• Journal of KSNVE
• /
• v.5 no.4
• /
• pp.453-460
• /
• 1995

본 논문에서는 정현파 소음을 제어하기 위한 filtered-x LMS에 바탕을 둔 새로운 적응 알고리듬을 제안하였다. 이러한 알고리듬은 두개의 연속적인 계수조정 식으로, 제어기의 계수를 조정한다. 서로 독립인 각 주파수별로 처리하기 때문에 빠른 수렴을 얻을 수 있다. 두번째식은 이차경로로 인한 위상지연을 추정한다. 정현 파 신호 주파수보다 4배 이상 빠른 표본화 주파수를 선택하여 추정된 위상지연 추정 값은 $2{\pi}f_0$만큼 오차를 나타내며, 이 값은 $\pi$2보다 작다. 정현파 신호의 주파수를 알면 이러한 오차는 $2{\pi}f_0$를 더함으로써 제거할 수 있다. 이러한 방법은 위상지연이 $\pi$2보다 큰 경우 수렴속도를 증가시킨다는 사실을 실험을 통하 여 알 수 있다. 추정된 위상지연은 제어기 계수값을 조정하는데 필요한 필터링된 참조신호를 발생시키믄데 사용된다. 참조신호의 위상지연이 각 주파수 성분별로 수행 되기 때문에, 콘볼루션 연산이 생략되어 계산량을 줄일 수 있다. 또한 연속적으로 위상지연을 추정하기 때문에 시변 상황에 적용이 가능하다. 조정식의 수렴조건을 유도하였다. 제안된 알고리듬은 제어기 계수를 추정하는데 바이어스가 없으며, 위상 지연추정을 위한 수렴상수의 최대허용치는 제어기계수에 대한 수렴상수에 반비례함을 이론적으로 분석을 통해 알 수 있다. 모의실험을 통하여 제안된 알고리듬이 filtered-x LMS 알고리듬에 바탕을 둔 다른 알고리듬보다 환경변화에 우수한 성능을 보임을 알 수 있다.

• Proceedings of the KIEE Conference
• /
• 2002.07d
• /
• pp.2055-2057
• /
• 2002

본 논문에서는 외란 및 파라미터 변동에 강건한 시간지연 제어 알고리즘을 전기식 구동장치의 위치 제어기에 적용하여 설계한 내용을 기술하였다. 시간지연 제어기는 일정하게 경과된 시간에서의 외란 및 동특성 변동의 양을 현재의 것으로 가정하여 전체 제어 시스템의 특성이 기준모델의 특성을 따르게 하는 제어기이다. 파라미터 변동 및 외란에 대해 시간지연 제어기의 강건성을 시뮬레이션을 통해 입증하였으며 PID 제어기와도 비교하였다. 또한 포화요소에 대한 와인드업 방지보상기와 모델링되지 않는 동특성에 대한 대향필터링 보상기를 설계하여 시스템의 안정성을 향상시켰다.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.21 no.7
• /
• pp.1261-1266
• /
• 2017

A random and systematic jitter suppressed delay locked loop (DLL)-based clock generator with a delay-time voltage variance converter (DVVC) and an averaging circuit (AC) is presented. The DVVC senses the delay variance of each delay stage and generates a voltage. The AC averages the output voltages of two consecutive DVVCs to suppress the systematic and random delay variance of each delay stage in the VCDL. The DVVC and AC averages the delay time of successive delay stages and equalizes the delay time of all delay stages. In addition, a capacitor with a switch working effectively as a negative feedback function is introduced to reduce the variation of the loop filter output voltage. Measurement results of the DLL-based clock generator fabricated in a one-poly six-metal $0.18{\mu}m$ CMOS process shows 13.4-ps rms jitter.

• The Journal of the Acoustical Society of Korea
• /
• v.18 no.7
• /
• pp.45-55
• /
• 1999

In this paper, we used the WLMS(Wavelet domain Least Mean Square) adaptive filter based on the wavelet transform to cancel grain noise. Usually, grain noise occurs in changes of the crystalline structure of metals in high temperature environment. It makes the detection of flaw difficult. The WLMS adaptive filtering algorithm establishes the faster convergence rate by orthogonalizaing the input vector of adaptive filter as compared with that of LMS adaptive filtering algorithm in time domain. We implemented the WLMS adaptive filter by using the delayed version of the primary input vector as the reference input vector and then implemented the CA-CFAR(Cell Averaging- Constant False Alarm Rate) threshold estimator. CA-CFAR threshold estimator enables to detect the flaw and back echo signals automatically. Here, we used the output signals of adaptive filter as its input signal. To Cow the statistical characteristic of ultrasonic signals corrupted by grain noise, we performed run test. The results showed that ultrasonic signals are nonstationary signal, that is, signals whose statistical properties vary with time. The performance of each filter is appreciated by the signal-to-noise ratio. After LMS adaptive filtering in time domain, SNR improves to about 2-3㏈ but after WLMS adaptive filtering in wavelet domain, SNR improves to about 4-6㏈.

