• Title, Summary, Keyword: LMS

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Deterministic Function Variable Step Size LMS Algorithm (결정함수 가변스텝 LMS 알고리즘)

  • Woo, Hong-Chae
    • Journal of the Institute of Convergence Signal Processing
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    • v.12 no.2
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    • pp.128-132
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    • 2011
  • Least mean square adaptive algorithms have played important role in radar, sonar, speech processing, and mobile communication. In mobile communication area, the convergence rate of a LMS algorithm is quite important. However, LMS algorithms have slow and non-uniform convergence rate problem For overcoming these shortcomings, various variable step LMS adaptive algorithms have been studied in recent years. Most of these recent LMS algorithms have used complex variable step methods to get a rapid convergence. But complex variable step methods need a high computational complexity. Therefore, the main merits such as the simplicity and the robustness in a LMS algorithm can be eroded. The proposed deterministic variable step LMS algorithm is based upon a simple deterministic function for the step update so that the simplicity of the proposed algorithm is obtained and the fast convergence is still maintainable.

Variable Step LMS Algorithm using Fibonacci Sequence (피보나치 수열을 활용한 가변스텝 LMS 알고리즘)

  • Woo, Hong-Chae
    • Journal of the Institute of Convergence Signal Processing
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    • v.19 no.2
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    • pp.42-46
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    • 2018
  • Adaptive signal processing is quite important in various signal and communication environments. In adaptive signal processing methods since the least mean square(LMS) algorithm is simple and robust, it is used everywhere. As the step is varied in the variable step(VS) LMS algorithm, the fast convergence speed and the small excess mean square error can be obtained. Various variable step LMS algorithms are researched for better performances. But in some of variable step LMS algorithms the computational complexity is quite large for better performances. The fixed step LMS algorithm with a low computational complexity merit and the variable step LMS algorithm with a fast convergence merit are combined in the proposed sporadic step algorithm. As the step is sporadically updated, the performances of the variable step LMS algorithm can be maintained in the low update rate using Fibonacci sequence. The performances of the proposed variable step LMS algorithm are proved in the adaptive equalizer.

Variable Step Size LMS Algorithm Using the Error Difference (오류 차이를 활용한 가변 스텝 사이즈 LMS 알고리즘)

  • Woo, Hong-Chae
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.3
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    • pp.245-250
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    • 2009
  • In communications and signal processing area, a number of least mean square adaptive algorithms have been used because of simplicity and robustness. However the LMS algorithm is known to have slow and non-uniform convergence. Various variable step size LMS adaptive algorithms have been introduced and researched to speed up the convergence rate. A variable step size LMS algorithm using the error difference for updating the step size is proposed. Compared with other algorithms, simulation results show that the proposed LMS algorithm has a fast convergence. The theoretical performance of the proposed algorithm is also analyzed for the steady state.

A Robustness Improvement of Adjoint-LMS Algorithms for Active Noise Control (능동소음제어를 위한 Adjoint-LMS 알고리즘의 강인성 개선)

  • Moon, Hak-ryong;Shon, Jin-geun
    • The Transactions of the Korean Institute of Electrical Engineers P
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    • v.65 no.3
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    • pp.171-177
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    • 2016
  • Noise problem that occurs in living environment is a big trouble in the economic, social and environmental aspects. In this paper, the filtered-X LMS algorithms, the adjoint LMS algorithms, and the robust adjoint LMS algorithms will be introduced for applications in active noise control(ANC). The filtered-X LMS algorithms is currently the most popular method for adapting a filter when the filter exits a transfer function in the error path. The adjoint LMS algorithms, that prefilter the error signals instead of divided reference signals in frequency band, is also used for adaptive filter algorithms to reduce the computational burden of multi-channel ANC systems such as the 3D space. To improve performance of the adjoint LMS ANC system, an off-line measured transfer function is connected parallel to the LMS filter. This parallel-fixed filter acts as a noise controller only when the LMS filter is abnormal condition. The superior performance of the proposed system was compared through simulation with the adjoint LMS ANC system when the adaptive filter is in normal and abnormal condition.

A Study on Applying the ${\mu}$-LMS Algorithm to the Adaptive Antenna Systems (${\mu}$-LMS 알고리즘의 적응 안테나 시스템에의 응용에 관한 연구)

  • Shin, Yoon-Ki
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.23 no.2
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    • pp.170-177
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    • 1986
  • The adaptive array antenna with the LMS algorithm has the advantage in that it can perform train't because of its slower convergencerate. In this paper, the \ulcornerLMS algorithm is applied to the adaptive array so that the convergence rate can be improved, and the performance of he adaptive array by the \ulcornerLMS algorithm is compared to, that of the LMS adaptive array. It is shown that the adaptive array by the \ulcornerLMS algorithm is superior to the LMS adaptive array in the narrow frequency band.

