• 한국전자통신학회논문지
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• 제2권3호
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• pp.174-179
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• 2007

본 논문에서는 유전자 알고리듬을 이용하여 전력, 면적, 속도를 고려한 저전력 모듈 선택을 제안한다. 제안한 알고리듬은 최적의 모듈 선택을 통해서 전력 소모를 최소화 할 수 있다. 비교 실험에서는 최적 모듈 선택을 고려한 알고리듬은 최대 전력 감소량은 26.9 %를 얻을 수 있었고, 반면에 최소 전력 감소량은 9.0% 얻었다. 모든 벤치마크 평균 전력 감소량은 15.525%가 되었다.

• 한국통신학회논문지
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• 제31권9C호
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• pp.876-881
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• 2006

In this paper, we propose a complex-valued IIR filter for digital VSB signals based on CMA in order to efficiently mitigate multipath distortions, especially the leakage from the quadrature component. The proposed equalizer overcomes the drawback of the conventional real-valued IIR equalizers that it attempts to equalize Hilbert transform of quadrature component. We demonstrate via simulation that the proposed complex IIR filter successfully mitigates the leakages from the quadrature component, while the conventional real IIR filter requires a longer IIR filter to achieve the same performance. We present cost function analysis for a simple two-tap case showing that the proposed IIR equalizer with CMA for VSB signals has a global minimum at the desired location.

• Journal of Advanced Marine Engineering and Technology
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• 제31권7호
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• pp.863-871
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• 2007

This paper proposes a methodology to estimate the system coefficients for the infinite impulse response(IIR) digital filters using real code GA. In the traditional real coded GA, it adapts the general genetic operations, whereas in this paper the proposed real coded GA applies improved genetic operations in order to search the optimal solution in given problems. Each of unknown IIR digital coefficients collected as forms of a chromosome. Two illustrative examples including the band pass and band stop IIR digital filters are demonstrated to verify the proposed method.

• ETRI Journal
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• 제25권5호
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• pp.345-355
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• 2003

In this paper, we propose a method for designing a class of M-channel, causal, stable, perfect reconstruction, infinite impulse response (IIR), and parallel uniform discrete Fourier transform (DFT) filter banks. It is based on a previously proposed structure by Martinez et al. [1] for IIR digital filter design for sampling rate reduction. The proposed filter bank has a modular structure and is therefore very well suited for VLSI implementation. Moreover, the current structure is more efficient in terms of computational complexity than the most general IIR DFT filter bank, and this results in a reduced computational complexity by more than 50% in both the critically sampled and oversampled cases. In the polyphase oversampled DFT filter bank case, we get flexible stop-band attenuation, which is also taken care of in the proposed algorithm.

• 전자공학회논문지B
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• 제32B권1호
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• pp.96-102
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• 1995

This paper presents the VHSIC Hardware Description Language(VHDL) implementation of the Fixed Point Covariance Lattice(FLAT) algorithm for an Linear Predictive Coding(LPC) analysis and its related algorithms, such as the forth order high pass Infinite Impulse Response(IIR) filter, covariance matrix calculation, and Spectral Smoothing Technique(SST) in the Vector Sum Exited Linear Predictive(VSELP) speech coder that has been Selected as the standard speech coder for the North America and Japanese digital cellular. Existing Digital Signal Processor(DSP) chips used in digital cellular phones are derived from general purpose DSP chips, and thus, these DSP chips may not be optimal and effective architectures are to be designed for the above mentioned algorithms. Then we implemented the VHDL code based on the C code, Finally, we verified that VHDL results are the same as C code results for real speech data. The implemented VHDL code can be used for performing logic synthesis and for designing an LPC Application Specific Integrated Circuit(ASOC) chip and DsP chips. We first developed the C language code to investigate the correctness of algorithms and to compare C code results with VHDL code results block by block.

