• 제목/요약/키워드: Adaptive frequency sampling

검색결과 48건 처리시간 0.023초

Performance Prediction of a Laser-guide Star Adaptive Optics System for a 1.6 m Telescope

  • Lee, Jun Ho;Lee, Sang Eun;Kong, Young Jun
    • Current Optics and Photonics
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    • 제2권3호
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    • pp.269-279
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    • 2018
  • We are currently investigating the feasibility of a 1.6 m telescope with a laser-guide star adaptive optics (AO) system. The telescope, if successfully commissioned, would be the first dedicated adaptive optics observatory in South Korea. The 1.6 m telescope is an f/13.6 Cassegrain telescope with a focal length of 21.7 m. This paper first reviews atmospheric seeing conditions measured over a year in 2014~2015 at the Bohyun Observatory, South Korea, which corresponds to an area from 11.6 to 21.6 cm within 95% probability with regard to the Fried parameter of 880 nm at a telescope pupil plane. We then derive principal seeing conditions such as the Fried parameter and Greenwood frequency for eight astronomical spectral bands (V/R/I/J/H/K/L/M centered at 0.55, 0.64, 0.79, 1.22, 1.65, 2.20, 3.55, and $4.77{\mu}m$). Then we propose an AO system with a laser guide star for the 1.6 m telescope based on the seeing conditions. The proposed AO system consists of a fast tip/tilt secondary mirror, a $17{\times}17$ deformable mirror, a $16{\times}16$ Shack-Hartmann sensor, and a sodium laser guide star (589.2 nm). The high order AO system is close-looped with 2 KHz sampling frequency while the tip/tilt mirror is independently close-looped with 63 Hz sampling frequency. The AO system has three operational concepts: 1) bright target observation with its own wavefront sensing, 2) less bright star observation with wavefront sensing from another bright natural guide star (NGS), and 3) faint target observation with tip/tilt sensing from a bright natural guide star and wavefront sensing from a laser guide star. We name these three concepts 'None', 'NGS only', and 'LGS + NGS', respectively. Following a thorough investigation into the error sources of the AO system, we predict the root mean square (RMS) wavefront error of the system and its corresponding Strehl ratio over nine analysis cases over the worst ($2{\sigma}$) seeing conditions. From the analysis, we expect Strehl ratio >0.3 in most seeing conditions with guide stars.

Probability Density Function of Samples' Amplitude of ASSS OFDM Signal

  • Wang, Lei;Yoon, Dong-Weon;Park, Sang-Kyu
    • Journal of electromagnetic engineering and science
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    • 제8권2호
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    • pp.59-63
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    • 2008
  • The adaptive symbol selection scheme(ASSS) is popular in reducing peak to average power ratio(PAPR) for orthogonal frequency division multiplexing(OFDM) signals. The probability density function(pdf) of the samples' amplitudes of the adaptively selected OFDM signal without over-sampling has been considered to be approximately equal to the Rayleigh pdf. In this paper, we derive a more precise pdf which shows the relationship between the probability distribution of the samples' amplitudes and the number of the candidate symbols for ASSS. Using the newly derived pdf in the theoretical analysis, more accurate calculation results can be obtained.

적응 웨이블릿 패킷을 이용한 오디오 부호화기와 심리음향 모델링 (Audio Coder Using an Adaptive Wavelet packet Decomposition and Psychoacoustic)

  • 김준성
    • 한국음향학회:학술대회논문집
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    • 한국음향학회 1998년도 학술발표대회 논문집 제17권 1호
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    • pp.245-248
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    • 1998
  • In this paper, a new variable wavelet packet decomposition audio coder, based on the time varying characteristic of the audio signals, is proposed and presents a technique to incorporate psychoacoustic models into an adaptive wave let packet scheme. The proposed filterbank improves the defect of the polyphase filterbank that could not properly represent the critical band and the defect of QMF-tree filter that need high complexity to implement. The filterbank consists of varying number of subband from 4 to 26 bands and use Daubechies 6-order wave let. The codec yields excellent quality at total bit rates of about 128kbps for monophonic CD-quality signals with an sampling frequency of 44.1kHz and reduces complexity by 19% for various bit-rates and sources with encoding and decoding process.

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NTSC 복합신호의 휘도 및 색도성분을 분리하기 위한 프레임간 적응 필터링 기법 (Adaptive Interframe Filtering Techniques for Separation of Luminance/Chrominance Components in NTSC Composite Signals)

  • 강철호;이정한
    • 대한전자공학회논문지
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    • 제25권1호
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    • pp.72-80
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    • 1988
  • In this paper, new adaptive interframe filtering methods have been proposed to separate the luminance and chrominance components in NTSC composite signals. In thess methods, the composite signals are adaptivelty processed in three dimensions according to the local change or movement of the picture. For interframe processing, two algorithms have been proposed which adapt three filters in the horizontal, vertical and temporal directions to the magnitude f detection signal dependent upon both the movement and local change of the picture. The three kind of filters have been used at the sampling rate of four times the subcarrier frequency. The various quantitative measures have been introduced to compare the objective performance of the conventional methods and that of proposed ones by computer simulation.

