• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.4
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• pp.565-570
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• 1986

In this paper, the implementation of a voiced/unvoiced/silence discrimination system is presented. The algorithm is based on the zerocrossing rate and the spectral energy distribution of speech. In measuring zerocrossing rate, a new frequency-to-voltage conversion type interval filter is used. Expermental results show that with the proposed algorithm the effect of impulse noise in voiced intervals can be removed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5C
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• pp.527-539
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• 2009

In this paper, we propose an asynchonous ICI (Inter-Cell Interference) mitigation techniques for 3GPP LTE MIMO-OFDMA down-link receiver. An increasing in symbol timing misalignments may occur relative to sychronous network as the result of BS (Base Station) timing differences. Such symbol synchronization errors that exceed the guard interval or the cyclic prefix duration may result in MAI (Multiple Access Interference) for other carriers. In particular, at the cell boundary, this MAI becomes a critical factor, leading to degraded channel throughput and severe asynchronous ICI. Hence, many researchers have investigated the interference mitigation method in the presence of asynchronous ICI and it appears that the knowledge of the SCM (Spatial Covariance Matrix) of the asynchronous ICI plus background noise is an important issue. Generally, it is assumed that the SCM estimated by using training symbols. However, it is difficult to measure the interference statistics for a long time and training symbol is also not appropriate for MIMO-OFDMA system such as LTE. Therefore, a noise reduction method is required to improve the estimation accuracy. Although the conventional time-domain low-pass type weighting method can be effective for noise reduction, it causes significant estimation error due to the spectral leakage in practical OFDM system. Therefore, we propose a time-domain sinc type weighing method which can not only reduce the noise effectively minimizing estimation error caused by the spectral leakage but also implement frequency-domain moving average filter easily. By using computer simulation, we show that the proposed method can provide up to 3dB SIR gain compared with the conventional method.

• Korean Journal of Optics and Photonics
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• v.11 no.6
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• pp.423-428
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• 2000

We have demonstrated a -8 dB additional reduction in the intensity sidelobe of an apodized-interaction-strength guide-wave acousto-optic filter with a center passband of 1551.6 nm. Acoustic-intensity weighting was achieved by launching a surface acoustic wave (SAW) beam in a straight acoustic waveguide, and gradually transferring this SAW intensity to the active device, and back out, by evanescent-wave coupling across a 50 !lm barrier over a 19 rom interaction length. The intensity sidelobe was -4.27 dB for an unapodized filter with abmpt onset and cutoff of the interaction, but sidelobes were reduced to at most -12.68 dB for a SAW intensity with raised-cosine weighting. The RF driving power was 17.78 mW. A linear tuning rate of 8.86 nmIMHz and a spectral width of -1.7 nm were demonstrated. rated.

• Journal of The Korean Astronomical Society
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• v.32 no.1
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• pp.65-73
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• 1999

In this study we present a new improved nonlinear calibration method for vector magnetograms made by the Solar Flare Telescope of BOAO. To identify Fe I 6302.5 line, we have scanned monochromatic images of the line integrated over filter passband, changing the location of the central transmission wavelength of a Lyot filter. Then we obtained a filter-convolved line profile, which is in good agreement with spectral atlas data provided by the Sacramento Peak Solar Observatory. The line profile has been used to derive calibration coefficients of longitudinal and transverse fields, employing the conventional line slope method under the weak field approximation. Our improved nonlinear calibration method has also been used to calculate theoretical Stokes polarization signals with various angles of inclination of magnetic fields. For its numerical test, we have compared input magnetic fields with the calibrated ones, which have been derived from the new improved non-linear method and the conventional method respectively. The numerical test shows that the calibrated fields obtained from the improved method are consistent with the input fields, but not with those from the conventional method. Finally, we applied our new improved method to a dipole model which characterizes a typical field configuration of a single, round sunspot. It is noted that the conventional method remarkably underestimates the transverse field component near the inner penumbra.

• International Journal of Internet, Broadcasting and Communication
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• v.13 no.1
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• pp.161-167
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• 2021

This paper proposes a new personalized HRTF estimation method which is based on a deep neural network (DNN) model and improved elevation reproduction using a notch filter. In the previous study, a DNN model was proposed that estimates the magnitude of HRTF by using anthropometric measurements [1]. However, since this method uses zero-phase without estimating the phase, it causes the internalization (i.e., the inside-the-head localization) of sound when listening the spatial sound. We devise a method to estimate both the magnitude and phase of HRTF based on the DNN model. Personalized HRIR was estimated using the anthropometric measurements including detailed data of the head, torso, shoulders and ears as inputs for the DNN model. After that, the estimated HRIR was filtered with an appropriate notch filter to improve elevation reproduction. In order to evaluate the performance, both of the objective and subjective evaluations are conducted. For the objective evaluation, the root mean square error (RMSE) and the log spectral distance (LSD) between the reference HRTF and the estimated HRTF are measured. For subjective evaluation, the MUSHRA test and preference test are conducted. As a result, the proposed method can make listeners experience more immersive audio than the previous methods.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.17 no.1
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• pp.280-294
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• 2023

