• 제목/요약/키워드: LMS method

검색결과 304건 처리시간 0.026초

개별탭 LMS 알고리듬을 이용한 이동통신 페이딩 채널의 적응 등화 (Adaptive Equalizations for Multipath Fading Channels in Mobile Communications Using the Individual Tap LMS Algorithm)

  • 김남용;강창언
    • 한국통신학회논문지
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    • 제16권8호
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    • pp.745-757
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    • 1991
  • 본 논문에서는 적응등화에 쓰이는 개별탭 조절 LMS 알고리즘의 수렴성실을 이론적으로 분석하고 여러 다경로 페이딩 이동체 채널에 대해 그 성능을 비교 부넛ㄱ하였다. LMS 알고리즘을 사용한 TDL 등화기의 개별 탭 조절 방법은 Wiener 최적 해를 가진다는 것을 보였으며 시변 또는 시불변 다겨로 이동체 채널에서 기존의 TDL-LMS 및 garient lattice 등화기보다 빠른 수렴속도와 더 낮은 비트 오차율을 나타내었다.

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

  • 문학룡;손진근
    • 전기학회논문지P
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    • 제63권3호
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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.

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

  • 최윤석;박형근
    • 한국통신학회논문지
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    • 제32권7C호
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    • pp.627-634
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    • 2007
  • 고효율 변조기법 (16 QAM or 64QAM)이 비선형 고전력 증폭기 (High-Power Amplifier; HPA)를 사용하는 통신시스템에 적용되었을 때 비선형 왜곡에 의해 성능저하가 발생할 수 있다. 이런 비선형 왜곡은 수신부에서 복잡도가 낮은 LMS 알고리즘을 적용한 적응적 비선형 Volterra 등화기를 사용하여 보상할 수 있지만, 매우 느린 수렴 속도를 가지는 단점이 있다. 본 논문에서는 수렴 속도를 향상시키기 위한 병렬 M대역 이산 웨이블릿 변환영역 LMS (Parallel M-band Discrete Wavelet Transform Least Mean Square) 알고리즘을 제안한다. 모의실험을 통하여 제안된 기법이 기존의 시간 영역 LMS 알고리즘과 변환 영역 LMS 알고리즘들에 비해 수렴 속도가 우수함을 보여준다.

웨이블릿 변환역 최소평균자승법을 이용한 능동 소음 제어 (Active Noise Control Using Wavelet Transform Domain Least Mean Square)

  • 김도형;박영진
    • 한국소음진동공학회:학술대회논문집
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    • 한국소음진동공학회 2000년도 춘계학술대회논문집
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    • pp.269-273
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    • 2000
  • This paper describes Active Noise Control (ANC) using Discrete Wavelet Transform (DWT) Domain Least Mean Square (LMS) Method. DWT-LMS is one of the transform domain input decorrelation LMS and improves the convergence speed of adaptive filter especially when the input signal is highly correlated. Conventional transform domain LMS's use Discrete Cosine Transform (DCT) because it offers linear band signal decomposition and fast transform algorithm. Wavelet transform can project the input signal into the several octave band subspace and offers more efficient sliding fast transform algorithm. In this paper, we propose Wavelet transform domain LMS algorithm and shows its performance is similar to DCT LMS in some cases using ANC simulation.

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새로운 반복 LMS 기반의 결정 궤환 등화기의 설계 (Design of Novel Iterative LMS-based Decision Feedback Equalizer)

  • 최윤석;박형근
    • 전기학회논문지
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    • 제56권11호
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    • pp.2033-2035
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    • 2007
  • This paper proposes a novel iterative LMS-based decision feedback equalizer for short burst transmission with relatively short training sequence. In the proposed equalizer, the longer concatenated training sequence can provide the more sufficient channel information and the reused original training sequence can provide the correct decision feedback information. In addition, the overall adaptive processing is performed using the low complexity LMS algorithm. The study shows the performance of the proposed method is enhanced with the number of iterations and, furthermore, better than that of the conventional LMS-based DFEs with the training sequence of longer or equal length. Computational complexity is increased linearly with the number of iterations.

LMS 기반 적응 등화기에서 빠른 수렴을 위한 기준신호 변형 (Modification of the Reference Signal for Fast Convergence in LMS-based Adaptive Equalizers)

  • 이기헌;최진호;박래홍;송익호;박재혁;이병욱
    • 한국통신학회논문지
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    • 제19권5호
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    • pp.939-951
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    • 1994
  • 최소 평균 제곱(least mean squares) 알고리듬에 기반을 둔 적응 동화기의 수렴속도는 입력신호의 공분산 행렬에 의해 결정된다. 공분산 행렬의 고유값 spread가 1에 가까울때 수렴속도는 매우 빠르다. 본 논문에서는 LMS 기반 적응 동화기의 빠른 수렴을 위해 통계적으로 주어진 채널에 적합한 변형된 기준신호를 제안하였다. 이론적인 분석과 모의실험을 통해 LMS 기반 적응 동화기의 빠른 수렴을 얻는데 이 변형 방법이 효과적임을 알 수 있었다.

