• 디지털산업정보학회논문지
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• 제10권1호
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• pp.131-138
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• 2014

This paper is on a stereo acoustic echo canceller with the blind signal separation for post processing. The convergence speed of the stereo acoustic echo canceller is deteriorated due to mixing two residual signals which are update signals of each echo canceller. To solve this problem, we are to use the blind signal separation(BSS) method separating the mixed signals after the echo cancellers. The blind signal separation method can extracts the source signals by means of the iterative computations with two input signals. We had verified performances of the proposed acoustic echo canceller for stereo through simulations. The results of simulations show that the acoustic echo canceller for stereo using this algorithm operates stably without divergence in the normal state. And, when the speech signals were inputted, this echo canceller achieved about 2dB higher ERLE with the BSS post processing method than without this method. This stereo echo canceller showed the best performance in the case of inputting the real voice signal.

• 디지털산업정보학회논문지
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• 제8권3호
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• pp.125-131
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• 2012

This paper is on a stereo acoustic echo canceller with the blind signal separation. The convergence speed of the stereo acoustic echo canceller is deteriorated due to mixing two residual signals in the update signal of each echo canceller. To solve this problem, we are to use the blind signal separation(BSS) method separating the mixed signals. The blind signal separation method can extracts the source signals by means of the iterative computations with two input signals. We had verified performances of the proposed acoustic echo canceller for stereo through simulations. The results of simulations show that the acoustic echo canceller for stereo using this algorithm operates stably without divergence in the normal state. And, when the speech signals were inputted, this echo canceller achieved about 3dB higher ERLE in the case of using the BSS algorithm than the case of not using the BSS algorithm. But this echo canceller didn't get good performances in the case of inputting the white noises as stereo signals.

• 디지털산업정보학회논문지
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• 제12권1호
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• pp.43-50
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• 2016

This paper is on an acoustic echo canceller solving the double-talk problem by using the blind signal separation technology. The acoustic echo canceller may be deteriorated or diverged during the double-talk period. So we use the blind signal separation to detect the double talking by separating the near-end speech signal from the mixed microphone signal. The blind signal separation extracts the near-end signal from dual microphones by the iterative computations using the 2nd order statistical character in the closed reverberation environment. By this method, the acoustic echo canceller operates irrespective of the double-talking. We verified performances of the proposed acoustic echo canceller in the computer simulations. The results show that the acoustic echo canceller with this algorithm detects the double-talk periods well, and then operates stably without diverging of the coefficients after ending the double-talking. The merits are in the simplicity and stability.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2001년도 제14회 신호처리 합동 학술대회 논문집
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• pp.99-102
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• 2001

The brief proposes to reduce the misadjustment of the adaptive filter coefficients that trace the interference signal in the sidelobe. The proposed sidelobe canceller that has the form of Griffiths-Jim sidelobe canceller with an adaptive compensator that reduces the misadjustment. The proposed sidelobe canceller updates the filter coefficients by the error Signet of the adaptive compensator instead of the output signal. This brief shows the Improvement of the performance by comparing the computer simulation of the output signal of the Griffiths-Jim sidelobe canceller to the output signal of the proposed sidelobe canceller.

• 한국정보통신학회논문지
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• 제16권2호
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• pp.237-245
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• 2012

본 논문은 암묵신호분리방법을 이용하여 동시통화를 가능하게 하는 음향반향제거기에 관한 것이다. 음향반향 제거기는 동시통화 구간에서 성능이 저하되거나 발산하게 된다. 그래서 근단화자신호를 추정해서 잔차신호로부터 차감하기 위하여 암묵신호분리방법을 사용한다. 암묵신호분리방법은 이중 마이크를 가지고 2차 통계적 성질을 이용한 반복적인 계산에 의해 근단화자신호를 추정해낸다. 그런데 폐쇄된 반향환경에서 암묵신호분리의 혼합모델은 다채널이기 때문에 분리계수를 직접 계산하지 않고 반향제거기의 계수를 복사하여 그대로 사용한다. 많은 시뮬레이션을 통하여 제안한 음향반향제거기의 성능을 검증하였다. 시뮬레이션 결과, 이 방법을 사용한 음향반향제거기는 동시통화의 유무에 상관없이 안전하게 동작하고, 일반적인 LMS 알고리즘에 비해 ERLE가 평균 20dB 향상되는 것으로 나타났다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1992년도 학술논문발표회 논문집 제11권 1호
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• pp.23-28
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• 1992

