• Title/Summary/Keyword: Adaptive Noise Cancellation

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Multi-Stage Adaptive Noise Cancellation Technique for Synthetic $Hard-{\alpha}$ Inclusion (합성 $Hard-{\alpha}$ Inclusion의 다단계 적응형 노이즈 제거기법 연구)

  • Kim, Jae-Joon
    • Journal of the Korean Society for Nondestructive Testing
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    • v.23 no.5
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    • pp.455-463
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    • 2003
  • Adaptive noise cancellation techniques are ideally suitable for reducing spatially varying noise due to the grain structure of material in ultrasonic nondestructive evaluation. Grain noises have an un-correlation property, while flaw echoes are correlated. Thus, adaptive filtering algorithms use the correlation properties of signals to enhance the signal-to-noise ratio (SNR) of the output signal. In this paper, a multi-stage adaptive noise cancellation (MANC) method using adaptive least mean square error (LMSE) filter for enhancing flaw detection in ultrasonic signals is proposed.

Local Adaptive Noise Cancellation for MCG Signals Based on Wavelet Transform (웨이브릿 변환을 기반으로 한 심자도 신호의 국소 적응잡음제거)

  • 김용주;박희준;원철호;이용호;김인선;김명남;조진호
    • Progress in Superconductivity
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    • v.5 no.1
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    • pp.26-30
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    • 2003
  • Magneto-cardiogram(MCG) signals may be highly distorted by the environmental noise, such as power-line interference, broadband white noise, surrounding magnetic noise, and baseline wondering. Several kinds of digital filters and noise cancellation methods have been designed and realized by many researchers, but these methods gave some problems that the original signal may be distorted by digital filter due to the wideband characteristics of background noise. To eliminate noise effectively without distortion of MCG signals, we performed multi-level frequency decomposition using wavelet packets and local adaptive noise cancellation in each local frequency range. In addition to the proposed wavelet filter to eliminate these various non-stationary noise elements, the local adaptive filter using the least mean square(LMS) algorithm and the soft threshold do-noising method are introduced in this paper. The signal to noise ratio(SNR) and the reconstruction square error(RSE) are calculated to evaluate the performance of the proposed method and compared with the results of the conventional wavelet filter and adaptive filter. The experimental results show that the proposed local adaptive filtering method is better than the conventional methods.

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Performance Analysis of Own Ship Noise Cancellation in Hull Mounted Sonar System Using Adaptive Filter (HMS시스템에서 적응필터를 이용한 자함의 소음감소 성능분석)

  • Yoon, Kyung-Sik;Jung, Tae-Jin;Lee, Kyun-Kyung
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.1
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    • pp.10-17
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    • 2010
  • In a passive sonar, the improvement of detection performance by using noise cancellation is usually a important problem. In this paper, we have analyzed the own-ship noise cancellation in the two operation modes which are used in the HMS system. In the operator mode, an adaptive line enhancer(ALE) is applied to improve the tonal detection by using broadband noise cancellation and the normalized least mean square(NLMS) algorithm is applied to the design of an adaptive filter. The reference input that is correlated with a primary input can be used to remove the noise incident on the observation directionin the automatic mode. Computer simulations with real sea that data show that the proposed adaptive noise canceller has good performance in passive detection under HMS operation.

Implementation of Real-Time Adaptive Noise Cancellation System Using DSP Processor (DSP 프로세서를 이용한 실시간 ANC 시스템 구현에 관한 연구)

  • Lee Young Il;Choi Hong Sub
    • MALSORI
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    • no.52
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    • pp.121-132
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    • 2004
  • This paper is aiming at real-time implementation of adaptive noise cancellation system using DSP processor. ACHARF algorithm, which guarantees stability and fast convergence by adaptive compensator, is used on this DSP system. For the experiments, TLV320AIC23 stereo CODEC of TI Inc. is used with TMS320C6413 DSP processor. Signals of primary input and reference input are obtained by two microphones. The primary input is the voice plus noise signal and the reference input is white noise or real noise. The experimental results show that ANC system using DSP processor with ACHARF is verified to be an effective speech enhancement method for various speech processing units.

