• International Journal of Fuzzy Logic and Intelligent Systems
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• v.5 no.1
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• pp.13-20
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• 2005

This paper addresses a new blind channel equalization method using fourth-order cumulants of channel inputs and a three-layer neural network equalizer. The proposed algorithm is robust with respect to the existence of heavy Gaussian noise in a channel and does not require the minimum-phase characteristic of the channel. The transmitted signals at the receiver are over-sampled to ensure the channel described by a full-column rank matrix. It changes a single-input/single-output (SISO) finite-impulse response (FIR) channel to a single-input/multi-output (SIMO) channel. Based on the properties of the fourth-order cumulants of the over-sampled channel inputs, the iterative algorithm is derived to estimate the deconvolution matrix which makes the overall transfer matrix transparent, i.e., it can be reduced to the identity matrix by simple recordering and scaling. By using this estimated deconvolution matrix, which is the inverse of the over-sampled unknown channel, a three-layer neural network equalizer is implemented at the receiver. In simulation studies, the stochastic version of the proposed algorithm is tested with three-ray multi-path channels for on-line operation, and its performance is compared with a method based on conventional second-order statistics. Relatively good results, withe fast convergence speed, are achieved, even when the transmitted symbols are significantly corrupted with Gaussian noise.

• Journal of Communications and Networks
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• v.12 no.2
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• pp.158-167
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• 2010

The energy efficiency of wireless infrastructure and terminals has been drawing renewed attention of late, due to their significant environmental cost. Emerging green communication paradigms such as cognitive radios, are also imposing the additional requirement of flexibility. This dual requirement of energy efficiency and flexibility poses new design challenges for implementing radio functional blocks. This paper focuses on the area vs. power trade-offs for the type of channel filters that are required in the digital frontend of a flexible, energy-efficient radio. In traditional CMOS circuits, increased area was traded for reduced dynamic power consumption. With leakage power emerging as the dominant mode of power consumption in nanoscale CMOS, these trade-offs must be revisited due to the strong correlation between area and leakage power. The current work discusses how the increased timing slacks obtained by increasing the parallelism can be exploited for overall power reduction even in nanoscale circuits. In this context the paper introduces the notion of 'area efficiency' and a metric for evaluating it. The proposed metric has also been used to compare the area efficiencies of different classes of time-shared filters.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.150-158
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• 2013

In this paper we present approximated constrained least squares filter for real-time directionally adaptive image restoration. The proposed method makes a hardware implementation easier for real-time image restoration because of reducing the filter size. Furthermore, for directional adaptive image restoration, this paper estimates the local orientation by analyzing the covariance matrix and applies to approximated constrained least squares filter. Experimental results show that the proposed method is sharper and less artifacts than existing methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1398-1415
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• 1996

Instead of using three charge-coupled devices (CCDs) for the corresponding color channels, most consumer's most consummer's color macmorders reconstruct color images by using only one CCD with a color filter array (CFA), which periodically samples different color signals. By this reson the resulting image cannot produce the full resolution of the input image. More sepecifically, a single-CCD color camcorder reconstructs red, greed, and blue color channels from a color filter array followed by a CCD. During the reconstruction process, color cross-talk among channels (interchannel distortion) and eriodically space-verying blur (intrachannel distortion) occur. The proposed restoration system reduces distortions due to interchannel interference, and then restores each color channel by removing the corresponding intrachannel distortion. Experimental results show that the proposedsystem provides the improved image in oth objective and subjective senses. A major advantage of the proposed system is feasible to real-time image improvement because it can be implemented by a finite impulse response (FIR) filter structure.

