• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2017.05a
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• pp.495-498
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• 2017

As the number of tourist has been increased recently, the interest level in the carriers was elevated. Hence, the technologies on the carriers have been developed variously, and smart carriers are widely used. However, noises are occurred with multiple causes since smart carrier uses the speed control system by gradient sensor. Because of this, the possibility of wrong operation is high by abrupt operation or cumulative error. Therefore, FIR low pass filter was designed and applied in the gradient sensor to realize the strong sensing system against the noise so as to supplement the weaknesses of the existing system in this article.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11C
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• pp.911-916
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• 2010

This paper proposes a new algorithm to cancel the acoustic feedback and noise signals in digital hearing aids. The proposed algorithm combines the feedback canceller to remove acoustic feedback signals and the noise canceller to reduce background noises. The feedback canceller is implemented by normal adaptive FIR filter, and the noise canceller is implemented by using the Wiener solution in frequency domain. This noise canceller has the transfer function presented by the power spectral density of signals. To verify the performances of the proposed algorithm, the simulations were carried out for the system. As the results of simulations, it was proved that we can advance 10.85dB output SNR on the average for the forward path gain of 0dB, and 11.04dB output SNR on the average for the forward path gain of 6dB, in the case of using the proposed algorithm.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.979-986
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• 2018

This paper deals with acoustic echo cancellation using blind signal separation method. This method does not degrade the echo cancellation performance even during double-talk. In the closed echo environment, the mixing model of acoustic signals is multi-channel, so the convolutive blind signal separation method is applied and the mixing coefficients are calculated by using the feedback model without directly calculating the separation coefficients for signal separation. The coefficient update is performed by iterative calculations based on the second-order statistical properties, thus estimates the near-end speech. A number of simulations have been performed to verify the performance of the proposed blind signal separation method. The simulation results show that the acoustic echo canceller using this method operates safely regardless of the presence of double-talk, and the PESQ is improved by 0.6 point compared with the general adaptive FIR filter structure.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.19 no.8
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• pp.116-121
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• 2005

To realize the tap coefficient of a finite impulse response(FIR) filter or the twiddle factor of a fast Fourier transform(FFT) using a current-mode analog circuit, a high accurate current-attenuator circuit is needed This paper introduces an accuracy enhancement technique in the current-mode signal processing. First of all, the DC of set-current error in a conventional current-attenuator using a gate-ratioed orient mirror circuit is analyzed and then, the current-attenuator circuit with a negligibly small DC offset-current error is introduced. The circuit consists of N-output current mirrors connected in parallel with me another. The output current of the circuit is attenuated to 1/N of the input current. On the basis of the Kirchhoff current law, the current scale ratio is determined simply by the number of the current mirrors in the N-current mirrors connected in parallel. In the proposed current-attenuator circuit the scale accuracy is limited by the ac gain error of the current mirror. Considering that a current mirror has a negligibly small ac gain error, the attainable maximum scale accuracy is theoretically -80[dB] to the input current.

• Journal of the Institute of Convergence Signal Processing
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• v.12 no.2
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• pp.145-149
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• 2011

Recently, as the design of a system gets larger and more complex, functional verification method based on system-level becomes more important. The verification of a functional block mainly uses BFM(bus functional model). The larger the burden on functional verification is, the more the importance of configuring a proper verification environment increases rapidly. SystemVerilog unifies hardware design languages and verification languages in the form of extensions to the Veri log HDL. The processing of design description, function simulation and verification using same language has many advantages in system development. In this paper, we design DUT that is composed of AMBA bus and function blocks using SystemVerilog and verify the function of DUT in verification environment using layered testbench. Adaptive FIR filter and Booth's multiplier are chosen as function blocks. We confirm that verification environment can be reused through a minor adaptation of interface to verify functions of other DUT.

• The Journal of the Korea institute of electronic communication sciences
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• v.17 no.3
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• pp.409-416
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• 2022

This paper relates to the blind noise separation method of time-delayed convolutive mixed signals. Since the mixed model of acoustic signals in a closed space is multi-channel, a convolutive blind signal separation method is applied and time-delayed data samples of the two microphone input signals is used. For signal separation, the mixing coefficient is calculated using an inverse model rather than directly calculating the separation coefficient, and the coefficient update is performed by repeated calculations based on secondary statistical properties to estimate the speech signal. Many simulations were performed to verify the performance of the proposed blind signal separation. As a result of the simulation, noise separation using this method operates safely regardless of convolutive mixing, and PESQ is improved by 0.3 points compared to the general adaptive FIR filter structure.

