• Title/Summary/Keyword: Input signal

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The Study on Reliability Improvement in Eddy Current Inspection by Signal Characteristic Optimization of Multi-coil Array Probe (다중센서 신호특성 최적화를 통한 와전류검사 신뢰성 개선연구)

  • Ahn, Y.S.;Gil, D.S.;Park, S.G.
    • Journal of Power System Engineering
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    • v.14 no.2
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    • pp.60-64
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    • 2010
  • This paper introduces reliability improvement and time saving in eddy current inspection by signal characteristic optimization of multi-coil eddy current array probe. In the past, Multi-coil array probe and single probe were used for the gas turbine rotor surface inspection & defect evaluation. The multi-coil array probe was used for the broad area inspection. But the signal deviations among multi-coil array probe are maximum 28% in commercial probe. This differences were considered to impedance differences among coils, so it is very difficult to evaluate exact defect size. The signal deviations among multi-coil array probe are maximum 28% in commercial probe. So, single coil inspection was used for exact defect sizing. The purpose of this study is to improve signal deviations of multi-coil array probe. The introduced new technology can improves this deviation by adjusting input voltage in each coil. At first, apply same voltage in each coil and collect signal amplitude of each coil. And calculate new input voltage based on signal amplitude of each coil. If the signal amplitude deviation is within 5% among multi-coil array probe, the signal amplitude of multi-coil array probe is reliable. The proposed technology gives 2% signal deviation among multi-coil array probe. The proposed new technology gives reliability improvement and inspection time saving in eddy current inspection.

Effects of interface delay in real-time dynamic substructuring tests on a cable for cable-stayed bridge

  • Marsico, Maria Rosaria
    • Smart Structures and Systems
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    • v.14 no.6
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    • pp.1173-1196
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    • 2014
  • Real-time dynamic substructuring tests have been conducted on a cable-deck system. The cable is representative of a full scale cable for a cable-stayed bridge and it interacts with a deck, numerically modelled as a single-degree-of-freedom system. The purpose of exciting the inclined cable at the bottom is to identify its nonlinear dynamics and to mark the stability boundary of the semi-trivial solution. The latter physically corresponds to the point at which the cable starts to have an out-of-plane response when both input and previous response were in-plane. The numerical and the physical parts of the system interact through a transfer system, which is an actuator, and the input signal generated by the numerical model is assumed to interact instantaneously with the system. However, only an ideal system manifests a perfect correspondence between the desired signal and the applied signal. In fact, the transfer system introduces into the desired input signal a delay, which considerably affects the feedback force that, in turn, is processed to generate a new input. The effectiveness of the control algorithm is measured by using the synchronization technique, while the online adaptive forward prediction algorithm is used to compensate for the delay error, which is present in the performed tests. The response of the cable interacting with the deck has been experimentally observed, both in the presence of delay and when delay is compensated for, and it has been compared with the analytical model. The effects of the interface delay in real-time dynamic substructuring tests conducted on the cable-deck system are extensively discussed.

Input Signal Model Analysis for Adaptive Beamformer (적응 빔형성기의 입력신호 모델 분석)

  • Mun, Ji-Youn;Hwang, Suk-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.12 no.3
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    • pp.433-438
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    • 2017
  • Containing an Angle-of-Arrival(: AOA) estimation and interference suppression techniques, an adaptive beamformer is one of core techniques for the Signal Intelligence(: SIGINT) which collect various intelligence utilizing cutting edge devices including the radar and satellite. It generates a beam with the directivity in a corresponding direction, to efficiently receive a signal from the specific direction, using antenna array. In this paper, we present the received signal model including interference signals and noise, which can be applied to an input of the signal intelligence satellite system equipped with the AOA estimation and the interference cancellation techniques, and analysis the characteristics of various signals, which can be included in the proposed received signal model. This proposed signal model can be directly applied to the performance evaluation for a variety of beamforming techniques. Also, we verify the spectrum characteristic of the presented received signal model in the frequency domain through computer simulation examples.

A User-friendly Remote Speech Input Method in Spontaneous Speech Recognition System

  • Suh, Young-Joo;Park, Jun;Lee, Young-Jik
    • The Journal of the Acoustical Society of Korea
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    • v.17 no.2E
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    • pp.38-46
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    • 1998
  • In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.

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High Voltage SMPS Design based on Dual-Excitation Flyback Converter (이중 여자 플라이백 기반 고압 SMPS 설계)

  • Yang, Hee-Won;Kim, Seong-Ae;Park, Seong-Mi;Park, Sung-Jun
    • Journal of the Korean Society of Industry Convergence
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    • v.20 no.2
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    • pp.115-124
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    • 2017
  • This paper aims to develop an SMPS topology for handling a high range of input voltages based on a DC-DC flyback converter circuit. For this purpose, 2 capacitors of the same specifications were serially connected on the input terminal side, with a flyback converter of the same circuit configuration serially connected to each of them, so as to achieve high input voltage and an effect of dividing input voltage. The serially connected flyback converters have the transformer turn ratio of 1:1, so that each coil is used for the winding of a single transformer, which is a characteristic of doubly-fed configuration and enables the correction of input capacitor voltage imbalance. In addition, a pulse transformer was designed and fabricated in a way that can achieve the isolation and noise robustness of the PWM output signal of the PWM controller that applies gate voltage to individual flyback converter switches. PSIM simulation was carried out to verify such a structure and confirm its feasibility, and a 100W class stack was fabricated and used to verify the feasibility of the proposed high voltage SMPS topology.

