• Journal of KIISE:Computer Systems and Theory
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• v.32 no.5
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• pp.232-242
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• 2005

This paper concerns automatic hardware synthesis from data flow graph (DFG) specification for fast HW/SW cosynthesis. A node in BFG represents a coarse grain block such as FIR and DCT and a port in a block may consume multiple data samples per invocation, which distinguishes our approach from behavioral synthesis and complicates the problem. In the presented design methodology, a dataflow graph with specified algorithm can be mapped to various hardware structures according to the resource allocation and schedule information. This simplifies the management of the area/performance tradeoff in hardware design and widens the design space of hardware implementation of a dataflow graph compared with the previous approaches. Through experiments with some examples, the usefulness of the proposed technique is demonstrated.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.11 no.4
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• pp.657-665
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• 2000

In this study, the bit error rate(BER) performance for the speech and data traffic is evaluated by results of the reverse link simulation of CDMA-based IMT-2000. Simulations in the reverse link are achieved for indoor, pedestrian, and vehicular environments, which are provided by ITU-R . Also, in the these simulations, the fast power control of 1.6kHz rate is applied. The amplitude and phase of the fading signal are estimated by using the 5-tap FIR filter, and the soft-decision Viterbi and Reed-Solomon (RS) decoding are applied. Simulation results provide the optimum ratio of pilot power to traffic power, the BER performance according to the number of fingers, and performance comparison between convolutional code and concatenated code at $10^-6$ BER in 5 MHz system.

• 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.

• Transactions of the Korean Society of Mechanical Engineers B
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• v.36 no.6
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• pp.655-662
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• 2012

Gas leakage in an oil refinery causes damage to the environment and unsafe conditions. Therefore, it is necessary to develop a technique that is able to detect the location of the leakage and to filter abnormal gas-leakage signals from normal background noise. In this study, the adaptation filter of the finite impulse response (FIR) least mean squares (LMS) algorithm and a cross-correlation function were used to develop a leakage-predicting program based on LABVIEW. Nitrogen gas at a high pressure of 120 kg/$cm^2$ and the assembled equipment were used to perform experiments in a reverberant chamber. Analysis of the data from the experiments performed with various hole sizes, pressures, distances, and frequencies indicated that the background noise occurred primarily at less than 1 kHz and that the leakage signal appeared in a high-frequency region of around 16 kHz. Measurement of the noise sources in an actual oil refinery revealed that the noise frequencies of pumps and compressors, which are two typical background noise sources in a petrochemical plant, were 2 kHz and 4.5 kHz, respectively. The fact that these two signals were separated clearly made it possible to distinguish leakage signals from background noises and, in addition, to detect the location of the leakage.

• The Journal of the Korea institute of electronic communication sciences
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• v.19 no.1
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• pp.31-38
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• 2024

The digital phase-locked loop(DPLL) is one of the circuits composed of a digital detector, digital loop filter, voltage-controlled oscillator, and divider as a fundamental circuit, widely used in many fields such as electrical and circuit fields. A state estimator using various mathematical algorithms is used to improve the performance of a digital phase-locked loop. Traditional state estimators have utilized Kalman filters of infinite impulse response state estimators, and digital phase-locked loops based on infinite impulse response state estimators can cause rapid performance degradation in unexpected situations such as inaccuracies in initial values, model errors, and various disturbances. In this paper, we propose a two-layer Frobenius norm-based finite impulse state estimator to design a new digital phase-locked loop. The proposed state estimator uses the estimated state of the first layer to estimate the state of the first layer with the accumulated measurement value. To verify the robust performance of the new finite impulse response state estimator-based digital phase locked-loop, simulations were performed by comparing it with the infinite impulse response state estimator in situations where noise covariance information was inaccurate.

• 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 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.