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Active Noise Control of Short Duct using Zero Acoustic Impedance Boundary (음향 임피던스 0의 경계면에 의한 짧은 덕트의 능동소음제어)

  • Cha, Kyung-Hwan;Lee, Chai-Bong;Kim, Chun-Duck
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.4
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    • pp.101-105
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    • 1997
  • The active noise control method that was developed for long duct has some problems to be applied for short duct. To overcome this problem, we apply the SISO(Single Input Single Output) algorithm for the active noise control of short duct using zero acoustic impedance boundary. The SISO algorithm can input noise signal and error signal with one microphone simultaneously. The real-time controller was implemented using TMS320E25 DSP(Digital Signal Processing) chip and it's performance was evaluated by experiment. As a result, we obtain total 4.7dBA noise reduction for 0.80m short duct.

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A convergence analysis of Block MADF algorithm for adaptive noise reduction

  • Min, Seung-gi;Young Huh;Yoon, Dal-hwan
    • Proceedings of the IEEK Conference
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    • 2002.07a
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    • pp.377-380
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    • 2002
  • When it calculates the optimum price of filter coefficient, the many operation quantity is necessary. Is like that the real-time control is difficult and the hardware embodiment expense is big. The case which does not know advance information of input signal or the case where the statistical nature changes with change of surroundings environment is necessary the adaptive filter. Every hour to change a coefficient automatically and system in order to reach to the condition of optimum oneself, the fact that is the adaptive filter. When it does not the quality of input signal or it does not know the environment of surroundings every hour changing, it does not emit not to be, in order to collect, the fact that is the adaptive filter. The case of the Acoustic Echo Canceler does thousands filter coefficients in necessity. It reduces a many calculation quantity to respect, it uses the IIR filter from hour territory. Also it uses the block adaptive filter which has a block input signal and a block output signal. The former there is a weak point where the stability discrimination is always demanded. Consequently, The block adaptive filter is researched plentifully. This dissertation planned the block MADF adaptive filter used to MADf algorithm.

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The Effects of Noise/Signal Ratios on Noise/Energy Source Identification in Linear Systems (선형계에 있어서의 잡음/신호비가 소음/진동원 규명에 미치는 영향)

  • 박정석;김광준;이종원
    • Transactions of the Korean Society of Mechanical Engineers
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    • v.15 no.6
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    • pp.1819-1830
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    • 1991
  • The problems associated with noise/energy source identification using multiple input/single output model in linear systems are investigated. Partial coherence function is formulated for the model introducing a virtual force and extraneous noises into the conventional two input/single output system. The analytical results show that the partial coherence function in two input/single output linear system is the function of noise/signal ratios when multiple inputs are mutually coherent and extraneous noises exist. Parametric studies for ordinary and partial coherence functions are carried out to demonstrate the effects of noise/signal ratios for these functions.

A Double Loop Control Model Using Leaky Delay LMS Algorithm for Active Noise Control (능동소음제어를 위한 망각형 지연 LMS 알고리듬을 이용한 이중루프제어 모델)

  • Kwon, Ki-Ryong;Park, Nam-Chun;Lee, Kuhn-Il
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.3
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    • pp.28-36
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    • 1995
  • In this paper, a double loop control model using leaky delay LMS algorithm are proposed for active noise control. The proposed double loop control model estimates the loudspeaker characteristic and the error path transfer function with on-line using only gain and acoustic time delay to reduce computation burden. The control of error signal through double loop control scheme makes the more robust cntrol system. The input signal of filter to estimate acoustic time delay is used difference between input signal of input microphone and adaptive filter output. And also, in nonstationary environments, the leaky delay LMS algorithm is employed to counteract parameter drift of delay LMS algorithm. For practical noise signal, the proposed double loop control model reduces noise level about 12.9 dB.

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Expandable Flash-Type CMOS Analog-to-Digital Converter for Sensor Signal Processing

  • Oh, Chang-Woo;Choi, Byoung-Soo;Kim, JinTae;Seo, Sang-Ho;Shin, Jang-Kyoo;Choi, Pyung
    • Journal of Sensor Science and Technology
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    • v.26 no.3
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    • pp.155-159
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    • 2017
  • The analog-to-digital converter (ADC) is an important component in various fields of sensor signal processing. This paper presents an expandable flash analog-to-digital converter (E-flash ADC) for sensor signal processing using a comparator, a subtractor, and a multiplexer (MUX). The E-flash ADC was simulated and designed in $0.35-{\mu}m$ standard complementary metal-oxide semiconductor (CMOS) technology. For operating the E-flash ADC, input voltage is supplied to the inputs of the comparator and subtractor. When the input voltage is lower than the reference voltage, it is outputted through the MUX in its original form. When it is higher than the reference voltage, the reference voltage is subtracted from the input value and the resulting voltage is outputted through the MUX. Operation of the MUX is determined by the output of the comparator. Further, the output of the comparator is a digital code. The E-flash ADC can be expanded easily.

ECG Identification Method Using Adaptive Weight Based LMSE Optimization (적응적 가중치를 사용한 LMSE 최적화 기반의 심전도 개인 인식 방법)

  • Kim, Seok-Ho;Kang, Hyun-Soo
    • The Journal of the Korea Contents Association
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    • v.15 no.4
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    • pp.1-8
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    • 2015
  • This paper presents a Electrocardiogram(ECG) identification method using adaptive weight based on Least Mean Square Error(LMSE) optimization. With a preprocessing for noise suppression, we extracts the average ECG signal and its standard deviation at every time instant. Then the extracted information is stored in database. ECG identification is achieved by matching an input ECG signal with the information in database. In computing the matching scores, the standard deviation is used. The scores are computed by applying adaptive weights to the values of the input signal over all time instants. The adaptive weight consists of two terms. The first term is the inverse of the standard deviation of an input signal. The second term is the proportional one to the standard deviation between user SAECGs stored in the DB. Experimental results show up to 100% recognition rate for 32 registered people.

