• The Journal of the Acoustical Society of Korea
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• v.17 no.1
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• pp.86-97
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• 1998

본 논문은 스테레오 음향 반향 제거기에 적용되는 고속 Affine Projection 알고리듬 을 제안한다. 최근 스테레오 원격 회의 시스템은 보다 현실감 있는 원격 회의를 가능케 하 는 장점으로 인해 많은 관심을 끌고 있다. 그러나, 회의실의 원단화자와 마이크로폰사이의 상호교차(cross-coupling)로 인해 음향 반향이 발생하게 된다. 만약 이 반향 신호가 제거되 지 않은채 수신 룸으로 전달되면 결국 음성 통화 품질이 저하된다. 이를 방지하기 위하여 추정 반향 신호를 만들어 내고 통신 품질의 손실 없이 이 반향을 제거하는 음향 반향 제거 기가 필수적이다. 단 채널 음향 반향 제거기와 다르게 스테레오 환경하에서의 음향 반향 제 거기는 전송실의 환경변화로 인한 성능 저하와 각 반향 경로를 추정하기 위해 사용하는 각 적응 필터의 임펄스응답이 반향 경로와 일치하지 않는 등의 각종 문제점들이 발생하게 된 다. 본 논문에서는 서로 상관관계 없는 입력신호를 만들어내고 전송실의 환경변화로 인한 성능저하를 보완하기 위해 전처리단(pre-processing block)을 제안하여 일반적인 방법에 대 해 3-10dB정도의 향상된 성능을 보이며 적은 계산량으로 빠른 수렴성능을 갖는 새로운 형 태의 스테레오 음향 반향 제거기를 제안한다.

• Journal of the Institute of Convergence Signal Processing
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• v.11 no.3
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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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.3
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• pp.443-448
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• 2022

This paper presents a subband affine projection algorithm with variable step size to improve convergence performance in adaptive filtering applications with long adaptive filters and highly correlated input signals. The proposed algorithm can obtain fast convergence speed and small steady-state error by using different step sizes for each adaptive sub-filter in the subband structure to which polyphase decomposition and noble identity are applied. The step size derived to minimize the mean square error of the adaptive filter at each update time shows better convergence performance than the existing algorithm using a variable step size. In order to confirm the convergence performance of the proposed algorithm, which is superior to the existing algorithm, computer simulations are performed for mean square deviation(MSD) for AR(1) and AR(2) colored input signals considering the system identification model.

• Journal of information and communication convergence engineering
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• v.8 no.1
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• pp.23-28
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• 2010

According to increasing projection order, the AP algorithm bas noise amplification problem in large background noise. This phenomenon degrades the performances of the AP algorithm. In this paper, we analyze convergence characteristic of the AP algorithm and then suggest a noise robust modified AP algorithm for reducing this problem. The proposed algorithm normalizes the update equation to reduce noise amplification of AP algorithm, by adding the multiplication of error power and projection order to auto-covariance matrix of input signal. By computer simulation, we show the improved performance than conventional AP algorithm.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.8 no.1
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• pp.45-50
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• 2015

In this paper, we confirm the effective adaptive algorithm for tha active noise contorl (ANC) though the performance comparison between adaptive algorithms. Generally, the normalized least mean square (NLMS) algorithm has been widely used for an adaptive algorithm thanks to its simplicity and having a fast convergence speed. However, the convergence performance of the NLMS algorithms is often deteriorated by colored input signals. To overcome this problem, the affine pojection (AP) algorithm that updates the weight vector based on a number of recent input vectors can be used for allowing a higher convergence speed than the NLMS algorithm, but it is computationally complex. Thus, the proper algorithm were determined by the comparison between NLMS and AP algorithms regarding as the convergence performance and complexity. Simulation results confirmed that the noise reduction performance of NLMS algorithm was comparable to AP algorithm with low complexity. Therefore the NLMS algorithm is more effective for ANC system.

• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.24-26
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• 2006

In this paper, a nonlinear echo cancellation approach, based on a Gauss-Seidel pseudo affine projection algorithm and Volterra filtering, is proposed to compensate for echo path nonlinearity in the telephone network. Simulation results demonstrate that the proposed approach yields reduction of computational complexity and improved convergence speed than the conventional nonlinear echo cancellation methods (NLMS, ECLMS, FAP, RLS).

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.129-132
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• 1999

스테레오 음향 반향 제거기는 입력 신호로 사용되는 두 채널간의 강한 상관관계로 인하여 수렴특성이 악화되는 문제점을 갖는다. 따라서 수렴속도를 향상시키기 위해 RLS또는 Affine Projection(AP) 알고리즘 같은 Least Square (LS)계 열의적응 알고리즘을 사용하는 것이 필요하다. 그러나, 이러한 알고리즘은 LMS 알고리즘과 같은 통계적 미분계열 알고리 즘에 비하여 과도한 계산량을 요구하므로 고속 알고리즘에 대한 연구가 진행되어왔다. 본 논문에서는 스테레오 환경에서 Gram-Schmidt(GS) 직교화를 이용하여 LMS 알고리즘 수준의 계산량을 갖는 근사화된 AP 알고리즘을 제안하였다. 제안한 알고리즘은 AP 알고리즘의 후행오차 성질을 이용하여 GS 직교화 구조로 구성되며, 계산량 감소를 위해 샘플단위의 재귀적 GS 직교화를 사용하였다. 또한 GS직교화를 스테레오 채널에 적용함으로써 적은 계산량으로 AS 알고리즘과 대등한 수렴 성능을 갖는다.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.423-426
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• 2003

Adaptive algorithm used in Acoustic Echo Canceller (AEC) needs fast convergence algorithm when reference signal is colored speech signal. Set-Membership Affine Projection (SMAP) algorithm is derived from the constraint, which is the minimum value adaptive filter coefficient error. In this paper, we test the characteristic about noise of the SMAP algorithm and proposed modified version of SMAP algorithm fur using at AEC. As the projection order increase, the convergence characteristic of the SMAP algorithm is improved where no noise space. But if the noise uncorrelated with input signal exists, the AEC shows bad performance. In this paper, we propose normalized version of adaptive constants using estimated error signal for robust to noise and show the good performance through AEC simulation.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.217-217
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• 2004

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.216-216
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• 2004