• Title/Summary/Keyword: 정규화 계수

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Combined Normalized and Offset Min-Sum Algorithm for Low-Density Parity-Check Codes (LDPC 부호의 복호를 위한 정규화와 오프셋이 조합된 최소-합 알고리즘)

  • Lee, Hee-ran;Yun, In-Woo;Kim, Joon Tae
    • Journal of Broadcast Engineering
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    • v.25 no.1
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    • pp.36-47
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    • 2020
  • The improved belief-propagation-based algorithms, such as normalized min-sum algorithm (NMSA) or offset min-sum algorithm (OMSA), are widely used to decode LDPC(Low-Density Parity-Check) codes because they are less computationally complex and work well even at low SNR(Signal-to-Noise Ratio). However, these algorithms work well only when an appropriate normalization factor or offset value is used. A new method that uses a CMD(Check Node Message Distribution) chart and least-square method, which has been recently proposed, has advantages on computational complexity over other approaches to get optimal coefficients. Furthermore, this method can be used to derive coefficients for each iteration. In this paper, we apply this method and propose an algorithm to derive a combination of normalization factor and offset value for a combined normalized and offset min-sum algorithm to further improve the decoding of LDPC codes. Simulations on the next-generation broadcasting standards, ATSC 3.0 LDPC codes, prove that a combined normalized and offset min-sum algorithm which takes the proposed coefficients as correction coefficients shows the best BER performance among other decoding algorithms.

Face Recognition with Normalized Wavelet Features (정규화된 웨이블렛 특징에 의한 얼굴 인식)

  • Lee, Chan-Ho;Park, Ju-Chul;Choi, Hyung-Il
    • Journal of KIISE:Software and Applications
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    • v.27 no.10
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    • pp.1046-1053
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    • 2000
  • 본 논문에서는 정규화된 웨이블렛 방법에 기반한 얼굴 인식 방법을 제안한다. 추출된 얼굴 영역의 크기를 정규화하고. 배경 영역을 제거하기 위해 이진화된 가우시안 윈도우를 사용하였으며, 또한 조명의 영향을 줄이기 위해 얼굴 영역의 히스토그램을 상세화하고, 왼쪽 부분의 밝기와 오른쪽 부분의 밝기를 평균하였다. 정규화된 얼굴 영역은 극좌표계로 표현하여 사각형의 형태를 가지게 하였다. 특징으로는 가보 웨이블렛 계수를 사용한다. 가보 웨이블렛 변환은 매개 변수와 정규화된 얼굴 영역의 해상도를 바꾸어 가며 여러 차례 적용하였다. 인식에 유용한 계수들을 선택하기 위해 FD 분석을 수행하였다. 선택된 특정들은 FD 값과 함께 인식에 사용되었다. 실험 결과를 보면 제안된 방법이 매우 유망하다는 것을 알 수 있다.

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A Fuzzy Controller using normalized Scale Factor (정규화 스케일계수를 이용한 퍼지제어기)

  • 정동화;이동욱;이상윤;신위재
    • Proceedings of the Korea Institute of Convergence Signal Processing
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    • 2003.06a
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    • pp.149-152
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    • 2003
  • 플랜트 모델이나 경험에 근거하여 설계된 퍼지제어기를 실제 플랜트에 적용할 경우, 모델링 오차와 플랜트에 대한 관련지식의 부족으로 만족할 만한 제어 결과를 나타내지 못할 경우가 있다. 이 경우 제어성능을 향상시키기 위해 제어기의 제어인자를 다시 조정하여야 하고, 이 조정과정은 시행착오 방법으로 수행되기 때문에 많은 시간과 비용을 필요로 한다. 본 논문에서는 정규화 된 오차와 오차 변화량를 사용하여 플랜트 응답에 따라 입력과 출력의 적절한 스케일 계수를 조정하는 퍼지제어기를 제안한다. 정규화 된 오차를 출력 소속함수의 중심과 폭에 곱해 출력 범위를 재조정하고, 플랜트 응답에 의해 입력의 스케일 계수를 결정한다. 이를 확인하기 위해 2차 플랜트에 적용하여 모의 실험을 수행하였다.