The purpose of this paper is to develop a new adaptive echo canceller improving convergence speed and near-end-talker detection performance of the conventional echo canceller. In a conventional adaptive echo canceller, an adaptive digital filter with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to cote the coefficients, and NET detector using energy comparison method prevents the adaptive digital filter to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF(Adaptive Digital Filter) output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yield more accurate detection of the start point of the NET signal.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1998년도 추계종합학술대회 논문집
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• pp.1183-1186
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• 1998

Adaptive nois canceller can extract the noiseremoved spech in noisy speech signal by adapting the filter-coefficients to the background noise environment. A kind of LMS algorithm is one of the most popular adaptive algorithm for noise cancellation due to low complexity, good numerical property and the merit of easy implementation. However there is the matter of increasing misadjustment at voiced speech signal. Therefore the demanded speech signal may be extracted. In this paper, we propose a fast and noise robust wavelet packet adaptive noise canceller with NLMS-SUM method LMS combined algorithm. That is, we decompose the frequency of noisy speech signal at the base of the proposed analysis tree structure. NLMS algorithm in low frequency band can efficiently dliminate the effect of the low frequency noise and SUM method LMS algorithm at each high frequency band can remove the high frequency nosie. The proposed wavelet packet adaptive noise canceller is enhanced the more in SNR and according to Itakura-Satio(IS) distance, it is closer to the clean speech signal than any other previous adaptive noise canceller.

• The Journal of the Acoustical Society of Korea
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• 제21권3E호
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• pp.119-125
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• 2002

In the conventional generalized side-lobe canceller (GSC), the output is an estimated error signal that causes the adaptive filter weights do not converge to the optimal value. This paper presents a new side-lobe canceller with adaptive compensator th it reduces the misadjustment of the adaptive filter coefficients for the structural problem in the GSC. The adaptive compensator separates the output signal from the estimated error. The newly estimated error signal converges to the zero while the output signal tracks the target signal. This paper shows improvement of the performance by comparing the computer simulation of the output signal of GSC with the output signal of the proposed algorithm.

• 한국전자통신학회논문지
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• 제12권3호
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• pp.419-426
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• 2017

본 논문은 기하학적 개념에 기반한 암묵신호분리를 이용하여 동시통화문제를 제어하는 음향반향제거기에 관한 것이다. 음향반향제거기는 동시통화 구간에서 성능이 저하되거나 발산하게 된다. 따라서 혼합된 마이크 입력신호로부터 근단화자신호를 분리해서 동시통화상태를 검출하기 위하여 암묵신호분리기술을 이용한다. 암묵신호분리는 미지의 입력신호들로부터 기하학적 개념에 기반하여 변형과 회전의 두 단계를 거쳐 근단화자신호를 추정해낸다. 컴퓨터 시뮬레이션을 통하여 이 음향반향제거기의 성능을 검증하였다. 동시통화 구간에서는 반향제거필터의 계수가 발산하는 것을 방지하기 위하여 계수 갱신작업을 중지하도록 하였다. 시뮬레이션 결과, 이 방법을 사용한 음향반향제거기는 암묵신호분리의 빠른 수렴속도로 인해 동시통화의 유무에 상관없이 안전하게 동작함을 확인하였다.

• 전자공학회논문지B
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• 제30B권12호
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• pp.12-20
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• 1993

In a conventional adaptive echo canceller, an ADF(Adaptive Digital Filter) with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to compute the coefficients, and NET detector using energy comparison method prevents the ADF to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yields more accurate detection of the start point of the NET signal.