with computer simulation and DSP board respectively. The results of the experiments show that the localization using the approximated HRTFs by IIR filters is the same accuracy as the case of FIR filters that simulate the HRTFs.

• Journal of the Institute of Electronics Engineers of Korea SD
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• 제45권9호
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• pp.65-70
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• 2008

In FIR filter design, multipliers occupy most of the area. To efficiently reduce the area occupied by multipliers, Common Subexpression Elimination (CSE) algorithm can be used instead of separate multipliers. However, the filter computation time can be increased due to the long carry propagation in CSE circuits. More specifically, when the difference of weights between the two inputs to an adder in CSE circuits is large, long carry propagation time is required due to large sign extension. In this paper, we propose a sign-extension reduction method in common subexpression elimination circuit. By Synopsys simulation using Samsung 0.35um library, it is shown that the proposed method leads to 17%, 31% and 12% reduction in the area, time delay and power consumption, respectively, compared with conventional method.

• Journal of Navigation and Port Research
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• 제28권6호
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• pp.491-496
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• 2004

The paper describes measurement techniques for an acoustic transfer characteristic of ship's bridge stimulated by a whistle sound The response sounds, according to the opening-shutting conditions of bridge doors for Training Ship ‘SAENURI’, are measured by B＆K 2260D equipment, and then the frequency responses are extracted by B＆K 7830 software. To evaluate the measured transfer characteristic, the 128th order FIR (Finite Impulse Response) filters, containing the different frequency characteristics, are constructed based on the frequency sampling-based design method Using evaluation indexes with six scales, psychological assessments by five subjects are carried out with test sounds which are obtained from convolving the source signal with FIR filters. As results of tests, the test sounds gives $A_S$ 3.3∼4.7 which means the psychological sense of ‘it is almost similar sound as original ones in a real world’, and thus it is clearly seen that the proposed method can be used for the measurement of an acoustic transfer characteristic of ship’s bridge.

• The Journal of the Acoustical Society of Korea
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• 제19권1호
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• pp.97-102
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• 2000

Subband coding is to divide the signal frequency band into a set of uncorrelated frequency bands by filtering and then to encode each of these subbands using a bit allocation rationale matched to the signal energy in that subband. The actual coding of the subband signal can be done using waveform encoding techniques such as PCM, DPCM and vector quantizer(VQ) in order to obtain higher data compression. Most researchers have focused on the error in the quantizer, but not on the overall reconstruction error and its dependence on the filter bank. This paper provides a thorough analysis of subband codecs and further development of optimum filter bank design using vector quantizer. We compute the mean squared reconstruction error(MSE) which depends on N the number of entries in each code book, k the length of each code word, and on the filter bank coefficients. We form this MSE measure in terms of the equivalent quantization model and find the optimum FIR filter coefficients for each channel in the M-band structure for a given bit rate, given filter length, and given input signal correlation model. Specific design examples are worked out for 4-tap filter in 2-band paraunitary filter bank structure. These optimum paraunitary filter coefficients are obtained by using Monte Carlo simulation. We expect that the results of this work could be contributed to study on the optimum design of subband codecs using vector quantizer.