• Journal of Korea Water Resources Association
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• v.32 no.4
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• pp.437-445
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• 1999

The present paper describes the problem of hydrologic response estimation using non-parametric ridge regression method. The method adapted in this work is based on the minimization of the $C_L$ statistics, which is an estimate of the mean square prediction error. For this method, effects of using both the identity matrix and the Laplacian matrix were considered. In addition, we evaluated methods for estimating the error variance of the impulse response. As a result of analyzing synthetic and real data, a good estimation was made when the Laplacian matrix for the weighting matrix and the bias corrected estimate for the error variance were used. The method and procedure presented in present paper will play a robust and effective role on separating hydrologic response.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.10a
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• pp.351-354
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• 2007

In recent years, digital processing techniques have been applied diversity of fields. Typical signal processing techniques are speech processing and image processing. And filters for the signal processing can be divided in FIR (finite impulse response) filter and IIR (infinite impulse response) filter. Compared with IIR filter, the FIR Filter has a defect of high-degree, but has a merit of stability and uses simply. Futhermore, FIR filter also has linear phase response characteristics, it is using in fields regarding wave information importantly. To FIR Filter design, the main issue is to remove the Gibbs phenomenon. Therefore, in this paper I was proposed a method using FIR digital filter applied a modified window function and the method was compared with conventional methods.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.1398-1403
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• 2003

In this paper, a simple method to design an MIMO sound field control system was proposed. The control system was designed to achieve 1) noise attenuation and 2) sound equalization by utilizing feedback and feedforward controllers. The method was based on partial model matching on frequency domain which only required measured frequency response data, or impulse response data in order to tune parameters of the controller. The proposed method was applied to a normal office room and results of experiment showed effectiveness of the proposed method.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.1191-1196
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• 2007

This research presents a laboratory study about an annoyance of impulsive sound caused by construction site(breaker and blasting). The sources are sampled from outdoor noise and their levels range from 40 to 75 dB at the interval of 5dB. The noise unit is based on A-weighted sound exposure level (ASEL; $L_{AE}$). To make equal ASEL of outdoor noise, finite impulse response (FIR) filter is applied to the originally sampled source to include the effect of distance attenuation. The evaluation method of jury test adopted a Semantic Difference Method (SDM). In the result of the Jury test for impulsive noise, the annoyance response of blasting noise was higher than that of breaker noise.

• 전기의세계
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• v.19 no.6
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• pp.1-7
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• 1970

In this paper, to determine process dynamic characteristics, the correlation method for measuring the impulse response of process using a pseudo-random binary signal as the test signal instead of white noise was studied. The error caused by using the signal of Mesquence signal generator which was built up by the authors was analysed. Experments were performed on the 1st and 2nd order lag systems and the results were in good coincidence with theoretical values. It is expected that applying these results, it may be possible to develop a continuous measuring method adaptable to modern control systems.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.2298-2303
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• 2003

In this paper, very simple methods based on the exponential analysis are presented by which transfer function models for processes can easily be obtained. These methods employ step responses or impulse responses of the processes. These can also give a more precise transfer function model compared to the well-known graphical methods. Transfer functions are determined based on Prony method, which is one of the oldest and the most representative methods in the exponential analysis. Here, the method is reformed and applied to obtain the so-called low-order transfer function with pure time delay from the data of the step response. The effectiveness of the proposed method is examined through several numerical examples and experiments of the 2-tank level control process.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.437-445
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• 2004

In this paper. we describe a novel method for transforming audio contents according to user's preference on sound field. Sound field effect technologies. which transform or simulate acoustic environments as user's preference, are very important for enlarging the reality of acoustic scene. However huge amount of computational power is required to process sound field effect in real time. so it is hard to implement this functionality at the portable audio devices such as MP3 player. In this paper, we propose an efficient method for providing sound field effect to audio contents independent of terminal's computational power through processing this functionality at the server using user's sound field preference, which is transfered from terminal side. To describe sound field preference, user can use perceptual acoustic parameters as well as the URI address of room impulse response signal. In addition, a novel fast convolution method is presented to implement a sound field effect engine as a result of convoluting with a room impulse response signal at the realtime application. and verified to be applicable to real-time applications through experiments. To verify the evidence of benefit of proposed method we performed two subjective listening tests about sound field descrimitive ability and preference on sound field processed sounds. The results showed that the proposed sound field preference can be applicable to the public.

• Journal of the Korean Institute of Intelligent Systems
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• v.11 no.6
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• pp.563-567
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• 2001

We generalize the widebane P0-weyl symbol (P0WS) and the widebane spreading function (WSF) using the generalized warping function . The new generalized P0WS and WSF are useful for analyzing system and communication channels producing generalized time shifts. We also investigated the relationship between the affine Wey1 symbol(AWS) and the P0WS. By using specific warping functions, we derive new P0WS and WSF as analysis tools for systems and communication channels with non-linear group delary characteristics. The new P0WS preserves specific types of changes imposed on random processes. The new WSF provides a new interpretation of output of system and communication channel as weighted superpositions of non-linear time shifts on the input. It is compared to the conventional method obtaining output of system and communication channel as a convention integration of the input with the impulse response of the system and the communication channel. The convolution integration can be interpreted as weighted superpositions of liner time shifts on the input where the weight is the impulse response of the system and the communication channel. Application examples in analysis and detection demonstrate the advantages of our new results.

• Smart Structures and Systems
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• v.17 no.5
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• pp.753-771
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• 2016

This paper addresses the free and transient responses of a SDOF linear complex stiffness system by making use of the Hilbert transform and the convolution integral. Because the second-order differential equation of motion having the complex stiffness give rise to the conjugate complex eigen values, its time-domain analysis using the standard time integration scheme suffers from the numerical instability and divergence. In order to overcome this problem, the transient response of the linear complex stiffness system is obtained by the convolution integral of a green function which corresponds to the unit-impulse free vibration response of the complex system. The damped free vibration of the complex system is theoretically derived by making use of the state-space formulation and the Hilbert transform. The convolution integral is implemented by piecewise-linearly interpolating the external force and by superimposing the transient responses of discretized piecewise impulse forces. The numerical experiments are carried out to verify the proposed time-domain analysis method, and the correlation between the real and imaginary parts in the free and transient responses is also investigated.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.310-312
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• 2011

In complex noise environment, digital filter is being used to obtain, transport and storage original voice or image signal. Digital filter can be largely separated FIR(Finite duration impulse response) filter and IIR(Infinite duration impulse response) filter. Among FIR filter, window function has characteristic of linear phase and as can be easily set pass-band frequency, cutoff frequency and so on. In this paper, We compared with established method using transient characteristic and peak side-lobe in order to check filter characteristics after we designed the existing variants of the window function.