• Title/Summary/Keyword: 암묵신호처리

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Multichannel Blind Deconvolution of Multistage Structure to Eliminate Interference and Reverberation Signals (간섭 및 반향신호 제거를 위한 다단계 구조의 다채널 암묵 디콘볼루션)

  • Lim, Joung-Woo;Jeong, Gyu-Hyeok;Joo, Gi-Ho;Kim, Young-Ju;Lee, In-Sung
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.44 no.1
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    • pp.85-93
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    • 2007
  • In case that multichannel blind deconvolution (MBD) applies to signals of which autocorrelation has a high level, separated signals are temporally whitened by diagonal elements of a separation filter matrix. In order to reduce this distortion, the algorithms, which are based on either constraining diagonal elements of a separation filter matrix or estimating a separation filter matrix by using linear prediction residual signals, are presented. Still, some problems are generated in these methods, when we separate reverberation of signals themselves or interference signals from mixed signals. To solve these problems, this paper proposes the multichannel blind deconvolution method which divides processing procedure into the stage to separate interference signals and the stage to eliminate a reverberation of signals themselves. In simulation results, we confirm that the proposed algorithm can solve the problems.

Initial Weighting Establishment Through Eigenanalysis for BSS in Two-by-two Delayed Mixture (2×2지연 혼합에서의 암문신호처리를 위한 고유값분석을 통한 초기값 설정)

  • Park, Keun-Soo
    • The Journal of the Korea institute of electronic communication sciences
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    • v.8 no.10
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    • pp.1451-1456
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    • 2013
  • This paper propose a method for fast convergence technique in frequency domain independent component analysis (FDICA) using eigenanalysis. It important, such as SONAR system, to eliminate the interference sources through fast algorithm. Through eigenanalysing a two-by-two delayed mixture case, information of delay can be used for initial weighting parameters. Simulations show the improved performances in convergence speed and noise rejection rate. The proposed method can present close weights for optimal convergence, noise can be diminished drastically about 3 times epoch, and get the better resultss with 1~3dB than the conventional method.

Non-hierarchical Clustering based Hybrid Recommendation using Context Knowledge (상황 지식을 이용한 비계층적 군집 기반 하이브리드 추천)

  • Baek, Ji-Won;Kim, Min-Jeong;Park, Roy C.;Jung, Hoill;Chung, Kyungyong
    • Journal of the Institute of Convergence Signal Processing
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    • v.20 no.3
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    • pp.138-144
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    • 2019
  • In a modern society, people are concerned seriously about their travel destinations depending on time, economic problem. In this paper, we propose an non-hierarchical clustering based hybrid recommendation using context knowledge. The proposed method is personalized way of recommended knowledge about preferred travel places according to the user's location, place, and weather. Based on 14 attributes from the data collected through the survey, users with similar characteristics are grouped using a non-hierarchical clustering based hybrid recommendation. This makes more accurate recommendation by weighting implicit and explicit data. The users can be recommended a preferred travel destination without spending unnecessary time. The performance evaluation uses accuracy, recall, F-measure. The evaluation result was shown 0.636 accuracy, 0.723 recall, and 0.676 F-measure.

Robust Blind Source Separation to Noisy Environment For Speech Recognition in Car (차량용 음성인식을 위한 주변잡음에 강건한 브라인드 음원분리)

  • Kim, Hyun-Tae;Park, Jang-Sik
    • The Journal of the Korea Contents Association
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    • v.6 no.12
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    • pp.89-95
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    • 2006
  • The performance of blind source separation(BSS) using independent component analysis (ICA) declines significantly in a reverberant environment. A post-processing method proposed in this paper was designed to remove the residual component precisely. The proposed method used modified NLMS(normalized least mean square) filter in frequency domain, to estimate cross-talk path that causes residual cross-talk components. Residual cross-talk components in one channel is correspond to direct components in another channel. Therefore, we can estimate cross-talk path using another channel input signals from adaptive filter. Step size is normalized by input signal power in conventional NLMS filter, but it is normalized by sum of input signal power and error signal power in modified NLMS filter. By using this method, we can prevent misadjustment of filter weights. The estimated residual cross-talk components are subtracted by non-stationary spectral subtraction. The computer simulation results using speech signals show that the proposed method improves the noise reduction ratio(NRR) by approximately 3dB on conventional FDICA.

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An Algorithm of Score Function Generation using Convolution-FFT in Independent Component Analysis (독립성분분석에서 Convolution-FFT을 이용한 효율적인 점수함수의 생성 알고리즘)

  • Kim Woong-Myung;Lee Hyon-Soo
    • The KIPS Transactions:PartB
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    • v.13B no.1 s.104
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    • pp.27-34
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    • 2006
  • In this study, we propose this new algorithm that generates score function in ICA(Independent Component Analysis) using entropy theory. To generate score function, estimation of probability density function about original signals are certainly necessary and density function should be differentiated. Therefore, we used kernel density estimation method in order to derive differential equation of score function by original signal. After changing formula to convolution form to increase speed of density estimation, we used FFT algorithm that can calculate convolution faster. Proposed score function generation method reduces the errors, it is density difference of recovered signals and originals signals. In the result of computer simulation, we estimate density function more similar to original signals compared with Extended Infomax and Fixed Point ICA in blind source separation problem and get improved performance at the SNR(Signal to Noise Ratio) between recovered signals and original signal.

Target Speaker Speech Restoration via Spectral bases Learning (주파수 특성 기저벡터 학습을 통한 특정화자 음성 복원)

  • Park, Sun-Ho;Yoo, Ji-Ho;Choi, Seung-Jin
    • Journal of KIISE:Software and Applications
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    • v.36 no.3
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    • pp.179-186
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    • 2009
  • This paper proposes a target speech extraction which restores speech signal of a target speaker form noisy convolutive mixture of speech and an interference source. We assume that the target speaker is known and his/her utterances are available in the training time. Incorporating the additional information extracted from the training utterances into the separation, we combine convolutive blind source separation(CBSS) and non-negative decomposition techniques, e.g., probabilistic latent variable model. The nonnegative decomposition is used to learn a set of bases from the spectrogram of the training utterances, where the bases represent the spectral information corresponding to the target speaker. Based on the learned spectral bases, our method provides two postprocessing steps for CBSS. Channel selection step finds a desirable output channel from CBSS, which dominantly contains the target speech. Reconstruct step recovers the original spectrogram of the target speech from the selected output channel so that the remained interference source and background noise are suppressed. Experimental results show that our method substantially improves the separation results of CBSS and, as a result, successfully recovers the target speech.

Double Talk Processing using Blind Signal Separation in Acoustic Echo Canceller (음향반향제거기에서 암묵신호분리를 이용한 동시통화처리)

  • Lee, Haengwoo
    • Journal of Korea Society of Digital Industry and Information Management
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    • v.12 no.1
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    • pp.43-50
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    • 2016
  • This paper is on an acoustic echo canceller solving the double-talk problem by using the blind signal separation technology. The acoustic echo canceller may be deteriorated or diverged during the double-talk period. So we use the blind signal separation to detect the double talking by separating the near-end speech signal from the mixed microphone signal. The blind signal separation extracts the near-end signal from dual microphones by the iterative computations using the 2nd order statistical character in the closed reverberation environment. By this method, the acoustic echo canceller operates irrespective of the double-talking. We verified performances of the proposed acoustic echo canceller in the computer simulations. The results show that the acoustic echo canceller with this algorithm detects the double-talk periods well, and then operates stably without diverging of the coefficients after ending the double-talking. The merits are in the simplicity and stability.