• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.46 no.3
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• pp.244-250
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• 2018

Multilateration acquires the transponder signal of target from receivers installed on the ground and calculates the position of the target using the difference of the signal acquisition time of each receiver. One of the factors that influence the positioning accuracy of Multilateration using the TDOA calculation method is the error due to the precision measurement of signal input time. When measuring the signal input time at the receiver, the input signal is sampled using the reference clock of the receiver and a reference sample having the same sampling rate is applied to the cross correlation technique. Therefore, the accuracy of the signal input time is proportional to the reference clock. In this paper, the algorithm for precisely measuring the signal input time by performing cross correlation between the input signal of the receiver and DRS(Delayed Reference Sample) is proposed. In order to verify this, we implemented the pulse signal of the transponder that is transmitted from the target using Matlab. Through the simulation, cross correlation between the proposed DRS and the input signal was performed. From this result, the performance of the precise measurement of signal input time was analyzed.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.293-298
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• 1999

본 논문은 수중음의 안정적 실시간 인식을 위한 새로운 음원 인식 알고리즘을 다루고 있다 본 논문에서 이용된 주파수인식 알고리즘은 크게 네 부분으로 구성되어 있는데 1)입력된 음파 신호를 duty cycle $50\%$의 디지털 신호로 바꾸고 기준 주파수의 음원을 duty cycle $50\%$, 위상차 0도 90도 180도 270도의 디지털 신호를 생성하는 부분, 2)입력된 음파신호를 4가지 위상의 각 기준신호와 배타적 논리합을 구하는 부분, 3)두 번째에서 만들어진 각 신호를 적분회로에 통과시키는 부분, 4)세 번째에서 발생한 각 신호중 최대값을 추출하여 입력된 음파신호의 주파수를 인식하는 부분으로 이루어져 있다. 이 회로에 대한 수치 해석을 통하여 각 부분의 특성치에 대한 최적 값 및 성능을 검증하였으며, 이의 결과를 각각 computer 수치 시험, 실제 회로 실험과 비교하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.1
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• pp.115-121
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• 2000

Command shaping is an important method to reduce vibration in flexible beam. This paper presents a very simple command control shaping which eliminates multiple mode residual vibration in a flexible beam in finite time. The command is constructed by solving linear equations. The finite time duration in which the desired motion of joint angle is achieved along with elimination of the residual vibration can be arbitrarily specified. The necessary conditions for using command as a reference input for the joint angle in a closed-loop configuration are also discussed. The effectiveness of Proposed scheme is demonstrated through computer simulation.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.48 no.11
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• pp.41-47
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• 2011

A novel digital instrumentation amplifier(DIA) converting universal signal inputs into 5V voltage-output for industry standard sensor signal processing was designed. The circuit consists of a commercial instrumentation amplifier, seven analog switches, two voltage references of 1.0V and -10.0V, and four resistors. The converting principle is the circuit reconstruction by switches for resistor values and reference voltages according to input signals. The simulation result shows that the DIA has a good output voltage characteristics of 0~5V for the input voltage of 0V~5V, 1V~5V, -10V~+10V, and 4mA~20mA. The nonlinearity error was less than 0.1% for the four type signal inputs.

• Proceedings of the IEEK Conference
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• 2003.07b
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• pp.1129-1132
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• 2003

