• The Journal of the Acoustical Society of Korea
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• v.25 no.2E
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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.

• Proceedings of the IEEK Conference
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• 2001.09a
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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.

• The Journal of the Acoustical Society of Korea
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• v.39 no.2
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• pp.120-130
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• 2020

This paper solves the problem in underdetermined convolutive mixture by improving the disadvantages of the multichannel nonnegative matrix factorization technique widely used in blind source separation. In conventional researches based on Spatial Covariance Matrix (SCM), each element composed of values such as power gain of single channel and correlation tends to degrade the quality of the separated sources due to high variance. In this paper, level and frequency normalization is performed to effectively cluster the estimated sources. Therefore, we propose a novel SCM and an effective distance function for cluster pairs. In this paper, the proposed SCM is used for the initialization of the spatial model and used for hierarchical agglomerative clustering in the bottom-up approach. The proposed algorithm was experimented using the 'Signal Separation Evaluation Campaign 2008 development dataset'. As a result, the improvement in most of the performance indicators was confirmed by utilizing the 'Blind Source Separation Eval toolbox', an objective source separation quality verification tool, and especially the performance superiority of the typical SDR of 1 dB to 3.5 dB was verified.

• The Journal of the Acoustical Society of Korea
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• v.25 no.1
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• pp.14-20
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• 2006

In this paper. a novel technique to set up the initial weights in BSS of delayed mixtures is proposed. After analyzing Eigendecomposition for the correlation matrix of mixing data. the initial weights are set from the Eigenvectors ith delay information. The Proposed setting of initial weighting method for conventional FDICA technique improved the separation Performance. The computer simulation shows that the Proposed method achieves the improved SIR and faster convergence speed of learning curve.

• Journal of Information Processing Systems
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• v.16 no.1
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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.

• Proceedings of the KIEE Conference
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• 2003.11c
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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.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.56 no.3
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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.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1E
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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.

• The Journal of the Acoustical Society of Korea
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• v.23 no.2E
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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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2013.11a
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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.