• 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.

• 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.

• 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.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2006년 학술대회 논문집 정보 및 제어부문
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• pp.416-418
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• 2006

In this paper, a new acoustic echo cancellation approach based on the DUET algorithm and scaling factor estimation is proposed to solve the scaling ambiguity in case of blind separation based acoustic echo cancellation in a noisy environment. In hands-free full-duplex communication system. acoustic noises picked up by the microphone are mixed with echo signal. For this reason, the echo cancellation system may provide poor performance. For that purpose, a degenerate unmixing estimation technique, adjusted in the time-frequency domain, is employed to separate undesired echo signals and noises. Also, since scaling and permutation ambiguities have not been solved in the blind source separation algorithm, kurtosis for the desired signal selection and a scaling factor estimation algorithm are utilized in this rarer for the separation of an echo signal. Simulation results demonstrate that the proposed approach yields better echo cancellation and noise reduction performances, compared with conventional methods.

• 한국음향학회지
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• 제22권5호
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• pp.351-359
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• 2003

본 논문에서는 독립성분분석을 이용한 음향 반향제거 방법을 제안하였다. 음향반향제거기의 마이크로폰에 반향 이외의 잡음이 부가될 경우 반향제거기의 성능은 저하된다. 이러한 문제를 해결하기 위해 본 연구에서는 두 개의 마이크로폰을 이용하여 반향과 선형으로 섞인 잡음을 받은 후 독립성분 분석 기법을 통해 반향과 잡음을 분리하였다. 그리고 분리된 반향 신호를 반향제거기에 사용되는 적응 알고리듬의 기준 신호로 이용함으로서 반향제거기의 성능을 향상시켰다. 컴퓨터 모의실험을 통해 제안한 방법의 타당성을 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.

• 한국통신학회논문지
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• 제23권7호
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• pp.1650-1658
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• 1998

본 논문에서는 이동통신 시스템의 반향을 제거하기 위해 음성부호화기에서 얻은 음성 분석 정보를 이용하여 반향을 제거하는 방법을 제시하였다. 반향 제거기 적응 알고리즘의 입력 신호로서 기존의 방법인 음성부호화기의 출력 음성신호를 사용하지 않고 음성 부호화기 디코더 과정에서 제공되어지는 여기 신호, 선형 예측 오차 신호를 사용하였다. 모의 실험을 위해 Normalized Least Mean Square(NLMS) 알고리즘을 이용한 적응 반향 제거기를 구성하였고, 기존의 음성신호를 사용하는 반향제거기에 비해 음성 부호화기에서 제공되어지는 음성의 여기 신호 성분을 적응 알고리즘 입력신호로 사용함으로써 40 dB Echo Return Loss Enhancement(ERLE)를 얻는데 걸리는 시간에 있어서 약 4배 정도의 빠른 속도를 얻을 수 있다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
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• pp.882-885
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• 2003

In this paper, nonlinear echo cancellation using a correlation LMS (CLMS) algorithm is proposed to cancel the undesired nonlinear echo signals generated in the hybrid system of the telephone network. In the telephone network, the echo signals may result the degradation of the network performance. Furthermore, digital to analog converter (DAC) and analog to digital converter (ADC) may be the source of the nonlinear distortion in the hybrid system. The adaptive filtering technique based on the nonlinear Volterra filter has been the general technique to cancel such a nonlinear echo signals in the telephone network. But in the presence of the double-talk situation, the error signal for tap adaptations will be greatly larger, and the near-end signal can cause any fluctuation of tap coefficients, and they may diverge greatly. To solve a such problem, the correlation LMS (CLMS) algorithm can be applied as the nonlinear adaptive echo cancellation algorithm. The CLMS algorithm utilizes the fact that the far-end signal is not correlated with a near-end signal. Accordingly, the residual error for the tap adaptation is relatively small, when compared to that of the conventional normalized LMS algorithm. To demonstrate the performance of the proposed algorithm, the DAC of hybrid system of the telephone network is considered. The simulation results show that the proposed algorithm can cancel the nonlinear echo signals effectively and show robustness under the double-talk situations.