• Title/Summary/Keyword: 주파수 복원

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Design of FIR Filters With Sparse Signed Digit Coefficients (희소한 부호 자리수 계수를 갖는 FIR 필터 설계)

  • Kim, Seehyun
    • Journal of IKEEE
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    • v.19 no.3
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    • pp.342-348
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    • 2015
  • High speed implementation of digital filters is required in high data rate applications such as hard-wired wide band modem and high resolution video codec. Since the critical path of the digital filter is the MAC (multiplication and accumulation) circuit, the filter coefficient with sparse non-zero bits enables high speed implementation with adders of low hardware cost. Compressive sensing has been reported to be very successful in sparse representation and sparse signal recovery. In this paper a filter design method for digital FIR filters with CSD (canonic signed digit) coefficients using compressive sensing technique is proposed. The sparse non-zero signed bits are selected in the greedy fashion while pruning the mistakenly selected digits. A few design examples show that the proposed method can be utilized for designing sparse CSD coefficient digital FIR filters approximating the desired frequency response.

A 3-stage Pipelined Architecture for Multi-View Images Decoder3 (단계 파이프라인 구조를 갖는 Multi-View 영상 디코더)

  • Bae, Chang-Ho;Yang, Yeong-Yil
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.39 no.4
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    • pp.104-111
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    • 2002
  • In this paper, we proposed the architecture of the decoder which implements the multi-view images decoding algorithm. The study of the hardware structure of the multi-view image processing has not been accomplished. The proposed multi-view images decoder operates in a three stage pipelined manner and extracts the depth of the pixels of the decoded image every clock. The multi-view images decoder consists of three modules, Node selector which transfers the value of the nodes repeatedly and Depth Extractor which extracts the depth of each pixel from the four values of the nodes and Affine transformer which generates the projecting position on the image plane from the values of the pixels and the specified viewpoint. The proposed architecture is designed and simulated by the Max+plus II design tool and the operating frequency is 30MHz. The image can be constructed in a real time by the decoder with the proposed architecture.

Random Vibration Analysis of Nonlinear Stochastic System under Earthquake Using Statistical Method (지진하중을 받는 비선헝 추계적 시스템의 불규칙진동해석)

  • Moon, Byung-Young;Kang, Gyung-Ju;Kang, Beom-Soo
    • Journal of the Earthquake Engineering Society of Korea
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    • v.5 no.6
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    • pp.55-64
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    • 2001
  • Industrial machines are sometimes exposed to the danger of earthquake. In the design of a mechanical system, this factor should be accounted for from the viewpoint of reliability to analyze a complex nonlinear structure system under random excitation is proposed. First, the actual random excitation, such as earthquake, is approximated to the corresponding Gaussian process for the statistical analysis. The modal equations of overall system are expanded sequentially. Then, the perturbed equations are synthesized into the overall system and solved in probabilistic way. Several statistical properties of a random process that are of interest in random vibration are evaluated in each substructure. Comparing with the results of the numerical simulation proved the efficiency of the proposed method.

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Sinusoidal Modeling of Polyphonic Audio Signals Using Dynamic Segmentation Method (동적 세그멘테이션을 이용한 폴리포닉 오디오 신호의 정현파 모델링)

  • 장호근;박주성
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.4
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    • pp.58-68
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    • 2000
  • This paper proposes a sinusoidal modeling of polyphonic audio signals. Sinusoidal modeling which has been applied well to speech and monophonic signals cannot be applied directly to polyphonic signals because a window size for sinusoidal analysis cannot be determined over the entire signal. In addition, for high quality synthesized signal transient parts like attacks should be preserved which determines timbre of musical instrument. In this paper, a multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in sinusoidal modeling a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. The improved dynamic segmentation is proposed which shows better performance about transients and reduced computation. For various polyphonic audio signals the result of simulation shows the suggested sinusoidal modeling can model polyphonic audio signals without loss of perceptual quality.

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3D motion aftereffect in a static region after adaptation to an adjacent counterphase flickering region (역 위상 깜박임 영역 순응에 의해 유도된 인접 영역의 3차원 운동잔여 효과)

  • 김정훈;남종호;정찬섭
    • Korean Journal of Cognitive Science
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    • v.10 no.3
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    • pp.29-37
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    • 1999
  • Murakami and Cavanagh (1998a,b, 1999) reported a jitter aftereffect in a static random noise after a period of adaptation to a patch of dynamic random noise. To a account for this phenomenon. they proposed the retinal slip caused by a small eye movements in the unadapted area, which is usually compensated by the visual system to stabilize images but is unsuppressed due to the adaptation. We tested this hypothesis with new experimental method and stimuli that were supposed to nullify or reduce the effect. However. the aftereffect was still observed even under these stimuls conditions More importantly, the perceived aftereffect was rather different from Murakami and Cavanagh's. After adaptation to a counterphase flickering cosine grating, the adjacent unadapted region seems to move away from the observer during the test period instead of jittering in the frontoparallel plane. We proposed a possible explanation for this new phenomenon noting the severe contrast reduction of the adapted region during flickering period. The aftereffect might be due to the flicker-inducing contrast reduction during adaptation that produces different depth planes for the adapted and unadapted region and its restoration during the test period.

