• Communications for Statistical Applications and Methods
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• v.25 no.1
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• pp.29-42
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• 2018

We combine the integer-valued GARCH(1, 1) model with a generalized regime-switching model to propose a dynamic count time series model. Our model adopts Markov-chains with time-varying dependent transition probabilities to model dynamic count time series called the generalized regime-switching integer-valued GARCH(1, 1) (GRS-INGARCH(1, 1)) models. We derive a recursive formula of the conditional probability of the regime in the Markov-chain given the past information, in terms of transition probabilities of the Markov-chain and the Poisson parameters of the INGARCH(1, 1) process. In addition, we also study the forecasting of the Poisson parameter as well as the cumulative impulse response function of the model, which is a measure for the persistence of volatility. A Monte-Carlo simulation is conducted to see the performances of volatility forecasting and behaviors of cumulative impulse response coefficients as well as conditional maximum likelihood estimation; consequently, a real data application is given.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9A
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• pp.755-761
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• 2005

In this paper, we proposed a channel estimation method by impulse signal train in OFDM. In order to estimate the channel response, 4 impulse signals are generated and transmitted during one OFDM (Orthogonal Frequency Division Multiplexing) symbol. The intervals between the impulse signals are all equal in time domain. At the receiver, the impulse response signals are summed and averaged. And then, the averaged impulse response signal is zero padded and fast Fourier transformed to obtain the channel estimation. The BER performance of the proposed method is compared with those of conventional estimation method using the long training sequence in fast fading environments. The simulation results show that the proposed method improves by 3 dB in terms of Eb/No, compared with the conventional method.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.1037-1041
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• 2007

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral psycho-acoustic filters considering masking effects are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for realtime implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.2
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• pp.231-237
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• 2008

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral real-time masking blocks are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for real-time implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2003.06a
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• pp.1131-1134
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• 2003

Unknown parameters of a nonlinear system were estimated using a signal compression method. The estimated parameters were natural frequency and tile damping coefficient. This study applied a algorithm using tile comparison of the cross-correlation coefficient between the impulse response from a model and it from the signal compression method. The impulse through linear element included in a nonlinear system could be obtained by the signal compression method. The unknown parameters of the linear element could be estimated by comparing the Bode plots of system's impulse response with them of model's response. In this study, a LSCM(LabVIEW-Signal-Compression-Method) was developed to identify a nonlinear system. The LSCM consisted of National Instrument's (NI) Data Acquisition (DAQ) Board (Model PCI-1200), a monitoring program using LabVIEW software package, DAQ Signal Accessory Board, and 2nd-order electric circuits. The designed electric circuits consisted of resistors, inductors and capacitors. To evaluate the performance of the LSCM, the response from model with known parameters is compared with the response from the real system using the monitoring program. The results from simulation of experiment showed that the developed LSCM provided a reliable estimation performance.

• Structural Engineering and Mechanics
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• v.42 no.3
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• pp.363-381
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• 2012

This study is aimed at providing an efficient analytical model to obtain pressure- impulse diagram of one-way reinforced concrete slabs subjected to different shapes of air blast loading using single degree of freedom method (SDOF). A tri-linear elastic perfectly plastic SDOF model has been used to obtain the pressure-impulse diagram to correlate the blast pressure and the corresponding concrete flexural damage. In order to capture the response history for the slab, a new approximately SDOF method based on the conventional SDOF method is proposed and validated using published test data. The influences of pulse loading shape on the pressure-impulse diagram are studied. Based on the results, a pressure-impulse diagram generation method using SDOF and an analytical equation for the pressure-impulse diagram is proposed to different damage levels and different blast loading shapes.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.3
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• pp.102-107
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• 2006

In this paper, we propose a method of a sound synthesis of Korean plucked string instrument, gayageum, by physical modeling which use impulse responses of body and Anjok. Gayageum consists of three kinds of systems: string, body, and Anjok. These are a serial combination of linear time invariant systems. String can be modeled by digital delay line. Body and Anjok can be estimated by their impulse responses. We found three resonance frequencies in the body impulse response, and implemented resonator as body. Anjok was implemented as high pass filter in fundamental frequency band of gayageum. RMSEs of synthesized sounds are distributed from 0.01 to 0.03. It was difficult to distinguish the resulting synthesized sounds from the originals sound by ear.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.8
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• pp.861-866
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• 2014

This study evaluates the modeling accuracy of the existing three phase shift models on which the time domain oscillator phase noise theories are based. For the evaluation, this study investigates how accurately the three models can model the phase shift impulse train response of a simple planar oscillator. Evaluation result reveals that Kaertner model most accurately reflects the oscillator's phase shift impulse train responses for five different impulse train inputs, whereas PP model exhibited the worst performance in modeling the phase shift impulse train responses.

• Geotechnical Engineering
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• v.13 no.1
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• pp.111-122
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• 1997

The calculation of phase velocities in Spectral-Analysis -of-Surface -Waves (SASW) meas urements requires unwrapping phase angles. In case of layered systems with strong stiffness contrast like a pavement system, conventional phase unwrapping algorithm to add in teger multiples of 2n to the principal value of a phase angle may lead to wrong phase volocities. This is because there is difficulty in counting the number of jumps in the phase spectrum especially at the receiver spacing where the measurements are in the transition Bone of defferent modes. A new phase interpretation scheme, called "Impulse Response Fil traction ( IRF) Technique," is proposed, which is based on the separation of wave groups by the filtration of the impulse response determinded between two receivers. The separation of a wave group is based on the impulse response filtered by using information from Gabor spectrogram, which visualizes the propagation of wave groups at the frequency -time space. The filtered impulse response leads to clear interpretation of phase spectrum, which eliminates difficulty in counting number of jumps in the phase spectrum. Verification of the IRF technique was performed by theoretical simulation of the SASW measurement on a pavement system which complicates wave propagation.opagation.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.4
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• pp.591-598
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• 2021

The digital phase-locked loops(DPLL) is a circuit used for phase synchronization and has been generally used in various fields such as communication and circuit fields. State estimators are used to design digital phase-locked loops, and infinite impulse response state estimators such as the well-known Kalman filter have been used. In general, the performance of the infinite impulse response state estimator-based digital phase-locked loop is excellent, but a sudden performance degradation may occur in unexpected situations such as inaccuracy of initial value, model error, and disturbance. In this paper, we propose a minimum variance finite impulse response filter with optimal horizon for designing a new digital phase-locked loop. A numerical method is introduced to obtain the measured value interval length, which is an important parameter of the proposed finite impulse response filter, and to obtain a gain, the covariance matrix of the error is set as a cost function, and a linear matrix inequality is used to minimize it. In order to verify the superiority and robustness of the proposed digital phase-locked loop, a simulation was performed for comparison and analysis with the existing method in a situation where noise information was inaccurate.