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A Study on LMS-MPC Method Considering Low Bit Rate (Low Bit Rate을 고려한 LMS-MPC 방식에 관한 연구)

  • Lee, See-Woo
    • Journal of Digital Convergence
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    • v.10 no.5
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    • pp.233-238
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    • 2012
  • In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of exist a voiced and an unvoiced consonants in a frame. To solve this problem, this paper present a method of LMS-MPC uses individual pitch and LMS(Least Mean Square). I evaluate the MPC and LMS-MPC using LMS. As a result, SNRseg of LMS-MPC was improved 1.5dB for female voice and 1.3dB for male voice respectively. Compared to the MPC, SNRseg of LMS-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

LEIOMYOSARCOMA OF THE MANDIBULAR GINGIVA: A CASE REPORT (하악치은에 발생한 평활근육종의 치험례)

  • Cho, Eun-Young;Ha, Jong-Woon;Kim, Eun-Cheol;Jeong, Yun-Shim
    • Journal of the Korean Association of Oral and Maxillofacial Surgeons
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    • v.27 no.4
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    • pp.362-366
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    • 2001
  • Leiomyosarcoma(LMS) is a malignant tumor from smooth muscle origin that arises most commonly in the gastrointestinal tract and uterus, but rarely in the oral and maxillofacial area which lacks smooth muscles. 63 cases of oral LMS have been reported, especially LMS that occurred in the mandibular gingiva is only 2 cases. Major symptom of oral LMS is painful or painless swelling. Besides it, oral LMS has no specific characteristics. So it is difficult to diagnose the case as LMS of the oral cavity. Certain cases report that LMS of the oral cavity was misdiagnosed as periodontitis and the patients were treated with unnecessary procedures. Conventional treatment of LMS is the radical surgical excision. LMS that is infiltrative and very malignant has poor prognosis despite of well-circumscribed boundary. LMS of the oral cavity is often recurred, has high rate of distant metastasis and 5-year-survival rate is as low as 23%. This article reports LMS of the mandibular gingiva that treated with surgical intervention, had local recurrence and metastasis to the lymph node after 16-month's follow-up examination.

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Parallel M-band DWT-LMS Algorithm to Improve Convergence Speed of Nonlinear Volterra Equalizer in MQAM System with Nonlinear HPA (비선형 HPA를 가진 M-QAM 시스템에서 비선형 Volterra 등화기의 수렴 속도 향상을 위한 병렬 M-band DWT-LMS 알고리즘)

  • Choi, Yun-Seok;Park, Hyung-Kun
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.7C
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    • pp.627-634
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    • 2007
  • When a higher-order modulation scheme (16QAM or 64QAM) is applied to the communications system using the nonlinear high power amplifier (HPA), the performance can be degraded by the nonlinear distortion of the HPA. The nonlinear distortion can be compensated by the adaptive nonlinear Volterra equalizer using the low-complexity LMS algorithm at the receiver. However, the LMS algorithm shows very slow convergence performance. So, in this paper, the parallel M-band discrete wavelet transformed LMS algorithm is proposed in order to improve the convergence speed. Throughout the computer simulations, it is shown that the convergence performance of the proposed method is superior to that of the conventional time-domain and transform-domain LMS algorithms.

Interference Cancellation System in Repeater Using Adaptive algorithm with step sizes (스텝사이즈에 따른 적응 알고리즘을 이용한 간섭제거 중계기)

  • Han, Yong-Sik
    • The Journal of the Korea institute of electronic communication sciences
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    • v.9 no.5
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    • pp.549-554
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    • 2014
  • In the paper, we propose a new Signed LMS(Least Mean Square) algorithm for ICS(Interference Cancellation System). The proposed Signed LMS algorithm improved performances by adjusting step size values. At the convergence of 1000 iteration state, the MSE(Mean Square Error) performance of the proposed Signed LMS algorithm with step size of 0.067 is about 3 ~ 18 dB better than the conventional LMS, CMA algorithm. And the proposed Signed LMS algorithm requires 500 ~ 4000 less iterations than the and LMS and CMA algorithms at MSE of -25dB.

A Study on the Co-LMS Algorithm Characteristics of Real-time Applicants for Road Environment Calming (도로환경 정온화의 실시간 적용을 위한 Co-LMS 알고리즘의 특성 고찰)

  • Moon, Hak-Ryong;Shon, Jin-Geun
    • The Transactions of the Korean Institute of Electrical Engineers P
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    • v.63 no.3
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    • pp.157-162
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    • 2014
  • The active noise control(ANC) method for noise problems solution generally uses filtered-X LMS algorithms. However, Filtered-X LMS algorithms were mainly used but these had a limitation that had to measure a transfer function of secondary noise path. However, newly proposed correlation-LMS algorithms have slightly much calculation and are minutely behind performance, these have a advantage not in measuring transfer function onerously so that we can easily adapt these in real time. Thus Co-LMS algorithm was developed to improve the real-time implementation performance under the variable input noise such as road noise environment. In this paper, the performance of the Co-LMS is presented in comparison with that of the Filtered-X LMS algorithm. Simulation results show that active noise control using Co-LMS have slightly much calculation and are minutely behind performance, these have a advantage not in measuring transfer function onerously so that we can easily adapt these in real time.