• The Journal of the Acoustical Society of Korea
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• v.18 no.2E
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• pp.45-52
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• 1999

This paper presents the integrated speech enhancement system for hands-free mobile communication. The proposed integrated system incorporates both acoustic echo cancellation and engine noise reduction device to provide signal enhancement of desired speech signal from the echoed plus noisy environments. To implement the system, a delayless subband adaptive structure is used for acoustic echo cancellation operation. The NLMS based adaptive noise canceller then applied to the residual echo removed noisy signal to achieve the selective engine noise attenuation in dominant frequency component. Two sets of computer simulations are conducted to demonstrate the effectiveness of the system; one for the fixed acoustical environment condition, the other for the robustness of the system in which, more realistic situation, the acoustic transmission environment change. Simulation results confirm the system performance of 20-25dB ERLE in acoustic echo cancellation and 9-19 dB engine noise attenuation in dominant frequency component for both cases.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.185-188
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• 2000

현대의 잡음이 많은 환경에서 적응 음향반향제거기는 배경잡음의 영향으로 원활한 통화환경을 제공할 수 없다. 이러한 문제점을 해결하기 위하여 음향반향 제거와 더불어 배경잡음을 제거하는 결합구조의 적응 음향반향제거기가 제안되었다. 본 논문에서는 기존의 결합구조가 가지는 단점을 보완하여 적응 음향반향제거기의 성능을 향상시켰다. 제안한 결합구조는 적응 음향잡음제거기의 기준입력 신호를 적응 음향잡음제거기의 오차신호와 같게 구성함으로서 배경잡음 신호뿐만 아니라 잔여반향 신호도 효율적으로 제거할 수 있다. 성능 평가를 위한 실험결과, 제안한 방법이 기존의 방법에 비하여 ERLE 성능이 수렴 구간에서 3㏈ 이상 향상되었음을 확인하였다.

• Journal of Korea Society of Digital Industry and Information Management
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• v.8 no.3
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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.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.425-433
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• 2003

Acoustic Echo Canceller (AEC) is a signal processing system for removing unwanted echo signals in teleconference and hands-free communication. Least mean square (LMS) algorithm is one of the adaptive echo cancellation algorithms and it has been most attractive because of its simplicity and robustness. However, the convergence properties of the LMS algorithm degrade with highly correlated input signals such as speech. For this reason, transform-domain adaptive filtering algorithm was introduced to decorrelate the colored input samples by using the orthogonal transform matrix such as DCT, DFT and then LMS adaptive filtering process is applied. In this paper, we propose a MLT domain adaptive echo canceller base on the MLT (Modulated lapped Transform) orthogonal transform matrix. The proposed algorithm achieves high decorrelation efficiency and fast convergence speed via modulated lapped transform of size 2NXN instead of NXN unitary transform such as DCT, DFT, Hadamad and it is applied to the acoustical echo cancellation system. Form the computer simulation with both synthesis and real speech, the proposed MLT domain adaptive echo canceller shows approximately twice faster convergence speed and 20∼30 ㏈ ERLE improvements over the DCT frequency domain acoustic echo cancellation system.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.41 no.9
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• pp.65-71
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• 2004

This paper describes a new adaptive algorithm which can reduce the required computation quantities in the adaptive filter. The proposed S-LMS algorithm uses only the signs of the normalized input signal rather than the input signals when coefficients of the filter are adapted. By doing so, there is no need for the multiplications and divisions which are mostly responsible for the computation quantities. To analyze the convergence characteristics of the proposed algorithm, the condition and speed of the convergence are derived mathematically. Also, we simulate an echo canceller adopting this algorithm and compare the performances of convergence for this algorithm with the ones for the other algorithm. As the results of simulations, it is proved that the echo canceller adopting this algorithm shows almost the same performances of convergence as the echo canceller adopting the SIA algorithm.