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MLMS-SUM Method LMS 결합 알고리듬을 적용한 웨이브렛 패킷 적응잡음제거기 (Wavelet Packet Adaptive Noise Canceller with NLMS-SUM Method Combined Algorithm)

  • 정의정;홍재근
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 1998년도 추계종합학술대회 논문집
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    • pp.1183-1186
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    • 1998
  • Adaptive nois canceller can extract the noiseremoved spech in noisy speech signal by adapting the filter-coefficients to the background noise environment. A kind of LMS algorithm is one of the most popular adaptive algorithm for noise cancellation due to low complexity, good numerical property and the merit of easy implementation. However there is the matter of increasing misadjustment at voiced speech signal. Therefore the demanded speech signal may be extracted. In this paper, we propose a fast and noise robust wavelet packet adaptive noise canceller with NLMS-SUM method LMS combined algorithm. That is, we decompose the frequency of noisy speech signal at the base of the proposed analysis tree structure. NLMS algorithm in low frequency band can efficiently dliminate the effect of the low frequency noise and SUM method LMS algorithm at each high frequency band can remove the high frequency nosie. The proposed wavelet packet adaptive noise canceller is enhanced the more in SNR and according to Itakura-Satio(IS) distance, it is closer to the clean speech signal than any other previous adaptive noise canceller.

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LMS 알고리즘을 이용한 RF 중계기의 궤환 간섭신호 제거 시스템(Interference Cancellation System) (Feedback Interference Cancellation System of RF Relay Utilizing the LMS Algorithm)

  • 김민수;안성수
    • 전자공학회논문지 IE
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    • 제45권1호
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    • pp.38-43
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    • 2008
  • 본 논문은 RF 중계기에서 발생되는 궤환 신호로 인한 발진현상을 제거하는 간섭제거 방법을 제시한다. 본 논문에서는 LMS(Least Mean Square) 알고리즘을 이용하여 궤환 신호를 추정하고, 그에 상응하는 감쇄신호를 만들어 간섭을 제거하도록 하였다. 본 제안 방법을 이용하여 도플러(Doppler) 주파수를 변화시키면서 성능시험을 수행한 결과 궤환 신호를 상쇄시키는 -30dB의 감쇄신호를 발생시켜 발진을 방지할 수 있었다.

후두 미세수술과 연관된 구강 및 후두 합병증 (Oropharyngeal Complications Associated with Laryngomicrosurgery(LMS))

  • 강진욱;최승효;남순열
    • 대한후두음성언어의학회지
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    • 제14권1호
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    • pp.5-9
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    • 2003
  • Introduction : Laryngomicrosurgery(LMS) is frequent procedure applying to benign and early malignant larygeal disease such as vocal cord polyp, nodule and early glottic cancer. LMS has been known as safe procedure and short time consuming treatment. So few reports about complications in LMS was done. In this study, complications and problems from LMS were investigated and reported. Method : From 2000, January to 2001, December, 180 patients who were treated with LMS in Asan medical center were studied by retrograde chart review. Results : In these patients, total 9 patients(5%) were suffered from complication. 4 patients (2%) had teeth injury and 4(2%) were suffered from foreign body sensation in tongue and 1(1%) had hypoglossal nerve injury. Main mechanism of complications is thought by pressure injury by laryngoscope blade. No definite correlation between procedure and complication was observed. Conclusion : There are few neural complications with LMS such as lingual and hypoglossal nerve injury. Before operation of LMS, warning and informing of complications by mechanical stress must be done. Gentle procedure and short operation time are necessary to avoid these problems. And patients who have risk factors of oral complications such as dental disease or dental prosthesis must have dental evaluation and treatment before LMS procedure.

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주파수 영역 적응 디지탈 필터를 이용한 Magnitude-Squared Coherence 함수 추정 (On Estimating Magnitude-Squared Coherence Functions Using Frequency-Domain Adaptive Digital Filters)

  • 김도년;차일환;윤대희
    • 한국음향학회지
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    • 제7권2호
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    • pp.39-50
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    • 1988
  • 한 쌍의 주파수 영역 적응 디지탈 필터를 이용하여 두 신호간의 magnitude squared coherence(MSC) 함수를 추정하는 방법을 제안하였다. 이와 같은 방법은 시간 영역에서 한 쌍의 적응디지탈 필터를 이용하는 LMS-MSC(least mean square-MSC)알고리즘에 비하여 적은 양의 계산으로 MSC 함수를 구할 수 있다. MSC함수 추정을 위하여 대표적인 주파수 영역 적응 필터링 알고리즘인 CFLMS(constrained frequency-domain LMS)와 UFLMS (unconstrained frequency-domain LMS)알고리즘을 사용하였으며, 컴퓨터 시뮬레이션을 통하여 LMS-MSC 알고리즘과 성능을 비교하였다.

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