• 제목/요약/키워드: Fast Fourier Transform Filter

검색결과 75건 처리시간 0.02초

능동소나 시스템을 위한 삼중 배열의 적응 빔형성 (Adaptive beamforming of triplet arrays for active sonar systems)

  • 안재균;류영우;천승용;김성일
    • 한국음향학회지
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    • 제37권1호
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    • pp.66-72
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    • 2018
  • 본 논문에서 우리는 능동소나 시스템에서 삼중 배열의 적응 빔형성 기법을 제안한다. 제안하는 기법은 정합필터, 카디오이드 빔형성, 선배열 빔형성으로 구성된 3단계의 과정을 통해 빔형성을 수행한다. 먼저, 송신 펄스에 대한 정합필터를 개별 수신 센서 신호에 적용하여 필터링된 신호를 획득한다. 그리고 정합필터 출력에 푸리에 변환을 적용하고, 각 삼각 센서 별 카디오이드 빔을 형성한다. 최종적으로 삼각 센서에 대한 카디오이드 빔들을 선배열 신호의 입력으로 가정하여 선배열 적응 빔형성을 적용한다. 실험 결과를 통해 제안하는 방법의 성능이 기존 기법보더 더 우수함을 확인한다.

EDLC 슈퍼 캐피시터 특성 분석을 위한 양산용 전기화학 분석 장치 개발 (Development of an Electro Impedance Spectroscopy device for EDLC super capacitor characterization in a mass production line)

  • 박찬희;이혜인;김상중;이정호;김성진;이희관
    • 한국산학기술학회논문지
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    • 제13권12호
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    • pp.5647-5654
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    • 2012
  • 본 논문에서는 기존에 주로 연료전지나 배터리 분석용으로 사용되었던 EIS 장비를 그 적용 범위를 넓혀 차세대 축전지인 EDLC 슈퍼커패시터까지도 분석할 수 있도록 개발하였다. 시스템은 다양한 패턴을 발생시킬 수 있는 신호 발생기, 정전위기, 수집된 신호처리를 위한 고속 디지털 필터 및 측정 프로그램으로 구성되었다. 개발된 시스템은 연구실 레벨에 그치지 않고 휴대용이면서 양산용에도 적합한 전기화학 분석 장치이다. 시스템의 특징으로는 0.01에서 20kHz까지 신호 출력이 가능하며, 동시에 수집된 전류신호는 푸리에변환을 이용하여 빠르게 분석할 수 있다. 장치를 이용하여 분석한 EDLC의 특성인 복소 저항 임피던스와, 등가이론을 적용한 등가회로 임피던스 데이터를 비교한 결과, 비슷한 결과를 확인 할 수 있었다. 이 시스템을 사용하면, 슈퍼커패시터의 생산 과정에서 시간에 따라 변하는 전기화학적 임피던스 변화를 빠른시간에 측정 할 수 있을 것으로 예상된다.

풍랑 모델링을 기반으로 한 실시간 파고 측정 시스템 (Wave Height Measurement System Based on Wind Wave Modeling)

  • 이중현;이동욱;허문범
    • 융합신호처리학회논문지
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    • 제13권4호
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    • pp.166-172
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    • 2012
  • 일반적인 파고 측정 시스템은 스펙트럼 분석을 통하여 파고를 측정한다. 하지만 스펙트럼 분석은 fast fourier transform(FFT)을 이용하기 때문에 구조가 복잡하며, 또한 FFT를 사용하기 위해서는 먼저 데이터를 저장해야 하기 때문에 실시간 처리가 어렵다. 본 논문에서는 칼만필터와 관성센서를 사용하여 실시간으로 데이터를 처리하는 알고리즘을 개발하고 성능을 평가하였다. 칼만필터 이론은 비교적 복잡한 알고리즘이지만 FFT를 사용하는 것에 비하여 간단하며, 가속도 데이터를 적분하여 위치를 효율적으로 추정할 수 있다. 하지만 가속도 데이터를 적분할 때 오차의 누적이 발생되는데, 파고의 특징을 이용한 알고리즘을 개발하여 누적오차 문제를 해결하였다. 본 논문에서는 다양한 조건의 해파를 모델링하여 성능평가를 수행하였으며, 실험을 통하여 풍랑측정에 한해서 평균 오차율 3.5%로 우수한 측정 성능을 확인할 수 있었다.

