• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1991.10a
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• pp.111-114
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• 1991

This paper proposes an efficient adaptive echo canceller using pilot filter approach to achieve improved convergence speed. The pilot filter is an adaptive filter with only a few filter coefficients to filter the received signal for the purpose of whitening the signal. Thus the convergence speed of the main LMS-TDL filter combined with the pilot filter is improved. In the proposed echo canceller, an adaptive lattice predictor as the pilot filter is used and its inverse filter is used to equalize the distorted near end talker signal. Simulation results for colored signal show that the convergence speed of the proposed echo cancellation algorithm is faster than that of the conventional LMS-TDL echo cancellation algorithm.

• Journal of information and communication convergence engineering
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• v.2 no.2
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• pp.89-92
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• 2004

An adaptive algorithm for reducing the hardware complexity is presented. This paper proposes a modified LMS algorithm for the adaptive system and analyzes its convergence characteristics mathematically. An objective of the proposed algorithm is to reduce the hardware complexity. In order to test the performances, it is applied to the echo canceller, and a program is described. The results from simulations show that the echo canceller adopting the proposed algorithm achieves almost the same performances as one adopting the NLMS algorithm. If an echo canceller is implemented with this algorithm, its computation quantities are reduced to the one third as many as the one that is implemented with the NLMS algorithm, without so much degradation of performances.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.1
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• pp.66-69
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• 2009

An adaptive echo canceller is necessary for an application such as a speakerphone, 3G image telephony and VoIP service system. These echo cancellers need to have many taps for filtering echo signals. Many taps cause computation data to increase and convergence speed to be low. To overcome these problems, An adaptive echo canceller with the advanced convergence speed is proposed in this paper. To improve the speed, we divide an echo band into subbands and place a subband filter to be adaptive for each subband. Each subband filter recognizes the echo signal as subband echo signals. So, dynamic range of subband is small, the convergence speed is fast. Moreover, as the number of Tap and weight update are estimated in each subband, the implementation complex of a adaptive filter is low.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.261-268
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• 2003

Conventional acoustic echo cancellers using ES algorithm have simple structure and fast convergence speed compared with those using NLMS algorithm, but they are very weak to external noise because ES algorithm updates the adaptive filter taps based on average energy reduction rate of room impulse response in specific acoustical condition. To solve this problem, in this paper, a new update algorithm for acoustic echo canceller with stepsize matrix generator is proposed. A set of stepsizes is determined based on residual error energy which is estimated by two moving average operators, and applied to the echo canceller in matrix from, resulting in improved convergence speed. Simulations in various noise condition show that the proposed algorithm improves the robustness of acoustic echo canceller to external noise.

• 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 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.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.2362-2367
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• 2005

This paper reports a technique for discriminating double talk and echo path change using the stochastic characteristics of power change for an adaptive noise canceller. The causes of rapid error increasing are double talk and echo path change. When the echo path is changed, the system corrects the impulse response in order to reduce the error. However, in the case of double talk, the system has to suspend the updating impulse response in order to maintain the quality of the voice signal. In the conventional system, it was difficult to discriminate between the two situations. In this research, the stochastic characteristics of the voice power change in the double talk period were experimentally verified to be different from the power change during echo path changing. Based on the results, a new double talk detection method is proposed.

• Journal of applied mathematics & informatics
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• v.20 no.1_2
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• pp.637-645
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• 2006

An adaptive algorithm for reducing the hardware complexity is presented. This paper proposes a simplified LMS algorithm for the adaptive system and analyzes its convergence characteristics mathematically. An objective of the proposed algorithm is to reduce the hardware complexity. In order to test the performances, it is applied to the echo canceller, and a program is described. The results from simulations show that the echo canceller adopting the proposed algorithm achieves almost the same performances as one adopting the NLMS algorithm. If an echo canceller is implemented with this algorithm, its computation quantities are reduced to the half as many as the one that is implemented with the LMS algorithm, without so much degradation of performances.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2E
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• pp.14-20
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• 1997

The future teleconferencing systems will need an appropriate system which controls properly the acoustic echo for the convenient communication. The conventional acoustic echo cancellation algorithms involve large adaptive filters identifying the impulse response of the echo path. The use of adaptive IIR filters appears to be a reasonable way to reduce computational complexity. Effective cancellation of acoustic echo presented in teleconferencing system requires that adaptive filters have a rapid convergence speed. One of the main problems of acoustic echo cancellation techniques is that the convergence properties degrade for an highly correlated signal input such as speech signals. By the way, the introduction of linear prediction filers onto the structure of the acoustic echo cancellation represents one approach to decorrelate the speech signal. And variable step-size LMS algorithm improves the convergence speed through a little increasing of computational complexity. In this paper, we applied these two methods to the acoustic echo canceller(AEC) and showed that these methods have better performances than the conventional AEC.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.35-46
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• 2000

In this paper, we propose a subband acoustic echo canceller using the weighted Overlap-add adaptive filter bank to prevent the decrease of convergence speed in full-band US processing, and make it possible to realize the adaptive filter in block-parallel processing, this paper introduces the weighted overlap-add technique for subband echo canceller. Moreover, we propose a new double-talk detector which employs dedicated filter in addition to the energy comparison method simultaneously. The computer simulation results show that the performance of the proposed subband adaptive echo canceller double-talk detection