• 소음진동
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• 제10권6호
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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.

• 대한전기학회논문지:시스템및제어부문D
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• 제49권8호
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• pp.442-450
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• 2000

Active noise control is an approach to noise reduction in which 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 a time-adaptive algorithm is used to adaptively estimate the parameters of the process. Based on these parameter estimates, a canceling signal is generated. Opppenheim model assumed that transfer function characteristics from the canceling source to the error sensor is only propagation delay. But this paper proposes a modified Oppenheim model by considering transfer characteristics of acoustic device and noise path. This transfer characteristics is adaptively cancelled by adaptive model. This is proved by computer simulation with artifically generated random noise and sine wave noise. The details of the proposed architecture, and theoretical simulation and experimental results of the noise cancellation system for three dimension enclosure are presented in the paper.

• 전자공학회논문지B
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• 제31B권7호
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• pp.33-41
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• 1994

A single channel adaptive noise canceling (ANC) technique with a recursive time delay estimator (RTDE) is presented for removing effects of additive noise on the speech signal. While the conventional method makes a reference signal for the adaptive filter using the pitch estimated on a frame basis from the input speech, the proposed method makes the reference signal using the delay estimated recursively on a sample-by-sample basis. As the RTDEs, the recursion formulae of autocorrelation function (ACF) and average magnitude difference function (AMDF) are derived. The normalized least mean square (NLMS) and recursive least square (RLS) algorithms are applied for adaptation of filter coefficients. Experimental results with noisy speech demonstrate that the proposed method improves the perceived speech quality as well as the signal-to-noise ratio and cepstral distance when compared with the conventional method.

• 전기학회논문지
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• 제59권3호
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• pp.669-673
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• 2010

This paper proposes a new active noise control (ANC) system for canceling broadband noise. The proposed ANC system is designed based on the multiband-structured delayless subband adaptive filter (MDSAF), which has advantages of fast-convergence speed and higher noise reduction performance by eliminating the aliasing and band-edge effects caused by band-partitioning. The simulation results show that the proposed ANC system has faster convergence speed as compared to the conventional ANC systems and effectively reduces the wideband noise.

• 대한전자공학회논문지SP
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• 제41권3호
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• pp.231-238
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• 2004

주파수 영역에서의 적응 신호처리는 입력의 자기 상관 행렬에 이산 퓨리에 변환(DFT Discrete Fourier Transform)을 이용할 때 거의 대각선화 되는 특성으로 인해 시간영역 적응필터보다 주파수 영역 적응 필터가 빠르게 적응한다. 본 논문에서는 변형된 이산 퓨리에 변환(MDFT: modified DFT)을 이용하여 주파수 영역 적응 필터를 설계함으로써 안정한 수렴 속도를 갖는 잡음 제거 시스템을 제안한다. 제안한 구조는 MDFT를 이용하여 연산수를 최소화하며, 안정한 수렴을 유지하면서 블록 없는 처리를 할 수 있고, 최적의 수렴 속도를 위해 입력 자기 상관 행렬에 MDFT를 사용해 근사적으로 대각화 시키고 시간적으로 변하는 스텝 크기를 정규화 하는 고속 적응 잡음 제거(HANR: high speed adaptive noise reduction) 알고리즘이다. HANR 알고리즘을 적용한 필터는 DFT변환법을 사용한 LMS방법(non-proposed)보다 30%정도의 속도 개선이 있다.

• 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.

• 대한의용생체공학회:의공학회지
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• 제34권4호
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• pp.204-211
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• 2013

In this paper, an algorithm for suppressing acoustic noises contained in stethoscope sound is proposed and implemented in real-time using an embedded DSP system. Sound collected by stethoscope is down-sampled and band-pass filtered, and later an NLMS adaptive filter is used to cancel the acoustic noise induced from external noise sources. Also, the unpredictable impulsive noises due to fabric friction and instantaneous tapping are detected using the SD-ROM algorithm, and suppressed using an algorithm approximating the morphology filter. The proposed algorithm was tested using signals collected with a digital stethoscope mockup, and implemented on an ARM920T-based DSP system.

• 한국음향학회지
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• 제13권3호
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• pp.16-24
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• 1994

음성신호에 내재한 배경잡음을 제거하는 단일입력 적응잡음제거 시스템을 구성하였다. 기존 방법에서는 프레임 단위로 분석된 음성신호의 피치 정보를 이용하여 적응여파기의 기준신호를 얻는데 비해 제안된 방법에서는 매 샘플마다 지연 정보를 추정하여 기준신호를 만든다. 입력되는 음성신호로부터 매 샘플시간마다 지연 정보를 구하기 위하여 일반적인 자기상관 함수와 평균절대차 함수로부터 재귀적 자기상관함수와 재귀적 평균절대차함수를 유도하였다. 정규화된 최소평균자승(NLMS) 적응알고리듬을 사용하는 단일입력 잡음제거 시스템에 제안된 지연추정 방법을 적용하여 백색 가우시안 잡음에 왜곡된 음성에 대해 음성개선 실험을 하였으며, 기존 방법과의 성능비교 실험을 하였다. 제안된 방법에 의한 음성개선이 기존 방법보다 음질 및 SNR면에서 더 좋은 결과를 보였다.

• Journal of Electrical Engineering and Technology
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• 제12권4호
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• pp.1627-1633
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• 2017

This paper proposes a voice enhancing ear-protection system which is based on feedback active noise control(FBANC). The proposed system selectively suppresses the background noise and preserves the talking voice by controlling the adaptive algorithm with the voice activity period detection module. The noise reduction performance of the proposed noise canceling algorithm is analytically derived for the two key performance affecting parameters, i.e., electro-acoustic coupling distance and noise bandwidth. The proposed system is also implemented with a floating-point DSP system and its performance is experimentally tested to compare with the analytically derived results. The achieved levels of noise reduction for the three different noise bandwidths cases, i.e., 10Hz, 50Hz, and 90Hz, are high to show 17.05dB, 10.54dB and 8.99dB, respectively. The feasibility of the proposed system is also shown by the peak noise reduction achieved more than 25dB while preserving the voice component in the frequency range between 200-800Hz.

• Progress in Superconductivity
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• 제4권1호
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• pp.53-58
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• 2002

Noise from electronic instrumentation is invariably present in biomedical signals, although the art of instrumentation design is such that this noise source may be negligible. And sometimes signals of interest are contaminated or degraded by signals of similar type from another source. Biomedical signals are omni-presently contaminated by these background noises that span nearly all frequency bandwidths. In the magneto-cardiogram (MCG), several digital filters have been designed for the elimination of the power-line interference, broadband white noise, surrounding magnetic noise, and baseline wondering. In addition to the introduced FIR filter, notch, adaptive filter using the least mean square (LMS) algorithm, and recurrent neural network (RNN) filter, a new filtering method for effective noise canceling in MCG signals is proposed in this paper, which is realized by the wavelet packets. The experimental results show that the proposed filter using wavelet packet performs efficiently with respect to noise rejection. To verify this, two characteristics were analyzed and compared with LMS adaptive filter, SNR of filtered signal and attractor pattern using the nonlinear dynamics.