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Adaptive inverse feedback control of periodic noise for systems with nonminimum phase cancellation path (비최소위상 상쇄계를 가진 시스템을 위한 주기소음의 적응 역 궤환 제어)

  • Kim, Sun-Min;Park, Young-Jin
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2000.11a
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    • pp.437-442
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    • 2000
  • An alternative inverse feedback structure for adaptive active control of periodic noise is introduced for systems with nonminimum phase cancellation path. To obtain the inverse model of the nonminimum phase cancellation path, the cancellation path model can be factorized into a minimum phase term and a maximum phase term. The maximum phase term containing unstable zeros makes the inverse model unstable. To avoid the instability, we alter the inverse model of the maximum phase system into an anti-causal FIR one. An LMS predictor estimates the future samples of the noise, which are necessary for causality of both anti-causal FIR approximation for the stable inverse of the maximum phase system and time-delay existing in the cancellation path. The proposed method has a faster convergence behavior and a better transient response than the conventional FX-LMS algorithms with the same internal model control structure since a filtered reference signal is not required. We compare the proposed methods with the conventional methods through simulation studies.

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Adaptive noise cancellation algorithm reducing path misadjustment due to speech signal (음성신호로 인한 잡음전달경로의 오조정을 감소시킨 적응잡음제거 알고리듬)

  • 박장식;김형순;김재호;손경식
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.21 no.5
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    • pp.1172-1179
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    • 1996
  • General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.

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Design of Reliable Adaptive Fitter with Fault Tolerance Using DSP (DSP를 이용한 고장허용을 갖는 신뢰 적응 필터 설계)

  • 유동완;이전우;서보혁
    • The Transactions of the Korean Institute of Electrical Engineers D
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    • v.50 no.1
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    • pp.8-13
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    • 2001
  • LMS algorithm has been used for plant identifier and noise cancellation. This algorithm has been researched for performance enhancement of filtering. The design and development of a reliable system has been becoming a key issue in industry field because the reliability of a system is considered as an important factor to perform the system's function successfully. And the computing with reliability and fault tolerance is a important factor in the case of aviation, system communication, and nuclear plant. This paper presents design of reliable adaptive filter with fault tolerance. Generally, redundancy is used for reliability. In this case it needs computing or circuit for voting mechanism, or fault detection. Therefore it has simple computing, and practicality for application. And in this paper, reliability of adaptive filter is analyzed. The effectiveness of the proposed adaptive filter is demonstrated to the case studies of plant identifier and noise cancellation by using DSP.

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Adaptive Inverse Feedback Control of Periodic Noise for Systems with Nonminimum Phase Cancellation Path (비최소위상 상쇄계를 가진 시스템을 위한 주기소음의 적응 역 궤환 제어)

  • Kim, Sun-Min;Park, Young-Jin
    • Journal of Institute of Control, Robotics and Systems
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    • v.7 no.11
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    • pp.891-895
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    • 2001
  • An alternative inverse feedback structure for adaptive active control of periodic noise is introduced for systems with nonminimum phase cancellation path. To obtain the inverse model of the nonminimum phase cancellation path, the cancellation path model can be factorized into a minimum phase term and a maximum phase term. The maximum phase term containing unstable zeros makes the inverse model unstable. To avoid the instability, we alter the inverse model of the maximum phase system into an anti-causal FIR one. An LMS predictor estimates the future samples of the noise, which are necessary for causality of both anti-causal FIR approximation for the stable inverse of the maximum phase system and time-delay existing in the cancellation path. The proposed method has a faster convergence behavior and a better transient response than the conventional filtered-x LMS algorithms with the same internal model control structure since a filtered reference signal is not required. We compare the proposed methods with the conventional methods through simulation studies.

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Adaptive Active Noise Control of Single Sensor Method (단일 센서 방식의 적응 능동 소음제어)

  • 김영달;장석구
    • Journal of KSNVE
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    • v.10 no.6
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    • pp.941-948
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    • 2000
  • Active noise control is an approach to reduce the noise by utilizing a secondary noise source that destructively interferes with the unwanted noise. In general, active noise control systems rely on multiple sensors to measure the unwanted noise field and the effect of the cancellation. This paper develops an approach that utilizes a single sensor. The noise field is modeled as a stochastic process, and an adaptive algorithm is used to adaptively estimate the parameters of the process. Based on these parameter estimates, a canceling signal is generated. Oppenheim assumed that transfer function characteristics from the canceling source to the error sensor is only a propagation delay. This paper proposes a modified Oppenheim algorithm by considering transfer characteristics of speaker-path-sensor This transfer characteristics is adaptively cancelled by the proposed adaptive modeling technique. Feasibility of the proposed method is proved by computer simulations with artificially generated random noises and sine wave noise. The details of the proposed architecture. and theoretical simulation of the noise cancellation system for three dimension enclosure are presented in the Paper.

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