• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.8
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• pp.2051-2056
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• 2014

This paper investigated the separation of magnetic signal from a ferro-magnetic object. The magnetic signals were ILM(induced longitudinal magnetization) and IVM(induced vertical magnetization), which were induced by earth magnetic field and PLM(permanent longitudinal magnetization) and PVM(permanent vertical magnetization), which were due to a permanent magnetization of the object, respectively. Magnetic signal separation was based on the fact that magnetization vector could be analyzed according to longitudinal and vertical directions. Also the influence of non-uniform magnetic field from a rectangular coil on the separation was examined. A military vessel with a size close to rectangular coil has more errors on the magnetic signal separation.

• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.416-418
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• 2006

In this paper, a new acoustic echo cancellation approach based on the DUET algorithm and scaling factor estimation is proposed to solve the scaling ambiguity in case of blind separation based acoustic echo cancellation in a noisy environment. In hands-free full-duplex communication system. acoustic noises picked up by the microphone are mixed with echo signal. For this reason, the echo cancellation system may provide poor performance. For that purpose, a degenerate unmixing estimation technique, adjusted in the time-frequency domain, is employed to separate undesired echo signals and noises. Also, since scaling and permutation ambiguities have not been solved in the blind source separation algorithm, kurtosis for the desired signal selection and a scaling factor estimation algorithm are utilized in this rarer for the separation of an echo signal. Simulation results demonstrate that the proposed approach yields better echo cancellation and noise reduction performances, compared with conventional methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.1
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• pp.67-79
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• 1996

This paper presents a composite luminance signal processing system for NTSC, PAL and SECAM standards. Eaxh of the three standards employs its own specifications of subcarmier bandwidth and luminance signal waveform. The proposed system, compatible to the specifications of the three standard and B/W TV, implements variable freqneucy characteristics by controlling filter coefficients. The major features of the system are a luminance/chroma separation unit and an aperture compensation unit. The luminance/chroma separation unit employes a notch filter selection a trap freqneyc to atenuate unwanted color signals in luminance signal bands. The aperture compensation unit comprises two subunits, to provide clear color definition for each of the three standards: a primary compensation circuit and a variable compensation circuits. The proposed system yields a 40 dB gain from the chroma/luminance separation and a 10 dB gain from the aperture compensation unit.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.41 no.9
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• pp.1051-1059
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• 1992

In this paper, two general methods using time-series analysis in the functional separation of the myoelectric signal of human arm movements are developed. Autocorrelation, covariance method and sequential least squares algorithm were used to determine the model parameters and the order of signal model to describe six arm movement patterns` the forearm flexion and extension, the wrist pronation and supination, rotation-in and rotation-out. The confidence interval to classify the functions of arm movement was defined by the mean and standard deviation of total squared error. With the error signals of autoregressive(AR) model, the result showed that the highest success rate was obtained in the case of 4th order, and success rate was decreased with increase of order. Autocorrelation was the method of choice for better success rate. This technique might be applied to biomedical and rehabilitation engineering.

• MALSORI
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• no.48
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• pp.81-91
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• 2003

In this paper, we propose a speech recognition method based on independent component analysis (ICA) and spectral enhancement techniques. While ICA tris to separate speech signal from noisy speech using multiple channels, some noise remains by its algorithmic limitations. Spectral enhancement techniques can compensate for lack of ICA's signal separation ability. From the speech recognition experiments with instantaneous and convolved mixing environments, we show that the proposed approach gives much improved recognition accuracies than conventional methods.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.4 no.2
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• pp.77-82
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• 2011

In this paper, we investigate source separation algorithms for stereo audio mixed using amplitude panning method. This source separation algorithms can be used in various applications such as up-mixing, speech enhancement, and high quality sound source separation. The methods in this paper estimate the panning angles of individual signals using the principal component analysis being applied in time-frequency tiles of the input signal and independently extract each signal through directional filtering. Performances of the methods were evaluated through computer simulations.

• Transactions on Control, Automation and Systems Engineering
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• v.2 no.1
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• pp.7-12
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• 2000

A new speech enhancement algorithm for speech corrupted by slowly varying additive colored noise is suggested based on a state-space signal model. Due to the FIR structure and the unimportance of long-term past information, the receding horizon (RH) FIR filter known to be a best linear unbiased estimation (BLUE) filter is utilized in order to obtain noise-suppressed speech signal. As a special case of the colored noise problem, the suggested approach is generalized to perform the single blind signal separation of two speech signals. It is shown that the exact speech signal is obtained when an incoming speech signal is noise-free.

• Proceedings of the KIEE Conference
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• 2001.07d
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• pp.2622-2624
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• 2001

In this thesis, speaker separation is employed when speech from two talkers has been summed into one signal and it is desirable to recover one or both of the speech signals from the composite signal. This paper proposed the method that separated the summed speeches and proved the similarity between the signals by the cross correlation between the signals for exact between original signal and separated signal. This paper uses frequency sampling method based on sinusoidal model to separate the composite signal with vocalic speech and vocalic speech and noise masking method based on psycoacoustics model to separate the composite signal with vocalic speech and nonvocalic speech.

• Journal of Communications and Networks
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• v.14 no.4
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• pp.410-417
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• 2012

One of the premier mechanisms used in extracting unobserved signals from observed mixtures in signal processing is employing a blind source separation (BSS) algorithm. Orthogonal frequency division multiplexing (OFDM) techniques are playing a prominent role in the sphere of multicarrier communication. A set of remedial solutions taken to mitigate deteriorative effects caused within the air interface of OFDM transmission with aid of BSS schemes is presented. Four energy functions are used in deriving the filter coefficients. Energy criterion functions to be optimized and the performance is justified. These functions together with iterative fixed point rule for receive signal are used in determining the filter coefficients. Time correlation properties of the channel are taken advantage for BSS. It is tried to remove colored noise and jamming components from themixture at the receiver. Themethod is tested in a slow fading channel with a receiver containing equal gain combining to treat the channel state information values. The importance is that, these are quite low computational complexity mechanisms.