본 논문에서는 위상/주파수 검출기을 설계시 문제가 되는 Reference Spur을 없게 하여 Low Noise를 구현할 수 있는 No Spike PFD(Phase Frequency Detector)를 제안한다. 위상동기루프의 특별한 형태로 차지 펌프 위상동기루프가 있다. 차지 펌프위상동기 루프는 일반적으로 3-state 위상/주파수 검출기를 이용한다. 이 3-state 위상/주파수 검출기는 기준 신호와 VCO 출력 신호의 위상차에 비례하는 디지털 파형으로 출력을 내보낸다. 차지 펌프 위상동기루프 그림 1 처럼 디지털 위상/주파수 검출기(PFD), 차지 펌프(CP), 루프 필터(LF), VCO로 구성된다. PFD 는 기준 신호와 VCO 에 의해 만들어진 출력 신호를 입력받아 각각의 위상과 주파수를 비교한다. 즉, 출력 신호가 기준 신호보다 느릴 때에는 출력 신호를 앞으로 당기기 위해서 up 신호를 넘겨주고, 출력 신호가 기준 신호보다 빠를 때에는 출력 신호를 뒤로 밀기 위해 down 신호를 넘겨준다. 차지 펌프(CP)의 전류를 Ip 라고 한다면, CP 에서 LF 로 흐르거나, LF에서 CP로 흐르는 전류 Ip의 평균량이 기준 신호와 VCO 출력 신호의 위상차에 비례하는 것이다.

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

• The Journal of the Acoustical Society of Korea
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• v.13 no.3
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• pp.16-24
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• 1994

A single channel adaptive noise canceling (ANC) technique is presented for removing effects of additive noise on the speech signal. The conventional method obtains a reference signal using the pitch estimated on a frame basis from the input speech. The proposed method, however, gets the reference signal using the delay estimated recursively on a sample by sample basis. To estimate the delay, we derive recursion formula of autocorrelation function and average magnitude difference function. The performance of the proposed method is evaluated for the speech signals distorted by the additive white Gaussian noise. Experimental results with normalized least mean square (NLMS) adaptive algorithm demonstrate that the proposed method improves the perceived speech quality quite well besides the signal-to-noise ratio.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.48 no.4
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• pp.80-85
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• 2011

A time-pickoff circuit used for time measurement suffers from a timing error due to the dependence of the generation time of a timing pulse on the size of the input signal, i.e., time walk. In this study, a time-pickoff method, which employs an automatic gain control (AGC) circuit, is proposed for reducing the timing error. The AGC circuit is added to the input of the comparator, and it renders the sizes of input signals of the comparator relatively uniform. The performance of the proposed time-pickoff method is analyzed using the SPICE simulation, and experiments are performed to confirm the analytical results. The measured time walk is reduced to 2.000 ns by 65% for input signals with a dynamic range of 20 dB as compared to a typical leading-edge discriminator.

• Proceedings of the KIEE Conference
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• 2000.07d
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• pp.3115-3117
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• 2000

기존의 LMS 알고리듬을 이용한 적응필터에 비해 연산횟수를 줄이고 입력신호의 통계적 특성에 덜 민감한 적응필터를 제안한다. 입력 신호와 기준신호에 대한 고속 하다마드 변환을 수행한 후 하다마드 변환 영역에서 LMS 알고리듬을 적용한다 기존의 적응필터와 비교하여 필터의 입력신호 추정 성능은 유지하면서 고속 하다마드 변환으로 인해 적응과정에서의 곱셈연산이 크게 줄어드며 잡음의 분산값 변화와 같은 입력신호의 변화에 대한 필터의 안정도와 강인성이 크게 향상됨을 보인다.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1004-1016
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• 2007

In this paper we proposed a Weight-Update-Control Adaptive Noise Canceller which improves speech when environmental noise is stationary and it is hard to acquire a reference signal. Adaptive Noise Canceller(ANC) needs a reference signal, but it is not easy to measure pure noise without voice for reference in factory. Because there are mixed various mechanical noise and workers' voice. Therefore ANC is not suitable to reduce background noise. So we proposed the method that uses an arbitrary constant as an input signal and inputs microphone signal to the reference signal. The noise is eliminated using updated weights in non-speech range. In speech range the weight is fixed and the modified voice is acquired then voice is restored through transversal filter. The proposed method is based on facts that the factory noise is stationary and the noise is not changed in short conversation range. As a result of simulation using MATLAB, we confirmed that the proposed method is effective for reducing factory noise and has high signal to noise ratio(SNR).