• Title/Summary/Keyword: adaptive signal processing

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A Study On ECLMS Using Estimated Correlation (추정상관을 이용한 ECLMS에 관한 연구)

  • 오신범;권순용;이채욱
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.27 no.7A
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    • pp.651-658
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    • 2002
  • Although least mean square(LMS) algorithm is known to one of the most popular algorithm in adaptive signal processing because of the simplicity and the small computation, the choice of the step size reflects a tradeoff between the misadjustment and the speed of adaptation. In this paper, we present a new variable step size LMS algorithm, so-called ECLMS(Estimated correlation LMS), using the correlation between reference input and error signal of adaptive filter. The proposed algorithm updates each weight of filter by different step size at same sample time. We applied this algorithm to adaptive multiple-notch filter. Simulation results are presented to compare the performance of the proposed algorithm with the usual LMS algorithm and another variable step algorithm.

Adaptive Noise Reduction Algorithm for an Image Based on a Bayesian Method

  • Kim, Yeong-Hwa;Nam, Ji-Ho
    • Communications for Statistical Applications and Methods
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    • v.19 no.4
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    • pp.619-628
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    • 2012
  • Noise reduction is an important issue in the field of image processing because image noise lowers the quality of the original pure image. The basic difficulty is that the noise and the signal are not easily distinguished. Simple smoothing is the most basic and important procedure to effectively remove the noise; however, the weakness is that the feature area is simultaneously blurred. In this research, we use ways to measure the degree of noise with respect to the degree of image features and propose a Bayesian noise reduction method based on MAP (maximum a posteriori). Simulation results show that the proposed adaptive noise reduction algorithm using Bayesian MAP provides good performance regardless of the level of noise variance.

Adaptive Noise Reduction of the Frequency Domain using the MDFT in the Biomedical Signal

  • Jung, Yen-Tae;Yoon, Dal-Hwan;Bae, Dong-Ju;Lee, Sun-Hyo;Jung, Ye-Heon;Kim, Seong-Woo
    • Proceedings of the IEEK Conference
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    • 2005.11a
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    • pp.505-508
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    • 2005
  • This paper presents the high speed noise reduction processing system using the MDFT on the frequency domain. The proposed system use the linear prediction coefficients of the AR method based on the SLMS(sign least mean square). The signals with a random noise tracking per-formance are examined through computer simu-lations. It is confirmed that the high speed adaptive noise reduction processing system is realized by the SLMS algorithms with rapid convergence on the FD(frequency domain).

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On the adaptive pre-processing technique for the linerization of weakly nonlinear volterra systems (볼테라 시스템 선형화를 위한 적응 선행처리 기법)

  • Choi, Bong-Joon;Kim, Yong-Nam;Chung, Ji-Hyun;Nam, Sang-Won
    • Journal of Institute of Control, Robotics and Systems
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    • v.3 no.5
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    • pp.450-454
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    • 1997
  • 본 논문에서는 볼테라 비선형 시스템의 선형화를 위한 새로운 적응 선행처리 기법을 제시한다. 특히, 제안된 적응 선행처리 기법은 (i) 순수 비선형 왜곡 보상을 위한 부분(pure nonlinear distortion compensator: PNDC)과, (ii) 선형 왜곡 보상을 위한 선형 역필터(linear inverse filter: LIF)의 두 부분으로 구성된다. 본 논문의 선형화 기법의 장점으로는 기존의 P차 역(Pth-order inverse) 기법에 비하여 계산량이 상당히 감소되며, 적응 알고리듬이 보다 빠르고 안정된 수렴 특성을 나타낸다. 끝으로, 모의실험을 통하여, 제안된 선행처리 기법의 성능및 실제 적용 가능성을 살펴본다.

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An Echo Canceller Robust to Noise and Residual Echo

  • Kim, Hyun-Tae;Park, Jang-Sik
    • Journal of information and communication convergence engineering
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    • v.8 no.6
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    • pp.640-644
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    • 2010
  • When we talk with hands-free in a car or noisy lobby, the performance of the echo canceller degrade because background noise added to echo caused by the distance from mouth to microphone is relatively long. It gives a reason for necessity of noise-robust and high convergence speed adaptive algorithm. And if acoustic echo canceller operated not perfectly, residual signal going through the echo canceller to far-end speaker remains residual echo, which degrade quality of talk. To solve this problem, post-processing needed to remove residual echo ones more. In this paper, we propose a new acoustic echo canceller, which has noise robust and high convergence speed, linked with linear predictor as a post-processor. By computer simulation, it is confirmed that the proposed algorithm shows better performance from acoustic interference cancellation (AIC) viewpoint.