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A Study on the Improvement of Image Fusion Accuracy Using Smoothing Filter-based Replacement Method (SFR기법을 이용한 영상 융합의 정확도 향상에 관한 연구)

  • Yun Kong-Hyun
    • Spatial Information Research
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    • v.14 no.1 s.36
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    • pp.85-94
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    • 2006
  • Image fusion techniques are widely used to integrate a lower spatial resolution multispectral image with a higher spatial resolution panchromatic image. However, the existing techniques either cannot avoid distorting the image spectral properties or involve complicated and time-consuming decomposition and reconstruction processing in the case of wavelet transform-based fusion. In this study a simple spectral preserve fusion technique: the Smoothing Filter-based Replacement(SFR) is proposed based on a simplified solar radiation and land surface reflection model. By using a ratio between a higher resolution image and its low pass filtered (with a smoothing filter) image, spatial details can be injected to a co-registered lower resolution multispectral image minimizing its spectral properties and contrast. The technique can be applied to improve spatial resolution for either colour composites or individual bands. The fidelity to spectral property and the spatial quality of SFM are convincingly demonstrated by an image fusion experiment using IKONOS panchromatic and multispectral images. The visual evaluation and statistical analysis compared with other image fusion techniques confirmed that SFR is a better fusion technique for preserving spectral information.

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A DCT Adaptive Subband Filter Algorithm Using Wavelet Transform (웨이브렛 변환을 이용한 DCT 적응 서브 밴드 필터 알고리즘)

  • Kim, Seon-Woong;Kim, Sung-Hwan
    • The Journal of the Acoustical Society of Korea
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    • v.15 no.1
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    • pp.46-53
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    • 1996
  • Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

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Performance Analysis of SDR Communication System Based on MTD Technology (MTD 기법이 적용된 SDR 통신 시스템의 성능 분석)

  • Ki, Jang-Geun;Lee, Kyu-Tae
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.17 no.2
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    • pp.51-56
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    • 2017
  • With the rapid increase in the number of mobile terminals, demand for wireless technologies has sharply increased these days. While wireless communication provides advantages such as ease of deployment, mobility of terminals, continuity of session, and almost comparable transmission bandwidth to the wired communication, it has vulnerability to malicious radio attacks such as eavesdropping, denial of service, session hijacking, and jamming. Among a variety of methods of preventing wireless attacks, the MTD(Moving Target Defense) is the technique for improving the security capability of the defense system by constantly changing the ability of the system to be attacked. In this paper, in order to develop a resilient software defined radio communication testbed system, we present a novel MTD approach to change dynamically and randomly the radio parameters such as modulation scheme, operating frequency, packet size. The probability of successful attack on the developed MTD-based SDR communication system has been analysed in a mathematical way and verified through simulation.

ECG Signal Compression using Feature Points based on Curvature (곡률을 이용한 특징점 기반 심전도 신호 압축)

  • Kim, Tae-Hun;Kim, Sung-Wan;Ryu, Chun-Ha;Yun, Byoung-Ju;Kim, Jeong-Hong;Choi, Byung-Jae;Park, Kil-Houm
    • Journal of the Korean Institute of Intelligent Systems
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    • v.20 no.5
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    • pp.624-630
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    • 2010
  • As electrocardiogram(ECG) signals are generally sampled with a frequency of over 200Hz, a method to compress diagnostic information without losing data is required to store and transmit them efficiently. In this paper, an ECG signal compression method, which uses feature points based on curvature, is proposed. The feature points of P, Q, R, S, T waves, which are critical components of the ECG signal, have large curvature values compared to other vertexes. Thus, these vertexes are extracted with the proposed method, which uses local extremum of curvatures. Furthermore, in order to minimize reconstruction errors of the ECG signal, extra vertexes are added according to the iterative vertex selection method. Through the experimental results on the ECG signals from MIT-BIH Arrhythmia database, it is concluded that the vertexes selected by the proposed method preserve all feature points of the ECG signals. In addition, they are more efficient than the AZTEC(Amplitude Zone Time Epoch Coding) method.

Digital Watermarking Algorithm for Copyright Protection of JPEG Image (JPEG 영상의 저작권 보호를 위한 Digital Watermarking 알고리즘)

  • Park, Eun-Suk;Woo, Jong-Won;Lee, Seok-Hee;Heo, Yoon-Seok;Cho, Ki-Hyung
    • The Transactions of the Korea Information Processing Society
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    • v.7 no.1
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    • pp.296-305
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    • 2000
  • In this paper, we propose the method of embedding the encrypted digital watermark in quantization coefficient when we encode the image data in the process of JPEC. The proposed method is as following. After a DCT coefficient of each block is quantized, we arrange the quantization coefficient as on dimension with a zigzag scan and replace each block. By applying even-odd feature of frequency of the encrypted watermark to a quantization coefficient of some fixed domain of replaced each block and embedding it, we obtain the compressed image data by encoding after placing it in the order prior to replacement. The advantages of the proposed method here are as follows: We can embed many information keeping a secret as much as possible by using the algorithm of block replacement. We can control the amount of embedding of each use, as we embed the encrypted information by selecting some fixed domain of a quantization coefficient, we can fix the embedding data regardless of the image and the value of quantization. We verified the results by experiments and analyzed the efficiency of them in comparison with the former study.

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