• 대한전자공학회논문지TC
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• 제43권8호
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• pp.69-76
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• 2006

이 논문에서는 조정 가능한 소프트 판정기를 이용하여 자력복구 판정궤한 채널등화기의 적응 모드를 포착모드와 추적 모드 사이에서 최적화하는 기법들을 제안한다. 제안된 기법들은 주어진 SNR에 따라 소프트 판정기를 최적화하여 DFE를 위한 궤환 신호를 생성하고 그에 따라 자력복구 IIR 필터 적응모드와 DD-LMS 적응모드를 결합한 적응방식을 적용한다. 제안된 기법들은 포착모드와 추적모드 사이의 최적화된 스위칭을 성취할뿐아니라 DFE 에러 propagation을 최소화하는데도 기여한다.

• 대한전자공학회논문지
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• 제24권4호
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• pp.719-726
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• 1987

In this paper, an adaptive IIR (infinite impulse response) notch filter of lattice type is constructed and its adaptation algorithm is proposed for the detection and retrieval of a sine wave signal embedded in noise. A modified method which adapts only one coefficient of the filter is also suggested. All these methods adapt the coefficients while keepting the poles of the filter inside the unit circle on z-plane, and thus they satisfy the condition on the stability of the IIR filter after it has converged. To investigate the convergence characteristics of these methods such as convergence speed and output S/N ratio, intensive computer simulation has been performed by varying the frequency of the sine wave and the input S/N ratio. And the results of the simulation have been compared to those of Rao and Kung's which shows relatively fast convergence speed. The methods proposed here, especially the second one. shows faster convergence speed and higher output S/N ratio than the Rao and Kung's.

• 전자공학회논문지S
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• 제34S권9호
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• pp.23-30
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• 1997

In this paper, a decision feedback recurrent neural equalization (DFRNE) scheme is proposed for adaptive equalization problems. The proposed equalizer models a nonlinear infinite impulse response (IIR) filter. The modified Real-Time recurrent Learning Algorithm (RTRL) is used to train the DFRNE. The DFRNE is applied to both linear channels with only intersymbol interference and nonlinear channels for digital video cassette recording (DVCR) system. And the performance of the DFRNE is compared to those of the conventional equalizaion schemes, such as a linear equalizer, a decision feedback equalizer, and neural equalizers based on multi-layer perceptron (MLP), in view of both bit error rate performance and mean squared error (MSE) convergence. It is shown that the DFRNE with a reasonable size not only gives improvement of compensating for the channel introduced distortions, but also makes the MSE converge fast and stable.

• 전기학회논문지
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• 제59권2호
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• pp.476-483
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• 2010

If the data are pre-whitened, then gradient adaptive algorithms which are simpler than higher order statistics algorithms can be used in adaptive blind signal estimation. In this paper, we propose a blind subband affine projection algorithm for multiple-input multiple-output adaptive equalization in the blind environments. All of the adaptive filters in subband affine projection equalization are decomposed to polyphase components, and the coefficients of the decomposed adaptive sub-filters are updated by defining the multiple cost functions. An infinite impulse response filter bank is designed for the data pre-whitening. Pre-whitening procedure through subband filtering can speed up the convergence rate of the algorithm without additional computation. Simulation results are presented showing the proposed algorithm's convergence rate, blind equalization and blind signal separation performances.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2014년도 춘계학술발표대회
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• pp.995-997
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• 2014

뇌파는 최근에 가장 많이 연구되고 있는 생체신호이다. 본 연구에서는 오픈 감정뇌파데이터인 DEAP Dataset를 활용한 데이터 분석 실험을 시행하였다. DEAP Dataset는 총 32개의 데이터이며, 32채널로 구성되어 있다. 전처리 과정에서는 디지털 필터인 IIR(Infinite Impulse Response) Filter를 사용하여 잡음을 제거하였고, 인공산물인 안구잡파(EOG: Electrooculograms) 제거에는 LMS(the Least Mean squares) 알고리즘을 사용하였다. 감정분류는 Valence-Arousal 평면을 사용하여 네 개의 감정으로 구분하였고, 분류 실험으로는 패턴인식 알고리즘인 SVM(support Vector Machine)를 사용하였다. 실험결과 SVM이 70%대의 결과를 도출하여 이전 실험결과보다 높은 정확도를 도출하였다.