• Journal of the Korean Institute of Intelligent Systems
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• v.22 no.2
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• pp.181-186
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• 2012

Recently there has been many active researches on searching similar sequences from data generated with the passage of time. Those data are classified as time series data or sequence data and have different semantics from scalar data of traditional databases. In this paper similar sequence search retrieves sequences that have a similar trend of value changes. At first we have transformed the original sequences by applying DFT. The converted data are more suitable for trend analysis and they require less number of attributes for sequence comparisons. In addition we have developed a region-based query and we applied bitmap indexes which could show better performance in data warehouse. We have built bitmap indexes with varying number of attributes and we have found the least cost query plans for efficient similar sequence searches.

• Journal of Information Processing Systems
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• v.13 no.5
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• pp.1135-1148
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• 2017

Speech recognition is one of the fascinating fields in the area of Computer science. Accuracy of speech recognition system may reduce due to the presence of noise present in speech signal. Therefore noise removal is an essential step in Automatic Speech Recognition (ASR) system and this paper proposes a new technique called combined thresholding for noise removal. Feature extraction is process of converting acoustic signal into most valuable set of parameters. This paper also concentrates on improving Mel Frequency Cepstral Coefficients (MFCC) features by introducing Discrete Wavelet Packet Transform (DWPT) in the place of Discrete Fourier Transformation (DFT) block to provide an efficient signal analysis. The feature vector is varied in size, for choosing the correct length of feature vector Self Organizing Map (SOM) is used. As a single classifier does not provide enough accuracy, so this research proposes an Ensemble Support Vector Machine (ESVM) classifier where the fixed length feature vector from SOM is given as input, termed as ESVM_SOM. The experimental results showed that the proposed methods provide better results than the existing methods.

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.4
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• pp.295-303
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• 2008

In general, transform domain adaptive filters show faster convergence speed than the time domain adaptive filters, but the amount of calculation increases dramatically as the filter order increases. This problem can be solved by making use of the subband structure in transform domain adaptive filters. In this paper, to increase the convergence speed on the generalized subband decomposition FIR adaptive filters, a structure of the adaptive filter with subfilter of dyadic sparsity factor in wavelet transform domain is designed. And, in this adaptive filter, the equivalent input in transform domain is derived and, by using the input, the convergence properties for the LMS algorithm is analyzed and evaluated. By using this sub band adaptive filter, the inverse system modeling and the periodic noise canceller were designed, and, by computer simulation, the convergence speeds of the systems on LMS algorithm were compared with that of the subband adaptive filter using DFT(discrete Fourier transform).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1A
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• pp.71-79
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• 2010

The algorithm of multiple channel signal processing requires the flexibility of variable frequency band, efficient allocation of transmission power, and flexible frequency band reallocation to satisfy various service types which requires different transmission rates and frequency band. This paper proposes an improved multiple channel signal processing for converting the frequency band of multiple carrier signals efficiently using a window function and DFT in the time domain. In contrast to the previous algorithm of multiple-channel signal processing performing band-pass signal processing in the frequency domain, the proposed algorithm is a method of block signal processing using a window function in the time domain. In addition, the complexity of proposed algorithm of the window function is lower than that of the previous algorithm performing signal processing in the frequency domain, and it performs the frequency band transform efficiently. The computer simulation result shows that the perfect reconstruction of output signal and the flexible frequency band reallocation is performed efficiently by the proposed algorithm.

• Journal of radiological science and technology
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• v.32 no.1
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• pp.17-24
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• 2009

Noise evaluation for an image has been performed by root mean square (RMS) granularity, autocorrelation function (ACF), and Wiener spectrum. RMS granularity stands for standard deviation of photon data and ACF is acquired by integration of 1 D function of distance variation. Fourier transform of ACF results in noise power spectrum which is called Wiener spectrum in image quality evaluation. Wiener spectrum represents noise itself. In addition, along with MTF, it is an important factor to produce detective quantum efficiency (DQE). The proposed evaluation method using Wiener spectrum is expected to contribute to educate the concept of Wiener spectrum in educational organizations, choose the appropriate imaging detectors for clinical applications, and maintain image quality in digital imaging systems.

