• Title/Summary/Keyword: Blind Signal Separation

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Post-Processing with Frequency Domain Wiener Filter for Blind Source Separation

  • Park, Keun-Soo;Park, Jang-Sik;Kim, Hyun-Tae;Son, Kyung-Sik
    • The Journal of the Acoustical Society of Korea
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    • 제25권2E호
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    • pp.36-42
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    • 2006
  • In this paper, a novel post processing using Wiener filtering technique is proposed to p rm further interference reduction in FDICA. Using the proposed method, the target signal components are remained with little attenuation while the interference components are drastically suppressed. The results of experiments show that the proposed method achieves a reduction of the residual crosstalk. Compared to the NLMS method, the proposed method has slightly better separation performance in SIR, and even requires much less computational complexity.

블라인드 동채널 신호 분리를 위한 순차적인 Joint Maximum Likelihood 알고리듬 (A Sequential Joint Maximum Likelihood Algorithm for Blind Co-Channel Signal Separation)

  • Inseon Jang;Park, Seungjin
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 2001년도 제14회 신호처리 합동 학술대회 논문집
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    • pp.85-88
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    • 2001
  • In this paper we consider a problem of blind co-channel signal separation, the goal of which is to estimate multiple co-channel digitally modulated signals using an antenna array. We employ the joint maximum likelihood estimation and present a sequential algorithm, which is referred to as sequential joint maximum likelihood (SJML) algorithm. It separates multiple co-channel signal on-line and converges fast in overdetermined noisy communication environment. And the computational complexity of SJML for M-QAM (M=8, 16, 64,...) signals is less expensive compared to the SLSP. Useful behavior of this algorithm are confirmed by simulations.

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멀티채널 비음수 행렬분해와 정규화된 공간 공분산 행렬을 이용한 미결정 블라인드 소스 분리 (Underdetermined blind source separation using normalized spatial covariance matrix and multichannel nonnegative matrix factorization)

  • 오순묵;김정한
    • 한국음향학회지
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    • 제39권2호
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    • pp.120-130
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    • 2020
  • 본 논문은 블라인드 소스 분리 분야에서 널리 사용되는 멀티채널 비음수 행렬 분해 기법의 단점을 개선하여 미결정 복잡한 혼합 환경에서 문제를 해결한다. 공간 공분산 행렬에 기반을 둔 기존의 연구들에서, 단일 채널의 파워게인 및 상관관계와 같은 값으로 구성된 행렬의 각 요소는 높은 분산으로 인해 분리된 소스의 품질을 저하시키는 경향이 있다. 이 논문에서는 추정된 소스들을 효과적으로 클러스터링하기 위해 레벨 및 주파수 정규화를 수행한다. 따라서 새로운 공간 공분산 행렬 및 효과적인 클러스터 쌍별 거리함수를 제안한다. 본 논문에서는 제안된 행렬을 공간 모델의 초기화에 활용하여 공간 모델의 향상된 추정과 이를 바탕으로 상향식 접근법에서의 계층적 응집 클러스터링에 활용함으로써 분리된 음원의 품질을 향상시켰다. 제안된 알고리즘은 'Signal Separation Evaluation Campaign 2008 development dataset'을 활용하여 실험을 하였다. 그 결과 객관적인 소스 분리 품질 검증 도구인 'Blind Source Separation Eval toolbox'를 활용하여 대부분의 성능향상지표에서의 향상을 확인하였으며, 특히 대표적인 수치인 SDR의 1 dB ~ 3.5 dB 정도의 성능우위를 검증하였다.

지연혼합에서의 초기 값으로 고유벡터를 이용하는 암묵신호분리 (Blind Signal Separation Using Eigenvectors as Initial Weights in Delayed Mixtures)

  • 박장식;손경식;박근수
    • 한국음향학회지
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    • 제25권1호
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    • pp.14-20
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    • 2006
  • 본 논문에서는 지연혼합에서의 암묵신호분리를 위해 분리행렬의 초기 값을 설정하는 방법을 제안한다. 혼합신호의 상호상관행렬에 대한 고유분리를 분석한 후, 고유벡터의 지연정보를 이용하여 초기 값으로 설정한다. 제안하는 방법을 기존의 주파수영역 독립성분분석 (FDICA: Frequency domain independent component analysis)에 초기 값으로 설정하여 분리 성능을 향상시킨다. 컴퓨터 시뮬레이션을 통해 제안하는 방법이 신호대간섭비 (SIR: Signal to Interference Ratio)가 우수하고 학습곡선의 수렴속도가 개선됨을 보인다.

Adaptive Signal Separation with Maximum Likelihood

  • Zhao, Yongjian;Jiang, Bin
    • Journal of Information Processing Systems
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    • 제16권1호
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    • pp.145-154
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    • 2020
  • Maximum likelihood (ML) is the best estimator asymptotically as the number of training samples approaches infinity. This paper deduces an adaptive algorithm for blind signal processing problem based on gradient optimization criterion. A parametric density model is introduced through a parameterized generalized distribution family in ML framework. After specifying a limited number of parameters, the density of specific original signal can be approximated automatically by the constructed density function. Consequently, signal separation can be conducted without any prior information about the probability density of the desired original signal. Simulations on classical biomedical signals confirm the performance of the deduced technique.

