• Proceedings of the KIEE Conference
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• 2001.11c
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• pp.142-145
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• 2001

In the double-talk situation where both the near-end and far-end signal present, the performance of echo cancellation using the NLMS algorithm is degraded easily since it freezes the adaptation in this situation. To solve this problem, which utilize the correlation function values of input signal instead of the input signal itself, have been proposed. Because this algorithm could be used to adapt the filter's parameters continuously even in the double-talk situation, give good convergence property compared with the NLMS. In this paper, we compare and analyze its performance. The computer simulation was performed and the results showed as that ECLMS algorithms were robust and kept the desirable performance even in the double-talk situation.

• Phonetics and Speech Sciences
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• v.5 no.4
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• pp.227-232
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• 2013

Since a reverberation filter is generally much shorter than the corresponding dereverberation filter, a single-channel speech dereverberation method based on reverberation filter estimation has been developed to improve its performance. Unfortunately, a typical reverberation filter still requires too many coefficients to be accurately estimated using limited speech observations. In order to exploit sparseness of reverberation filter coefficients, in this paper, we present an algorithm to impose a sparse prior to the process of reverberation filter estimation. Simulation results demonstrate that the sparse prior imposition further improves performance of the speech dereverberation method based on reverberation filter estimation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.6B
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• pp.746-751
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• 2001

일반적으로 음향 반향 제거기에서 적응필터의 안정성을 보장하기 위해서 동시통화 검출기(DTD)를 사용하여 원단화자 신호와 근단화자 신호의 존재 여부에 따라 적응필터 계수의 적응 여부를 결정한다. 본 논문에서는 두 개의 상호상관계수를 이용하여 계산량과 메모리 소자수 면에서 효율적인 동시통화 검출 알고리즘을 제안하였다. 제안된 알고리즘은 마이크로폰의 입력신호와 추정된 반향 신호간의 상호상관계수, 그리고 마이크로폰의 입력신호와 오차신호간의 상호상관계수를 이용하여, 주행 상황과 같이 심한 잡음 환경에서도 동시통화 구간의 시작점과 끝점 검출에 강인한 특성을 갖는다. 또한, 기존의 상호상관도를 이용하는 방법에 비하여 적은 양의 계산과 메모리를 필요로 하여 저가의 고정소수점 DSP를 이용한 실시간 구현에 적합하다. 성능 평가를 위하여 차내 핸즈프리 통신 환경에서 얻은 실측 데이터를 사용하여 기존의 방법과 비교하였다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.1
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• pp.85-93
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• 2007

In case that multichannel blind deconvolution (MBD) applies to signals of which autocorrelation has a high level, separated signals are temporally whitened by diagonal elements of a separation filter matrix. In order to reduce this distortion, the algorithms, which are based on either constraining diagonal elements of a separation filter matrix or estimating a separation filter matrix by using linear prediction residual signals, are presented. Still, some problems are generated in these methods, when we separate reverberation of signals themselves or interference signals from mixed signals. To solve these problems, this paper proposes the multichannel blind deconvolution method which divides processing procedure into the stage to separate interference signals and the stage to eliminate a reverberation of signals themselves. In simulation results, we confirm that the proposed algorithm can solve the problems.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.147-150
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• 2008

The explanation of Tourism Guide of present whole country main tourist resort helps to understand the tourist resort. However, activity space of Tourism Guide is not state that can be understood all by upbringing by a natural voice. Because action of Tourism Guide is much in case of most sightseeing explanation to use microphone and speaker etc., as sticking that attach and uses to clothing and so on uses, there are much vexatious. Treatise that see hereupon makes use of establishment style fixing microphone and embody inputted obscene sounds by On-board system inflecting MCU (ATmega128), MSM7731-02 Oki-Dual Codec to minimize noise using ecad filter, and embodied a control program by serial communication method with filter codec. The resultant audible direction the maximum 59ms, the line echo maximum 27ms, the echo decrease maximum 35dB, it embodied the system which removes the adaptation elder brother noise of the back.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.7-10
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• 2000

This paper focuses on the development of speech enhancement techniques for hands-free audio terminals, including two major problems : noise cancellation and acoustic echo cancellation. The objective is to find a joint structure to get a near-end speech signal with minimum distortion and low levels of echo and noise. To solve the two problems, a new promising technique is studied and tested in computer simulation conditions.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.193-196
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• 2000

An stereophonic acoustic echo canceller cannot exactly estimate the echo path in the receiving room, because of the cross-correlation between stereo signals. In this paper, the new preprocessing filter is proposed to reduce the cross-correlation between the signals without influence on stereophonic sound. Two channel signals are linearly decorrelated by using orthogonality principles and the attenuated absolute values of the decorrelated signals are added to each channel input signals. Assuming that the power of each channel signal is larger than that of the cross-correlation, computational burden is reduced.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.261-264
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• 2004

본 논문에서 는 muti-channel lattice 예측기를 사용하여 AP(affine projection) 알고리듬을 근사적으로 구현하는 알고리즘을 제안한다. Lattice 예측기의 예측 오차를 사용하여 TDL 필터 계수를 적응적으로 조정함으로써 AP 알고리듬을 근사화한다. 또한 전처리단으로 사용된 lattice 예측기를 TDL 필터와 결합함으로써 기존의 방법보다 계산량을 더욱 줄일 수 있는 알고리듬을 제안한다. 제안된 알고리즘은 기존의 알고리즘보다 적은 계산량을 필요로 하지 만 AP 알고리즘을 보다 근사적으로 구현할 수 있다는 장점이 있다.

• Information and Communications Magazine
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• v.15 no.12
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• pp.85-96
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• 1998

본 논문에서는 현재 연구 개발되고 있는 HDSL/ADSL 시스템의 동작환경과 채널(선로감쇄(propagation loss), NEXT, 충격잡음 등)에 관하여 언급한 후, 전체 시스템의 구조 및 동작 원리를 기술하였다. 연구 개발된 HDSL/ADSL 전체시스템은 64-CAP 전송방식을 기반으로 Tomlinson precoder, Trellis코드, modulo Viterbi복호기, Convolutional-interleaved R-S 부호기/복호기, Hybrid-DFE, 반향 제거기, 타이밍 복원회로 등으로 구성되어 있다. HDSL/ADSL 전송시스템의 모든 기능들은 디지털로 구현되었으며, 칩 구현을 위하여 각 기능들의 VLSI 시스템 파라메터들을 추출하고 성능시험을 실시하였다.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.10
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• pp.639-642
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• 2005

In this paper, a robust nonlinear echo cancellation is proposed, where a third-order adaptive Volterra filtering is employed along with an expanded correlation LMS (ECLMS) algorithm to compensate for nonlinear distortion in the echo path. (e.g., DAC of the hybrid network). Finally, the robustness in the echo cancellation of the proposed approach is demonstrated using computer simulations, where high attenuation of echo signals is achieved even in the double-talk situation (e.n., BdB improvement in ERLE).