• 제목/요약/키워드: estimated signals

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GPS AOA Choosing Algorithm in Environment of High-Power Interference Signals (고 전력 간섭 환경에서의 GPS AOA 선택 알고리즘)

  • Hwang, Suk-Seung
    • Journal of Advanced Navigation Technology
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    • v.16 no.4
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    • pp.649-656
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    • 2012
  • The Global Positioning System (GPS) is widely utilized for commercial and military applications to estimate the location of the user or object. The GPS suffers from various intentional or unintentional interferers and it requires estimating the accurate angle-of-arrival (AOA) of the GPS signal to suppress interference signals and to efficiently detect GPS data. Since the power of GPS signal is very low comparing with the noise and interference signals, it is extremely difficult to estimate GPS AOA before despreading. Although AOA of GPS signal is usually estimated after despreading, it requires choosing the GPS AOA among results of AOA estimation because they include AOAs of interference and GPS signals when existing high-power interferers. In this paper, we propose the efficient choosing algorithm of the GPS signal among the estimated AOAs. The proposed algorithm compares the estimated results before despreading and after despreading for choosing AOA of GPS signal. Computer simulation examples are presented to illustrate the performance of the proposed algorithm.

A study on the SIC with the improved delay time in CDMA System (CDMA시스템에서 지연 시간을 향상시킨 순차적 간섭 제거기에 관한 연구)

  • Choe, Byeong-Gu;Park, Yong-Wan
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.37 no.7
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    • pp.1-8
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    • 2000
  • In this paper, we introduce a modified interference cancellation scheme for multiuser detection in CDMA(Code Division Multiple Access). This detector uses SIC(Successive Interference Canceller) scheme and divides the received signals to reduce the delay time. In this proposed structure, the active users are divided into a number of groups. Within each group, parallel detection is performed to estimate the output signal of that group. The estimated output signal due to that group is then subtracted from the received signal and the resulting residual signal is used for the parallel detection of the next group. This parallel and serial cancellation process is repeated until the last group in the stage is completed. The estimated output signals due to all groups except -th group are MAI for the user signals in -th group. Therefore, the estimated output signals due to all groups except th group are subtracted from the received signal, and then the obtained signal becomes the input signal of -th SIC. The proposed RDSIC (Reduced Delay time of Successive Interference Canceller) has performance and complexity close to the SIC, but with much less detection delay.

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A New Double-Talk Detection Algorithm (새로운 동시통화 검출 알고리즘)

  • Jung, Hong-Hee;Kim, Hyun-Tae;Park, Jang-Sik;Son, Kyung-Sik
    • Journal of Korea Multimedia Society
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    • v.11 no.3
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    • pp.281-291
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    • 2008
  • In this paper, we propose a new double talk detection algorithm which detects near end signals with less degradation, tracking echo path variation of echo canceler simultaneously. Our method makes use of a cross-correlation between channel input signals and estimated error signals and a normalized cross-correlation between microphone input signals and estimated error signals. By combing thresholds for these cross-correlations pertinently, this algorithm discriminates between variation of echo path and occurrence of double talk. These two cross-correlation are used to detect double talk periods, tracking echo path variation. During the detection period, adjustive adaptive filter is ceased to prevent the echo canceler from being disturbed by near end signals. Also, the echo canceler will still be kept on for tracking any variation in echo path. Through computer simulation results, it was confirmed that the proposed algorithm shows better performance, tracking echo path variation and detecting the double talk periods, than the Ye et. al's and the NLMS algorithms from ERLE viewpoint.

