• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06d
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• pp.80-83
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• 1998

In this paper, we propose a new filtered-x least mean fouth (LMF) algorithm where the error raised to the power of four is minimized and analyze its convergence behavior or a multiple sinusoidal acoustic noise and Gaussian measurement noise. Application of the filtered-x LMF adaptive filter to active noise cancellation (ANC) requires estimating of the transfer characteristic of the acoustic path between the ouput and error signal of the adaptive canceller. The results of the convergence analysis of the filtered-x LMF algorithm indicates that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct component . Phase estimation error and estimated again. In particular , the convergence is shown to be strongly affected by the accuracy of the phase response estimate. Also, we newly show that convergence behavior can differ depending on the relative sizes of the Gaussian measurement noise and convergence constant.

• Proceedings of the KSPS conference
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• 2005.11a
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• pp.147-150
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• 2005

The performance of speech recognition in car environment is severely degraded when there is music or news coming from a radio or a CD player. Since reference signals are available from the audio unit in the car, it is possible to remove them with an adaptive filter. In this paper, we present experimental results of speech recognition in car environment using the echo canceller. For this, we generate test speech signals by adding music or news to the car noisy speech from Aurora2 DB. The HTK-based continuous HMT system is constructed for a recognition system. In addition, the MMSE-STSA method is used to the output of the echo canceller to remove the residual noise more.

• The Journal of the Acoustical Society of Korea
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• v.18 no.2E
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• pp.45-52
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• 1999

This paper presents the integrated speech enhancement system for hands-free mobile communication. The proposed integrated system incorporates both acoustic echo cancellation and engine noise reduction device to provide signal enhancement of desired speech signal from the echoed plus noisy environments. To implement the system, a delayless subband adaptive structure is used for acoustic echo cancellation operation. The NLMS based adaptive noise canceller then applied to the residual echo removed noisy signal to achieve the selective engine noise attenuation in dominant frequency component. Two sets of computer simulations are conducted to demonstrate the effectiveness of the system; one for the fixed acoustical environment condition, the other for the robustness of the system in which, more realistic situation, the acoustic transmission environment change. Simulation results confirm the system performance of 20-25dB ERLE in acoustic echo cancellation and 9-19 dB engine noise attenuation in dominant frequency component for both cases.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1E
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• pp.84-88
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• 1996

In this paper, we propose a novel adaptive digital filter for tonal noise cancellation, with a frequency tracking capability. The proposed algorithm not only estimates the magnitude and phase of the tonal disturbance but also tracks its frequency, which changes in quasi-static manner. The algorithm uses the steepest descent method and the instantaneous frequency approach for the phase/magnitude estimation and frequency tracking, respectively. A number of computer simulations have been carried out in order to demonstrate the feasibility of the proposed ANC algorithm under various conditions.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.04a
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• pp.939-946
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• 2008

본 논문은 시청자의 위치에 최적화된 입체음향을 제공하는 TV 개발에 관한 것으로 2 개의 TV 스피커만으로 5.1 채널 스피커가 주는 입체음향 효과를 제공해준다. 기존의 Speaker Virtualizer 기술은 시청자가 특정 위치(Sweet Spot)를 벗어나면 입체음향 성능이 현저히 저하된다. 반면, 본 논문에서 제안하는 Adaptive Virtualizer 기술은 초음파가 장착된 리모콘을 사용하여 시청자의 위치를 인식하고 인식된 시청자의 위치 정보를 활용하여 청취위치에 해당하는 HRTF로부터 설계된 Filter를 Update 하고 두 스피커의 출력레벨 및 시간지연 값을 보정함으로써 최적의 입체음향을 재현한다. 본 논문에서는 실시간 구현을 위해 Speaker Virtualizer의 계산량을 최소화하는 기술을 제안하고 다양한 청취 위치에 해당하는 Filter를 설계하고 설계된 Filter를 효율적으로 Update 하는 Adaptive Virtualizer 기술을 제안한다. 또한, 초음파를 이용한 시청자 위치 인식 기술 및 전체 시스템 통합 기술을 제시한다.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.2
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• pp.61-70
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• 1984

This paper examines the array antenna theory, basic relations between the array size (aperture) and its beamwidth and resultant patterns. This paper also provides array antenna system design criteria, mainly maximizing the signal-to-noise ratio (SNR) and its corresponding optimum array structure and weight functions. Explicit new expressions for array performance are also illustrated in terms of the array output SNR. An example is provided for a 37-element planar array to explicitly illustrate the beam-forming and nulling operations of the array. Fundamentals of sidelobe canceller (SLC) systems have been discussed along with a derivation of new SLC equations for optimum weights.

• 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.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.3
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• pp.145-155
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• 1990

In this paper, an adaptive Infinite Impulse Response (IIR) filter algorithm using output error method, which prevents poles of a system transfer function from being out of unit circle, is proposed, and it is proved that the proposed algorithm always satisfies hyperstability. The proposed algorithm is applied to an Adaptive Noise Canceller (ANC), and compared with a Least Square (LS) method adaptive IIR filter algorithm and an adaptive Finite Inpulse Response (FIR) filter algorithm. As a result, the validity of the proposed algorithm is proved.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.4
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• pp.639-644
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• 2001

Sequence Code Division Multiple Access) and TDMA(Time Division Multiple Access) system. In this system, the narrowband TDMA signals can cause intolerant interferences to CDMA signals. In this paper, DS-CDMA/TDMA frequency sharing system have been analyzed in AWGN(Additive white Gaussian Noise), MAI (Multi Access Interference) and NI(Narrowband Interference) environment. Also, performance improvement has been obtained by adopting an adaptive notch filtering scheme using complex filter bank and CCI canceller.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.10
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• pp.144-155
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• 1998

The array weights that will maximize the SNR for any type of noise environment are determined by the function of the antenna design configuration and the directions of receiving target and interference signals. The conventional SLCs(sidelobe cancellers) using the SNR maximization perform worst from the saturation of the receiving system of main channel when the main antenna has pattern with high gain at the arrival angle of strong interference. In this paper, the new SLC is accomplished by using two independent antenna architecture. Main antenna is implemented with adaptive nulling, which is used for rejecting high-power interference primarily. Auxiliary antenna is realized with adaptive array for receiving interference signal to be suppressed completely, which has a characteristics of sufficient gain for every direction. The new SLC is implemented with above both antennas. We show that the new SLC, which consists of the adaptive nulling main antenna and the adaptive array auxiliary antenna, is useful in reducing the effect of strong interference like jammer, because the adaptive nulling at main antenna prevents its receiver and signal processor for saturation by strong interference. The proposed SLC has improved SNR over the conventional SLCs. The improved SNR at sidelobe region is typically more than 7 dB for a given test signal. Moreover, it improves the SNR of about 20 dB under strong interference at mainlobe.