• Structural Engineering and Mechanics
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• v.68 no.3
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• pp.377-384
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• 2018

Bi-stable structure can be stable in both its extended and coiled forms. For the un-stressed thin cylindrical shell, the strain energy expressions are deduced by using a theoretical model in terms of only two parameters. Based on the principle of minimum potential energy, the bi-stable behaviors of the cylindrical shells are investigated. The results indicate that the isotropic cylindrical shell does not have the second stable configuration and laminated cylindrical shells with symmetric or antisymmetric layup of fibers have the second stable state under some confined conditions. In the case of antisymmetric laminated cylindrical shell, the analytical expressions of the stability are derived based on the extremal principle, and the shell can achieve a compact coiled configuration without twist deformation in its second stable state. In the case of symmetric laminated cylindrical shell, the explicit solutions for the stability conditions cannot be deduced. Numerical results show that stable configuration of symmetric shell is difficult to achieve and symmetric shell has twist deformation in its second stable form. In addition, the roll-up radii of the antisymmetric laminated cylindrical shells are calculated using the finite element package ABAQUS. The results show that the value of the roll-up radii is larger from FE simulation than from theoretical analysis. By and large, the predicted roll-up radii of the cylindrical shells using ABAQUS agree well with the theoretical results.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.150-155
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• 2014

본 논문에서는, 원거리 음성인식을 위한 서브밴드 필터링 기반의 빠르고 효율적인 에코제거 시스템을 제안한다. 제안하는 에코제거 시스템은 우선 채널간 유사도 (correlation) 가 높을 경우 적응필터가 오작동하는 것을 방지하기 위해 spatial decorrelation 을 적용하게 된다. 그리고 tree 형태를 가지는 IIR filterbank 기반의 subband 구조를 채택함으로써, 적은 차수로도 효과적인 analysis, synthesis 필터링을 수행할 수 있도록 한다. 이 과정에서 불가피하게 발생하는 서브 밴드간 spectral aliasing은 notch filter를 적용해 해결할 수 있다. 또한 적응 필터로는 improved proportionate normalized least-mean-square (IP-NLMS) 알고리즘을 사용해 수렴속도 및 에코제거 성능에서 우수함을 확인하였다. 마지막으로 decision-directed estimation 기반의 residual echo suppressor를 적용해 잔여 에코를 제거하게 된다. 본 논문에서는 각 단계를 구성하게 된 이론적인 배경을 소개하고, 실제 에코가 존재하는 환경에서 ERLE, 원거리 음성 인식률, computational complexity를 통해 제안하는 에코제거 시스템의 효과를 입증하도록 한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.7
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• pp.1650-1658
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• 1998

This paper describes how to remove echoes effectively using speech parameter information provided form speech coder. More specially, the proposed adaptive echo canceller utilizes the excitation signal or linearly predicted error signal instead of output speech signal of vocoder as the input signal for adaptation algorithm. The normalized least mean ssquare(NLMS) algorithm is used for the adaptive echo canceller. The proposed algorithm showed a fast convergece charactersitcis in the sinulatio compared to the conventional method. Specially, the proposed echo canceller utilizing the excitation signal of speech coder showed about four times fast convergence speed over the echo canceller utilizing the output speech signal of the speech coder for the adaptation input.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.4C
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• pp.343-349
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• 2010

In this paper, we study real time AEC (acoustic echo canceller) based on NLMS adaptive filter. Proposed method improves conversation quality by enhancing the performance of AEC during double talk section and reduces the power consumption by controling the adaption operation of NLMS adaptive filter. Proposed method examines the convergence of the NLMS adaptive filter, stores the estimated echo path and chooses operation of NLMS adaptive filter. Furthermore if double talk is detected, the proposed AEC utilizes the stored echo path optionally considering missed double talk time. When the proposed AEC is used, the performance of the AEC is enhanced although the simple double talk detector is used and the power consumption of the AEC is reduced.

• 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%.