• Journal of the Korea Institute of Military Science and Technology
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• v.24 no.5
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• pp.475-481
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• 2021

The on-board processor of satellite Synthetic Aperture Radar(SAR) digitizes the back-scattered echoes and transmits them to the ground. As satellite SAR image of various operating conditions including broadband and high resolution is required, an enormous amount of SAR data is generated. Decimation filter is used for data compression to improve the transmission efficiency of these data. Decimation filter is implemented with the FIR(Finite Impulse Response) filter and here, the decimation ratio and tap length are constrained by resource requirements of FPGA used for implementation. This paper suggests to use a non-integer ratio decimation filter in order to optimize the data transmission efficiency. Also, it proposes a filter design method that remarkably reduces the resource constraints of the FPGA in-use via applying a polyphase filter structure. The required resources for implementing the proposed filter is analysed in this paper.

• Journal of the Korean Institute of Electrical and Electronic Material Engineers
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• v.32 no.1
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• pp.13-19
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• 2019

This paper proposes an easy algorithm for finding tapped-delay-line (TDL) filter coefficients in an adaptive filter algorithm using orthogonal input signals. The proposed algorithm can be used to obtain the coefficients and errors of a TDL filter without using an inverse orthogonalization process for the orthogonal input signals. The form of the proposed algorithm in this paper has the advantages of being easy to use and similar to the familiar recursive least-squares (RLS) algorithm. In order to evaluate the proposed algorithm, system identification simulation of the $11^{th}$-order finite-impulse-response (FIR) filter was performed. It is shown that the convergence characteristics of the learning curve and the tracking ability of the coefficient vectors are similar to those of the conventional RLS analysis. Also, the derived equations and computer simulation results ensure that the proposed algorithm can be used in a similar manner to the Levinson-Durbin algorithm.

• Journal of Biosystems Engineering
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• v.33 no.5
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• pp.317-325
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• 2008

The electrocardiogram (ECG) measurement system consists of I/O interface to input the ECG signals from two electrodes, FPGA (Field programmable gate arrays) module to process the signal conditioning, and real time module to control the system. The algorithms based on wavelet transform were developed to remove the noise of the ECG signals and to determine the QRS-waves. Triangular wave tests were conducted to determine the optimal factors of the wavelet filter by analyzing the SNRs (signal to noise ratios) and RMSEs (root mean square errors). The hybrid rule, soft method, and symlets of order 5 were selected as thresholding rule, thresholding method, and mother wavelet, respectively. The developed wavelet filter showed good performance to remove the noise of the triangular waves with 10.98 dB of SNR and 0.140 mV of RMSE. The ECG signals from a total of 6 subjects were measured at different measuring postures such as lying, sitting, and standing. The durations of QRS-waves, the amplitudes of R-waves, the intervals of RR-waves were analyzed by using the finite impulse response (FIR) filter and the developed wavelet filter. The wavelet filter showed good performance to determine the features of QRS-waves, but the FIR filter had some problems to detect the peaks of Q and S waves. The measuring postures affected accuracy and precision of the ECG signals. The noises of the ECG signals were increased due to the movement of the subject during measurement. The results showed that the wavelet filter was a useful tool to remove the noise of the ECG signals and to determine the features of the QRS-waves.

• 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 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.6A
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• pp.341-350
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• 2003

In this paper, we evaluate the performance of a moving average (MA) channel estimation filter when space-time block coding transmit diversity (STBC-TD) is applied to the wideband direct sequence code division multiple access (WCDMA) forward link. And we present the infinite impulse response (IIR) filter scheme that can reduce the required memory buffer and the channel estimation delay time. This paper also compares the performance between MA filter scheme and IIR filter scheme in various Rayleigh fading channel environments through the bit error rate (BER) and the frame error rate (FER). Extensive computer simulation results show that transmission with STBC-TD provides a significant gain in performance over no transmit diversity technique, particularly at pedestrian speeds. If STBC-TD technique is employed in the channel estimator based on MA filter, it provides considerable performance gains against Rayleigh fading and reduces the optimum filter tap number. Consequently, the channel estimation delay time and the complexity of the receiver are reduced. In addition, the channel estimator based on IIR filter has the advantages such as little memory requirement and no delay time compared to the MA scheme. However, IIR filter coefficients is very sensitive to the mobile speed change and it exerts a serious influence upon the performance. For that reason, it is important to set uP the optimum IIR filter coefficients.