• Title/Summary/Keyword: 주파수 수렴

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Inductance Extraction of Microstrip Lines using Adaptive PEEC Grid (적응 PEEC 격자를 이용한 마이크로스트립의 인덕턴스 계산)

  • Kim, Han;Ahn, Chang-Hoi
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.14 no.8
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    • pp.823-829
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    • 2003
  • For high frequency microstrip line modelling, a fast inductance extraction technique using an adaptive PEEC(partial element equivalent circuit) grid is proposed. The grid refinement technique is based on the current distribution depend on the excitation frequencies and the geometry of the microstrip lines. The adaptive ids are refined mainly in the area where heavy currents reside. This technique is applied to the inductance extraction of the microstrip lines. The results show fast convergence, and this adaptive technique is efficient to reduce computing time and the number of grids.

A Study on the Generation and Processing of Depth Map for Multi-resolution Image Using Belief Propagation Algorithm (신뢰확산 알고리즘을 이용한 다해상도 영상에서 깊이영상의 생성과 처리에 관한 연구)

  • Jee, Innho
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.15 no.6
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    • pp.201-208
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    • 2015
  • 3D image must have depth image for depth information in order for 3D realistic media broadcasting. We used generally belief propagation algorithm to solve probability model. Belief propagation algorithm is operated by message passing between nodes corresponding to each pixel. The high resolution image will be able to precisely represent but that required much computational complexity for 3D representation. We proposed fast stereo matching algorithm using belief propagation with multi-resolution based wavelet or lifting. This method can be shown efficiently computational time at much iterations for accurate disparity map.

A Method of Learning and Recognition of Vowels by Using Neural Network (신경망을 이용한 모음의 학습 및 인식 방법)

  • Shim, Jae-Hyoung;Lee, Jong-Hyeok;Yoon, Tae-Hoon;Kim, Jae-Chang;Lee, Yang-Sung
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.27 no.11
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    • pp.144-151
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    • 1990
  • In this work Ohotomo et al., neural network model for learning and recognizing vowels is modified in order to reduce the time for learning and the possibility of incorrect recognition. In this modification, the finite bandwidth of formant frequencies of vowels are taken into consider-ations in coding input patterns. Computer simulations show that the modification reduces not only the possibility of incorrect recognition by about $30{\%}$ but also the time for learning by about $7{\%}$.

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Implementation of the 155.52 MHz Clock Recovery Receiver for the Fiber Optic Modules (광통신 모듈용 155.52 MHz 클럭복원 리시버의 구현)

  • 이길재;채상훈
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.26 no.12C
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    • pp.249-254
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    • 2001
  • A receiver ASIC for fiber optic modules of STM-1 optical communication has been fabricated with 0.65 $\mu\textrm{m}$ CMOS technology. The ASIC has a limit amplifier circuit for the 155.52 Mbps data reshaping, and a clock extraction circuit for the 155.52 MHz clock recovery. The ASIC has an acquisition aid and LOS monitoring circuit for properly operation with near 155.52 MHz clock frequency in case of the data loss due to transmission line open or data transfer fail. Measured results show that the circuit reshapes data from 5 mV to 1 V wide range of input voltage condition, add it recovers system clock with stable on any condition.

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Blind Signal Separation Using Eigenvectors as Initial Weights in Delayed Mixtures (지연혼합에서의 초기 값으로 고유벡터를 이용하는 암묵신호분리)

  • Park, Jang-Sik;Son, Kyung-Sik;Park, Keun-Soo
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.1
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    • pp.14-20
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    • 2006
  • In this paper. a novel technique to set up the initial weights in BSS of delayed mixtures is proposed. After analyzing Eigendecomposition for the correlation matrix of mixing data. the initial weights are set from the Eigenvectors ith delay information. The Proposed setting of initial weighting method for conventional FDICA technique improved the separation Performance. The computer simulation shows that the Proposed method achieves the improved SIR and faster convergence speed of learning curve.

Computational Efficiency on Frequency Domain Analysis of Large-scale Finite Element Model by Combination of Iterative and Direct Sparse Solver (반복-직접 희소 솔버 조합에 의한 대규모 유한요소 모델의 주파수 영역 해석의 계산 효율)

  • Cho, Jeong-Rae;Cho, Keunhee
    • Journal of the Computational Structural Engineering Institute of Korea
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    • v.32 no.2
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    • pp.117-124
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    • 2019
  • Parallel sparse solvers are essential for solving large-scale finite element models. This paper introduces the combination of iterative and direct solver that can be applied efficiently to problems that require continuous solution for a subtly changing sequence of systems of equations. The iterative-direct sparse solver combination technique, proposed and implemented in the parallel sparse solver package, PARDISO, means that iterative sparse solver is applied for the newly updated linear system, but it uses the direct sparse solver's factorization of previous system matrix as a preconditioner. If the solution does not converge until the preset iterations, the solution will be sought by the direct sparse solver, and the last factorization results will be used as a preconditioner for subsequent updated system of equations. In this study, an improved method that sets the maximum number of iterations dynamically at the first Krylov iteration step is proposed and verified thereby enhancing calculation efficiency by the frequency domain analysis.

