• The Transactions of The Korean Institute of Electrical Engineers
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• v.58 no.11
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• pp.2247-2252
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• 2009

Since the dimensions of MEMS gyroscope are very small compared to those of conventional gyroscope, MEMS gyroscope should be able to measure charge of pico-coulomb caused by very small change of electrodes gap. However, feed-through signal from driving electrodes to the sensing electrodes due to the electromagnetic coupling is much greater than the sensing signal, which degrades the sensitivity of MEMS gyroscope. This paper introduces the feed-through noise canceling technique using dummy port and confirms the feasibility of feed-through noise canceling experimentally. Experimental results shows that, when driving signal is 6 Vpp, 30 kHz, feed-through signal of vacuum packaged Si Gyroscope decreases from -53.2 dBm to -77.1 dBm by using feed-through reduction technique. Q-factor that could not be measured without noise reduction is measured to be about 2500 and resonance frequency to be 7.018 kHz.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.7
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• pp.33-41
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• 1994

A single channel adaptive noise canceling (ANC) technique with a recursive time delay estimator (RTDE) is presented for removing effects of additive noise on the speech signal. While the conventional method makes a reference signal for the adaptive filter using the pitch estimated on a frame basis from the input speech, the proposed method makes the reference signal using the delay estimated recursively on a sample-by-sample basis. As the RTDEs, the recursion formulae of autocorrelation function (ACF) and average magnitude difference function (AMDF) are derived. The normalized least mean square (NLMS) and recursive least square (RLS) algorithms are applied for adaptation of filter coefficients. Experimental results with noisy speech demonstrate that the proposed method improves the perceived speech quality as well as the signal-to-noise ratio and cepstral distance when compared with the conventional method.

• Journal of KSNVE
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• v.10 no.6
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• pp.941-948
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• 2000

Active noise control is an approach to reduce the noise by utilizing a secondary noise source that destructively interferes with the unwanted noise. In general, active noise control systems rely on multiple sensors to measure the unwanted noise field and the effect of the cancellation. This paper develops an approach that utilizes a single sensor. The noise field is modeled as a stochastic process, and an adaptive algorithm is used to adaptively estimate the parameters of the process. Based on these parameter estimates, a canceling signal is generated. Oppenheim assumed that transfer function characteristics from the canceling source to the error sensor is only a propagation delay. This paper proposes a modified Oppenheim algorithm by considering transfer characteristics of speaker-path-sensor This transfer characteristics is adaptively cancelled by the proposed adaptive modeling technique. Feasibility of the proposed method is proved by computer simulations with artificially generated random noises and sine wave noise. The details of the proposed architecture. and theoretical simulation of the noise cancellation system for three dimension enclosure are presented in the Paper.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.4
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• pp.897-904
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• 2011

In this paper, a novel circuit topology of a low-noise amplifier tunable at 1.8GHz band for PCS and 2.4GHz band for WLAN using a CMOS active inductor is proposed. This circuit topology to reduce higher noise figure of the low noise amplifier with the CMOS active load is analyzed. Furthermore, the noise canceling technique is adopted to reduce more the noise figure. The noise figure of the proposed circuit topology is analyzed and simulated in $0.18{\mu}m$ CMOS process technology. Thus, the simulation results exhibit that the noise performance enhancement of the tunable low noise amplifier is about 3.4dB, which is mainly due to the proposed new circuit topology.

• Proceedings of the KIEE Conference
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• 1992.07a
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• pp.323-326
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• 1992

In this paper, the active noise control(ANC) system using the inverse modeling techiniques is presented. The nonlinearity and time delay of the transfer function from the secondary speaker to the error microphone makes the ANC system have poor performance. To solve this problem, the inverse model technique and filtered-X LMS algorithm is using proposed. This proposed ANC system is implemented using DSP chip and operated in on-line. The experimental results show that this ANC system has better noise canceling performance than that used LMS only about 5-15[db]

• 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 and Information Engineers
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• v.49 no.12
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• pp.242-246
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• 2012

This paper presents an integrated low noise and highly linear wideband RF front-end for a digital terrestrial and cable TV tuner, which are used as a part of double-conversion TV tuner. The low noise amplifier (LNA) has a low noise figure and high linearity by adopting a noise canceling technique based on current amplification. The up-conversion mixer and SAW buffer have high linearity by employing a third order intermodulation cancellation technique. The proposed RF front-end is designed in a $0.18{\mu}m$ CMOS and draws 60 mA from a 1.8 V supply voltage. The RF front-end shows a voltage gain of 30 dB, an average single side-band noise figure of 4.2 dB, an IIP2 of 40 dBm, and an IIP3 of -4.5 dBm for the entire band from 48 MHz to 862Hz.

