• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.10
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• pp.1867-1872
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• 2016

In the area of digital signal processing, it is necessary to improve the quality of the speech signal after removing the background noise which exists in a various real environments. The important thing to consider when removing the background noise acoustically is that to solve the problem, depending on the information of the human auditory mechanism is mainly the amplitude spectrum of the speech signal. This paper introduces the characteristics of a frequency-weighted spectrum shaping filter for the extraction of the amplitude spectrum of the speech signal with the primary purpose. Therefore, this paper proposes an algorithm using the methods of a Wiener filter and the frequency-weighted spectrum shaping filter according to the acoustic model, after extracted the amplitude spectral information in the noisy speech signal. The spectral distortion (SD) output of the proposed algorithm is experimentally improved more than 5.28 dB compared to a conventional method.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.6
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• pp.19-26
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• 2005

Image interpolation in the wavelet domain usually takes advantage of the probabilistic models for the intrascale statistics and the interscale dependency. In this paper, we adopt the linear model for the absolute values of wavelet coefficients of interpolated image across scale to estimate the variances of extrapolated bands. The proposed algorithm uses randomly generated wavelet coefficients based on the estimated parameters for probabilistic model. Random number generation according to the estimated probabilistic model may induce the 'salt and pepper' noise in subbands. We reduce the noise power by Wiener filtering. We observe that the proposed method generates the histogram of the subband coefficients similar to the that of original image. Experimental results show that our method outperforms the previous wavelet-domain interpolation method as well as the conventional bicubic method.

• 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.

• Journal of the Korean Data and Information Science Society
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• v.13 no.1
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• pp.165-175
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• 2002

We consider the problem of estimating the change-point in mean change model with the one change-point. Lombard (1987) suggested change-point estimation based on score functions. Gombay and Huskova (1998) derived a class of change-point estimators with the score function of rank. Various change-point estimators with the log score functions of ranks are suggested and compared via simulation.

• Journal of applied mathematics & informatics
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• v.27 no.1_2
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• pp.175-182
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• 2009

Microscopic imaging system often requires the algorithm to adjust location of camera lenses automatically in machine level. An effort to detect the best focal point is naturally interpreted as a mathematical inverse problem [1]. Following Wiener's point of view [2], we interpret the focus level of images as the quantified factor appeared in image degradation model: g = $f{\ast}H+{\eta}$, a standard mathematical model for understanding signal or image degradation process [3]. In this paper we propose a simple, very fast and robust method to compare the degradation parameters among the multiple images given by introducing outlier analysis of histogram.

• 한국해양학회지
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• v.18 no.2
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• pp.117-124
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• 1983

Pizza's synthetic model, a modified Wiener filter method, was tested to establish the procedure of desirable interpretive wavelet extraction and its application to the marine seismic exploration using several approaches with a real offshore seismic data of the southeast Asia. Noise spectrum acquisition is difficult and any assumptions for it do not produce interpretive wavelets as good as synthetic model result by Piazza (1979). however the resolution could be improved with spiking deconvoultion and following zero phase bandpass filter, and the testing procedure and evaluatttion of results can hopefully contribute in future study and practical evaluation of Piazza's method.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.295-302
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• 2004

On the field of speech processing the treatment of noise is still important problems for speech research. Especially, it has been noticed that the background noise causes remarkable reduction of speech recognition ratio. As the examples of the background noise, there are such various non-stationary noises existing in the real environment as driving noise of automobiles on the road or typing noise of printer. The treatment for these kinds of noises is not so simple as could be eliminated by the former Wiener filter, but needs more skillful techniques. In this paper as one of these trials, we show an algorithm which is a speech enhancement method using a model of mutual inhibition for noise reduction in speech which is contaminated by white noise or background noise mentioned above. It is confirmed that the proposed algorithm is effective for the speech degraded not only by white noise but also by colored noise, judging from the spectral distortion measurement.

• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.124.3-124
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• 2001

The nonlinearity of several chemical processes is usually approximated by a series of the nonlinear static element and the linear subsystem. In the case of the model that the nonlinear static element precedes the linear subsystem, it is called a Hammerstein model. It is a Wiener model when the order is reserved. Here we investigate a relay feedback identification method for Hammerstein type nonlinear processes. The proposed method separates the identification of the nonlinear static function from that of the linear subsystem by using a relay feedback method. From two times activation of nonlinear processes, we identify he whole range of the nonlinear static function as well as the ultimate information of the linear subsystem.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1211-1214
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• 2004

In this paper, we propose to use variable inlet guide vane as a means of active surge control to solve above problems. There is some advantage. For example, since the inlet guide vanes are already present in POSCO, no additional actuation device is required. We can collect all data easily which is related to the test and can simulate new model using our compressors. We can obtain the result that blow-off valve is opened less 5% and can operate air compressor automatically and more efficiently. Through a simulation example, the effectiveness of the proposed schemes is illustrated.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.122-127
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• 2003

In this paper, an adaptive predistortion scheme is proposed to compensate nonlinear distortions caused by high power amplifiers (HPA) in OFDM systems. A complex Wiener-Hammerstein model (WHM) is used to describe input-output relationship of HPA with linear dynamics. The predistorter is directly identified by complex power series model with memory, which is an approximate inverse of the HPA expressed by the WHM. The effectiveness of the proposed adaptive compensation scheme is validated by numerical simulation for 64QAM-OFDM systems.