• Journal of the Korea Academia-Industrial cooperation Society
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• v.21 no.1
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• pp.753-759
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• 2020

A modified Chebyshev lowpass filter function with progressively diminishing ripples in the passband is proposed and analyzed in the frequency domain. Owing to the diminishing ripples, the passband magnitude characteristic of the proposed Chebyshev function has improved compared to the classical Chebyshev function. In addition, the phase characteristics of the proposed Chebyshev function were improved considerably compared to that of the Chebyshev function, and the time delay of the proposed function was much simpler and flatter. In addition, the proposed Chebyshev filter was realizable by the passive doubly terminated ladder network delivering maximum power transfer for the order n, even or odd, thus making themselves amenable to low-sensitivity active RC or switched capacitor filters through the simulation techniques. To verify the proposed Chebyshev filter characteristics, a 6th order passive doubly terminated ladder lowpass filter was designed and analyzed using the MATLAB and SPICE program. Thus, the proposed Chebyshev function can remove the drawbacks of the classical Chebyshev function and could be applicable to the design of a filter with an improved filter size, phase, and time delay characteristics for various signal processing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.4C
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• pp.505-513
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• 2004

Most video codecs including the international standards use the block-based hybrid structure for efficient compression. But for low bit-rate applications such as video transmission through wireless channels, the blocking artifacts degrade image qualify seriously. In this paper, we propose an adaptive de-blocking algorithm using characteristics of the block boundaries. Blocking artifacts contain the high frequency components near the block boundaries, therefore the lowpass filtering can remove them. However, simple lowpass filtering results into blurring by removing important information such as edges. To overcome this problem, we determine the modes depending upon the characteristics of pixels adjacent to block boundary then proper filter is applied to each area. Simulation results show that proposed method improves de-blocking performance compared to that of MPEG-4.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.12
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• pp.1274-1285
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• 1991

This paper presents a statistical convergence analysis of the normalized least mean square(NLMS)algorithm that employs a single-pole lowpass filter, In this algorithm the lowpass filter is used to adjust its output towards the estimated value of the input signal power recursively. The estimated input signal power so obtained at each time is then used to normalize the convergence parameter. Under the assumption that the primary and reference inputs to the adaptive filter are zero mean wide sense stationary, and Gaussian random processes, and further making use of the independence assumption. we derive expressions that characterize the mean and maen squared behavior of the filter coefficients as well as the mean squared estimation error. Conditions for the mean and mean squared convergence are explored. Comparisons are also made between the performance of the NLMS algorithm and that of the popular least mean square(LMS) algorithm Finally, experimental results that show very good agreement between the analytical and emprincal results are presented.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.4
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• pp.214-219
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• 2009

This paper proposes a watermarking algorithm considering the frequency characteristics of Haegeum sounds for copyright protection of digital Haegeum sound contents. The harmonics of Haegeum sounds commonly have large magnitude values in 1500Hz~2000Hz and 2800Hz~3500Hz so that those bands are selected to embed a watermark. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and embeds the watermark bits generated by PN (pseudo noise) sequence into the harmonics in the selected bands. Furthermore, the proposed method is robust to lowpass filter, bandpass filter, cropping, noise addition, MP3 compression attacks and the maximum BER (bit error rate) is 1.41% after lowpass filter attack. To measure the quality of the watermarked sound, subjective listening test, MUSHRA (multiple stimuli with hidden reference and anchor), was conducted. The mean value of MUSHRA listening test is bigger than 98 and 96.67 for every Haegeum sounds and Korean classical music with Haeguem, respectively.

