• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.11 no.5
• /
• pp.59-63
• /
• 2011

We present a lattice transversal joint (LTJ) adaptive filter with fixed reflection coefficients to achieve fast convergence with low complexity. The reflection coefficients of the filter are given by the statistics of speech signals, and the proposed order of the lattice predictor is one. Experimental results confirm that as compared to the adaptive transversal filter, the proposed adaptive filter achieves fast convergence with a negligible increase in complexity. The proposed adaptive filter converges around six times faster than the adaptive transversal filter in case of the band-limited voiced signal from the ITU-T G.168 standard.

• The Journal of the Acoustical Society of Korea
• /
• v.15 no.1
• /
• pp.46-53
• /
• 1996

Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

• Journal of the Institute of Convergence Signal Processing
• /
• v.9 no.4
• /
• pp.295-303
• /
• 2008

In general, transform domain adaptive filters show faster convergence speed than the time domain adaptive filters, but the amount of calculation increases dramatically as the filter order increases. This problem can be solved by making use of the subband structure in transform domain adaptive filters. In this paper, to increase the convergence speed on the generalized subband decomposition FIR adaptive filters, a structure of the adaptive filter with subfilter of dyadic sparsity factor in wavelet transform domain is designed. And, in this adaptive filter, the equivalent input in transform domain is derived and, by using the input, the convergence properties for the LMS algorithm is analyzed and evaluated. By using this sub band adaptive filter, the inverse system modeling and the periodic noise canceller were designed, and, by computer simulation, the convergence speeds of the systems on LMS algorithm were compared with that of the subband adaptive filter using DFT(discrete Fourier transform).

• Journal of the Korean Institute of Telematics and Electronics S
• /
• v.36S no.4
• /
• pp.83-92
• /
• 1999

본 논문에서는 Jue Chang 과 John R. Glover 가 1993년에 제안한 회귀적 적응 주기 신호 검출기[1]를 소개하고 이를 구현하기 위한 최적의 실시간 알고리즘을 제안하여 회귀적 주기 신호 검출기의 실용적인 응용 예를 제시하였다. 회귀적 적응 주기신호 검출기(FALE:Feedback Adaptive Line Enhancer)는 기존의 적응 주기 신호 검출기에 회귀 경로를 달아줌으로써, 필터 차수를 같게 했을 때 낮은 신호 대 잡음비 환경 하에서 더 높은 필터 이득과 더 낮은 추정 오차를 얻을 수 있다. 회귀 경로를 통해 들어오는 필터 출력 신호는 회귀 이득 상수 값에 따라 전체 시스템의 성능이 달라지므로 최적의 회귀 이득 상수를 찾아내는 것이 중요하며 이는 회귀 이득 상수를 변화시키며 최적의 결과값(최소 추정오차)을 유도하는 실험을 통해 얻을 수 있다. 한편, 이를 구현하는 문제에 있어서는 일잔 최적의 회귀 이득 상수 값이 정해지면 회귀 이득 상수가 초기 값으로부터 최적 값에 도달하는 변화율과 변화 유형이 시스템의 실시간 구현 및 성능에 중요한 영향을 미치게 된다. 본 논문에서는 실험을 통해 최적의 구현 알고리즘을 찾아냄으로써 Jue Chang 과 John R, Glover가 제시한 이론적인 수렴율과 수렴 성능을 유지하면서 실시간으로 동작하는 시스템을 구현하고 모의실험을 통한 성능분석 결과를 제시하였다.

• The Journal of the Acoustical Society of Korea
• /
• v.13 no.6
• /
• pp.15-23
• /
• 1994

This paper described adaptive ANC (Active Noise Control) systems for noise canacelling in multiple inputs and multiple outputs channel. Until this time, it was reported that the study of AMC for the multiple inputs and the multiple outputs had no correlation between input signals of adaptive filters. In fact, it is difficult independently to check the noise source signals and insufficient about this study. In this paper we proposed to use the adaptive filter for eliminating the correlation between input signals. The simulation results showed that the convergence rate improved about two times with this filter than that one without this filter at the rate $b_{12}=0.95$.