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A Neuro-Fuzzy Inference System for Sensor Failure Detection Using Wavelet Denoising, PCA and SPRT

  • Na, Man-Gyun
    • Nuclear Engineering and Technology
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    • v.33 no.5
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    • pp.483-497
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    • 2001
  • In this work, a neuro-fuzzy inference system combined with the wavelet denoising, PCA (principal component analysis) and SPRT (sequential probability ratio test) methods is developed to detect the relevant sensor failure using other sensor signals. The wavelet denoising technique is applied to remove noise components in input signals into the neuro-fuzzy system The PCA is used to reduce the dimension of an input space without losing a significant amount of information. The PCA makes easy the selection of the input signals into the neuro-fuzzy system. Also, a lower dimensional input space usually reduces the time necessary to train a neuro-fuzzy system. The parameters of the neuro-fuzzy inference system which estimates the relevant sensor signal are optimized by a genetic algorithm and a least-squares algorithm. The residuals between the estimated signals and the measured signals are used to detect whether the sensors are failed or not. The SPRT is used in this failure detection algorithm. The proposed sensor-monitoring algorithm was verified through applications to the pressurizer water level and the hot-leg flowrate sensors in pressurized water reactors.

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Enhancing Nearfield Acoustic Holography using Wavelet Transform

  • Ko, ByeongSik
    • Journal of Mechanical Science and Technology
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    • v.18 no.10
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    • pp.1738-1746
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    • 2004
  • When there are low signal to noise relationships or low coherences between measured pressure and a reference sensor, a pressure field measured and estimated by NAH (Nearfield Acoustic Holography) becomes noisy on the hologram and source planes. This paper proposes a method to obtain the high coherent de-noised pressure signals from low coherent noisy ones by combining a wavelet algorithm with NAH. The proposed method obtains the de-noised field from acoustic fields on a noise source plane reconstructed through backward propagation of NAH. Thus this method does not need high coherent pressure signals on the hologram surface while the conventional nearfield acoustic holography requires high-coherent signals. The proposed method was verified by numerical simulation using noisy signals, composed of original signals and imposed noises distributed on the hologram surface.

A Fault Location Algorithm Using Adaptively Estimated Local Source Impedance for a Double-Circuit Transmission Line System (자기단 전원 임피던스 추정 기법을 사용한 병행 2회선 송전선로 고장점 표정 알고리즘)

  • Park, Gun-Ho;Kang, Sang-Hee;Kim, Sok-Il;Shin, Jonathan H.
    • The Transactions of The Korean Institute of Electrical Engineers
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    • v.61 no.3
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    • pp.373-379
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    • 2012
  • This paper presents a fault location algorithm based on the adaptively estimated value of the local sequence source impedance for faults on a parallel transmission line. This algorithm uses only the local voltage and current signals of a faulted circuit. The remote current signals and the zero-sequence current of the healthy adjacent circuit are calculated by using the current distribution factors together with the local terminal currents of the faulted circuit. The current distribution factors consist of local equivalent source impedance and the others such as fault distance, line impedance and remote equivalent source impedance. It means that the values of the current distribution factors can change according to the operation condition of a power system. Consequently, the accuracy of the fault location algorithm is affected by the two values of equivalent source impedances, one is local source impedance and the other is remote source impedance. Nevertheless, only the local equivalent impedance can be estimated in this paper. A series of test results using EMTP simulation data show the effectiveness of the proposed algorithm. The proposed algorithm is valid for a double-circuit transmission line system where the equivalent source impedance changes continuously.

In-process Immersion Ratio Estimation Using Spindle Motor Current during Face Milling (정면밀링공정중 추축모터전류를 이용한 절입비의 실시간 추정)

  • 조규진;오영탁;권원태;주종남
    • Journal of the Korean Society of Manufacturing Technology Engineers
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    • v.9 no.6
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    • pp.57-64
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    • 2000
  • In order to regulate cutting torque in milling, monitoring system should be set to a certain threshold. Radial immersion ratio is an important factor to determine the threshold and should be estimated in process for automatic regulation. In this paper, on-line estimation of the radial immersion ratio using spindle motor current in face milling is presented. When a tooth finishes sweeping, a sudden drop of cutting torque occurs. This torque drop is equal to the cutting torque acting on a single tooth at the swept angle of cut and can be acquired from cutting torque signals. Average cutting torque per revolution can also be calculated from cutting torque signals. The ratio of cutting torque acting on a single tooth at the swept angle of cut to the average cutting torque per revolution is a function of the swept angle of cut and the number of teeth. Using the magnitude of this ratio, the radial immersion ratio is estimated. Identical algorithm is adopted to estimate the immersion ratio based on the spindle motor current measurement. The experiments performed under different cutting conditions show that the radial immersion ratio can be estimated within 10% error range by the proposed method using spindle motor current. Varying immersion ratio is also estimated well using the presented algorithm.