• Journal of the Institute of Convergence Signal Processing
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• v.1 no.1
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• pp.7-14
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• 2000

Recently, the technologies of adaptive noise cancellation(ANC) are developed fast and widely due to the highly sophisticated digital signal processing algorithms and the high-speed communication networks and devices. But, thousand numbers of the adaptive filter taps are required to obtain the satisfying results in the fields of the adaptive noise cancellation and echo cancellation. In the paper, performance comparisons based on the real-time processing between frequency domain adaptive filter(FDAF) and multi-delay frequency domain adaptive filter(MDF) are carried. Those algorithms provide us with the reductions of the computational burdens and the increase of the convergence rate for the lengthy Fill adaptive filters. The time delay due to the long taps of FDAF can be reduced by adopting the MDF algorithms. The conventional ANC and cross talks ANC using FDAF are implemented on the dSP ACE 1103 real-time signal processing board.

• Journal of the Korean Geotechnical Society
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• v.18 no.5
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• pp.261-269
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• 2002

The objectives of this paper are to study the use of geotextile tribes for dewatering high water content sludges and sediments and to evaluate their feasibility and affecting fsctors. To accomplish these objectives, pressure filtration tests were conducted on woven geotextile (Geotex$\circledR$ 46T and 1212T) fir high water content materials with a modified experimental apparatus. Test results indicate that 1) the filter cake formed on the inside of the geotextile tube is the major contributor to the retention of fine particles, but also causes a decrease in permeability, 2) controlling the formation of the filter cake and thus achieving a balance between soil retention and permeability is vital to a successful project, and 3) geotextiles, sludge properties, and filtration pressures have some effects on the dewatering efficiency and dewatering rate.

• Journal of Biomedical Engineering Research
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• v.17 no.4
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• pp.449-458
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• 1996

Most of the conventional electrocardiowaphs foil to detect signals other than P-QRS-T due to the limited SNR and bandwidth. High-resolution electrocardiography(HRECG) provides better SNR and wider bandwidth for the detection of micro-potentials with higher frequency components such as vontricular late potentials(LP). We have developed a HRECG using uncorrected XYZ lead for the detection of LPs. The overall gain of the amplifier is 4000 and the bandwidth is 0.5-300Hz without using 60Hz notch filter. Three 16-bit A/D converters sample X, Y, and Z signals simultaneously with a sampling frequency of 2000Hz. Sampled data are transmitted to a PC via a DMA-controlled, optically-coupled serial communication channel. In order to further reduce the noise, we implemented a signal averaging algorithm that averaged many instances of aligned beats. The beat alignment was carried out through the use of a template matching technique that finds a location maximizing cross-correlation with a given beat tem- plate. Beat alignment error was reduced to $\pm$0.25ms. FIR high-pass filter with cut-off frequency of 40Hz was applied to remove the low frequency components of the averaged X, Y, and Z signals. QRS onset and end point were determined from the vector magnitude of the sigrlaIL and some parameters needed to detect the existence of LP were estimated. The entire system was designed for the easy application of the future research topics including the optimal lead system, filter design, new parameter extraction, etc. In the developed HRECG, without signal averaging, the noise level was less than 5$\mu$V$_rms RTI$. With signal averaging of at least 100 beats, the noise level was reduced to 0.5$\mu$V$_rms RTI$, which is low enough to detect LPs. The developed HRECG will provide a new advanced functionality to interpretive ECG analyzers.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.10
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• pp.1916-1923
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• 2007

We have studied to obtain the SAW filter for the passband was formed on the Langasite substrate and was evaporated by Aluminum-Copper alloy and thin we performed computer-simulated by simulator. We cm fabricate that the block weighted type IDT as an input transducer of the filter and the withdrawal weighted type IDT as an output transducer of the filter from the results of our computer-simulation. Also, we have performed to obtain the properly design conditions about phase shift of the SAW filter for WCDMA. We have employed that the number of pairs of the input and output IDT are 50 pairs and the thickness and the width of reflector are $5000\;{\AA}$ and $3.6{\mu}m$ respectively. And we have employed that the distances from the hot electrode to the reflector are $2.0{\mu}m$, $2.4{\mu}m$ and the distance from the hot electrode to the ground is $1.5{\mu}m$ respectively. Frequency response of the fabricated SAW filter has the property that the center frequency is about 190MHz and bandwidth at the 3dB is probably 7,8MHz. And we could obtain that return loss is less then -18dB, ripple characteristics is probably 3dB and triple transit echo is less then -25dB after when we have matched impedance.