• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.4
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• pp.703-711
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• 2005

This paper proposes the improved orthogonal projection method as a new technique advancing the performance of the echo cancellation for speeches in the acoustic echo canceller. Comparing with the used NLMS adaptive algorithm, it shows that this method improves the performance of the echo cancellation for signals with the large auto-correlation. In order to testify performances of the orthogonal projection method whom this paper proposes, we have coded a simulation program and executed computer simulations. We observed convergence curves by using two adaptive algorithm for noises and speeches. From simulation results for two input signals, the proposed method shows the high ERLE and the fast convergence and the stable operation in case of using speeches as well as noises.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2388-2396
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• 2012

This paper is on an simplified orthogonal projection method which cancel the acoustic echo signals in the stereo acoustic echo canceller. Comparing with the NLMS algorithm which is widely used for simplicity and stability, it shows that this method has the improvement of the convergence performances for signals with the high auto-correlation, and has small computational quantities. To verify the convergence characteristics of the proposed algorithm, we simulated about various input signals. And we compared the results of simulation for this algorithm with the ones for the NLMS algorithm. By these works, it was proved that the stereo acoustic echo canceller adopting the proposed algorithm shows about 3dB more high ERLE than the NLMS algorithm for the white noise signals, and 5dB for the colored voice signals.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.979-986
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• 2018

This paper deals with acoustic echo cancellation using blind signal separation method. This method does not degrade the echo cancellation performance even during double-talk. In the closed echo environment, the mixing model of acoustic signals is multi-channel, so the convolutive blind signal separation method is applied and the mixing coefficients are calculated by using the feedback model without directly calculating the separation coefficients for signal separation. The coefficient update is performed by iterative calculations based on the second-order statistical properties, thus estimates the near-end speech. A number of simulations have been performed to verify the performance of the proposed blind signal separation method. The simulation results show that the acoustic echo canceller using this method operates safely regardless of the presence of double-talk, and the PESQ is improved by 0.6 point compared with the general adaptive FIR filter structure.

• 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.73-76
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• 2000

본 논문에서는 음향반향제거기를 구현하기 위한 적응알고리즘을 제안한다 특히 긴 임펄스 응답을 가지는 시스템의 식별을 위해 웨이블릿 필터를 사용하여 입력신호를 서브밴드로 분해함으로써 기존의 RLS알고리즘의 계산량을 줄여 수렴속도를 향상시켰다. 이 과정에서 적응필터를 다위상 구조로 구성하여 컨벌루션 과정을 병렬처리가 가능하도록 하였다. 제안된 알고리즘의 성능분석을 위하여 실제 음성신호를 입력신호로 하여 컴퓨터 모의실험을 수행하였으며 전대역 RLS알고리즘과 비교하였다.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.623-626
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• 1998

NLMS algorithm is degraded by the ambient noises and the near-end speech signals. In this paper, a robust acoustic echo cancellation algorithm is proposed. To enhance the echo cancellation performance, the step size of the proposed algorithm is normalized by the sum o fthe power of the reference signals and the primary signals. as results of comparing the excess mean square errors, it is shown that the proosed algorithm can enhance the performance of cancelling the echo signals. Some experiments, which is used multimedia personal computer, are carried out. As results of experiments, the proposed algorithm shows better performance than conventional ones.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1A
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• pp.58-65
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• 2002

This paper deals with a delay problem in the endpoint detection of double-talk detection algorithm using correlation coefficient in the acoustic echo canceller. In case that past power is much bigger than current power like at the end of double-talking period, the power, estimated using forgetting factor, decreases slowly to cause a delay problem in the endpoint detection. In this paper, two methods are proposed to solve this problem. One is that the current power is periodically replaced by a new average power and the other is that the past power in recursive equation is periodically removed or replaced by other values. The simulation results show that proposed methods outperform conventional method in the endpoint of double-talking periods without increasing the computational burden much more.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.161-164
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• 2000

본 논문에서 제안한 동시통화 검출기는 기존의 전대역에서 이루어지던 상호상관계수를 이용한 동시통화 검출기의 검출성능을 향상시키기 위하여 웨이브릿변환된 각각의 서브밴드 내에서 동시통화 및 반향경로를 구별하여 효율적으로 검출할 수 있도록 구성하였다. 서브밴드 동시통화 검출기 사용으로 동시통화 시에 발생하는 적응필터의 계수 발산을 막음으로써 시스템의 안정성을 높이고, 근단화자 신호가 원단화자에게 더 유쾌하게 들릴 수 있게 함으로써 원활한 통화환경을 제공할 수 있도록 구현하였다.

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.438-445
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• 2002

In this paper, the necessity of the filter coefficients compensation for the lattice transversal joint (LTJ) adaptive filter was explained in general and with ease by analyzing it with respect to the time-varying transform domain adaptive filter. And also the reduction method of computational complexity for filter coefficients compensation was proposed using the property that speech signal is stationary during a short time period and its effectiveness was verified through experiments using artificial and real speech signals. The proposed adaptive filter reduces the computational complexity for filter coefficients compensation by 95%, and when the filter is applied to the acoustic echo canceller with 1000 taps, the total complexity is reduced by 82%.

• Speech Sciences
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• v.9 no.3
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• pp.17-24
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• 2002

The goal of this research is the real-time implementation of an AEC (Acoustic Echo Canceller) using the floating-point digital signal processor of TMS320C31. We employ an FIR-type adaptive filter with the conventional NLMS (Normalized Least Mean Square) algorithm for the adaptation of filter coefficients. We program and optimize the system in the assembler level to make it run in real-time. With 8 kHz sampling rate, the implemented AEC requires $46\;\mu$sec and $77\;\mu$sec computational time per sample for 128-and 256-tap filter, respectively. It corresponds to 37% and 62% of maximum computational ability of TMS320C31 DSP.