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Narrow Band Interference Suppression In Multiuser CDMA System By Linear Prediction In Subband

  • Yoon-Gi Yang
    • 인터넷정보학회논문지
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    • 제2권3호
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    • pp.27-36
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    • 2001
  • Recently much attention has been paid for interference mitigation technique for the COMA system, since more capacity is available with same bandwidth. In this paper, we introduces a novel adaptive interference suppression techniques for the CDMA system with narrow band interference. The proposed interference rejection scheme employs the adaptive linear prediction techniques in the subband. In each subband, we can more easily find and cancel the narrow band signal as compared to the full band. Thus, the proposed interference rejection can be classified as another time-frequency techniques for the narrow band interference rejection(10). Computer simulation is conducted for the 3-G COMA system with IF band sampling techniques, yielding better interference rejection and bit error rate performance as compared to conventional one. Also, optimum filter is analyzed and from the analysis, it can be shown the subband prediction techniques can suppress narrow band interference more efficiently.

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HEVC RExt RGB 영상의 색평면 간 예측 향상을 위한 적응적 필터링 기법 (An Adaptive Filtering Method for Enhancement of Inter-color Plane Estimation in HEVC RExt RGB Images)

  • 최장원;최윤식
    • 방송공학회논문지
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    • 제18권4호
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    • pp.647-650
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    • 2013
  • HEVC RExt(High Efficiency Video Coding Range Extension)는 RGB/YUV 4:2:2 4:4:4 색 샘플링 영상과 10비트 심도 이상의 영상 지원을 목표로 한다. RGB 영상은 YUV 4:2:0 색 샘플링 영상과는 달리 색평면 간 높은 상관도를 갖고 있으며, 이를 이용하여 화소값을 예측하는 기법들이 JCT-VC 표준화 회의에서 기고되었다. 하지만 일반적으로 RGB 영상의 고주파수 성분은 색평면 간 낮은 상관도를 갖고 있으며, 이는 색평면 간 예측 시 부호화 효율 저하의 원인이 된다. 따라서 본 논문에서는 색평면 간 예측 시 고주파수 성분을 저역통과필터를 통해 적응적으로 제거하는 기법을 제안한다. HEVC RExt의 RGB 영상을 통한 실험 결과, 본 논문에서 제안하는 기법은 기존 색평면 간 예측 기법에 비해 큰 복잡도의 증가 없이 평균 0.6%의 BD(Bjontegaard Distortion)-율 이득을 얻을 수 있었다.

DTV시스템에서 평균 파워 조절기와 추정 옵셋 변화율에 따른 대역폭 조절 필터를 이용한 동기 성능 최적화 (Synchronization performance optimization using adaptive bandwidth filter and average power controller over DTV system)

  • 남완주;이성준;손성환;김재명
    • 대한전자공학회논문지SP
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    • 제44권5호
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    • pp.45-53
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    • 2007
  • DTV수신기에서 송신신호를 완벽하게 복원하기 위해서는 채널의 영향으로 인해 파일롯의 위치가 바뀌고 위상이 틀어지는 것을 보상해주는 반송파 주파수 동기와 샘플링 클락 주파수와의 위상오차로 인해 발생하는 샘플링 타이밍 오차를 보상하는 심볼 타이밍 동기가 모두 획득되어야 한다. 심볼 타이밍 동기부는 일반적으로 다중레벨을 가지는 신호에 사용되는 가드너(Gardner)방법을 사용한다. 가드너 방법은 매 심볼마다 타이밍 에러성분을 추출하므로 다중경로 채널에서 타이밍동기를 추적하면서 유지하는데 유리한 방식이다. 본 논문에서는 가드너 방법에서 에러를 검출하기 위해 사용되는 가드너 타이밍 에러 검출기(Timing Error Detector)가 수신파워레벨이 기준 파워레벨에서 크게 벗어날 경우 동기를 획득할 수 없는 문제점을 해결하기 위해 1단계로 가드너 타이밍 에러 검출기 블록 앞에 수신파워레벨을 계산하여 보정하는 블록을 추가하여 수신파워레벨을 보정한다. 2단계로 반송파 주파수동기와 심볼타이밍동기에 사용되는 PLL(Phase Locked Loop)회로의 빠른 동기 획득과 동기 획득 후 지터량을 줄이기 위하여 루프필터의 출력 값의 평균을 이용하여 옵셋량을 추정하여 추정된 옵셋의 변화율에 따라 단계적 대역폭을 가지는 적응적인 루프필터를 반송파 주파수 동기 회로와 심볼 타이밍동기 회로에 적용함으로써 최적의 동기성능을 얻는다.