QPSK 신호 입력시스템에서의 유한 알파벹 기반 ML 블라인드 신호 추정 비교 (A Consideration on ML Blind Signal Estimation based on Finite-Alphabet Characteristic in QPSK Modulation)

  • 권순만;김석주;이종무;김춘경;천종민
    • 대한전기학회:학술대회논문집
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    • 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
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    • pp.685-688
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    • 2003
  • In this paper, a performance comparison between two blind signal estimation algorithms in a LTI channel is considered. The two algorithms, Iterative Least-Squares with Projection (ILSP) and a modified ILSP, are based on the finite-alphabet property of input symbols. This case typically arises in a multiple access system with a sensor array antenna at the receiving end. We start with the formulation of a maximum-likelihood (ML) estimation problem under an additive white Gaussian noise assumption. A blind ML estimator is derived with its iterative algorithm for calculation. Then we narrow down the consideration of this problem to QPSK case so that a modified algorithm is proposed for $\pi$/4-QPSK case. The modified version is compared with the original ILSP algorithm in terms of the rate of the convergence to global minima. A computer simulation shows that the modified algorithm gives a better performance. This result implies that the performance of the blind separation algorithms may be greatly improved by adopting a smart coding scheme with rich structure.

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시간-주파수 마스킹과 고차 신호 통계를 이용한 음향 반향신호 제거 (Acoustic Echo Cancellation using Time-Frequency Masking and Higher-order Statistics)

  • 김경재;남상원
    • 전기학회논문지
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    • 제56권3호
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    • pp.629-631
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    • 2007
  • In hands-free full-duplex communication systems, acoustic signals picked up by the microphones can be mixed with echo signals as well as noises, which may result in poor performance of the corresponding communication system. Also, the system performance may decrease further if the reverberation occurs since it is harder to estimate the impulse response of the demixing system. For blind source separation (BSS) in such cases, a time-frequency masking approach can be employed to separate undesired echo signals and noises, but, permutation ambiguities also should be solved for the echo cancellation. In this paper, we propose a new acoustic echo cancellation (AEC) approach utilizing the time-frequency masking and higher-order statistics, whereby a desired signal selection, based on coherence and third-order statistics (i.e., kurtosis), is introduced along with output signal normalization. Simulation results demonstrate that the proposed approach yields better echo and noise cancellation performances than the conventional AEC approaches.

Overlapped Subband-Based Independent Vector Analysis

  • Jang, Gil-Jin;Lee, Te-Won
    • The Journal of the Acoustical Society of Korea
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    • 제27권1E호
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    • pp.30-34
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    • 2008
  • An improvement to the existing blind signal separation (BSS) method has been made in this paper. The proposed method models the inherent signal dependency observed in acoustic object to separate the real-world convolutive sound mixtures. The frequency domain approach requires solving the well known permutation problem, and the problem had been successfully solved by a vector representation of the sources whose multidimensional joint densities have a certain amount of dependency expressed by non-spherical distributions. Especially for speech signals, we observe strong dependencies across neighboring frequency bins and the decrease of those dependencies as the bins become far apart. The non-spherical joint density model proposed in this paper reflects this property of real-world speech signals. Experimental results show the improved performances over the spherical joint density representations.

Multi-channel Speech Enhancement Using Blind Source Separation and Cross-channel Wiener Filtering

  • Jang, Gil-Jin;Choi, Chang-Kyu;Lee, Yong-Beom;Kim, Jeong-Su;Kim, Sang-Ryong
    • The Journal of the Acoustical Society of Korea
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    • 제23권2E호
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    • pp.56-67
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    • 2004
  • Despite abundant research outcomes of blind source separation (BSS) in many types of simulated environments, their performances are still not satisfactory to be applied to the real environments. The major obstacle may seem the finite filter length of the assumed mixing model and the nonlinear sensor noises. This paper presents a two-step speech enhancement method with multiple microphone inputs. The first step performs a frequency-domain BSS algorithm to produce multiple outputs without any prior knowledge of the mixed source signals. The second step further removes the remaining cross-channel interference by a spectral cancellation approach using a probabilistic source absence/presence detection technique. The desired primary source is detected every frame of the signal, and the secondary source is estimated in the power spectral domain using the other BSS output as a reference interfering source. Then the estimated secondary source is subtracted to reduce the cross-channel interference. Our experimental results show good separation enhancement performances on the real recordings of speech and music signals compared to the conventional BSS methods.

PCA 기반 오디오 신호 분리 알고리즘 구현 (Audio signal separation Algorithm Implementation based PCA)

  • 전재현;조두리;정제창
    • 한국방송∙미디어공학회:학술대회논문집
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    • 한국방송공학회 2013년도 추계학술대회
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    • pp.151-154
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    • 2013
  • 다수의 음원이 특정한 공간에 산재하고 있을 때, 그 중 특정 음원에 주목하면 다른 음원과 분리되어 특정 음원만 들리는 현상을 칵테일파티 현상이라고 한다. 심리적인 이 현상에 영감을 받아 음원을 분리하는 알고리즘이 만들어졌다. 이런 음원 분리방법을 Blind Source Separation(BSS) 이라고 하는데, 여러 신호가 섞이는 과정을 모르는 상태에서 음원을 분리한다는 뜻에서 Blind Source Separation 이라고 한다. BSS에 사용되는 알고리즘으로 주로 PCA, ICA이 있다. PCA는 2차원의 경우를, ICA는 그 이상의 고차원의 통계적 특성을 이용한다. 이에 본 논문은 PCA를 이용하여 두 음원을 분리하는 알고리즘을 구현하는데 역점을 두었다. PCA는 주로 음원보다는 이미지 신호 처리에 초점이 맞추어져 있지만, 음원 분리에 있어서도 충분한 성능을 보여주므로, ICA를 이용한 음원 분리 알고리즘과의 비교를 통하여 장, 단점을 알아보고 추후 PCA의 응용 가능성을 알아보았다.

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