Feasibility of hearing aid gain self-adjustment using speech recognition (말소리 인지를 이용한 보청기 이득 자가 조절의 실현)

  • Yun, Donghyeon;Shen, Yi;Zhang, Zhuohuang
    • The Journal of the Acoustical Society of Korea
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    • v.41 no.1
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    • pp.76-86
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    • 2022
  • Personal hearing devices, such as hearing aids, may be fine-tuned by allowing the users to conduct self-adjustment. Two self-adjustment procedures were developed to collect the listener preferred gains in six octave-frequency bands from 0.25 kHz to 8 kHz. These procedures were designed to allow rapid exploration of a multi-dimensional parameter space using a simple, one-dimensional user control interface (i.e., a programmable knob). The two procedures differ in whether the user interface controls the gains in all frequency bands simultaneously (Procedure A) or only the gain in one frequency band (Procedure B) on a given trial. Monte-Carlo simulations suggested that for both procedures the gain preference identified by simulated listeners rapidly converged to the ground-truth preferred gain profile over the first 20 trials. Initial behavioral evaluations of the self-adjustment procedures, in terms of test-retest reliability, were conducted using 20 young, normal-hearing listeners. Each estimate of the preferred gain profile took less than 20 minutes. The deviation between two separate estimates of the preferred gain profile, conducted at least a week apart, was about 10 dB ~ 15 dB.

Inverse Estimation of Geoacoustic Parameters in Shallow Water Using tight Bulb Sound Source (천해환경에서 전구음원을 이용한 지음향인자의 역추정)

  • 한주영;이성욱;나정열;김성일
    • The Journal of the Acoustical Society of Korea
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    • v.23 no.1
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    • pp.8-16
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    • 2004
  • An inversion method is presented for the determination of the compressional wave speed, compressional wave attenuation, thickness of the sediment layer and density as a function of depth for a horizontally stratified ocean bottom. An experiment for estimating those properties was conducted in the shallow water of South Sea in Korea. In the experiment, a light bulb implosion and the propagating sound were measured using a VLA (vertical line array). As a method for estimating the geoacoustic properties, a coherent broadband matched field processing combined with Genetic Algorithm was employed. When a time-dependent signal is very short, the Fourier transform results are not accurate, since the frequency components are not locatable in time and the windowed Fourier transform is limited by the length of the window. However, it is possible to do this using the wavelet transform a transform that yields a time-frequency representation of a signal. In this study, this transform is used to identify and extract the acoustic components from multipath time series. The inversion is formulated as an optimization problem which maximizes the cost function defined as a normalized correlation between the measured and modeled signals in the wavelet transform coefficient vector. The experiments and procedures for deploying the light bulbs and the coherent broadband inversion method are described, and the estimated geoacoustic profile in the vicinity of the VLA site is presented.

A Study on the Vulcanization Reaction of Modified NR Blends by In-Situ Electrical Property Measurement (전기적 특성의 in-situ 측정에 의한 개질된 NR 블랜드의 가황 반응에 관한 연구)

  • Ha, Ki-Ryong;Suh, Soong-Hyuck;Rho, Seung-Baik;Lee, Seung-Hyun;Ahn, Won-Sool
    • Elastomers and Composites
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    • v.38 no.3
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    • pp.235-242
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    • 2003
  • A vulcanization reaction characteristics of an isoprene rubber (IR)-modified natural rubber/carbon black (NR/CB) composite was studied using in-situ electrical property measuring technique. Since the electrical conductivity of the sample composite would be changed continuously during the vulcanization reaction by rearranging of the carbon black particles within the sample, volume resistivity (${\rho}$) might be obtained as a function or reaction time. A stabilization time ($t_i$), maximum reaction speed time ($t_p$), and volume resistivity at that time(${\rho}_p$) were defined from the data for the Arrhenius analysis. Volume resistivity ${\rho}$ showed a comparatively high value of ${\sim}10^8$ order before the reaction started, and dramatically decreased to be stabilized within $1{\sim}2$ minutes as soon as the reaction started. As the more time elapsed, thereafter, ${\rho}$ decreased monotonously to a certain constant value through a peak, ${\rho}_p$ at time $t_p$, which was considered as the maximum reaction rate. As a result, while $t_i$ values were comparatively constant as $1{\sim}2$ minutes, $t_p$ values showed to become shorter and shorter as the reaction temperature.

Noise-Biased Compensation of Minimum Statistics Method using a Nonlinear Function and A Priori Speech Absence Probability for Speech Enhancement (음질향상을 위해 비선형 함수와 사전 음성부재확률을 이용한 최소통계법의 잡음전력편의 보상방법)

  • Lee, Soo-Jeong;Lee, Gang-Seong;Kim, Sun-Hyob
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.1
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    • pp.77-83
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    • 2009
  • This paper proposes a new noise-biased compensation of minimum statistics(MS) method using a nonlinear function and a priori speech absence probability(SAP) for speech enhancement in non-stationary noisy environments. The minimum statistics(MS) method is well known technique for noise power estimation in non-stationary noisy environments. It tends to bias the noise estimate below that of true noise level. The proposed method is combined with an adaptive parameter based on a sigmoid function and a priori speech absence probability (SAP) for biased compensation. Specifically. we apply the adaptive parameter according to the a posteriori SNR. In addition, when the a priori SAP equals unity, the adaptive biased compensation factor separately increases ${\delta}_{max}$ each frequency bin, and vice versa. We evaluate the estimation of noise power capability in highly non-stationary and various noise environments, the improvement in the segmental signal-to-noise ratio (SNR), and the Itakura-Saito Distortion Measure (ISDM) integrated into a spectral subtraction (SS). The results shows that our proposed method is superior to the conventional MS approach.