0.11-2.5 GHz All-digital DLL for Mobile Memory Interface with Phase Sampling Window Adaptation to Reduce Jitter Accumulation

  • Chae, Joo-Hyung;Kim, Mino;Hong, Gi-Moon;Park, Jihwan;Ko, Hyeongjun;Shin, Woo-Yeol;Chi, Hankyu;Jeong, Deog-Kyoon;Kim, Suhwan
    • JSTS:Journal of Semiconductor Technology and Science
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    • 제17권3호
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    • pp.411-424
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    • 2017
  • An all-digital delay-locked loop (DLL) for a mobile memory interface, which runs at 0.11-2.5 GHz with a phase-shift capability of $180^{\circ}$, has two internal DLLs: a global DLL which uses a time-to-digital converter to assist fast locking, and shuts down after locking to save power; and a local DLL which uses a phase detector with an adaptive phase sampling window (WPD) to reduce jitter accumulation. The WPD in the local DLL adjusts the width of its sampling window adaptively to control the loop bandwidth, thus reducing jitter induced by UP/DN dithering, input clock jitter, and supply/ground noise. Implemented in a 65 nm CMOS process, the DLL operates over 0.11-2.5 GHz. It locks within 6 clock cycles at 0.11 GHz, and within 17 clock cycles at 2.5 GHz. At 2.5 GHz, the integrated jitter is $954fs_{rms}$, and the long-term jitter is $2.33ps_{rms}/23.10ps_{pp}$. The ratio of the RMS jitter at the output to that at the input is about 1.17 at 2.5 GHz, when the sampling window of the WPD is being adjusted adaptively. The DLL consumes 1.77 mW/GHz and occupies $0.075mm^2$.

최대 도플러 주파수 추정 시 대역 조절을 통한 부가 잡음의 영향 완화 기법 (SNR-independent Methods for Estimating Maximum Doppler Frequency)

  • 유현규;박구현;오성목;강창언;홍대식
    • 한국통신학회논문지
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    • 제30권6C호
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    • pp.475-480
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    • 2005
  • 이동 통신 시스템에서 최대 도플러 주파수 정보는 여러 가지 채널 적응 기술들과 무선 자원 관리 방법들의 무선 채널에 대한 최적화를 가능하게 한다. 본 논문에서는 레벨 교차율(LCR: level crossing rate) 및 공분산 함수(COV: covariance)에 기반을 둔 두 가지 최대 도플러 주파수 추정 기법을 제안한다. 부가 잡음의 영향을 제거하기 위해 신호 대 잡음비(SNR)에 독립적인 성능을 위한 조건을 분석하고, 제시한 조건을 만족시키기 위하여 하향 표본화 (downsampling) 방법을 이용한 도플러 주파수 추정 알고리즘을 새로이 제안하다. 모의실험을 통해, 제안된 기법이 SNR 변화에 상관없이 좋은 성능을 나타냄을 확인할 수 있다.

Improvement in Computation of Δ V10 Flicker Severity Index Using Intelligent Methods

  • Moallem, Payman;Zargari, Abolfazl;Kiyoumarsi, Arash
    • Journal of Power Electronics
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    • 제11권2호
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    • pp.228-236
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    • 2011
  • The ${\Delta}\;V_{10}$ or 10-Hz flicker index, as a common method of measurement of voltage flicker severity in power systems, requires a high computational cost and a large amount of memory. In this paper, for measuring the ${\Delta}\;V_{10}$ index, a new method based on the Adaline (adaptive linear neuron) system, the FFT (fast Fourier transform), and the PSO (particle swarm optimization) algorithm is proposed. In this method, for reducing the sampling frequency, calculations are carried out on the envelope of a power system voltage that contains a flicker component. Extracting the envelope of the voltage is implemented by the Adaline system. In addition, in order to increase the accuracy in computing the flicker components, the PSO algorithm is used for reducing the spectral leakage error in the FFT calculations. Therefore, the proposed method has a lower computational cost in FFT computation due to the use of a smaller sampling window. It also requires less memory since it uses the envelope of the power system voltage. Moreover, it shows more accuracy because the PSO algorithm is used in the determination of the flicker frequency and the corresponding amplitude. The sensitivity of the proposed method with respect to the main frequency drift is very low. The proposed algorithm is evaluated by simulations. The validity of the simulations is proven by the implementation of the algorithm with an ARM microcontroller-based digital system. Finally, its function is evaluated with real-time measurements.