• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.24-26
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• 2006

In this paper, a nonlinear echo cancellation approach, based on a Gauss-Seidel pseudo affine projection algorithm and Volterra filtering, is proposed to compensate for echo path nonlinearity in the telephone network. Simulation results demonstrate that the proposed approach yields reduction of computational complexity and improved convergence speed than the conventional nonlinear echo cancellation methods (NLMS, ECLMS, FAP, RLS).

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.53 no.6
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• pp.446-452
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• 2004

Under the condition of a common narrowband target signal and interference signals from several directions, the linearly constrained minimum variance (LCMV) method using the generalized sidelobe canceller (GSC) for adaptive beamforming has been exploited successfully However, in the case of wideband signals, the length of the adaptive filter must be extended. As a result, the complexity of the beamformer increases, which makes real-time implementation difficult. In this paper, we improve the convergence characteristics of the adaptive filter using the transform domain normalized least mean square (NLMS) approach based on the subband GSC structure without the increase of complexity. Besides, the M-MAX algorithm, which is one of various selective coefficient updating methods, is employed in order to remarkably reduce the computational cost without decreasing the convergence quality. With the combination of these methods, we propose a computationally efficient wideband adaptive beamformer and verify its efficiency through a series of simulations.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.3-11
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• 1999

The conventional stereophonic acoustic echo cancellers need two adaptive filters to estimate one channel echo signal. Since the two channel signals are strongly correlated, the ESR of the input signals is considerably increased whatever the input signals may be. This causes the slow convergence of the adaptive filter for echo cancellation. To speed up the convergence, the AP algorithm is frequently used for the stereophonic acoustic echo canceller although there isn't a fast version for 2-channel case. The AP algorithm can be approximated with the Gram-Schmidt orthogonalization and a TDL structure. We propose a two channel algorithm for stereophonic acoustic echo canceller with the approximated AP algorithm.

• 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%.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.2
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• pp.128-136
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• 2002

In this paper, we developed an acoustic echo canceller in real-time using TeakLite DSP Core, which will be placed in the vocoder chip of a mobile handset. Considering the limited computational capacity given to the acoustic echo canceller in a vocoder chip, we employed a FIR-type adaptive filter using a conventional NLMS algorithm. To begin with, we designed and implemented an acoustic echo canceller with floating-point format C-source code, and then converted it into fixed-point format through integer simulation. Then we programmed and optimized it in the assembler level to make it run ill real-time. After optimization procedure, the implemented echo canceller has approximately 624 words of program memory and 811 words of data memory. With 8 KHz sampling rate and 256 filter taps in the echo canceller that corresponds to 32 msec of echo delay, it requires 14.12 MIPS of computational capacity. For coverage of 16 msec echo delay, i.e., 128 filter taps, 9 MIPS is requited.

• MALSORI
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• no.62
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• pp.85-96
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• 2007

In this paper, we carried out recognition experiments for noisy speech having various levels of car noise and output of an audio system using the speech interface. The speech interface consists of three parts: pre-processing, acoustic echo canceller, post-processing. First, a high pass filter is employed as a pre-processing part to remove some engine noises. Then, an echo canceller implemented by using an FIR-type filter with an NLMS adaptive algorithm is used to remove the music or speech coming from the audio system in a car. As a last part, the MMSE-STSA based speech enhancement method is applied to the out of the echo canceller to remove the residual noise further. For recognition experiments, we generated test signals by adding music to the car noisy speech from Aurora 2 database. The HTK-based continuous HMM system is constructed for a recognition system. Experimental results show that the proposed speech interface is very promising for robust speech recognition in a noisy car environment.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.11a
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• pp.163-166
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• 2010

본 논문에서는 현재 상용화 되어있는 WCDMA용 ICS (Interference Cancellation System) 중계기의 성능 개선을 위한 개선된 LMS 알고리즘을 연구하였다. ICS 중계기의 수신으로 입력되는 신호는 수신 신호와 궤환되어 입력되는 신호로 구성된다. 이렇게 입력되는 궤환 신호를 LMS와 같은 적응형 채널 추정 알고리즘을 통해 제거하는 기술이 ICS 중계기의 핵심 요소이다. 중계기의 저비용 및 단순화를 위해서는 기존에 사용되어온 적응형 채널 추정 알고리즘의 단순화가 필요하다. 실험을 통해 기존 NLMS 알고리즘 및 계산 복잡도 감소를 위해 수정된 LMS 알고리즘을 MSE (Mean Square Error) 기준에서 성능 비교를 하였다.

• 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.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.56 no.2
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• pp.420-423
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• 2007

In this paper, an efficient nonlinear echo cancellation method is proposed, whereby the fast block exact Gauss-Seidel pseudo affine projection (FBEGS-PAP) is further utilized for adaptive Volterra filtering. In particular, the proposed nonlinear echo cancellation approach requires lower computational complexity as in the conventional linear adaptive echo cancellation methods based on NLMS and GS-PAP, and still provides nonlinear echo cancellation performance similar to the GS-PAP method. Finally, echo cancellation performance of the proposed approach is demonstrated by providing some simulation results.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.357-358
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• 2006

We propose the robust PBFLMS in noises, which is the enhanced acoustic echo canceller using ACPBF-LMS(Alternative Constrained Partitioned Block Frequency domain Least Mean Square) algorithm. The defect of the block structure filtering is the deterioration of convergence efficiency from noise and interference. To improve the performance of convergence efficiency, noise effect should be reduced. The new method of reducing noise effect is proposed, which apply the estimated background noise to adaptive filter step size. By experiments, the proposed acoustic echo canceller has TCL of 50dB, and always provides faster convergence speed and lower complexity than the full-tap NLMS. We also carried out an implementation of PBFLMS using ARM9EJ-S.