• Journal of IKEEE
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• v.24 no.1
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• pp.120-126
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• 2020

We present a user recognition method using human body impulse response signals. The body compositions vary from person to person depending on the portion of water, muscle, and fat. In the body communication study, the body has been interpreted circuit models using capacitance and resistances, and its characteristics are determined by the body compositions. Therefore, the individual body channel is unique and can be used for user recognition. In this paper, we applied pseudo impulse signals to the left hand and recorded received signals from the right hand. The empirical mode decomposition (EMD) method removed noise from the received signals and 10 peak values are extracted. We set the differences between peak amplitudes as a key feature to identify individuals. We collected data from 6 subjects and achieved accuracy of 97.71% for the user recognition application.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.3
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• pp.32-38
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• 2001

Multi-carrier data transmission system performance is mostly limited by Inter- symbol-interference that is caused by a dispersive characteristic of the transmission channel. In order to enhance the system performance to meet the service requirements of local access, the channel impulse response shortening method incorporated with a channel frequency response compensation method is proposed. For a fast and efficient implementation of the equalizer proposed, Kalman and LMS algorithms are successively used. To verify the channel equalization performance, a set of computer simulation is performed on a filter bank based multitone system operating in a typical high-speed local area data transmission environment. The results showed us a comparable signal-to-interference improvement over the conventional multitone equalization scheme.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.8 no.3
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• pp.9-14
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• 2008

In this paper, we propose the channel sounding scheme which is made for ideal communication between some application as well as the short distance of high speed data transmission in MIMO-OFDM system for Wireless PAN. This method is able to perceive the duration of the impulse response through the delaying of power delay profile, modeled a power delay profile which has an attenuate characteristic, and obtained the coefficient of channel response by ML (maximum likelihood). Through the amplitudes, phases and delays associated with each multipath component which were acquired from this channel sounding scheme, we can describe the wave propagation characteristics of channels between the transmitter and receiver so that the receiver could enhance not only the reliability but also the ability of communication link.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39A no.6
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• pp.333-342
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• 2014

The pilot assisted channel frequency response (CFR) estimation schemes for an OFDM-based system with virtual subcarriers are analyzed under the assumption that pilot symbols are located according to a comb-type pattern in the OFDM block. In particular, as the minimum mean square error (MMSE) based scheme aiming to directly predict the channel impulse response and the MMSE based scheme aiming to suppress the leakage have not been clearly compared, by proving that the mean square errors (MSEs) of the latter scheme is always larger than that of the former scheme, this paper shows that the former scheme is superior to the latter scheme. Moreover, the impact of the number of pilots on the performances of the MMSE and least-square based channel estimation schemes are investigated. The performance analyses of the presented schemes are confirmed by computer simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.6A
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• pp.676-684
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• 2008

In digital video broadcasting-terrestrial (DVB-T), which is the European digital terrestrial television standard, the orthogonal frequency division multiplexing (OFDM) has been adopted for signal transmission. The main reasons using OFDM are to increase the robustness against the frequency selective fading and impulse noise, and to use available bandwidth efficiently. However, channel variation within an OFDM symbol destroys orthogonality between subcarriers, resulting in inter-carrier interference (ICI), which increases an error floor in proportional to maximum Doppler spread. This paper provides an ICI analysis in both time and frequency domains while existing literatures analyze the ICI effects mainly in frequency domain and proposes the algorithms that estimate the channel impulse response and channel variation using least square (LS) algorithm which is the most simple channel estimation technique. And we propose adaptive channel estimation algorithm that estimates the velocity of terminals. The simulation results show that proposed algorithm has similar performance with about 1.5% computational complexity of noise and ICI reduction LS algorithm in low speed environments.

• Proceedings of the KIEE Conference
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• 2005.07d
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• pp.2986-2988
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• 2005

To improve the reliability of power-line communication systems, the measurement and analysis has been proceeded in many power-line channel environments. In spite of the wired channel, power line channel has many multi-paths that are changing with load-variation, line-interconnection, impedance mismatching and so on. We accordingly need an analysis method based on the multi-path channel impulse response. Recently, a method to describe the homogeneous Power-line network has been published[1]. In this paper the modified method that can describe both the homogeneous and non -homogeneous power-line network has presented.

• Proceedings of the IEEK Conference
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• 2003.07a
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• pp.418-421
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• 2003

In this thesis, a new channel estimation technique is proposed for orthogonal frequency division multiplexing (OFDM) over time varying channels. The channel estimation algorithm exploits the fact that the estimated channel impulse response (CIR) by using pilot signal is the average value of the CIR variation within an OFDM symbol period. With this fact, the CIR variation is simply estimated through lowpass interpolation of the CIRs of the adjacent OFDM symbols. For signal detection, a time domain equalizer is used in this thesis. Simulation results show that the proposed system improves the bit error rate (BER) over time varying channels.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.6
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• pp.32-41
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• 1996

We propose the improved GMSK (gaussian-filtered minimum shift keying) quadrature modulator using the FIR(finite impulse response )filter whose coefficients are obtained form the differnce of phase response, and design its ASIC (applicaton specific integrated circuit) which can be used for GSM (global system for mobile communication) digital cellular system and DCS 1800 (digital cellular system at 1800MHz) personal communication system. Input data become quantized I and Q channel 10 bit signal through cosine and sine ROM mapping after being filtered by the FIR filter whose normalized bandwidth is 0.3 and designed by considering intersymbol interference as well as sampling ratio. These two signals become the GMSK modulated I and Q channel signal through DAC (digital-to-analog converter) and 7th order analog chebyshev LPF(low pass filter) respectively. The difference between the ideal analog signal and its digitized signal is analyzed in terms of sampling noise, quantization noise, truncation noise and coefficient noise. And the effect of the LPF following the DAC is considered. The ASIC design of the GMSK quadrature modulator is also confirmed by an experiment.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.