A Block-Based Adaptive Data Hiding Approach Using Pixel Value Difference and LSB Substitution to Secure E-Governance Documents

  • Halder, Tanmoy;Karforma, Sunil;Mandal, Rupali
    • Journal of Information Processing Systems
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    • v.15 no.2
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    • pp.261-270
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    • 2019
  • In order to protect secret digital documents against vulnerabilities while communicating, steganography algorithms are applied. It protects a digital file from unauthorized access by hiding the entire content. Pixel-value-difference being a method from spatial domain steganography utilizes the difference gap between neighbor pixels to fulfill the same. The proposed approach is a block-wise embedding process where blocks of variable size are chosen from the cover image, therefore, a stream of secret digital contents is hidden. Least significant bit (LSB) substitution method is applied as an adaptive mechanism and optimal pixel adjustment process (OPAP) is used to minimize the error rate. The proposed application succeeds to maintain good hiding capacity and better signal-to-noise ratio when compared against other existing methods. Any means of digital communication specially e-Governance applications could be highly benefited from this approach.

A VLSI Architecture for the Real-Time 2-D Digital Signal Processing (실시간 2차원 디지털 신호처리를 위한 VLSI 구조)

  • 권희훈
    • Information and Communications Magazine
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    • v.9 no.9
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    • pp.72-85
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    • 1992
  • The throughput requirement for many digital signal processing is such that multiple processing units are essential for real-time implementation. Advances in VLSI technology make it feasible to design and implement computer systems consisting of a large number of function units. The research on a very high throughput VLSI architecture for digital signal processing applications requires the development of an algorithm, decomposition scheme which can minimize data communication requirements as well as minimize computational complexity. The objectives of the research are to investigate computationally efficient algorithms for solution of the class of problems which can be modeled as DLSI systems or adaptive system, and develop VLSI architectures and associated multiprocessor systems which can be used to implement these algorithms in real-time. A new VLSI architecture for real-time 2-D digital signal processing applications is proposed in this research. This VLSI architecture extends the concept of having a single processing units in a chip. Because this VLSI architecture has the advantage that the complexity and the number of computations per input does not increase as the size of the input data in increased, it can process very large 2-D date in near real-time.

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Temporal Filter for Image Data Compression (영상 데이터 압축을 위한 Temporal Filter의 구성)

  • 김종훈;김성대
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.18 no.11
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    • pp.1645-1654
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    • 1993
  • Unlike a noise removal recursive temporal filter, this paper presents a temporal filter which improves visual quality and data compression efficiency. In general, for the temporal band-limitation, temporal aliasing should be considered. Since most of a video signal has temporally aliased components, it is desirable to consider them. From a signal processing point of view, it is impossible to realize the filtering not afeced by the aliasings. However, in this paper, efficient filtering with de-aliasing characteristics is proposed. Considering the location of a video signal, temporal filtering can be accomplished by the spatial filtering along the motion vector trajectory (Motion Adaptive Spatial Filter). This filtered result dose not include the aliasings. Besides the efficient band-limitation, temporal noise is also reduced. For the evaluation of the MASF, its realization and filtering characteristics will be discussed in ditail.

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Study on the Simultaneous Control of the Seam tracking and Leg Length in a Horizontal Fillet Welding Part 1: Analysis and Measurement of the Weld Bend Geometry

  • Moon, H.S.;Na, S.J.
    • International Journal of Korean Welding Society
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    • v.1 no.1
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    • pp.23-30
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    • 2001
  • Among the various welding conditions, the welding current that is inversely proportional to the tip-to-work-piece distance is an essential parameter as to monitor the GMAW process and to implement the welding automation. Considering the weld pool surface geometry including weld defects, it should modify the signal processing method for automatic seam tracking in horizontal fillet welding. To meet the above necessities, a mathematical model related with the weld pool geometry was proposed as in a conjunction with the two-dimensional heat flow analysis of the horizontal fillet welding. The signal processing method based on the artificial neural network (Adaptive Resonance Theory) was proposed for discriminating the sound weld pool surface from that with the weld defects. The reliability of the numerical model and the signal processing method proposed were evaluated through the experiments of which showed that they are effective for predicting the weld bead shape with or without the weld defects in a horizontal fillet welding.

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The cancellation performance of loop-back signal in wireless USN multihop relay node (무선 USN 멀티홉 중계 노드에서 루프백 신호의 제거 성능)

  • Lim, Seung-Gag;Kang, Dae-Soo
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.9 no.4
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    • pp.17-24
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    • 2009
  • This paper deals with the cancellation performance of loop back interference signal in the case of multihop relay of 16-QAM received signal at the USN radio network. For this, it is necessary to the exchange of information with long distance located station by means of the relay function between the node in the USN environment. In the relay node, the loop-back interference signal which the retransmitting signal is feedback to the receiver side due to the antenna of transmitter and receiver are co-used or very colsely located or using the nonlinear device. Due to this signal, the performance of USN system are degraded which are using the limited resource of frequency and power. For improve this, it is necessary to applying the adaptive signal processing algorithm in order to cancellating the unwanted loop-back interference signal at the frontend of receiver in relaying node, we can get the better system and multi hop performance. In the adaptive signal processing, we considered the 16-QAM signal which has a good spectral efficiency, firstly, than, the QR-Array RLS algorithm was used that has a fairly good convergence property and the solving the finite length problem in the H/W implementation. Finaly, we confirmed that the good elimination performanc was confirmed by computer simulation in the learing cuved and received signal constellation compared to the conventional RLS.

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