Implementation of sigma-delta A/D converter IP for digital audio

  • Park SangBong;Lee YoungDae
    • Proceedings of the IEEK Conference
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    • summer
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    • pp.199-203
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    • 2004
  • In this paper, we only describe the digital block of two-channel 18-bit analog-to-digital (A/D) converter employing sigma-delta method and xl28 decimation. The device contains two fourth comb filters with 1-bit input from sigma­delta modulator. each followed by a digital half band FIR(Finite Impulse Response) filters. The external analog sigma-delta modulators are sampled at 6.144MHz and the digital words are output at 48kHz. The fourth-order comb filter has designed 3 types of ways for optimal power consumption and signal-to-noise ratio. The following 3 digital filters are designed with 12tap, 22tap and 116tap to meet the specification. These filters eliminate images of the base band audio signal that exist at multiples of the input sample rate. We also designed these filters with 8bit and 16bit filter coefficient to analysis signal-to-noise ratio and hardware complexity. It also included digital output interface block for I2S serial data protocol, test circuit and internal input vector generator. It is fabricated with 0.35um HYNIX standard CMOS cell library with 3.3V supply voltage and the chip size is 2000um by 2000um. The function and the performance have been verified using Verilog XL logic simulator and Matlab tool.

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A Reconsideration of the Causality Requirement in Proving the z-Transform of a Discrete Convolution Sum (이산 Convolution 적산의 z변환의 증명을 위한 인과성의 필요에 대한 재고)

  • Chung Tae-Sang;Lee Jae Seok
    • The Transactions of the Korean Institute of Electrical Engineers D
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    • v.52 no.1
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    • pp.51-54
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    • 2003
  • The z-transform method is a basic mathematical tool in analyzing and designing digital signal processing systems for discrete input and output signals. There are may cases where the output signal is in the form of a discrete convolution sum of an input function and a designed digital processing algorithm function. It is well known that the z-transform of the convolution sum becomes the product of the two z-transforms of the input function and the digital processing function, whose proofs require the causality of the digital signal processing function in the almost all the available references. However, not all of the convolution sum functions are based on the causality. Many digital signal processing systems such as image processing system may depend not on the time information but on the spatial information, which has nothing to do with causality requirement. Thus, the application of the causality-based z-transform theorem on the convolution sum cannot be used without difficulty in this case. This paper proves the z-transform theorem on the discrete convolution sum without causality requirement, and make it possible for the theorem to be used in analysis and desing for any cases.

Design of an Adaptive Filter for Noise Cancdlation of ECG's (심전도 신호의 잡음 제거를 위한 적응 필터 설계)

  • 이재준;송철규
    • Journal of Biomedical Engineering Research
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    • v.13 no.2
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    • pp.107-114
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    • 1992
  • An adaptive filter for noise cancellation of ECG Is proposed. An adaptive noise canceller using the least mean squares algorithm Is used to reduce unwanted noise. An adaptive filter for nolse cancella lion minimizes the mean-square error between a primary input and a reference input. A primary input is the noisy ECG, and a reference input is a noise that Is correlated in some way with the noise in the primary input or a signal that is correlated only with ECG in the primary input.

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LONGITUDINAL WAVES, STORING AND AMPLIFYING CAPABILITY OF INFORMATION IN WATER MOLECULES AND QUANTUM RESONANCE SPECTROMETER

  • Oh, Hung-Kuk
    • Proceedings of the Korea Society for Energy Engineering kosee Conference
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    • 1996.10b
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    • pp.18-28
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    • 1996
  • The outer-most electrons of metal atoms and the remaining valence electrons of any molecular atoms make three-dimensional crystallizing $\pi$-bondings. The rotating electrons on the three-dimensional crystallizing $\pi$-bonding orbitals of atoms make $\pi$-far infrared rays. Longitudinal wave is a propagation of a bundle of $\pi$-far infrared rays, which are produced by a dynamic impact on a solid bar. The $\pi$-far infrared rays make three-dimensional crystallizing $\pi$-bondings in the material, which reproduce the same $\pi$-far infrared rays. If a current signal is input into water molecules under a given electric potential field with $\pi$-far infrared rays (input information), the signal can be amplified because the $\pi$-far infrared rays make the $\pi$-bondings, which reduce electric resistance. The three-dimensional crystallizing $\pi$-bondings can induce normal electrons to move from one orbital to next one with a aid of potential electric field. Quantum Resonance Spectrometer is composed of tesla coil absorbing $\pi$-far infrared rays, tesla coil emitting varying electromagnetic waves signal generator, signal storage, human body amplifier, signal analyzer and data indicator. The absorbing tesla coil making varying magnetic field and downward and upward electric field, which resonates the $\pi$-far infrared rays coming out from specimen and absorbs them. The modulated current signal from the input square signal can generate and emit varying electromagnetic waves from the tesla coil. The varying electro-magnetic waves make the three-dimensional crystallizing $\pi$-bondings and the $\pi$-far infrared rays in the water molecules.

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