• Journal of the Institute of Electronics Engineers of Korea CI
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• v.41 no.4
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• pp.31-39
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• 2004

The ARM7 TDMI microprocessor employ a software routine iteration method in order to handle integer division operation, but this method has long execution time and many execution instruction. In this paper, we proposed ARM7 TDMI microprocessor with integer division instruction. To make this, we additionally defined UDIV instruction for unsigned integer division operation and SDIV instruction for signed integer division operation, and proposed ARM7 TDMI microprocessor data Path to apply division algorithm. Applied division algorithm is nonrestoring division algorithm and additive hardware is reduced using existent ARM data path. To verify the proposed method, we designed proposed method on RTL level using HDL, and conducted logic simulation. we estimated the number of execution cycles and the number of execution instructions as compared proposed method with a software routine iteration method, and compared with other published integer divider from the number of execution cycles and hardware size.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.4
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• pp.45-53
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• 2003

A new smart antenna receiver which incorporates the spatial fourier Transform and the maximal ratio combining(MRC) is proposed. By adapting the spatial fourier transform, the proposed method could separate the received signal into several spatial frequency components which correspond to the arrival angles of signal components, which means the beam focusing. By using the MRC, the proposed method could achieve the maximum signal to noise ratio for the signal of interest. The proposed algorithm is integrated to the CDMA reverse link receiver and simulations are performed to confirm the performance. As a result, the beam focusing effect is confirmed and the performance gain with the proposed algorithm is comparable to ordinary smart antenna receivers. The simulations are performed over the additive white gaussian noise (AWGN) environments and the results are obtained for the beam focusing capability according to the angle of arrival of a signal and the bit error performance improvement according to the number of combining branches in the MRC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10C
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• pp.975-981
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• 2007

In this paper, we propose a spatially adaptive image restoration algorithm using local and statistics and estimated noise. The ratio of local mean, variance, and maximum values with different window size is used to constrain the solution space, and these parameters are computed at each iteration step using partially restored image. In addition, the additive noise estimated from partially restored image and the local constraints are used to determine a parameter for controlling the degree of local smoothness on the solution. The resulting iterative algorithm exhibits increased convergence speed when compared to the non-adaptive algorithm. In addition, a smooth solution with a controlled degree of smoothness is obtained without a prior knowledge about the noise. Experimental results demonstrate that the proposed algorithm requires the similar iteration number to converge, but there is the improvement of SNR more than 0.2 dB comparing to the previous approach.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1
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• pp.46-53
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• 1996

Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

• Journal of the Computational Structural Engineering Institute of Korea
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• v.31 no.5
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• pp.243-250
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• 2018

In this paper, design technique using genetic algorithms(GA) for design optimization of composite leaf springs is presented here. After the initial design has been validated by the car plate spring as a finite element model, the genetic algorithm suggests the process of optimizing the number of layers of composite materials and their angles. Through optimization process, the weight reduction process of leaf springs and the number of repetitions are compared to the existing algorithm results. The safety margin is calculated by organizing a finite element model to verify the integrity of the structure by applying an additive sequence optimized through the genetic algorithm to the structure. When GA is applied, layer thickness and layer angle of complex leaf springs have been obtained, which contributes to the achievement of minimum weight with appropriate strength and stiffness. A reduction of 65.6% original weight is reached when a leaf steel spring is replaced with a leaf composite spring under identical requirement of design parameters and optimization.

• Transactions on Control, Automation and Systems Engineering
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• v.2 no.1
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• pp.7-12
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• 2000

A new speech enhancement algorithm for speech corrupted by slowly varying additive colored noise is suggested based on a state-space signal model. Due to the FIR structure and the unimportance of long-term past information, the receding horizon (RH) FIR filter known to be a best linear unbiased estimation (BLUE) filter is utilized in order to obtain noise-suppressed speech signal. As a special case of the colored noise problem, the suggested approach is generalized to perform the single blind signal separation of two speech signals. It is shown that the exact speech signal is obtained when an incoming speech signal is noise-free.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2E
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• pp.24-32
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• 1996

A rigorous theory for modeling, analysis, optimum nonseparable filter bank in multidimensional M-band quantized subband codec are developed in this paper. Each pdf-optimized quantizer is modeled by a nonlinear gain-plus-additive uncorrelated noise and embedded into the subband structure. We then decompose the analysis/synthesis filter banks into their polyphase components and shift the down-and up-samplers to the right and left of the analysis/synthesis polyphase matrices respectively. Focusing on the slow clock rate signal between the samplers, we derive the exact expression for the output mean square quantization error by using spatial-invariant analysis. We show that this error can be represented by two uncorrelated components : a distortion component due to the quantizer gain, and a random noise component due to fictitious uncorrelated noise at the uantizer. This mean square error is then minimized subject to perfect reconstruction (PR) constraints and the total bit allocation for the entire filter bank. The algorithm gives filter coefficients and subband bit allocations. Numerical design example for the optimum nonseparable orthonormal filter bank is given with a quincunx subsampling lattice.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.36 no.5
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• pp.367-376
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• 1987

In this paper we have disigned and implemented a real time ALE (adaptive line enhancer) using a high speed digital processor,NEC 7720. For the ALE system, we have employed an adaptive LMS(least mean square) algorithm proposed by Widrow and Hoff and a 32-order FIR(finite impulse response) filter. Extensive computer simulations have been performed to investigate the peformance of the ALE and to determine necessary parameters for hardware design. The developed software for an NEC 7720 was tested in real time operation using an NEC7720 hardware emulator. The ALE has been tested by sinusoidal waves and real CW (continuous wave) signals. It was found that the experimental results were well agreed with the computer simulation results. Thus it can be concluded that the ALE is useful for detection and enhancement of a sinusoidal signal which is corrupted by an additive Gaussian noise.

• Speech Sciences
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• v.9 no.1
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• pp.39-47
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• 2002

This paper proposes an effective method to remedy the acoustic modeling problem inherent in the usual log-normal Parallel Model Composition intended for achieving robust speech recognition. In particular, the Gaussian kernels under the prescribed log-normal PMC cannot sufficiently express the corrupted speech distributions. The proposed scheme corrects this deficiency by judiciously selecting the 'fairly' corrupted component and by re-estimating it as a mixture of two distributions using data-driven PMC. As a result, some components become merged while equal number of components split. The determination for splitting or merging is achieved by means of measuring the similarity of the corrupted speech model to those of the clean model and the noise model. The experimental results indicate that the suggested algorithm is effective in representing the corrupted speech distributions and attains consistent improvement over various SNR and noise cases.

• Journal of Korea Society of Digital Industry and Information Management
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• v.11 no.3
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• pp.45-51
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• 2015

In this paper, we discuss the belief propagation decoding algorithm for polar codes. The performance of Polar codes for shorter lengths is not satisfactory. Motivated by this, we propose a novel technique to improve its performance at short lengths. We showed that the probability of messages passed along the factor graph of polar codes, can be increased by multiplying the current message of nodes with their previous message. This is like a feedback path in which the present signal is updated by multiplying with its previous signal. Thus the experimental results show that performance of belief propagation polar decoder can be improved using this proposed technique. Simulation results in binary-input additive white Gaussian noise channel (BI-AWGNC) show that the proposed belief propagation polar decoder can provide significant gain of 2 dB over the original belief propagation polar decoder with code rate 0.5 and code length 128 at the bit error rate (BER) of $10^{-4}$.