• Title/Summary/Keyword: 신호적응필터

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Proposal of High Quality Audio DSP System using Flexible Filterbank for Pro-Audio Equipment (Pro-Audio 장비용 가변형 필터뱅크 기반 고품질 음향 DSP 시스템 개발을 위한 제안)

  • Song, Chai-Jong;Yang, Chang-Mo;Lim, Tea-Beom
    • Annual Conference of KIPS
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    • 2013.11a
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    • pp.1450-1451
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    • 2013
  • 본 논문에서는 음향 확성 환경과 음향신호의 입 출력 조건에 최적화된 음향 시스템을 실시간으로 생성 및 적용 가능한 음향 신호처리용 시스템으로서, 가변형 필터뱅크 기술 및 상황 적응적 필터 재조합 재배열 기술을 음향 신호처리용 DSP에 적용함으로서 Pro-Audio 장비, 방송 음향장비, 산업 음향장비와 같은 다양한 음향관련 장비에서 고품질 음향 서비스를 제공하기위한 핵심 기술인 가변형 필터뱅크 기반 고품질 음향 DSP 시스템 개발을 제안한다.

Noise Reduction of the Distribution Line Communication System using NLMS Adaptive Filters (NLMS 적응필터를 이용한 배전선 반송통신 시스템의 잡은 저감)

  • 정원용;최태원;김석순;이수흠;고희석
    • The Proceedings of the Korean Institute of Illuminating and Electrical Installation Engineers
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    • v.8 no.6
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    • pp.63-70
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    • 1994
  • 컴퓨터나 정밀기기기의 빈번한 운용으로 배전선 반송통신 시스템의 신뢰도는 그 어느 때보다 크게 요구된다. 배전선에 나타나는 여러 종류의 잡음은 반송통신 정보를 쉽게 오염시켜 전체 시스템에 큰 영향을 미치게 된다. 본 논문에서는 평활 잡음과 60[Hz] 전원 주파수에 동기된 고조파 잡음제거를 위해 LMS와 NLMS알고리즘을 이용한 적응필터를 사용하였고 컴퓨터 시뮬레이션을 수행해 본 결과 신호 대 잡음비의 관점에서 LMS보다 NLMS의 탁월한 효과를 확인하였다.

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RLSLTDE Algorithm for Bearing Estimation of the Underwater Acoustic Signal (수중음향신호 입사방위 추정을 위한 RLSLTDE 알고리즘)

  • Choi, Jae-Yong;Son, Kweon;Dho, Kyeong-Cheol
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.5
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    • pp.84-90
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    • 2000
  • The bearing detection of radiated target noise is very important at underwater acoustic measurement and passive detection. It differs the arrival tines of received signal at each sensor. Therefore, the bearing can be obtained from the time delay. This paper proposes a new algorithm using the RLSL adaptive filter for TDE. The proposed method is particularly attractive when there is a limitation of priori information about the received signal spectra and when the delay is subject to variation. As the simulation results, it is shown that the proposed algorithm has better convergence characteristics and TDE speed, and so that the usefulness of proposed algorithm is confirmed.

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A Robust Frequency-Domain Multi-Reference Narrowband Adaptive Noise Canceller (여러 개의 참고입력 신호를 사용하는 강인한 주파수 영역 협대역 잡음 제거기)

  • Kim, Seong-Woo;Seo, Ji-Ho;Ryu, Young-Woo;Park, Young-Cheol;Youn, Dae Hee
    • The Journal of the Acoustical Society of Korea
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    • v.34 no.2
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    • pp.163-170
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    • 2015
  • In this paper, it is shown that the performance of the frequency-domain multi-reference narrowband noise canceller is determined by the narrowband component to the broadband disturbance power ratio in the reference signals. To overcome this problem, a new narrowband ANC is proposed, where the update of the adaptive filter is determined based on SNR of the reference inputs being measured using the magnitude squared coherence (MSC) between the primary and the reference signals. Simulation results show that the proposed ANC has superior performance over the conventional one.

A Study on the Adaptive Neural Network Filter for Signal Detection (신호 검출을 위한 적응형 신경망 필터에 관한 연구)

  • 안종구;추형석
    • Journal of the Institute of Convergence Signal Processing
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    • v.5 no.2
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    • pp.132-137
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    • 2004
  • In this paper, the adaptive noise canceler using neural network with backpropagation is designed. The adaptive noise canceler using the least mean square algorithm has the large correlativity of the reference signal. The performance of the adaptive noise canceler shows the limitation when the information signal is relatively small to the noise. The system proposed in this paper plays an important role in denoising these signals. In addition, the experiments are carried out to analyze the effects of the number of hidden layers and nodes about the system. The performance of the proposed adaptive noise canceler is compared with that of the system which is used the least mean square algorithm.