• The Journal of the Convergence on Culture Technology
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• v.9 no.4
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• pp.563-568
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• 2023

ECG(electrocardiogram) is used to measure the rate and regularity of heartbeats, as well as the size and position of the chambers, the presence of any damage to the heart, and the cause of all heart diseases can be found. Because the ECG signal obtained using the ECG-KIT includes noise in the ECG signal, noise must be removed from the ECG signal to apply to the deep learning. In this paper, Noise included in the ECG signal was removed by using a lowpass filter of the Digital FIR Hamming window function. When the performance evaluation of the three activation functions, sigmoid(), ReLU(), and tanh() functions, which was confirmed that the activation function with the smallest error was the tanh() function, the elapsed time was longer when the batch size was small than large. Also, it was confirmed that result of the performance evaluation for the GRU model was superior to that of the LSTM model.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.60 no.12
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• pp.2374-2379
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• 2011

The fundamental frequency of the voiced speech is estimated using the instantaneous frequency based on the symmetric higher order differential energy operator. The instantaneous frequency based on the symmetric higher order energy operator shows better frequency estimation result since it is aligned to the time instance of the signal. The speech is pre-processed by a lowpass filter to remove higher frequency components. Then, it is processed by the instantaneous frequency to obtain the fundamental frequency estimates. The symmetric higher order energy operator has been used as an indicator to determine the voiced/unvoiced speech. The fundamental frequency estimates are further processed by a moving average filter to obtain the monotonically changed estimates. The obtained fundamental frequency estimates have been compared with the spectrogram of the speech to confirm its accuracy.

• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.203-206
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• 2001

This Paper suggests real-time video streaming method by using DCT-based scalability, and evaluates and analyzes the function. It is similar to using lowpass filter. That is, as following figure, this method is to split the encoded data in splitter and transmit it, and to decode the data according to the situation. This method can be applied to any video CODEC which is based on DCT. Therefore, this thesis suggests easy video streaming method by using DCT-based scalability, and shows the result of experiment. By using suggested scalability, calculations are reduced, and spacial scalability is realized. Moreover, the objective data which meet user's need according to the network condition and choose the appropriate scalability according to the capability of terming can be extracted. And it is possible to apply any resources according to the specificity of image.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.694-697
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• 2002

Sampling rate conversion widely used in subband coding, A/D and D/A transitions etc. is an important techniques. Nyquist filters and the filter banks have been used far the sampling converter. However, they need many memories and, whenever the sampling rate is changed it is necessary to redesign filters. Then we propose design method of the new interpolation kernel. Design method of the new interpolation kernel is approximated each piecewise of lowpass filter by n th polynomials. The proposed kernel is not redesigned, whenever the sampling rate is changed. The proposed kernel is a continuous function, the sampling rate of the rational number can be converted.

• The Journal of the Acoustical Society of Korea
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• v.17 no.3E
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• pp.58-66
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• 1998

In this paper, we analyze the effects of interference canceller on the performance of multitone DS/CDMA system proposed by Vandendorpe[5]. There are various kinds of interference canceller suggested by different researchers including parallel and successive cancellers and we adopt a canceller used by Yoon et al.[9] which is a kind of parallel canceller. We consider three kinds of interferences, that is, multipath interference(MPI), interchannel interference(ICI) and multiple access interference(MAI). The ICI is the interference between multitones. The equations for variances. are derived for the inteferences and thermal noise used for signal to noise ratio calculation. We also consider RAKE reception over multipath channel which is modeled as lowpass equivalent linear filter and three stage interference canceller used for performance improvement. We show the performance results for number of canceller stage, diversity order and number of users and draw some conclusions that interference canceller is effective in multitone DS/CDMA system and the performance is further improved with the higher order of diversity and larger number of PN chips.

• Proceedings of the Korea Institute of Applied Superconductivity and Cryogenics Conference
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• 2002.02a
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• pp.358-361
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• 2002

The millirneterwave high-T$_{c}$ superconducting(HTS) down-converter sub-system with the HTS/III-V integrated mixer as the central device is demonstrated first. The constituent components of HTS down-converter sub-system such as a single balanced type integrated mixer with rat-race coupler, a cavity type bandpass filter (26 GHz), and a HTS planar lowpass filter(1 GHz), semiconductor LNA and IF-power amplifier, a driving electronic module for A/D converter, and a Stirling type mini-cooler module were combined into an International stand- and rack of 19-inch. From the RF(-61 dBm, 26.5GHz)and LO signal(-1 dBm, 25.6 GHz), IF signal(0dBm, 0.9 GHz) agreed with simulated results is obtained.d.