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An Acoustic Echo Cancellation Algorithm Using the Correlation of Input Signals and Error Signals (입력신호와 오차신호의 상관도를 이용한 음향반향제거 알고리즘)

  • 류종훈
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1998.08a
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    • pp.432-437
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    • 1998
  • NLMS 알고리즘을 채용한 음향반향제거기는 주변잡음에 대해서 적응필터의 계수가 오조정되어 반향제거기의 성능이 저하된다. 본 논문에서 음향반향제거기의 마이크 입력신호와 추정 오차신호의 상관도를 이용해서 주변 잡음신호에 의한 계수 오조정이 작은 적응 알고리즘과 잔여반향을 제거하기 위한 후처리기로 구성된 음향 반향 제거기를 제안한다. 기존의 NLMS 알고리즘이 입력신호의전력으로 적응상수를 정규화하지만 제안하는 알고리즘은 마이크 입력신호와 추정 오차신호의상관도와 입력신호 전력의 합으로 정규화한다. 적응필터가 반향 경로를 추정한 경우, 추정 오차신호에는 근단화자 신호가 대부분을 차지한다. 따라서 근단화자 신호가 있는 경우에는 상관도 값이 커져서 적응 상수가 작아지고 근단화자 신호에 의한 계수의 오조정을 줄일 수 있다. 후처리기도 마이크 입력신호와 추정 오차신호의 상관도를 마이크 입력신호의 전력으로 정규화한 값으로 추정 오차신호를 감쇠시킴으로써 근단화자 신호는 감쇠를 적게 하고 잔여반향을 감쇠시킨다. 멀티미디어 PC를 이용한 실험을 통해서 제안하는 알고리즘이 기존의 알고리즘에 비해서 우수한 성능을 보임을 확인했다.

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Application of Effective Regularization to Gradient-based Seismic Full Waveform Inversion using Selective Smoothing Coefficients (선택적 평활화 계수를 이용한 그래디언트기반 탄성파 완전파형역산의 효과적인 정규화 기법 적용)

  • Park, Yunhui;Pyun, Sukjoon
    • Geophysics and Geophysical Exploration
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    • v.16 no.4
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    • pp.211-216
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    • 2013
  • In general, smoothing filters regularize functions by reducing differences between adjacent values. The smoothing filters, therefore, can regularize inverse solutions and produce more accurate subsurface structure when we apply it to full waveform inversion. If we apply a smoothing filter with a constant coefficient to subsurface image or velocity model, it will make layer interfaces and fault structures vague because it does not consider any information of geologic structures and variations of velocity. In this study, we develop a selective smoothing regularization technique, which adapts smoothing coefficients according to inversion iteration, to solve the weakness of smoothing regularization with a constant coefficient. First, we determine appropriate frequencies and analyze the corresponding wavenumber coverage. Then, we define effective maximum wavenumber as 99 percentile of wavenumber spectrum in order to choose smoothing coefficients which can effectively limit the wavenumber coverage. By adapting the chosen smoothing coefficients according to the iteration, we can implement multi-scale full waveform inversion while inverting multi-frequency components simultaneously. Through the successful inversion example on a salt model with high-contrast velocity structures, we can note that our method effectively regularizes the inverse solution. We also verify that our scheme is applicable to field data through the numerical example to the synthetic data containing random noise.

Speech Recognition Using Formant Bandwidth Normalization (포만트 밴드폭 정규화를 이용한 음성인식)

  • 홍종진;강석건;박군작;박규태
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.16 no.5
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    • pp.458-467
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    • 1991
  • In this paper, the cause of linear prediction error is analysed and the theoretical basis for nomalizing the format bandwidth to 0is given and its validity is verified. The formant and bandwidth in relation to the position of the poles of AR filter are measured for an alaysis of the relation between the pole position and the formant bandwidth. By changing the glottis reflection coefficient to 1. the pole position and the formant bandwidth. By changing the glottis reflection coefficient to 1. the effect of the glottis is eliminated and as the result a new linear preiction coefficients are obtained by normalizing the formant bandwidth of the signal to 0. since these coefficients are symmetrical, the standard deviation is larger than the coefficients with fixed glottis reflection coefficient. The bit rate for speech coding can be reduced by a factor of 2 without any loss of information. Through computer simulation, recognition rate of 96.7% is botained by using the proposed algorithm in recognizing 5 Korean vowels in noisy environment.

