• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1533-1545
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• 1996

In this paper, the design and implementation of echo canceller with new variable step size algorithm is discussed. The method used in the new algorithm is to periodically adopt the test function which helps an optimal coefficient tracking. This algorithm outperforms LMS and VS algorithms in convergence speed and steady state error. As the period of test function is decreased, the speed of convergence is improved, but the number of calculation is increased, then the trade off between these parameters must be considered. Simulation results show new algorithm outperforms LMS and VS algorithms in convergence rate. For the design of hardware, circuit is designed with VHDL, and synthesized with Act1 withc is a FPGA library of ActelTM in use of synovation of InterGraph$^{TM}$. Verification of the synthesized circuit is carried out with simulator DLAB. The circuit based on the algorithm which is suggested in this paper calculated 7 radix places of inary number. A simulation data for the verification is based on the data of algorithm simulation. When the same input data is applied to the both simulation, output results of circuit simulation had slight difference in compare with that of algorithm simulation. The number of used gate is about 5,500 and We have 5.53MHz in maximum frequency.y.

• 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%.

• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.98.6-98
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• 2001

In the double-talk situation where both the near-end and far-end signal present, the performance of echo cancellation using the conventional LMS algorithm is easily degraded because it freezes the adaptation in this situation. Recently CLMS and ECLMS algorithms were proposed to solve this problem. These algorithms could be used to adapt the filter´s parameters continuously even in the double-talk situation. In this paper, we propose new recursion formulas to calculate the ECLMS algorithm. And we compare and analyze the performances of double-talk echo canceller according to changing the value of channel tracking factors ${\alpha}$, ${\beta}$ and forgetting factor λ. The computer simulation was performed and the results showed that, ...

• 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 Korea Multimedia Society
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• v.3 no.3
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• pp.263-271
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• 2000

In this paper, an acoustic echo canceller(AEC) is implemented by ADSP-2181. This AEC uses a noise robust adaptive algorithm and a postprocessing method which attenuates residual echo using cross-correlation between estimated error signal and microphone input signal. We propose new postprocessing method that uses two thresholds to prevent signal distortion after postprocessing and to improve the performance of AEC without extra computational burden. Through experiments using a 16 bit fixed-point DSP board (ADSP-2181 EZ-KIT Lite board), it is shown that the noise robust adaptive algorithm performs well in the double-talk situations and the convergence speed is comparable to NLMS. Using the postprocessor, ERLE is improved about 20 dB. As a result, the AEC with a postprocessor shows better performance than conventional ones.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.3
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• pp.97-104
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• 2007

Adaptive echo canceller(AEC) has become an important component in speech communication systems, including mobile phones and speech recognition. In these applications, the acoustic echo path has a long impulse response. We propose a moving-averge least mean square(MVLMS) algorithm with a detection method for acoustic echo cancellation. Using, the result of the tests that used colored input models clearly shows that the MVLMS detection algorithm has convergence performance superior to the least mean square(LMS) detection algorithm alone. Although the computational complexity of the new MVLMS algorithm is only slightly greater than that of the standard LMS detection algorithm, the new algorithm confers a significant improvement in stability.

• Journal of Korea Multimedia Society
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• v.3 no.6
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• pp.604-610
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• 2000

A stereo acoustic echo canceller has significant cross-correlation between each channel signal. As the result, the adaptive filter coefficient that estimates the acoustic echo path of a receiving room can misconverge to the path or converge slowly. In this paper, a new preprocessor using absolute difference in stereo signals is proposed to reduce cross-correlations and to improve the removal efficiency of the stereo acoustic echo. Compared to the previous preprocessor using a half wave rectifier, the newly proposed preprocessor showed better performance according to computer simulation. In addition, when the paths of transmitting room were changed the performance of the proposed preprocessor was not degraded.

• Journal of Broadcast Engineering
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• v.5 no.1
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• pp.123-129
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• 2000

This paper addresses a new double detection algorithm which is based on the fuzzy control in the adaptive echo canceller of communication system. In this method, the two input of the fuzzy inference for detecting double talk condition are used. The one is the cross-correlation coefficient between the error signal and the primary signal which is the summed signal of the real echo signal and the near-end signal. The other is the cross-correlation coefficient between the estimation error signal and the primary signal. The fuzzy controller made a fuzzification for two inputs by the membership functions of trapezoid and them became the composition using inference rules. The composed result is defuzzificated by the center gravity method. The output is compared with two threshold values to detect double talk and echo path variation effectively. It is confirmed by computer simulation that this fuzzy double talk detector is able to track echo path variation accurately.

• Proceedings of the Korea Multimedia Society Conference
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• 2001.11a
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• pp.46-50
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• 2001

Adaptive signal processing is used to acoustic echo canceller. adaptive noise canceller and adaptive algorithm among adaptive algorithms is mainly used because the structure is simple and computa LMS algorithm has trade-off between the converge speed and the steady state error. In this paper, step-size of adaptive algorithm is varied with orthogonality Principles of optimal filter to get fasts though small steady state error. Time varying step-size is determined proportional to the maximum vector of LMS algorithm. As results of simulations, the adaptive algorithm with proposed time-v compared with conventional ones.

• Proceedings of the KIEE Conference
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• 2005.07d
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• pp.2995-2997
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• 2005

통신망에서 최적 적응 반향제거기(Echo Canceller; EC)는 반향성분이 길게 존재하는 환경에서도 실시간으로 동작할 수 있도록 알고리즘이 간결하여야 하며, 시간에 따라 빠르게 변하는 동특성의 반향경로에서도 동작을 보장할 수 있도록 빠른 수렴특성을 갖아야 한다. 또한, 전화망에서 수십 [ms] 이상의 지연이 발생 할 경우에도 반향제거 성능이 우수해야 한다. 본 논문에서는 이러한 조건을 만족시키기 위해 오차의 크기에 따라 수렴속도를 가변시키는 오차적응 NLMS(Error-Adaptive NLMS) 알고리즘을 제안하였으며, 시뮬레이션을 통해 일반적으로 사용되는 LMS(Least Mean Square) 알고리즘과 이를 개선한 NLMS(Normalized LMS) 알고리즘과 성능을 비교하였다.