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ECG Filtering using Empirical Mode Decomposition Method (EMD 방법을 이용한 ECG 신호 필터링)

  • Lee, Geum-Boon;Cho, Beom-Joon
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.13 no.12
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    • pp.2671-2676
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    • 2009
  • Empirical mode decomposition (EMD) is new time-frequency analysis method to decompose the signal adaptively and efficiently. The key idea of EMD is to decompose the signal into a set of functions defined by the signal itself, named Intrinsic Mode Functions (IMFs), which preserve the inherent properties of the original signal. Since the decomposition is based on the local time scale of the signal, it is not only applicable to nonlinear and non-stationary processes but also useful in biomedical signals like electrocardiogram (ECG). Traditional low-pass filter uses fourier transform to analysis signal in frequency domain, but EMD is filtered to maintain signal properties in time domain. This paper performed signal decomposition and filtering for noisy ECGs using EMD method. The proposed method is presented and compared with traditional low-pass filter by two performance indices. Our results show effectiveness for enhancement of the noisy ECG waveforms.

Design of an adaptive IIR notch filter to reject the interference in GPS Receiver (GPS 수신기 간섭 제거를 위한 적응 IIR 노치 필터 설계)

  • Lim, Deok-Won;Lee, Geon-Woo;Park, Chan-Sik;Hwang, Dong-Hwan;Lee, Sang-Jeong
    • Journal of the Institute of Electronics Engineers of Korea SC
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    • v.44 no.3
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    • pp.58-63
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    • 2007
  • GPS signal is vulnerable to intentional and unintentional interferences since it has a very weak signal power and its structure is well-known. Among the interference rejecting techniques, the ATF is being generally used as a pre-correlation technique. However, it does not have a design parameter relating to the notch width, resulting in the spectral loss around frequency of the interference. The IIR notch filter has a design parameter relating to the notch width and can generate a sharp notch for the CW interference. In this paper, an adaptive IIR notch filter is proposed and the performance is evaluated using software signal generator and real measurements.

Denosing of images using locally adaptive wiener filter in wavelet domain (웨이브렛 변환 영역에서의 국부적응 Wiener 필터에 의한 영상 신호의 잡음 제거)

  • 장익훈;김남철
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.22 no.12
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    • pp.2772-2782
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    • 1997
  • In this paepr, a Wiener filtering method in wavelet domain is proposed for restoring an image corrupted by additive white noise. The proposed method utilizes the characteristics of wavelet transform signals and the local statistics of each subband. When estimating the local statistics in each subband, the size of filter window is varied according to each scale. At this point, the local statistics in each wavelet subband is estimated only by using pixedls which have similar statistical property. Experimental results show that the proposed method has better performance over the conventional Lee filter with a window of fixed size.

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Performance of Adaptive Correlator using Recursive Least Square Backpropagation Neural Network in DS/SS Mobile Communication Systems (DS/SS 이동 통신에서 반복적 최소 자승 역전파 신경망을 이용한 적응 상관기)

  • Jeong, Woo-Yeol;Kim, Hwan-Yong
    • The Journal of the Acoustical Society of Korea
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    • v.15 no.2
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    • pp.79-84
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    • 1996
  • In this paper, adaptive correlator model using backpropagation neural network based on complex multilayer perceptron is presented for suppressing interference of narrow-band of direct sequence spread spectrum receiver in CDMA mobile communication systems. Recursive least square backpropagation algorithm with backpropagation error is used for fast convergence and better performance in adaptive correlator scheme. According to signal noise ratio and transmission power ratio, computer simulation results show that bit error ratio of adaptive correlator uswing backpropagation neural network improved than that of adaptive transversal filter of direct sequence spread spectrum considering of co-channel and narrow-band interference. Bit error ratio of adaptive correlator using backpropagation neural network is reduced about $10^{-1}$ than that of adaptive transversal filter where interference versus signal ratio is 5 dB.

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Subband Affine Projection Algorithm (부밴드 인접투사 알고리즘)

  • Choi, Hun;Bae, Hyeon Deok
    • The Journal of the Acoustical Society of Korea
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    • v.23 no.3
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    • pp.221-227
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    • 2004
  • This paper presents the subband affine projection algorithm(SAPA). The improved performance of SAPA is achieved by applying the affine projection algorithm to the subband adaptive structure. In this algorithm, the weight updating formula of adaptive filter is simply derived by using the orthogonal quadrature filter(OQF) as an analysis filter bank for subband filtering. The derived SAPA has the fast convergence speed and small computational complexity. The efficiency of the proposed algorithm for colored input signal is evaluated through some experiments.