• Journal of the Institute of Electronics Engineers of Korea SC
• /
• v.39 no.1
• /
• pp.76-84
• /
• 2002

In this paper, it is studied to remove the baseline wander and to minimize the distortion of ST segment in the noise filtering of ECG. In general, the standard filter and adaptive filter are used to remove the baseline wander of the ECG. But the standard filter is limited because the frequency of the baseline signal is variable and the apative filter is difficult to select the reference signal in case of using the adaptive filter. So we proposed a new method of the structure without reference signal using neural networks. To be convinced of the performance of this method, we used ECG data of MIT-BIHs. and obtained the result of the high performance,(-53.3[dB]) than standard filter(-16.3[dB]) and adaptive filter (-44.9[dB]).

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.18 no.9
• /
• pp.1248-1256
• /
• 1993

The propagation and delay properties in opical fiber are particularly attractive because digital signal processing and conventional analog signal processing techniques such as those using surface acoustic wave devices are limited In their usefulness for signal bandwidth exceeding one or two GHz, although they are very effective at lower frequencies. Since an accurate, low loss and short time delay elements can be obtained by using such an optical fiber, optical signal precessing has attracted much attention for high speed and broad-band signal precessing in particular channel separation filtering for optical FDM signals. In this paper, we consider a coherent optical lattice filter, which uses coherent light sources and consists of directional couplers and optical fiber delay elemnts. The optical fiber fitters are more restricted than the usual digital filters. The reasons are as follows. 1) the coupling coefficients of directional couplers are restricted to the number between 0 and 1. 2) optical signal E(complex amplitude) is divided into J If-$\boxUl$ and J L/7$\div$$\boxUl$ at the directional coupler. Considering these restrictions and in this case all the coupling coefficients of summing and branching elements are set to be equal, we have given design formulae for optical lattice filter, which make the best use of optical signal energy.

• Journal of the Korean Institute of Telematics and Electronics D
• /
• v.35D no.10
• /
• pp.39-50
• /
• 1998

Wide-band sharp-transition filters are widely used in applications such as wireless CODEC design or medical systems. Since these filters suffer from large sensitivity and roundoff noise, large word-length is required for the VLSI implementation, which increases the hardware size and the power consumption of the chip. In this paper, a low-power implementation technique for digital filters with wide-band sharp-transition characteristics is proposed using CPL (Complementary Pass-Transistor Logic), LWDF (Lattice Wave Digital Filter) and a modified DIFIR (Decomposed & Interpolated FIR) algorithm. To reduce the short-circuit current component in CPL circuits due to threshold voltage reduction through the pass transistor, three different approaches can be used: cross-coupled PMOS latch, PMOS body biasing and weak PMOS latch. Of the three, the cross-coupled PMOS latch approach is the most realistic solution when the noise margin as well as the energy-delay product is considered. To optimize CPL transistor size with insight, the empirical formulas for the delay and energy consumption in the basic structure of CPL circuits were derived from the simulation results. In addition, the filter coefficients are encoded using CSD (Canonic Signed Digit) format and optimized by a coefficient quantization program. The hardware cost is minimized further by a modified DIFIR algorithm. Simulation result shows that the proposed method can achieve about 38% reductions in power consumption compared with the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.39C no.5
• /
• pp.418-424
• /
• 2014

We propose and experimentally demonstrate a photonic microwave notch filter with a negative coefficient. The negative coefficient is obtained by using cross gain modulation (XGM) in a reflective semiconductor optical amplifier (RSOA). When the RSOA is operated in saturated region, the signal carried on the pump wavelength is inversely copied to the probe wavelength by the XGM effect, showing a negative coefficient. Time delay between pump signal and probe signal is provided by single mode fiber (SMF) with wavelength dependent time delay. The proposed microwave notch filter shows notch dips of more than 35.1 dB and free spectral range (FSR) of about 380.6 MHz.

• Journal of the Korean Institute of Intelligent Systems
• /
• v.24 no.6
• /
• pp.628-633
• /
• 2014

This paper proposes a performance improvement method of the conventional double talk detection method which can operate before convergence of the echo canceller. The proposed method estimates the coefficients of the linear predictive coding(LPC) filter by using the primary input signal. The time-varying threshold for double talk detection is determined based on the LPC filter gain of the primary input signal level. The proposed method can reduce not only false detection rate which means wrong detection of single talk as double talk but also double talk detection delay. Computer simulation was performed using a long-term real speech signals. It is shown that the proposed method improves the conventional method in terms of lowering the false detection rate and shortening the detection delay.

• The Journal of the Acoustical Society of Korea
• /
• v.22 no.8
• /
• pp.680-686
• /
• 2003

Digital waveguide model is a general method that is used in physical modeling of musical instruments. Wave motion is analyzed by time or by space in digital waveguide model. Because sampling is made via time, it is general that musical instrument model is described by wave motion of time. In this paper, we synthesized the musical instrument sound by adding instrument body model to the spatial based string model. In this way, we could improve sound quality and process musical instrument model's tone control variables effectively. We explained about delay error that happens in string and body in space based sampling and showed method to process fractional delay using FD (Fractional Delay)filter. Finally, we explained the relation between tone quality and number of delays. And we also compared the result with time base digital waveguide model.