• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2002년도 정기총회 및 학술대회
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• pp.9-14
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• 2002

Direct-conversion receiver(DCR) architecture has superior advantages in size, cost, and power over superheterodyne receiver architectures. However, the use of direct-conversion receiver architecture has been limited due to the direct-current offset noise. The ac coupling, which is used to overcome the direct-current offset noise, causes an inter-symbol interference(ISI), whose effects can be effectively mitigated using an equalizer. In this paper, the performance of a direct-conversion receiver with ac coupling in the presence of direct-current offset is analyzed via computer simulation. The simulation result shows that by using decision feedback equalizer with LMS(Least Mean Square) algorithm, signal-to-noise ratio loss of the direct-conversion receiver compared to the idea receiver can be reduced to less than 1㏈ for corner frequencies as large as 10% of the symbol rate.

• 융합신호처리학회 학술대회논문집
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• 한국신호처리시스템학회 2006년도 하계 학술대회 논문집
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• pp.67-72
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• 2006

적응신호처리 분야에서 LMS(Least Mean Square) 알고리즘은 수식이 간단하고, 적은 계산 량으로 인해 널리 사용되고 있지만, 시간영역의 적응알고리즘은 입력신호의 고유치 분포 폭이 넓게 분포할 때는 수렴속도가 느려지는 단점이 있다. 본 논문에서는 적응 신호처리의 수렴속도를 향상 시키고 복잡한 계산 량을 줄이는 새로운 fast running FIR 필터 구조를 제안한다. 그리고 제안한 알고리즘을 가변스텝 사이즈 웨이블렛 기반 적응 알고리즘에 적용한다. 실제로 합성 음성을 사용하여 적응 잡음 제거기에 적용하여 컴퓨터 시뮬레이션을 통해 제안한 알고리즘과 기존 알고리즘과의 성능을 비교한다.

• 음성과학
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• 제8권1호
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• pp.159-165
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• 2001

In this paper, we make an experiment on speech recognition systems with barging-in and non-barging-in utterances. Barging-in capability, with which we can say voice commands while voice announcement is coming out, is one of the important elements for practical speech recognition systems. Barging-in capability can be realized by echo cancellation techniques based on the LMS (least-mean-square) algorithm. We use three kinds of telephone interface boards with barging-in capability, which are respectively made by Dialogic Company, Natural MicroSystems Company and Korea Telecom. Speech database was made using these three kinds of boards. We make a comparative recognition experiment with this speech database.

• The Journal of the Acoustical Society of Korea
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• 제24권3E호
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• pp.99-108
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• 2005

To investigate the equalizer performance in underwater acoustic communication m the presence of intersymbol interference (ISI) due to multipath, computer simulations are carried out in discrete multipath shallow water channels for three different horizontal ranges. For the purpose of computation simplicity, least mean square (LMS) algorithm is adopted both in linear equalizer and nonlinear equalizer, decision feedback equalizer (DFE) to cancel out ISI effects. Binary phase shift keying (BPSK) signals have been transmitted with high data rate of 2000bps through the use of equalization technique. The results demonstrate that equalization is an efficient way to achieve high transmission data rate in the shallow water channel.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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• pp.357-358
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• 2006

We propose the robust PBFLMS in noises, which is the enhanced acoustic echo canceller using ACPBF-LMS(Alternative Constrained Partitioned Block Frequency domain Least Mean Square) algorithm. The defect of the block structure filtering is the deterioration of convergence efficiency from noise and interference. To improve the performance of convergence efficiency, noise effect should be reduced. The new method of reducing noise effect is proposed, which apply the estimated background noise to adaptive filter step size. By experiments, the proposed acoustic echo canceller has TCL of 50dB, and always provides faster convergence speed and lower complexity than the full-tap NLMS. We also carried out an implementation of PBFLMS using ARM9EJ-S.