Power Factor Correction Technique of Boost Converter Based on Averaged Model (평균화 모델을 이용한 역률개선 제어기법)

  • 정영석;문건우;이준영;윤명중
    • Proceedings of the KIPE Conference
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    • 1996.06a
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    • pp.85-88
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    • 1996
  • New power factor correction(PFC) technique based on the averaged model of boost converter is proposed. Without measurement of input current, power factor correction scheme derived from the averaged model is presented. With the measurements of input voltage and output voltage, the control signal is generated to make the shape of the line current same as the input voltage. The characteristics of input line current distortion is analyzed by considering the generation of duty cycle.

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Development of a neural network with fuzzy preprocessor (퍼지 전처리기를 가진 신경회로망 모델의 개발)

  • 조성원;최경삼;황인호
    • 제어로봇시스템학회:학술대회논문집
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    • 1993.10a
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    • pp.718-723
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    • 1993
  • In this paper, we propose a neural network with fuzzy preprocessor not only for improving the classification accuracy but also for being able to classify objects whose attribute values do not have clear boundaries. The fuzzy input signal representation scheme is included as a preprocessing module. It transforms imprecise input in linguistic form and precisely stated numerical input into multidimensional numerical values. The transformed input is processed in the postprocessing module. The experimental results indicate the superiority of the backpropagation network with fuzzy preprocessor in comparison to the conventional backpropagation network.

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A Design of Noise Reduction Circuit for A radio Telephonic System (무선전화 시스템용 잡음억제회로의 설계)

  • Moon, Jong-Kyu;Kim, Duk-Gyoo
    • Journal of the Institute of Electronics Engineers of Korea TE
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    • v.39 no.2
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    • pp.84-89
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    • 2002
  • In this paper, we present the design method of noise reduction circuit in telephonic system. The circuit consists of compressor, expander and a filter. The basic idea of a proposed method compresses the audible signal in order to mask the channel noise during transmission and then expand at the reverse rate the transmitted signal to naturally recover the original signal. Of course, there should be no distortion or other degradation of the audio itself in passing through companding(compress/expand) cycle. In the compressing process, the gain of compressor is automatically controlled by the envelope level of input signal in order to increase the effective dynamic range of input signal and to improve the signal to noise ratio. The compressed rate is the root time of a audible signal. The compressed signal should be expanded at the square time of the signal to recover a original signal. Simulation shows the proposed method improves the performance of the noise reduction of a channel noise as well as stability. 

Design of Boundary Filter in Subband Coding using M-band Orthogonal Wavelet Filter (M-대역 직교 웨이브렛 필터를 이용한 부대역 부호화에서 경계 필터의 설계)

  • 권상근
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.4 no.5
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    • pp.997-1003
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    • 2000
  • When finite length signal is the input of QMF banks which are composed of the M-band orthogonal wavelet filter, the boundaries of reconstructed signal are not reconstructed perfectly. For boundary signals to be reconstructed perfectly, different type filters or methods are applied to them. In this paper, the design method of boundary filters which can be reconstructed the boundary signal perfectly was proposed, in case the dimension of M-band decomposed signal is the same as that of input signal. The boundary filters were designed using the perfect reconstruction condition of paraunitary matrix. In an application of subband coding of still image, the proposed boundary filters achieve better PSNR about 5% in reconstructed image than reflected method at the same bit rate.

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Debonding monitoring of CFRP strengthened RC beams using active sensing and infrared imaging

  • Sohn, Hoon;Kim, Seung Dae;In, Chi Won;Cronin, Kelly E.;Harries, Kent
    • Smart Structures and Systems
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    • v.4 no.4
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    • pp.391-406
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    • 2008
  • This study attempts to develop a real-time debonding monitoring system for carbon fiber-reinforced polymer (CFRP) strengthened structures by continuously inspecting the bonding condition between the CFRP layer and the host structure. The uniqueness of this study is in developing a new concept and theoretical framework of nondestructive testing (NDT), in which debonding is detected without relying on previously-obtained baseline data. The proposed reference-free damage diagnosis is achieved based on the concept of time reversal acoustics (TRA). In TRA, an input signal at an excitation point can be reconstructed if the response signal measured at another point is reemitted to the original excitation point after being reversed in the time domain. Examining the deviation of the reconstructed signal from the known initial input signal allows instantaneous identification of damage without requiring a baseline signal representing the undamaged state for comparison. The concept of TRA has been extended to guided wave propagations within the CFRP-strengthened reinforced concrete (RC) beams to improve the detectibility of local debonding. Monotonic and fatigue load tests of large-scale CFRP-strengthened RC beams are conducted to demonstrate the potential of the proposed reference-free debonding monitoring system. Comparisons with an electro-mechanical impedance method and an inferred imaging technique are provided as well.