해금 사운드 콘텐츠의 저작권 보호를 위한 워터마킹 알고리듬 (Watermarking Algorithm for Copyright Protection of Haegeum Sound Contents)

  • 홍연우;강명수;조상진;정의필
    • 융합신호처리학회논문지
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    • 제10권4호
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    • pp.214-219
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    • 2009
  • 본 논문에서는 해금 사운드 콘텐츠의 저작권 보호를 위해 해금의 주파수 특성을 고려한 워터마킹 알고리듬을 제안한다. 해금 음은 1500Hz~2000Hz 대역과 2800Hz~3500Hz 대역에서 배음 성분의 크기가 공통적으로 크게 나타나는데 이 대역을 워터마크 삽입대역으로 결정하였다. 제안한 방법은 원 신호에 푸리에 변환을 하여 주파수 영역으로 변환하고 삽입대역에 속한 배음 성분에 PN 시퀀스를 통해 생성한 워터마크 비트를 삽입하였다. 또한 저역통과 필터, 대역통과 필터, 잘라내기, 노이즈 추가, MP3 압축과 같은 공격에서 최대 1.41%의 BER을 보였고, 클래식, 팝, 가야금 등의 연주곡과 비교 실험 결과 해금에서 가장 뛰어난 성능을 보였다. 주관 청취 테스트인 MUSHRA 테스트 결과 해금 음에 대해서는 평균 98점 이상을 보였고 국악연주곡에서는 96.67점을 보여 모두 Excellent의 평가 결과를 보였다.

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중공 미소 유리구를 이용한 LED 스크린 모아레 억제 (Suppression of Moiré Fringes Using Hollow Glass Microspheres for LED Screen)

  • 홍송은;나정필;정모세;김기은;박종운
    • 반도체디스플레이기술학회지
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    • 제22권3호
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    • pp.28-35
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    • 2023
  • Moiré patterns emerge due to the interference between the non-emission area of the LED screen and the grid line in an image sensor of a video recording device when taking a video in the presence of the LED screen. To reduce the moiré intensity, we have fabricated an anti-moiré filter using hollow glass microspheres (HGMs) by slot-die coating. The LED screen has a large non-emission area because of a large pitch (distance between LED chips), causing more severe moiré phenomenon, compared with a display panel having a very narrow black matrix (BM). It is shown that HGMs diffuse light in such a way that the periodicity of the screen is broken and thus the moiré intensity weakens. To quantitatively analyze its moiré suppression capability, we have calculated the spatial frequencies of the moiré fringes using fast Fourier transform. It is addressed that the moiré phenomenon is suppressed and thus the amplitude of each discrete spatial frequency term is reduced as the HGM concentration is increased. Using the filter with the HGM concentration of 9 wt%, the moiré fringes appeared depending sensitively on the distance between the LED screen and the camera are almost completely removed and the visibility of a nature image is enhanced at a sacrifice of luminance.

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디지털신호처리 칩과 마이크로 컨트롤러를 이용한 자동 조정 양변위 되먹임 제어기의 구현 (Implementation of Auto-tuning Positive Position Feedback Controller Using DSP Chip and Microcontroller)

  • 곽문규;김기영;방세윤
    • 한국소음진동공학회논문집
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    • 제15권8호
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    • pp.954-961
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    • 2005
  • This paper is concerned with the implementation of auto-tuning positive position feedback controller using a digital signal processor and microcontroller. The main advantage of the positive position feedback controller is that it can control a natural mode of interest by tuning the filter frequency of the positive position feedback controller to the natural frequency of the target mode. However, the positive position feedback controller loses its advantage when mistuned. In this paper, the fast fourier transform algorithm is implemented on the microcontroller whereas the positive position feedback controller is implemented on the digital signal processor. After calculating the frequency which affects the vibrations of structure most, the result is transferred to the digital signal processor. The digital signal processor updates the information on the frequency to be controlled so that it can cope with both internal and external changes. The proposed scheme was installed and tested using a beam equipped with piezoceramic sensor and actuator. The experimental results show that the auto-tuning positive position feedback controller proposed in this paper can suppress vibrations even when the target structure undergoes structural change thus validating the approach.

디지털신호처리 칩과 마이크로 컨트롤러를 이용한 적응 양변위 되먹임 제어기의 구현 (Implementation of Adaptive Positive Popsition Feedback Controller Using DSP chip and Microcontroller)

  • 곽문규;김기영;방세윤
    • 한국소음진동공학회:학술대회논문집
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    • 한국소음진동공학회 2005년도 춘계학술대회논문집
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    • pp.498-503
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    • 2005
  • This paper is concerned with the implementation of adaptive positive position feedback controller using a digital signal processor and microcontroller The main advantage of the positive position feedback controller is that it can control a natural mode of interest by tuning the filter frequency of the positive position feedback controller to the natural frequency of the target mode. However, the positive position feedback controller loses its advantage when mistuned. In this paper, the fast fourier transform algorithm is implemented on the microcontroller whereas the positive position feedback controller is implemented on the digital signal processor. After calculating the frequency which affects the vibrations of structure most the result is transferred to the digital signal processor. The digital signal processor updates the information on the frequency to be controlled so that it can cope with both internal and external changes. The proposed scheme was installed and tested using a beam equipped with piezoceramic sensor and actuator. The experimental results show that the adaptive positive position feedback controller proposed in this paper can suppress vibrations even when the target structure undergoes structural change thus validating the approach.

