• 한국통신학회논문지
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• 제21권6호
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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.

• 전기학회논문지
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• 제62권12호
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• pp.1744-1749
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• 2013

Acoustic feedback is perceived as whistling or howling, which is a major complaint of hearing-aids users. Acoustic feedback cancellation is important in hearing-aids because acoustic feedback degrades performance of the hearing aid device by reducing maximum insertion gain. Adaptive systems for estimate acoustic feedback path and feedback suppression algorithms have been proposed in order to solve this problem. A typical feedback cancellation algorithm is LMS(least mean squares) because of its computational efficiency. However it has problem of convergence performance in high correlated input signal. In this paper, we propose a new variable step-size normalized LMS(least mean squares) algorithm using VAD(voice activity detection) to overcome the limitation of the LMS algorithm. The VAD algorithm is GSAP(global speech absence probability) and the feedback cancellation algorithm is normalized LMS. The proposed algorithm applies different step-size between voice and non-voice using VAD, for high stability, fast convergence speed and low misalignment when correlated inputs, such as speech. The result of simulation with white noise mixed speech signal, the proposed algorithm shows high performance then traditional algorithm in terms of stability, convergence speed and misalignment.

• 융합신호처리학회논문지
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• 제10권3호
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• pp.158-163
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• 2009

LMS(Least Mean Square) 알고리즘은 식이 간단하고 계산량이 비교적 적기 때문에 널리 사용되고 있다. 그러나 LMS 알고리즘은 수렴속도를 높이게 되면 주위 환경이 급격하게 변하는 경우 발산하게 되고, 안정성을 높이게 되면 수렴속도가 떨어지는 수렴속도와 안정성 사이에 trade-off가 발생한다. 이러한 LMS 알고리즘의 단점을 개선하기 위해 VSSLMS(Variable Step Size LMS) 알고리즘이 개발되었다. VSSLMS 알고리즘은 에러 신호를 이용하여 스텝사이즈를 조절하여 수렴속도와 성능을 동시에 향상시킨다. 본 논문에서는 실시간 환경에서 적응잡음 제거기의 성능을 개선시키기 위해 I-VSSLMS(Instantaneous Signal to Noise Radio VSSLMS)을 제안한다. 제안한 I-VSSLMS을 TMS320C6713 DSP 보드를 이용하여 적응 잡음 제거기에 적용하여, 실시간으로 시뮬레이션을 하였고, 각 알고리즘의 성능을 비교 분석하여 제안한 알고리즘의 성능이 우수함을 입증하였다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
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• pp.689-692
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• 2003

In this paper, we propose an adaptive beamforming algorithm for array antenna. The proposed beamforming algorithm, based on Block LMS (Block - Least Mean Squares) algorithm, has a variable step size from coefficient update. This method shows some advantages that the convergence speed is fast and the calculation time can reduced using a block LMS algorithm from frequency domain. As the adaptive parameter approaches a stationary state, it could reduce the number of filter coefficient update with the help of various step size. In this paper we compared the efficiency of the proposed algorithm with a standard LMS algorithm which is a representative method of adaptive beamforming.

• 전자공학회논문지S
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• 제34S권4호
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• pp.42-49
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• 1997

A conventional adaptive noise canceller (ANC) using LMS algorithm suffers from the misadjustment of adaptive filter weights due to the gradient-estimate noise by input speech signal at steady state. In this paper, an ANC is proposed which uses the combination of VSLMS (variable step size LMS) and SA (sign algorithm) to improve steady state performance and convergence rate. SA algorithm is applied in speech region to prevent the weights from perturbing by output speech of ANC and VSLMS algorithm is applied to improve convergence rate and channel tracking ability in silence region and adaptive transient region. In compute rsimulation, the performance of the proposed VSLMS-SA combination algorithm is much better than LMS algorithm and the algorithm, recently proposed by greenberg, with adaptation step-size parameter determine dby sum method in convergence rate, channel tracking and steady state performance.

• The Journal of the Acoustical Society of Korea
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• 제16권2E호
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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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2009년도 정보 및 제어 심포지움 논문집
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• pp.70-72
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• 2009

In this paper, nonlinear active noise control(ANC) of a 3rd-order Volterra system, which is robust to background noise variation, is proposed along with a variable step-size Gauss-Seidel pseudo affine projection(VSSGS-PAP) algorithm. Simulation results demonstrate that the proposed approach yields better nonlinear ANC performance in a background noise variation environment, compared with the conventional methods for the nonlinear ANC.

• 한국콘텐츠학회:학술대회논문집
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• 한국콘텐츠학회 2005년도 춘계 종합학술대회 논문집
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• pp.369-373
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• 2005

본 논문은 기존의 주파수 영역 독립 성분 분석(FDICA : frequency domain independent component analysis) 방법에 가변 적응 상수를 이용한 알고리즘을 제안한다. 여러 반향 환경에 대한 모의실험으로 기존의 temporal structure(TA)알고리즘, FDICA알고리즘과 비교하여 그 분리 성능을 평가하고 비교한다. 실제 녹음한 혼합 음원에 대한 분리정도를 비교 분석한다.

• 한국멀티미디어학회논문지
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• 제2권1호
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• pp.99-108
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• 1999

본 논문에서는 계산량이 크게 증가하지 않으면서 음향 반향 제거기의 성능을 향상시키는 잡음에 강건한 알고리즘을 제안한다. 제안하는 알고리즘은 NLMS 알고리즘을 기반으로 적응 상수를 기준 입력신호와 마이크 입력신호의 전력합으로 정규화한다. 마이크 입력으로 근단화자의음성신호와 잡음이 입력되면 그 전력만큼 적응 상수가 작아지기 때문에 이들 주변 잡음신호에 의한 계수 오조정을 줄일 수있다. 따라서 제안하는 알고리즘은 동시통화와 주변 잡음이 있는 상황에서 반향 제거 기능을 잘 수행한다. 제안하는 알고리즘의 수렴 속도는 반향 신호가 10~20 dBSPL 정도 감쇠되기 때문에 NLMS 알고리즘에 비해서 크게 느려지지 않는다. 본 논문에서는 제안하는 알고리즘의 성능을 분석하고 기존의 알고리즘과 비교한다.

• 한국전자파학회논문지
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• 제19권9호
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• pp.1058-1064
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• 2008

본 논문에서는 RF(Radio Frequency) 중계기에서 그룹화 CMA(Constant Modulus Algorithm)와 LMS(Least Mean Square) 알고리즘을 이용하여 적응 필터를 적용시킨 새로운 혼합 간섭 제거기를 제안한다. 송신 안테나에서 수신안테나로 궤환되는 신호는 수신 시스템의 성능을 저하시킨다. 제안한 간섭 제거기는 그룹화 CMA 알고리즘 간섭 제거 기법을 적용시키기 때문에 기존 구조보다 나은 채널 적응 성능과 낮은 MSE(Mean Square Error)을 가진다. 이 구조는 기존 비선형 간섭 제거기에 비해 같은 MSE(Mean Square Error)에 대한 반복수와 하드웨어 복잡도를 줄여준다. 즉, 제안한 알고리즘은 LMS 알고리즘에 비해 평균 자승 에러가 적응 상수에 따라 2.5 dB 또는 4 dB 정도 낮은 값을 보였다. 또한, VSS(Variable Step Size)-LMS 알고리즘에 비해 수렴 속도가 빠르고, 비슷한 평균 자승 에러를 가진다.