• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.8-14
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• 2001

In this paper, we present a new cost function for subband block adaptive algorithm and block conjugate gradient algorithm for noise cancellation of acoustic signal. For the cost function, we process the subband signals with data blocks for each subbands and recombine it a whole data block. After these process, the cost function has a quadratic form in adaptive filter coefficients, it guarantees the convergence of the suggested block conjugate gradient algorithm. And the block conjugate gradient algorithm which minimizes the suggested cost function has better performance than the case of full-band block conjugate gradient algorithm, the computer simulation results of noise cancellation show the efficiency of the suggested algorithm.

• The Journal of the Acoustical Society of Korea
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• v.17 no.1E
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• pp.20-29
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• 1998

This paper proposes a new stereophonic acoustic echo canceler with deploying delayless subband adpative filters. Due to the storong correlation between stereo signals, a stereophonic acoustic echo canceler is suffering from the slow convergence and the misalignment for estimating impulse responses corresponding to true echo paths at receiving room. Specially, dual adaptive filters for echo cancellation are significantly affected by the abrupt change of the transmission room environment so that the impariments on voice communication could be experienced. To prevent these performance degradations, this paper proposes a robust subband echo canceler employing pre-processing block so as to enhance the convergence speed and provide the insusceptibility to the environment change at transmission room.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.43-49
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• 1997

Generally, echo cancelers of a TV conference system or a audio conference system are to carry out a real time processing in the case of the closed room having long reverberation time because the system requires much time to modify filter coefficients to environmental changes. Therefore this paper proposes a new subband adaptive filtering method using polyphase filter banks of MPEG(Moving Picture Experts Group) audio system to solve the problems. This method divides signal spectra of input and output into several frequency bands, and each band is adaptively filtered by using ES-NLMS (Exponential Step-Normalized Least Mean Square) algorithm. The optimal number of subband is determined by computational simulations. According to the results of simulation, ERLE of the subband model is 2dB smaller than general full band, calculation rate's of the subband model is decreased about 88%.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.5 s.305
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• pp.145-156
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• 2005

In application for acoustic echo cancellation and adaptive equalization, input signal is highly correlated and the long length of adaptive filter is needed. Affine projection algorithms, in these applications, can produce a good convergence performance. However, they have a drawback that is a complex hardware implementation. In this paper, we propose a new subband affine projection algorithm with improved convergence and reduced computational complexity. In addition, we suggest a good approach to implement the proposed method. In this method by applying polyphase decomposition, noble identity and critical decimation to the anne projection algorithm the number of input vectors for decorrelation can be reduced. The weight-updating formula of the proposed method is derived as a simple form that compared with the NLMS(normalized least mean square) algorithm by the reduced projection order The efficiency of the proposed algorithm for a colored input signal was evaluated by using computer simulations.

• Proceedings of the Acoustical Society of Korea Conference
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• 1996.10a
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• pp.42-45
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• 1996

잔향시간이 긴 실내의 전달함수의 모델링에 있어서 , 일반적인 플 밴드 MA 모델에 기초한 적응 필터를 이용한 경우에는, 많은 필터 차수를 필요로 하고 적응 시간이 길어지는 문제점이 있다. 본 논문에서는 필터 차수를 감소시키고 수렴 특성을 향사시키기 위해서, 각 입출력 신호를 몇 개의 주파수 대역으로 나우어서 각각의 주파수 대역에 대새서 적응 처리 과정을 행하는 서브밴드 MA 모델을 이용한 적응디지털 필터 처리 방법을 제안한다. 컴퓨터 시뮬레이션 서브밴드MA 모델을 이용한 디지털 적응 필터 처리과정의 유효성을 나타냈었다.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.40 no.4
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• pp.342-350
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• 1991

This paper describes an adaptive beamformer which improves the frequency characteristics of the AMNOR system. The proposed beamformer uses a bank of filters to calculate the frequency degradation in each subband. In the process ofdesigning the beamformer, the filter coefficients are updated to minimize the power of the difference between a fictitious desired signal and the output under the permissible degradation value in each subband. Simulation results demonstrating the improved performances of the proposed method are presented.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.56 no.4
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• pp.795-798
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• 2007

In this paper, we propose an image reconstruction method utilizing an optimized adaptive interpolation kernel along with a 2D M-channel perfect reconstruction filter bank (M-ch PR-FB) structure. In particular, the proposed approach leads to construction of a sharper image than a direct conversion, still preserving high frequency components of the original image through the subband processing of the 2D M-ch PR-FB. Finally, the image quality of the proposed approach is demonstrated by comparing with those of the direct methods using conventional interpolation kernels.

• The Journal of the Acoustical Society of Korea
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• v.26 no.2
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• pp.69-74
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• 2007

In signal processing applications with highly correlated input signals, subband affine projection algorithm and step size controlling is a good solution for improving the slow convergence rate and large computational complexity of LMS-type algorithms. This paper proposes a subband affine projection algorithm using a variable step size. The proposed method achieves fast convergence rate and small steady-state error with a small computational complexity by combining the SAP and step size controlling in a subband structure. Experimental results on highly correlated input signal show that the proposed method is superior to the conventional methods.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.11
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• pp.1545-1550
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• 2021

We propose an optimal variable step size adjustment method for the simplified subband affine projection algorithm (Simplified SAP; SSAP) in a subband structure based on a polyphase decomposition technique. The proposed method provides an optimal step size derived to minimize the mean square deviation(MSD) at the time of updating the coefficients of the subband adaptive filter. Application of the proposed optimal step size in the SSAP algorithm using colored input signals ensures fast convergence speed and small steady-state error. The results of computer simulations performed using AR(2) signals and real voices as input signals prove the validity of the proposed optimal step size for the SSAP algorithm. Also, the simulation results show that the proposed algorithm has a faster convergence rate and good steady-state error compared to the existing other adaptive algorithms.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.645-651
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• 2003

The system in which a microphone and a loudspeaker are simultaneously used can cause an echo. The echo is caused by feedback between the output of the loudspeaker and the input of the microphone. The acoustic echo canceller is a device to cancel the echo in a communication system. Its general procedure for cancellation is first estimating the plant response of the feedback path and then eliminating the feedback signal from the input signal. In this paper, full-band and sub-band approaches are compared by using some simulation examples.