• The Journal of the Acoustical Society of Korea
• /
• v.19 no.8
• /
• pp.54-62
• /
• 2000

Adaptive filtering method is one of signal processing area which is frequently used in the case of statistical characteristic change in time-varing situation. The performance of adaptive filter is usually evaluated with complexity of its structure, convergence speed and misadjustment. The structure of adaptive filter must be simple and its speed of adaptation must be fast for real-time implementation. In this paper, we propose chirp discrete cosine transform (CDCT), which has the characteristics of CZT (chrip z-transform) and DCT (discrete cosine transform), and then CDCTLMS (chirp discrete cosine transform LMS) using the above mentioned algorithm for the improvement of its speed of adaptation. Using loaming curve, we prove that the proposed method is superior to the conventional US (normalized LMS) algorithm and DCTLMS (discrete cosine transform LMS) algorithm. Also, we show the real application for the ultrasonic signal processing.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.225-228
• /
• 2000

본 논문에서는 2차의 격자 IIR 노치 필터를 이용하여DS-CDMA 시스템에서 협대역 간섭신호를 제거한다. 본 노치필터는 구조가 간단하고 다양한 간섭 신호 모델에 따라 필터의 계수를 조절할 수 있어 간섭신호의 효과적인 제거가 가능하다. 노치필터의 앞단에는 주파수 추정부를 평행하게 구성하여 간섭신호 주파수의 위치와 파워를 검출한다. 제안된 적응 필터링 알고리듬은 이러한 정보를 이용하여 노치필터의 주파수를 조절하고 간섭 신호의 파워와 대역폭에 따라 노치의 넓이와 깊이를 조절한다. 즉, IIR 노치 필터의 특성을 이론적으로 분석하여 출력 신호대 잡음비를 입력과 필터의 파라미터에 관한 수식으로 유도하였고 이를 이용하여 주어진 입력에 따른 최적의 필터 파라미터를 구하여 적용시키는 것이다. 입력의 간섭 주파수와 파워를 검출하는 방법에는 여 러가지가 있지만 본 논문에서는 비교적 간단한 IIR ALE[6]를 사용하였다. 제안된 알고리듬을 사용한 결과 신호대 잡음비와 에러율에 있어서 FIR 필터를 사용한 기존의 방법[4]에 비하여 좋은 성능을 보임을 알 수 있다.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.15-18
• /
• 2000

회전기에서 발생하는 소음성 잡음의 경우 능동잡음 제어를 이용 진폭이 상대적으로 큰 주기성 저주파 잡음 제어로 상당한 감쇄효과를 얻을 수 있다. 본 논문에서는 주기적 저주파 잡음을 효과적으로 제거하는 부밴드 능동잡음제어 구조를 제안하였다 이 구조에서는 QMF를 이용, 진폭이 큰 주기적 저주파신호를 분리하여 적응측엽제거 원리를 적용하여 저주파 신호를 제거한다. QMF에서 발생하는 지연은 각 필터를 통과한 신호경로에 적응필터를 사용하여 보상하였다. 그리고 적응필터 성능향상을 위해 최적 적응이득을 해석적으로 구하여 사용하였다.

• Journal of Korea Society of Industrial Information Systems
• /
• v.12 no.1
• /
• pp.39-45
• /
• 2007

In this paper, we present a new postprocessing algorithm using interpolation and signal adaptive filter according to the each block characteristic which is acquired in block classification process. We applied blocking artifact reduction algorithm for four neighbor low frequency block and ringing artifacts is removed with preserving edges by applying a signal adaptive filter in high frequency block based on edge map. The computer simulation results confirmed a better performance by the proposed method in both the subjective and objective image qualities.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.45 no.2
• /
• pp.90-94
• /
• 2008

It is well-blown that the convergence rate gets worse when an input signal to an adaptive filter is correlated. In this paper we propose a new adaptive filtering algorithm that makes the convergence rate much improved even for highly correlated input signals. By introducing an orthogonal constraint between successive input signal vectors we overcome the slow convergence problem of the LMS algorithm with the correlated input signal. Simulation results show that the proposed algerian yields fast convergence speed and excellent tracking capability under both time-invariant and time-varying environments, while keeping both computation and implementation simple.