• 전기학회논문지P
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• 제63권4호
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• pp.260-265
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• 2014

Recently, many active noise control (ANC) systems, which employ the adaptive filter controlling method, have been reported for eliminating unwanted noise. ANC systems based on the filtered-X least mean square (FXLMS) algorithm have a problem with compensating the acoustic feedback of secondary route. It is difficult to apply the real time, because transfer function of secondary route must be measured by off-line method to solve this problem. In this paper, we propose the ANC system that applies a correlation LMS(CLMS) algorithm for improving a problem of transfer function measurement. The proposed algorithm is based on input of road noise. The proposed ANC systems have an advantage of real-time process without degradation of performance, although there are many calculation compared with FXLMS algorithm.

• 전기학회논문지P
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• 제64권4호
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• pp.267-272
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• 2015

Active noise control (ANC) is a method of cancelling a noise signal in an acoustic cavity by generating an appropriate anti-noise signal via canceling loudspeakers. The continuous progress of ANC involves the development of improved adaptive signal processing algorithms, transducers, and DSP hardware. In this paper, the convergence behavior and the stability of the FxLMS algorithm in ANC systems with real-time implementation is proposed. Specially, The advanced DSP H/W with dual core(DSP+ARM) and API(application programming interface) S/W programming was developed to improve the real-time implementation performance under the FxLMS algorithms of input noise such as road noise environment. The experimental results are found to be in good agreement with the theoretical predictions.

• 전자공학회논문지
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• 제50권9호
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• pp.172-176
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• 2013

This paper presents a modified filtered-x least mean square (FxLMS) algorithm to improve the stability of active noise control (ANC) system in realistic environment. A real-time ANC system employing modified FxLMS is designed and implemented on digital signal processor (DSP) board. The ANC system is evaluated for cancelling various tonal frequency noises in the range from 100 to 500 Hz and the performance is measured in terms of sound pressure level (SPL) attenuation. Experiment results show that a quiet zone with maximum 20 dB SPL attenuation can be generated around the location of error microphone.

• 한국통신학회논문지
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• 제33권4C호
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• pp.305-311
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• 2008

본 논문에서는 실시간 자동차 환경에서 VSSNLMS(variable step size normalized least mean square)를 이용하여 적응 잡음 제거 알고리즘을 제안한다. 기본적인 ANC(adaptive noise canceller)알고리즘인 LMS알고리즘은 알고리즘의 간단성 때문에 가장 많이 사용되고 있다. 그러나 LMS알고리즘은 수렴율과 실시간 환경에서의 정확성 사이에서 문제를 가지고 있다. 이러한 문제를 풀기 위해, 비정장성 환경에서 잡음제거를 위해 VSSLMS알고리즘이 사용된다. 본 논문에서 실시간 데이터 입력 시스템을 사용하여 컴퓨터 시뮬레이션 함으로써, VSSLMS알고리즘이 LMS알고리즘에 비해 수렴율과 정확성 이 모두에 더 효율적이라는 것을 입증한다.

• 전기학회논문지P
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• 제58권4호
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• pp.385-390
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• 2009

This paper presents a novel active noise control (ANC) system using least mean square (LMS) algorithm and neural network approach for decreasing acoustic noise signals inside high-speed trains. We construct a LMS framework as a nominal ANC system and additionally design an artificial single-layered perceptron model as an auxiliary ANC which is aimed to reduce real-time residuary noise due to its nonstationary and uncertain nature. Parameter vector of the hybrid ANC is determined through online estimation to realize an adaptive ANC configuration by means of the steepest descent algorithm. We achieve simulation experiment to demonstrate the proposed ANC system employing realistic acoustic noise signals measured in Korea Train eXpress (KTX).

• 전기학회논문지P
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• 제64권3호
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• pp.169-174
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• 2015

Noise problem that occurs on the road is raising a lot of problems in the economic, social and environmental aspects. The active noise control (ANC) systems based on the filtered-X least mean square(FxLMS) algorithm have a problem with compensating the acoustic feedback of secondary route. However, newly proposed correlation-LMS(CLMS) and expanded CLMS algorithms have slightly much calculation and are minutely behind performance, these have a advantage not in measuring transfer function onerously so that we can easily adapt these in real time. The CLMS and expanded CLMS algorithm was developed to improve the real-time implementation performance under the variable input noise such as road noise environment. In this paper, we compared and analyzed their performance. From the results of the Matlab simulation for an ANC system, it is shown that expanded CLMS algorithms are more convergence speed and keep the desirable performance even in the input of road noise situation.

• 한국통신학회논문지
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• 제36권6C호
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• pp.343-348
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• 2011

이 논문은 NFxLMS (normalized filtered-x least mean square) 적응 필터기반의 실시간 ANC (active noise control) 무선 헤드셋에 관해서 연구한다. RMS (root mean square) 지연 분포를 측정한 후 채널을 보정해서 학습시간을 줄이고, 학습 시간동안 NFxLMS 필터 계수를 갱신해서 잡음 제거 필터의 수렴속도를 개선하는 방법을 제안한다. 제안한 방법을 실제 잡음 환경에서 이용할 경우에 짧은 학습 시간과 빠른 수렴속도를 가지면서 기존 잡음 제거기와 비슷한 성능을 가지는 잡음 제거기를 구성할 수 있다.

• 음성과학
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• 제8권2호
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• pp.73-81
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• 2001

In this paper, we focus on the real time implementation of the single channel adaptive noise canceller(ANC) by using TMS320C30 EVM board. The implemented single channel adaptive noise canceller is based on a reference paper [1] in which it is simulated by using the recursive average magnitude difference function(AMDF) to get a properly delayed input speech on a sample basis as a reference signal and normalized least mean square(NLMS) algorithm. To certify results of the real time implementation, we measured the processing time of the ANC and enhancement ratio according to various signalto-noise ratios(SNRs). Experimental results demonstrate that the processing time of the speech signal of 32ms length with delay estimation of every 10 samples is about 26.3 ms, and almost the same performance as given in [1] is obtained with the implemented system.

• 대한음성학회지:말소리
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• 제52호
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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.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1998년도 제13차 학술회의논문집
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• pp.81-84
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• 1998

For ANC systems applied to aircrafts or passenger ships, engines from which reference signals are usually measured are located so far from seats where main part of controllers are placed. It can make feedforward ANC scheme difficult to implement or very costly. Feedback ANC algorithms which do not require reference signals and use error signals alone to update the filter, are usually sensitive to measurement noise ' and impulse noise. In this paper, reference signal needed for the feedforward control is not measured directly but generated with the estimated frequencies. Cascade adaptive notch filter (ANF), which has the low computational burden, is used to implement ANC system in real time. Several ANFs of order 2 are connected in series to estimate multiple sinusoids. Computer simulations and experiments in the laboratory for verifying efficacy of the proposed algorithm are carried out.