• The Transactions of the Korean Institute of Power Electronics
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• v.2 no.2
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• pp.35-40
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• 1997

The hysteresis control method has been frequently used for current control of power conversion equipments or motor drive systems. This method makes the measured signal follow the reference signal by changing the control signal whenever the error signal exceeds the preset band width. In this paper, hysteresis control method with fast response characteristics is applied for active noise control to suppress acoustic noise. Both Pentium processor and sound blaster 16 are used for experimental implementation, which executes A/D, D/A conversion and also is used as operating source of loudspeaker for audible noise cancellation.

• The Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.272-279
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• 2016

In this paper, the realization of a hearing aid adaptively cancelling feedback noise was considered. Conventional least mean square method in time domain was transformed into frequency domain in order to minimize computational burden. The adaptive filter algorithm was evaluated by Matlab (Matrix laboratory), and it was confirmed by CSR 8675 Bluetooth DSP IC (Digital Signal Processor Integrated Circuit) chip firmware realization. Some remote control features by a smart phone was added to the smart hearing aid for user interface easiness.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.64 no.12
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• pp.1756-1760
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• 2015

This paper focuses on a priori signal to noise ratio (SNR) estimation method for the speech enhancement. There are many researches for speech enhancement with several ambient noise cancellation methods. The method based on spectral subtraction (SS) which is widely used in noise reduction has a trade-off between the performance and the distortion of the signals. So the need of adaptive method like an estimated a priori SNR being able to making a high performance and low distortion is increasing. The decision directed (DD) approach is used to determine a priori SNR in noisy speech signals. A priori SNR is estimated by using only the magnitude components and consequently follows a posteriori SNR with one frame delay. We propose a modified a priori SNR estimator and the weighted rational transfer function for speech enhancement with wind noises. The experimental result shows the performance of our proposed estimator is better Perceptual Evaluation of Speech Quality scores (PESQ, ITU-T P.862) compare to the conventional DD approach-based systems and different noise reduction methods.

• Proceedings of the KOSOMBE Conference
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• v.1996 no.05
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• pp.143-146
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• 1996

An adaptive digital filtering of the electroencophalogram(EEG) is a successful way of suppressing mains interference, but it affects some of the frequency components of the signal, an artifact which not be acceptable in some cases of automatic EEG processing. The types of electrical artifact seen on EEG records is described. Those are the EOG and the PVC roller pump noise. And we study the method for cancelling these artifacts. The method does not need the reference channel, and are obtained by cascading the linear predictor and the noise canceller. The simulation results illustrate the performances of the presented method in terms of the capability of interferences suppression.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2018.11a
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• pp.73-76
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• 2018

A Gaussian noise is caused by surrounding environment or channel interference when transmitting image. The noise reduces not only image quality degradation but also high-level image processing performance. The Non-Local Means (NLM) filter finds similarity in the neighboring sets of pixels to remove noise and assigns weights according to similarity. The weighted average is calculated based on the weight. The NLM filter method shows low noise cancellation performance and high complexity in the process of finding the similarity using weight allocation and neighbor set. In order to solve these problems, we propose an algorithm that shows an excellent noise reduction performance by using Summed Square Image (SSI) to reduce the complexity and applying the weighting function based on a cosine Gaussian kernel function. Experimental results demonstrate the effectiveness of the proposed algorithm.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.14 no.11
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• pp.1191-1197
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• 2003

In the multi-carrier OFDM communication system, the inter-carrier-interference(ICI) produced by phase noise in the transceiver local oscillator makes a severe influence on the system performance. In this paper, a new ICI self-cancellation scheme in the data-conversion type is proposed to reduce effectively the ICI. Also, the common phase error(CPE), ICI and carrier to interference power ratio(CIR) are found by the linear approximation of the phase noise. Then, the proposed method is compared with the conventional OFDM to analyze the efficiency of system performance improvement. When the number of subcarriers is 64, there are respectively the SNR gain of 0.6 ㏈ in the phase noise variance of 0.3 with QPSK and 1.5 ㏈ in the phase noise variance of 0.1 with 16 QAM at BER=10$\^$-3/. As a result, the performance degradation by ICI can be effectively lowered in the proposed system with ICI self. cancellation scheme, compared with the conventional OFDM system.

• The Journal of the Acoustical Society of Korea
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• v.13 no.3
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• pp.16-24
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• 1994

A single channel adaptive noise canceling (ANC) technique is presented for removing effects of additive noise on the speech signal. The conventional method obtains a reference signal using the pitch estimated on a frame basis from the input speech. The proposed method, however, gets the reference signal using the delay estimated recursively on a sample by sample basis. To estimate the delay, we derive recursion formula of autocorrelation function and average magnitude difference function. The performance of the proposed method is evaluated for the speech signals distorted by the additive white Gaussian noise. Experimental results with normalized least mean square (NLMS) adaptive algorithm demonstrate that the proposed method improves the perceived speech quality quite well besides the signal-to-noise ratio.

• Journal of KSNVE
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• v.8 no.6
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• pp.1037-1042
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• 1998

This paper discusses the dependence of the convergence rate on the acoustic error path in these popular algorithms and introduces new algorithms which increase the convergence region regardless of the time-delay in the acoustic error path. We also Propose a novel control algorithm (AFC/CAFC) for tonal noise cancellation. The proposed algorithm estimates the magnitude and phase of the tonal noise. The algorithm uses the steepest descent method for the phase/magnitude estimation. Performances of tile CAFC algorithm are presented in comparison with those by the AFC algorithm based on computer simulations and experiments.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4E
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• pp.178-182
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• 2002

In this paper, we propose an adaptive wavelet method for car interior noise cancellation. For this purpose, we use a node dependent threshold which minimizes the Bayesian risk. We propose a noise estimation method based on spectral entropy using histogram of intensity and a candidate best basis instead of Donoho's best bases. And we modify the hard threshold function. Experimental results show that the proposed algorithm is more efficient, especially to heavy noisy signal than conventional one.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.13 no.3
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• pp.17-24
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• 1999

This paper presents a method of ANC for transfonrer noise control utilizing a sproker and microphone pair. In this study, the main focus is on identifying the dynamic characteristics of speaker - amplifier microphone path. This study presents a theoretical method to identify the dynamic characteristics of speaker-microphone pairs. The transfer functions of microphone - speaker pair have been estimated utilizing sequential least square(SLS) algorithm. We identified the estimated transfer function has stable JXlles and zeros in z-plane. This paper also propose an architecture far the noise cancellation to which we applied the estimated transfer function.nction.