• Journal of the Korea Institute of Military Science and Technology
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• v.24 no.5
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• pp.545-557
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• 2021

Engine is the main source of vibration that generates unwanted noise and vibration of vehicle chassis. Especially, in submarine applications, radiation of noise signatures can be detected at some distance away from the submarine using a sonar array. Thus quiet operation is crucial for submarine's survivability. This study addresses reduction of the force transmissibility originating from engines and transmitted to hull through engine mounts. An inertial damper, as an actuator of hybrid mount system, is addressed to reduce even further the level of vibration. Narrow band FxLMS algorithms are broadly used to cancel the vibration of engine mount because of its excellent performance of canceling narrow band noise. However, in real active dampers, the maximum displacement of damper mass is kinematically restricted. When the control input signal from the FxLMS algorithm exceeds this limitation, the damper mass will collide with the mechanical stops and results in many problems. Originated from these, a modified narrow band FxLMS algorithm based on the equalizer technique with the maximum allowable displacement of active damper mass is proposed in this study. Some simulation results showed that the propose algorithm is effective to suppress vibration of engine mount while ensuring given displacement constraint.

• Proceedings of the KIEE Conference
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• 2009.07a
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• pp.1503_1504
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• 2009

MEMS 구조물은 ${\mu}m$단위의 크기로 만들어지므로 각속도계와 같이 정밀한 센서를 만들 때에는 노이즈 문제를 해결하지 않으면 신호를 측정할 수가 없다. MEMS 구조물의 미세한 진동에 의해 발생되는 수 pico-coulomb의 전하를 측정해야하므로 구동 신호가 검출 전극에서 Feed-through되어 나타나는 경우 그 크기가 구동에 의한 신호보다 100배 이상 크기 때문에 원하는 신호를 검출할 수 없다. 본 논문에서는 이러한 Feed-through 현상에 의한 노이즈를 줄이기 위하여 Guard-ring을 이용한 blocking 방법과 dummy port를 이용한 canceling 방법을 고안하고 Feed-through reduction 회로를 설계, 제작, 실험하여 그 효과를 확인하였다. 그 결과 구동신호가 6Vpp, 30kHz일 때, -53.186dBm이었던 Feed-through 신호가 -77.107dBm으로 줄어드는 것을 확인하였다. 또한 노이즈를 제거하지 않은 경우 측정할 수 없었던 Q-factor를 Feed-through reduction 회로를 사용하여 측정한 결과 진공 패키징된 Si 기반 자이로스코프가 공진주파수 약 7.018kHz에서 Q-factor가 약 2500임을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1082-1091
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• 2000

In this paper, the efficient equalization method for OFDM(Orthogonal Frequency Division Multiflexing) System using the QAM(Quadrature Amplitude Modulation) in multipath fading channel is proposed in order to faster and more efficiently equalize the received signals that are sent over real channel. In generally, the one-tap linear equalizers have been used in the frequency-domain as the existing equalization method for OFDM system. In this technique, if characteristics of the channel are changed fast, the one-tap linear equalizers cannot compensate for the distortion due to time variant multipath channels. Therefore, in this paper, we use one-tap non-linear equalizers instead of using one-tap linear equalizers in the frequency-domain, and also use the linear equalizer in the time-domain to compensate the rapid performance reduction at the low SNR(Signal-to-Noise Ratio) that is the disadvantage of the non-linear equalizer. In the frequency-domain, when QAM signals, consisting of in-phase components and quadrature (out-phase) components, are sent over the complex channel, the only in-phase and quadrature components of signals distorted by the multipath fading are changed the same as signals distorted by the noise. So the cross components are canceled in the frequency-domain equalizer. The time-domain equalizer and the adaptive algorithm that has lower-error probability and fast convergence speed are applied to compensate for the error that is caused by canceling the cross components in the frequency-domain equalizer. In the time-domain, To compensate for the performance of frequency-domain equalizer the time-domain equalizes the distorted signals at a frame by using the Gold-code as a training sequence in the receiver after the Gold-codes are inserted into the guard signal in the transmitter. By using the proposed equalization method, we can achieve faster and more efficient equalization method that has the reduced computational complexity and improved performance.