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Relationship between Hydraulic Conductivity and Electrical Conductivity in Sands (사질토의 투수계수와 전기전도도 간의 상관관계)

  • Kim, Jinwook;Choo, Hyunwook;Lee, Changho;Lee, Woojin
    • Journal of the Korean Geotechnical Society
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    • v.31 no.6
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    • pp.45-58
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    • 2015
  • The aim of this study is to suggest a semi-empirical equation for estimating the hydraulic conductivity of sands using geoelectrical measurements technique. The suggested formula is based on the original Kozeny-Carman equation; therefore varying factors affecting the Kozeny-Carman equation were selected as the testing variables, and six different sands with varying particle sizes and particle shapes were used as the testing materials in this study. To measure both hydraulic and electrical conductivities, a series of constant head permeameter tests equipped with the four electrodes conductivity probe was conducted. Test results reveal that the effects of both pore water conductivity and flow rate in relation between hydraulic conductivity and formation factor (=pore water conductivity / measused conductivity of soil) of tested materials are negligible. However, because the variations of hydraulic conductivity of the tested sands according to particle sizes are significant, the estimated hydraulic conductivity using the formation factor varies with particle sizes. The overall comparison between the measured hydraulic conductivity and the estimated hydraulic conductivity using the suggested formula shows a good agreement, and the variation of hydraulic conductivity with varying Archie's m exponents is smaller compared with varying porosities.

A Method of Adaptive ISF Split Vector Quantization Using Normalized Codebook (정규화 코드북을 이용한 분할 벡터 구조의 ISF 적응적 양자화 기법)

  • Piao, Zhigang;Lim, Jong-Ha;Hong, Gi-Bong;Lee, In-Sung
    • The Journal of the Acoustical Society of Korea
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    • v.30 no.5
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    • pp.265-272
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    • 2011
  • In most of the ISF (or LSF) based real time speech codec, SVQ (split vector quantization) method is used to decrease the quantizer complexity and memory size of codebook. However, it produces drawback that the level of correlation between code vectors can not be used during vector splits. This paper presents a new method of adaptive ISF vector quantization, which compensates the drawbacks of SVQ structured quantizer for wideband speech codec. In each different frame, the proposed method makes use of the correlation between splitted vectors by adaptively changing codebook distribution according to ordering property of ISF. The algorithm is evaluated in AMR-WB, and shows about 1.5 bit per frame improvement.

The Effect of Laser Geometry and Material Parameters on the Single Mode Gain Difference in Quarter Wavelength Shifted DFB Laser above Threshold Current (문턱전류이상에서 구조 및 재료 변수들이 $\lambda$/4위상천이 DFB 레이저의 단일모드 이득차에 미치는 영향)

  • 이홍석;김홍국;김부균;이병호
    • Journal of the Korean Institute of Telematics and Electronics D
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    • v.36D no.3
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    • pp.75-84
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    • 1999
  • Systematic studies for the effect of the linewidth enhancement factor, the confinement factor, the internal loss and the cavity length on the single mode gain difference and the frequency detuning are performed for $\lambda$/4 phase shifted DFB lasers above threshold. The above threshold characteristics are mainly determined by the linewidth enhancement factor, not by the confinement factor or the parameter defined by the product of the linewidth enhancement factor and the confinement factor. The normalized internal loss defined by the product of the internal loss and the cavity length mainly determines the above threshold characteristics compared to that of the internal loss or the cavity length alone. The effect of the cavity length on threshold characteristics is larger than that of the internal loss in the case of the same normalized internal loss. The above threshold characteristics of quantum well lasers are more resistant to the variations of the confinement factor and the normalized internal loss than those of bulk lasers due to the small linewidth enhancement factor.

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Quantization Based Speaker Normalization for DHMM Speech Recognition System (DHMM 음성 인식 시스템을 위한 양자화 기반의 화자 정규화)

  • 신옥근
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.4
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    • pp.299-307
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    • 2003
  • There have been many studies on speaker normalization which aims to minimize the effects of speaker's vocal tract length on the recognition performance of the speaker independent speech recognition system. In this paper, we propose a simple vector quantizer based linear warping speaker normalization method based on the observation that the vector quantizer can be successfully used for speaker verification. For this purpose, we firstly generate an optimal codebook which will be used as the basis of the speaker normalization, and then the warping factor of the unknown speaker will be extracted by comparing the feature vectors and the codebook. Finally, the extracted warping factor is used to linearly warp the Mel scale filter bank adopted in the course of MFCC calculation. To test the performance of the proposed method, a series of recognition experiments are conducted on discrete HMM with thirteen mono-syllabic Korean number utterances. The results showed that about 29% of word error rate can be reduced, and that the proposed warping factor extraction method is useful due to its simplicity compared to other line search warping methods.