• 한국지능시스템학회논문지
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• 제13권5호
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• pp.606-613
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• 2003

As a technical method for controlling chaotic dynamics, this paper presents a predictive control for chaotic systems based on radial basis function networks(RBFNs). To control the chaotic systems, we employ an on-line identification unit and a nonlinear feedback controller, where the RBFN identifier is based on a suitable NARMA real-time modeling method and the controller is predictive control scheme. In our design method, the identifier and controller are most conveniently implemented using a gradient-descent procedure that represents a generalization of the least mean square(LMS) algorithm. Also, we introduce a projection matrix to determine the control input, which decreases the control performance function very rapidly. And the effectiveness and feasibility of the proposed control method is demonstrated with application to the continuous-time and discrete-time chaotic nonlinear system.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송∙미디어공학회 2016년도 추계학술대회
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• pp.163-164
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• 2016

이전까지 많은 연구자들은 적응 신호처리(Adaptive Signal Process)를 이용한 잡음 제거 방법을 연구해 왔다. 그러나, 최근 발전하고 있는 멀티콥터는 프로펠러 모터의 RPM(Revolution Per Minute)이 실시간으로 변하기 때문에 적응 신호처리를 이용하여도 깔끔한 결과를 얻어 내기가 어렵다는 한계가 존재한다. 또한, 특정 주파수를 기준으로 형성되는 고조파(Harmonics)는 적응 알고리즘인 (N)LMS 를 이용한 예측에서 오차를 발생시키는 문제를 발생시킨다. 따라서, 본 논문에서는 멀티콥터를 이용한 음향 취득에 대한 소음 저감 방법으로 회전 속도계(Tachometer), 콤 필터(Comb Filter), NLMS 알고리즘(Normalized Least Mean Square Algorithm)을 이용한 방법을 제안한다.

• 전자공학회논문지B
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• 제32B권9호
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• pp.62-70
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• 1995

In some practical applications of the LMS Algorithm the coefficient adaptation can be performed only after some fixed delay. The resulting algorithm is known as the Delayed Least Mean Square (DLMS) algorithm in the literature. There exist analyses for this algorithm, but most of them are based on the unrealistic independence assumption between successive input vectors. Inthis paper we consider the DLMS algorithm with decreasing step size .mu.(n)=n/a, a>0 and prove the almost-sure convergence ofthe weight vector W(n) to the Wiener solution W$_{opt}$ as n .rarw. .inf. under the mixing unput condition and the satisfaction of the law of large numbers. Computer simulations for decision-directed adaptive equalizer with decoding delay are performed to demonstrate the functioning of the proposed algorithm.m.

• 한국음향학회지
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• 제17권8호
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• pp.19-24
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• 1998

이 논문에서는 능동 소음 제어 시스템을 구성하는 요소들인 증폭기와 저주파 필터 와 같은 소자들의 비선형 특성과 공간에서의 주파수 대역에 따른 비선형 특성을 보상하여, 보다 효과적인 능동 소음 제어기를 설계하기 위해 시간 지연 신경 회로망을 이용하는 새로 운 방법을 제안한다. 공간을 포함한 2차 경로 함수를 모델링하여 보다 나은 성능을 갖는 능 동 소음 제어기를 구성하기 위한 기존의 최소 자승 오차 알고리듬에 기반한 filtered-x least mean square(LMS) 알고리듬과 오차 역전달 학습 알고리듬을 갖는 시간 지연 다층 구조 인 식자를 이용한 결과를 간단한 실험을 통하여 그 성능을 비교 분석한다.

• 대한전기학회논문지:시스템및제어부문D
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• 제53권6호
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• pp.446-452
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• 2004

Under the condition of a common narrowband target signal and interference signals from several directions, the linearly constrained minimum variance (LCMV) method using the generalized sidelobe canceller (GSC) for adaptive beamforming has been exploited successfully However, in the case of wideband signals, the length of the adaptive filter must be extended. As a result, the complexity of the beamformer increases, which makes real-time implementation difficult. In this paper, we improve the convergence characteristics of the adaptive filter using the transform domain normalized least mean square (NLMS) approach based on the subband GSC structure without the increase of complexity. Besides, the M-MAX algorithm, which is one of various selective coefficient updating methods, is employed in order to remarkably reduce the computational cost without decreasing the convergence quality. With the combination of these methods, we propose a computationally efficient wideband adaptive beamformer and verify its efficiency through a series of simulations.