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Graphical Representation of the Instantaneous Compensation Power Flow for Single-Phase Active Power Filters

  • Jung, Young-Gook
    • Journal of Electrical Engineering and Technology
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    • 제8권6호
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    • pp.1380-1388
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    • 2013
  • The conventional graphical representation of the instantaneous compensation power flow for single-phase active power filters(APFs) simply represents the active power flow and the reactive power flow which flowing between the power source and the active filter / the load. But, this method does not provide the information about the rectification mode and the compensation mode of APFs, especially, the loss for each mode was not considered at all. This is very important to understand the compensation operation characteristics of APFs. Therefore, this paper proposes the graphical representation of the instantaneous compensation power flow for single-phase APFs considering the instantaneous rectification mode and the instantaneous inversion mode. Three cases are verified in this paper - without compensation, with compensation of the active power 'p' and the fundamental reactive power 'q', and with compensation of only the distorted power 'h'. To ensure the validity of the proposed approach, PSIM simulation is achieved. As a result, we could confirm that the proposed approach was easy to explain the instantaneous compensation power flow considering the instantaneous rectification mode and the instantaneous inversion mode of APFs, also, Total Harmonic Distortion(THD)/Power Factor (P.F) and Fast Fourier Transform(FFT) analysis were compared for each case.

Long Term Average Spectrum Characteristics of Head and Chest Register Sounds of Western Operatic Singers - Possibility of a Second Singer's Formant-

  • Jin, Sung-Min;Kwon, Young-Kyung;Song, Yun-Kyung
    • 음성과학
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    • 제10권2호
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    • pp.99-109
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    • 2003
  • The purpose of this study was to analyze and compare head register with chest register of singers acoustically. Fifteen healthy tenor major students were participated. Fifteen healthy untrained adults were chosen as the control group for this study. Long term average (LTA) power spectrum using the Fast Fourier transform (FFT) algorithm and Linear predictive coding (LPC) filter response were made with /a/ sustained in both head (G4, 392 Hz) and chest registers (C3, 131 Hz). Statistical analysis was performed using the Mann-Whitney test. In the LTA power spectrum, head register of singers increased in the level of energy gain within the frequency of 2.2-3.4 kHz (p<0.01), and 7.5-8.4 kHz (p<0.01, p<0.05). Chest register of singers increased in the frequency of 2.2-3.1 kHz (p<0.01), 7.8-8.4 kHz (p<0.05) and around 9.6 kHz (p<0.01). The LTA power spectrum revealed a peak of acoustic energy around 2,500 Hz, known as the singer's formant and another peak of acoustic energy around 8,000 Hz in the singer's voice.

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Blood Pressure Simulation using an Arterial Pressure-volume Model

  • Yoon, Sang-Hwa;Kim, Jae-Hyung;Ye, Soo-Young;Kim, Cheol-Han;Jeon, Gye-Rok
    • Transactions on Electrical and Electronic Materials
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    • 제9권1호
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    • pp.38-43
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    • 2008
  • Using an arterial pressure-volume (APV) model, we performed an analysis of the conventional blood pressure estimation method using an oscillometric sphygmomanometer with computer simulation. Traditionally, the maximum amplitude algorithm (MAA) has been applied to the oscillation waveforms of the APV model to obtain the mean arterial pressure and the characteristic ratio. The estimation of mean arterial pressure and characteristic ratio was significantly affected by the shape of the blood pressure waveforms and the cutoff frequency of high-pass filter (HPF) circuitry. Experimental errors result from these effects when estimating blood pressure. To determine an algorithm independent of the influence of waveform shapes and parameters of HPF, the volume oscillation of the APV model and the phase shift of the oscillation with fast Fourier transform (FFT) were tested while increasing the cuff pressure from 1 mmHg to 200 mmHg (1 mmHg/s). The phase shift between ranges of volume oscillation was then only observed between the systolic and the diastolic blood pressures. The same results were obtained from simulations performed on two different arterial blood pressure waveforms and one hyperthermia waveform.