• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2005년도 ICCAS
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• 전기학회논문지
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• 제64권1호
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• pp.149-158
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• 2015

This paper proposes a subband filtering technique to the MUSIC and the ESPRIT algorithm for estimating the magnitude and frequency of the harmonics of power signal. In proposed method, the input power signal is decomposed to the odd harmonics and the even harmonics respectively by the filter bank system. The amplitude and the frequency estimation of the decomposed harmonics are carried out using the MUSIC and the ESPRIT method. Subband filtering can reduce the autocorrelation matrix size of input data, and spectrum leakage between adjacent harmonics. Therefore, this subband technique has advantage in computational cost and estimation accuracy compared to fullband MUSIC and ESPRIT. To demonstrate the performance of the method, computer simulations are performed to the synthesized input signal, and experiment results are compared in subband and fullband cases.

• 전기학회논문지
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• 제58권4호
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• pp.679-687
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• 2009

In this paper, a subband filtering and adaptive prediction technique for analyzing harmonics in power systems is presented. In this method, the filter banks are designed to decompose odd and even order harmonics separately. The adaptive prediction has been employed reduce the transient and white noise effect in time varying harmonics detecting and tracking. The frequencies and amplitudes of the decomposed harmonics are estimated by recursive algorithm. To demonstrate the performances of the developed technique, computer simulations to the signal with the time-varying frequency and THD are carried out.

• 융합신호처리학회논문지
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• 제11권3호
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• pp.238-243
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• 2010

A new implementation method for MPDSAP(Maximally Polyphase Decomposed Subband Affine Projection) adaptive filter is proposed. The affine projection(AP) adaptive filter achieves fast convergence speed, however, its implementation is so expensive because of the matrix inversion for a weight-updating of adaptive filter. The maximally polyphase decomposed subband filtering allows the AP adaptive filter to avoid the matrix inversion, moreover, by using a pipelining technique, the simple subband structured AP is suitable for VLSI implementations concerning throughput, power dissipation and area. Computer simulations are presented to verify the performance of the proposed algorithm.

• 전기학회논문지
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• 제60권2호
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• pp.423-430
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• 2011

In adaptive filtering, the sparseness of impulse response and input signal characteristics are very important factors of it's performance. This paper presents a subband improved proportionate normalized least square (SIPNLMS) algorithm which combines IPNLMS for impulse response sparseness and subband filtering for prewhitening the input signal. As drawing and combining the advantage of conventional approaches, the proposed algorithm, for impulse responses exhibiting high sparseness, achieve improved convergence speed and tracking ability. Simulation results, using colored signal(AR(4)) and speech input signals, show improved performance compared to fullband structure of existing methods.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2001년도 제14회 신호처리 합동 학술대회 논문집
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• pp.907-910
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• 2001

In this paper, we address a new time delay estimation method using third-order statistics and subband adaptive filtering to improve the accuracy of target detection for acoustic backscattered signals in a noise interference environment. Each reference and primary signals are decorrelated using the multiresolution analysis framework through a M-band discrete wavelet transform(M-DWT). Then noise effect can be reduced. Here, time delays are estimated iteratively in each subband using two different adaptation mechanisms that minimize the mean squared error (MSE) between the references and primary signal. More specifically, third-order cumulants and projection cross-correlation(PCC) criterion are utilized to achieve an effective SNR improvement for the time delay estimation.

• 한국정보통신학회논문지
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• 제4권2호
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• pp.407-416
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• 2000

본 논문은 트리구조의 필터뱅크를 이용하여 필터링을 할 때 데시메이션으로 인해서 생기는 왜곡을 보상하기 위해서 보조 필터(엘리어싱 제거기)를 사용하는 방법을 제안하고 이를 이용하여 잡음을 제거하여 음질을 향상시키는 실험을 행했다. 입력신호는 두 채널의 분석필터에 의해 서브밴드로 분해된 후 데시메이션되고 다시 각 서브밴드로 분해된 신호를 적응필터를 이용하여 필터링을 한다. 이러한 기법으로 각 주파수 대역을 균일한 대역으로 분해할 수 있으며 각 서브밴드별로 각기 다른 수렴상수를 사용할 수 있어서 수렴속도를 향상시킬 수 있는 장점이 있다. 특히 잡음 제거나 음질 향상에 서브밴드 적응 필터뱅크를 응용할 경우 수렴속도가 빨라야 하며, 임펄스 응답의 길이가 긴 시스템을 모델링하는 데 응용할 수 있는 장점이 있다. 적응필터 계수의 갱신 방법으로는 VSS(Variable Step Size) LMS 알고리듬을 사용하였으며, 실험 결과, 기존의 서브밴드 구조보다 제안한 구조가 최소 평균 오차나 수렴성능 면에서 우수한 성능을 보였다.

• 한국음향학회지
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• 제23권3호
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• pp.221-227
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• 2004

본 논문은 부밴드 인접투사 알고리즘 (SAPA)을 소개한다. SAPA의 개선된 성능은 인접투사 알고리즘에 부밴드 구조를 적용함으로써 가능하다. 본 알고리즘에서 적응 필터의 계수 갱신식은 부밴드 필터링을 위한 분해 필터뱅크로서 OQF (Orthogonal Quadrature Filter)를 사용함으로써 간단하게 유도된다. 유도된 SAPA는 빠른 수렴속도와 적은 계산량을 갖는다. 유색 입력신호에 대한 제안한 알고리즘의 효과를 실험을 통해 평가하였다.

• 전기학회논문지
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• 제59권2호
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• pp.476-483
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• 2010

If the data are pre-whitened, then gradient adaptive algorithms which are simpler than higher order statistics algorithms can be used in adaptive blind signal estimation. In this paper, we propose a blind subband affine projection algorithm for multiple-input multiple-output adaptive equalization in the blind environments. All of the adaptive filters in subband affine projection equalization are decomposed to polyphase components, and the coefficients of the decomposed adaptive sub-filters are updated by defining the multiple cost functions. An infinite impulse response filter bank is designed for the data pre-whitening. Pre-whitening procedure through subband filtering can speed up the convergence rate of the algorithm without additional computation. Simulation results are presented showing the proposed algorithm's convergence rate, blind equalization and blind signal separation performances.

• 한국정보통신학회논문지
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• 제26권3호
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• pp.443-448
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• 2022

긴 길이의 적응 필터와 높은 상관도의 입력신호를 사용하는 적응 필터링 응용에서 적응 필터의 수렴성능을 향상 시키기 위해 가변 스텝사이즈를 이용하는 부밴드 인접투사 알고리즘을 제안한다. 제안한 알고리즘은 다위상 분해와 노블아이덴티티를 적용한 M-부밴드 구조에서 각 적응 부필터별 서로 다른 스텝사이즈를 사용함으로써 빠른 수렴속도와 작은 정상상태오차를 얻을 수 있다. 각 갱신시점에서 적응 필터의 평균자승오차를 최소화하도록 유도된 스텝사이즈는 가변 스텝사이즈를 사용하는 기존 알고리즘에 비해 좋은 수렴성능을 보인다. 기존 알고리즘에 비해 우수한 제안한 알고리즘의 수렴성능을 확인하기 위해 시스템 식별 모델을 고려하여 AR(1)과 AR(2) 유색 입력 신호에 대한 최소자승편차에 대한 컴퓨터 시뮬레이션을 수행한다.