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Technical Status of LED Traffic Signals made in Korea (국내의 LED 교통신호등의 기술현황 분석연구)

  • Jeong, Hak-Geun;Jung, Bong-Man;Han, Su-Bin;Park, Suk-In;Kim, Kyu-Deok;Yu, Seong-Won
    • Proceedings of the Korean Institute of IIIuminating and Electrical Installation Engineers Conference
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    • 2004.05a
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    • pp.171-174
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    • 2004
  • Concerning Korean energy consumption, 20% of the total electrical energy (96% is now exported) is consumed in lighting area. Accordingly, it is one of the most important governmental policies to efficiently utilize electricity due to development and application of high-efficiency lighting equipment. In Korea, widely-built traffic signals employ an incandescent and are of 100W/300 mm, and 280,080set/801,932lamps have been installed and operated by 1999. Of them, 58%(161,181set), 25%(69,655set) and 18%(49,244set) correspond to traffic signals, pedestrian signals and other supplementary signals respectively. It was estimated that electrical energy consumed 28MW instantaneously, 245GWH annually. On the other hand, the LED traffic signal is expected to be a future traffic signal since if traffic signals are replaced by LEDs, energy saving effect of 85% and drop of 75% in maintenance fee will be obtained. In this paper, the performance and characteristics of many LED traffic signals made in Korea are summarized in order to reform Korean standard of LED traffic signals.

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Thickness Measurements of the Base Concrete by the Impact-resonance Test (탄성파 공진법에 의한 기초 콘크리트의 두께 측정)

  • 김영환
    • Proceedings of the Korea Concrete Institute Conference
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    • 1989.10a
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    • pp.53-58
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    • 1989
  • Thicknesses of the base concrete blocks for large machines were estimated by analyzing the resonance modes of mechanical vibrations. The vibration was produced by the mechanical impact and detected by a wideband conical transcuder. There signals were analyzed by FET and thicknesses were obtained by the peaks of frequency spectrum. The estimated thickness upto 100cm are in good agreement with the real ones. For the layered concrete block, the estimated thickness is dependent on the acoustic reflective index at the boundary of the two layers.

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AM-FM Decomposition and Estimation of Instantaneous Frequency and Instantaneous Amplitude of Speech Signals for Natural Human-robot Interaction (자연스런 인간-로봇 상호작용을 위한 음성 신호의 AM-FM 성분 분해 및 순간 주파수와 순간 진폭의 추정에 관한 연구)

  • Lee, He-Young
    • Speech Sciences
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    • v.12 no.4
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    • pp.53-70
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    • 2005
  • A Vowel of speech signals are multicomponent signals composed of AM-FM components whose instantaneous frequency and instantaneous amplitude are time-varying. The changes of emotion states cause the variation of the instantaneous frequencies and the instantaneous amplitudes of AM-FM components. Therefore, it is important to estimate exactly the instantaneous frequencies and the instantaneous amplitudes of AM-FM components for the extraction of key information representing emotion states and changes in speech signals. In tills paper, firstly a method decomposing speech signals into AM - FM components is addressed. Secondly, the fundamental frequency of vowel sound is estimated by the simple method based on the spectrogram. The estimate of the fundamental frequency is used for decomposing speech signals into AM-FM components. Thirdly, an estimation method is suggested for separation of the instantaneous frequencies and the instantaneous amplitudes of the decomposed AM - FM components, based on Hilbert transform and the demodulation property of the extended Fourier transform. The estimates of the instantaneous frequencies and the instantaneous amplitudes can be used for modification of the spectral distribution and smooth connection of two words in the speech synthesis systems based on a corpus.

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