• ETRI Journal
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• v.40 no.4
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• pp.537-545
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• 2018

In this paper, we present a comprehensive analytical study of the symbol error rate (SER) of single-carrier frequency-division multiple access (SC-FDMA) with zero-forcing frequency domain equalization (ZF-FDE) over a Rayleigh fading channel. SC-FDMA is considered as a potential waveform candidate for fifth-generation (5G) radio access networks (RANs). First, the $N_C$ fold convolution of the noise distribution of an orthogonal frequency-division multiplexing (OFDM) system is computed for each value of the signal-to-noise ratio (SNR) in order to determine the noise distribution of the SC-FDMA system. $N_C$ is the number of subcarriers assigned to a user or the size of the discrete Fourier transform (DFT) precoding. Here, we present a simple alternative method of calculating the SER by simplifying the $N_C$ fold convolution using time and amplitude scaling properties. The effects of the $N_C$ fold convolution and SNR over the computation of the SER of the SC-FDMA system has been separated out. As a result, the proposed approach only requires the computation of the $N_C$ fold convolution once, and it is used for different values of SNR to calculate the SER of SC-FDMA systems.

• Proceedings of the KIEE Conference
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• 2005.11b
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• pp.283-285
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• 2005

This paper presents a new two-terminal numerical algorithm for fault location estimation and for faults recognition using the synchronized phasor in time-domain. The proposed algorithm is also based on the synchronized voltage and current phasor measured from the PMUs(Phasor Measurement Units) installed at both ends of the transmission lines. Also the arc voltage wave shape is modeled numerically on the basis of a great number of arc voltage records obtained by transient recorder. From the calculated arc voltage amplitude it can make a decision whether the fault is permanent or transient. In this paper the algorithm is given and estimated using DFT(Discrete Fourier Transform) and the LES(Least Error Squares Method). The algorithm uses a very short data window and enables fast fault detection and classification for real-time transmission line protection. To test the validity of the proposed algorithm, the Electro-Magnetic Transient Program(EMTP/ATP) and MATLAB is used.

• Journal of computational fluids engineering
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• v.18 no.2
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• pp.78-84
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• 2013

In this paper, numerical simulations of both low- and high-frequency buzz phenomena at the throttle ratios (T.R.) in Nagashima's experiment are performed. The dominant frequencies of the low-and high-frequency buzz in the experiment are about 109 Hz with T.R.=0.97 and 376 Hz with T.R.=0.55, respectively. An axisymmetric solver with the S-A turbulence model is used for the simulations, and DFT(Discrete Fourier Transform) on pressure histories is conducted for the buzz frequency analysis. In the present simulations, the free-stream Mach number and the Reynolds number based on the inlet diameter are 2 and $10^7$, respectively. Both the low- and high-frequency buzz phenomena are accomplished without the changes in the grid topology. The dominant frequency of the simulation is about 125 Hz with T.R.=0.97, while it is 399 Hz with T.R.=0.55.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.14 no.5
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• pp.2275-2293
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• 2020

This paper focuses on the simplified receiver design for multi-user circular block oversampled filter bank (CB-OSFB) uplink systems. Through application of discrete Fourier transform (DFT), the special banded structure and circular properties of the modulation matrix in the frequency domain of each user are derived. By exploiting the newly derived properties, a simplified zero-forcing (ZF) receiver is proposed for multi-user CB-OSFB uplink systems in the multipath channels. In the proposed receiver, the matrix inversion operation of the large dimension multi-user equivalent channel matrix is transformed into DFTs and smaller size matrix inversion operations. Simulation is given to show that the proposed ZF receiver can dramatically reduce the computational complexity while with almost the same symbol error rate as that of the traditional ZF receiver.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.10
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• pp.21-31
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• 1994

This paper describes the multichannel combined R2MFC/DTMF/CCT reciver which provides a signaling service functions for call processing control in digital switching system. Using the TMS320C25 DSP chip, we have implemented multi-function receriver shich processes 8 channels of R2MFC, DTMF, and CCT signals simultaneously. In order to increase the channel multiplicity of the combined receiver. R2MFC and CCT receiver were employed by discrete Fourier transform(DFT) method using Goertzel algorithm, and DTMFreceiver was employ by infinite impulse reponse(IIR) filtering method using 4KHz subsampling technique. The combined receiver has 4 function modes for each channel such as R2MFC, DTMF, CCT, and Idle modes. The function mode of each channel may be selected at any time by single-chip micro-controller(.mu.C). Hence, the number of channels assigned for each function mode can be adjusted dynamically according to the signaling traffic variations. From the experimental test results using the test-bed, it has been proved that the combined receiver statisfies all receiver satisfies all receiver specifications, and provides good channel multiplicity and performance, Therefore, it may give a great improvement than existing receiver in cost, reliability, availability, and serviceability.