• 한국소음진동공학회논문집
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• 제22권2호
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• pp.146-155
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• 2012

The filtered-x LMS(FX-LMS) algorithm has been applied to the active noise control(ANC) system in an acoustic duct. This algorithm is designed based on the FIR(finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filtered-u LMS algorithm(FU-LMS) based on infinite impulse response(IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

• 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.

• 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.

• 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.

• 대한전자공학회논문지
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• 제24권2호
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• pp.196-200
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• 1987

The following methods are proposed for implementing digital dual tone multi-frequency (DTMF) receiver: using infinite impulse response(IIR) digital filters, period-counting algorithm, discrete Fourier transform(DFT), and fast Fourier transform(FFT)[2]. The PARCOR(Partical Correlation) analysis method which has been widly used in the speech signal processing area is applied to the dual tone multi-frequency(DTMF) signal detection. This method is easy to implement digitally and stronger to digit simulation of speech than any other methods proposed up to date. Since sampling rate of 4KHz is used in the DTMF receiver for the detection of input DTMF signal originally sampled at 8KHz, it effects two times higher multiplexing efficiency.

• ETRI Journal
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• 제35권4호
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• pp.697-705
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• 2013

Emerging trends in the area of digital very large scale integration (VLSI) signal processing can lead to a reduction in the cost of the cochlear implant. Digital signal processing algorithms are repetitively used in speech processors for filtering and encoding operations. The critical paths in these algorithms limit the performance of the speech processors. These algorithms must be transformed to accommodate processors designed to be high speed and have less area and low power. This can be realized by basing the design of the auditory filter banks for the processors on digital VLSI signal processing concepts. By applying a folding algorithm to the second-order digital gammatone filter (GTF), the number of multipliers is reduced from five to one and the number of adders is reduced from three to one, without changing the characteristics of the filter. Folded second-order filter sections are cascaded with three similar structures to realize the eighth-order digital GTF whose response is a close match to the human cochlea response. The silicon area is reduced from twenty to four multipliers and from twelve to four adders by using the folding architecture.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2011년도 제42회 하계학술대회
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• pp.1794-1795
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• 2011

본 연구에서는 가속도 센서의 값을 디지털 신호 처리 과정을 통하여 저역통과 필터(low pass filter), 벡터의 크기(vector magnitude), 롤(roll) 그리고 피치(pitch)를 계산하는 알고리즘을 적용하였다. 필터의 경우 IIR(Infinite Impulse Response)을 이용하였으며 차수는 9차로 하였다. 피험자의 연령은 $25{\pm}5$세의 10명을 기준으로 실험하였으며 앞, 뒤, 좌, 우 방향으로 직각 낙하하도록 하였고 센서 모듈은 오른쪽 허리의 정중앙에 착용하도록 하여 피험자간의 오차가 발생하지 않도록 하였다. 환자의 낙상을 검출하기 위해서 벡터의 크기를 사용하였고 롤과 피치를 이용하여 환자의 낙상 방향을 검출하였다. 결과적으로 피험자 10명의 경우 낙상의 검출률은 100% 였으며 낙상 방향에 따른 앞, 뒤, 좌, 우 판별 정확도는 95% 정도이다. 낙상 방향의 판별은 사고 후 환자를 다룰 때의 주의할 신체부위를 참고하며 재활 운동 시 하체의 어느 쪽이 낙상의 주요인인지 분석하는 보조 자료가 될 수 있다.

• 한국정보과학회:학술대회논문집
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• 한국정보과학회 2003년도 가을 학술발표논문집 Vol.30 No.2 (3)
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• pp.115-117
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• 2003

ea­RED(Efficient Adaptive RED)[1][2] 라우터 버퍼 관리 알고리즘 성능 향상을 위해서 ea­RED 라우터 버퍼 사이즈 변화를 예측할 수 있는 예측 알고리즘 모듈의 추가 필요성을 느낀다. 그래서 본 논문에서는 ea­RED 라우터 버퍼 관리 알고리즘의 원형인 RED 라우터 버퍼 관리 알고리즘에 AR(AutoRegression Analysis), IIR(Infinite Impulse Response) MACD(Moving Average Convergence & Divergence), LR_Lines(Linear Regression Lines)등의 예측 알고리즘 모듈을 적용하여 변화를 살펴보고. 결과를 비교. 분석하여 ea­RED 라우터 버퍼 관리 알고리즘 성능 향상에 가장 적합한 예측 알고리즘으로 LR_Lines를 선정했다. ea­RED 라우터 버퍼 관리 알고리즘에 적합한 예측 알고리즘 선정을 위해서 RED 라우터 버퍼 관리 알고리즘을 대신 이용한 이유는 ea­RED 라우터 버퍼 관리 알고리즘의 경우 네트워크 상황에 따라, 버퍼 관련 파라미터 값을 수시로 바꾸기 때문에 예측 알고리즘의 정확성을 판단하는데 어려움이 있지만, RED 라우터의 경우는 버퍼 관련 파라미터 값을 변화시키지 않기 때문에, 좀 더 일관성 있고 정확한 분석을 수행할 수 있기 때문이다.