• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.205-209
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• 2013

A conventional acoustic echo suppressor (AES) considering only room impulse response between a loudspeaker and a microphone eliminates acoustic echo from the microphone input. However, in a nonlinear acoustic echo environment, the conventional AES degraded because of a nonlinearity of the loudspeaker. In this paper, we adopt AES based on the frequency-domain second-order Volterra filter using Least Square method. For comparing performances, we conduct objective tests including Echo Return Loss Enhancement (ERLE) and Speech Attenuation (SA). The proposed algorithm shows better performance than the conventional in both linear and nonlinear acoustic echo environments.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1E
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• pp.17-24
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• 2008

Blind speech separation (BSS) is well-known as a powerful technique for speech enhancement in many real world environments. In this paper, we propose a new application of BSS - acoustic echo cancellation (AEC) in a car environment. For this purpose, we develop a block-online BSS algorithm which provides robust separation than a batch version in changing environments with moving speakers. Simulation results using real world recordings show that the block-online BSS algorithm is very robust to speaker movement. When combined with AEC, simulation results using real audio recording in a car confirm the expectation that BSS improves double talk detection and echo suppression.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3
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• pp.57-63
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• 1997

This paper proposes a new stereophonic acoustic echo canceller to preven the performance degradation at the instant of far-end talker's movement or change in transmission room environment in stereophonic teleconferencing system. In stereophonic acoustic echo canceller, the filter coefficients of the adaptive filters for echo cancellation do not have unique solutions and not converge to their optimum values. Therefore, the change of the far-end transmission room environment leads to degradation of ERLE(Echo Return Loss Enhanceement). Moreover, their computational complexity increases as the number of the filter coefficient increases, and their adaptive filters converge very slowly. So the real-time implementation is very difficult. To overcome these problems we propose a pre-processor consisting of an adaptive filter for making pseudo stereophonic signal, and it results in improved performance at the instant of environment change at far-end side and reduction of the total complexity.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.42-46
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• 2002

In this paper, a new pre-processing algorithm is proposed to improve the performance of stereo acoustic echo canceller. The proposed algorithm has the improved performance by the estimation error reduction of filter coefficient using input signal which was reduced reverberation of room in the basis MINT (Mu1tip1e-input/output Inverse Theorem) filtering. For real stereo speech signal and real room impulse response the results of simulation, we showed that the proposed method could improved 3∼5 dB ERLE (Echo Return Loss Enhancement) regardless of NLMS (Normalized Least Mean Square) and Projection adaptive algorithm.

• 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.

• 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 Korean Institute of Telematics and Electronics S
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• v.36S no.4
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• pp.93-100
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• 1999

확성회의 시스템에서 응용되는 반향신호 제거기는 긴 잔향시간을 갖는 실내 공간의 환경변화에 따라 필터 계수의 갱신에 많은 시간이 요구되어 실시간 처리에 문제점으로 지적되고 있다. 본 논문에서는 연산량 저감을 통한 실시간 처리를 위하여 중첩가산방식의 SSB(Single Side Band) 필터뱅크를 사용한 서브밴드 적응 신호처리법을 제안한다. 이 방법은 입력과 출력의 스펙트럼을 몇 개의 주파수 밴드로 분할하여, 각 밴드를 ES-NLMS(Exponential Step-Normalized Least Mean Square) 알고리즘을 이용하여 적응 처리하는 것이다. 시뮬레이션 결과 중첩가산방식의 SSB 필터뱅크가 풀밴드 보다 ERLE(Echo Return Loss Enhancement)가 1∼2㏈ 정도 작을 때 연산량이 풀밴드 보다 약95%, QMF(Quadrature Mirror Filter)필터뱅크보다 약50% 정도 감소하여 우수한 것으로 나타났다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.5
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• pp.556-563
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• 2002

In this paper, a novel method for designing an M-channel, causal, stable IIR cosine-modulated filter bank (CMFB) with near PR property is proposed. The IIR prototype filter is designed with a simple constraint using lattice stucture with 1st order allpass filter components. The IIR prototype filter which is designed by the proposed method has higher stopband attenuation and sharper roll-off characteristic than the one which is designed by the previously proposed method with similar complexity. The proposed M-channel IIR CMFB which is designed from this IIR prototype filter is applied to subband acoustic echo canceller (AEC). We obtained about 15dB higher ERLE using this subband AEC than when M-channel FIR subband AEC with similar complexity.