Unmanned aerial vehicles (UAVs) and millimeter-wave frequencies play key roles in supporting 5G wireless communication systems. They expand the field of wireless communication by increasing the data capacities of communication systems and supporting high data rates. However, short wavelengths, owing to the high millimeter-wave frequencies can cause problems, such as signal attenuation and path loss. To address these limitations, research on high directional beamforming technologies continue to garner interest. Furthermore, owing to the mobility of the UAVs, it is essential to track the beam angle accurately to obtain full beamforming gain. This study presents a beam tracking method based on the unscented Kalman filter using hybrid beamforming. The simulation results reveal that the proposed beam tracking scheme improves the overall performance in terms of the mean-squared error and spectral efficiency. In addition, by expanding analog beamforming to hybrid beamforming, the proposed algorithm can be used even in multi-user and multi-stream environments to increase data capacity, thereby increasing utilization in new-radio multiple-input multiple-output orthogonal frequency-division multiplexing systems.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.4
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• pp.286-291
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• 2003

In this Paper, we designed an LSP(Line Spectral Pairs) parameter quantizer with cascaded structure of vector quantizer and scalar quantizer for the wideband speech coder. We have chosen the 16th-order of the LP coefficients. These coefficients are then transformed into the LSP parameters which have the excellent properties for quantization and easy stability checking condition of synthesis filter. In the first stage of quantization, input LSP parameters are split-vector-quantized using two 8-th order codebooks. In the second stage, the components of residual vector are individually quantized by the scalar quantizer utilizing the ordering property of LSP parameters. The designed adaptive VQ-SQ quantizer using 35 bits/frame shows the wideband transparency that the average spectral distortion should be less than 1.6 ㏈ and less than 4% of the frames should have SD above 3 ㏈. The simulation results show that the designed quantizer provides a 2-3 bits/frame saving over the typical vector-scalar quantizer.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.1
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• pp.225-230
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• 2017

In this paper, a highly efficient GMR(guided-mode resonant) color filter is proposed and implemented. The GMR color filter consists of $Si_3N_4/air$ layers containing complementary fixed and mobile gratings. The device is designed using RETT(rigorous equivalent transmission-line theory) and a grating structure operating in subwavelength. The numerical result reveals that the color filter has a tuning capability of about 35 nm over the $0.45{\mu}m{\sim}0.55{\mu}m$ range for blue-green color and across $0.6{\mu}m{\sim}0.7{\mu}m$ range for red color. Furthermore, The color filters have a spectral bandwidth of about 8 nm with efficiencies of 99%, 98%, and 99% at the center wavelength of blue, green, and red color, respectively, and these are higher efficiencies than reported in the literature previously.

• The Journal of the Korea Contents Association
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• v.6 no.12
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• pp.89-95
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• 2006

The performance of blind source separation(BSS) using independent component analysis (ICA) declines significantly in a reverberant environment. A post-processing method proposed in this paper was designed to remove the residual component precisely. The proposed method used modified NLMS(normalized least mean square) filter in frequency domain, to estimate cross-talk path that causes residual cross-talk components. Residual cross-talk components in one channel is correspond to direct components in another channel. Therefore, we can estimate cross-talk path using another channel input signals from adaptive filter. Step size is normalized by input signal power in conventional NLMS filter, but it is normalized by sum of input signal power and error signal power in modified NLMS filter. By using this method, we can prevent misadjustment of filter weights. The estimated residual cross-talk components are subtracted by non-stationary spectral subtraction. The computer simulation results using speech signals show that the proposed method improves the noise reduction ratio(NRR) by approximately 3dB on conventional FDICA.

• Korean Journal of Optics and Photonics
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• v.31 no.3
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• pp.148-154
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• 2020

In this paper we propose a plasmonic color filter with a novel nanopattern. The suggested pattern, called a "polyperiodic hole array" (PPHA), is introduced to solve the angle dependence of the optical response that originates from the periodic structure. We set the diameter and period of the hole to make a green color filter, and set the unit-cell size and metal and dielectric thicknesses in consideration of the propagation length and skin depth. The periodic hole arrays are locally rotated to make a PPHA pattern, resulting in a globally aperiodic yet partially periodic pattern. As a result, compared to a general pattern, the PPHA nanostructured color filter has a maximum 40% improvement in spectral shift when the angle of incidence is increased from 0° to 30°. Transmittance reduction was also alleviated by 30%. This work will improve the performance of nanostructured color filters and help with nanotechnology being applied industrially to imaging devices, including displays and image sensors.