• ETRI Journal
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• v.46 no.3
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• pp.392-403
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• 2024

Even though generalized frequency division multiplexing is an alternative waveform method expected to replace the orthogonal frequency division multiplexing in the future, its implementation must alleviate channel effects. Least-squares (LS), a low-complexity channel estimation technique, could be improved by using the discrete Fourier transform (DFT) without increasing complexity. Unlike the usage of the LS method, the DFT-based method requires the receiver to know the channel impulse response (CIR) length, which is unknown. This study introduces a simple, yet effective, CIR length estimator by utilizing LS estimation. As the cyclic prefix (CP) length is commonly set to be longer than the CIR length, it is possible to search through the first samples if CP is larger than a threshold set using the remaining samples. An adaptive scale is also designed to lower the error probability of the estimation, and a simple signal-to-interference-noise ratio estimation is also proposed by utilizing a sparse preamble to support the use of the scale. A software simulation is used to show the ability of the proposed system to estimate the CIR length. Due to shorter CIR length of rural area, the performance is slightly poorer compared to urban environment. Nevertheless, satisfactory performance is shown for both environments.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.69-72
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• 1999

In this paper, an efficient time domain equalization algorithm for discrete wavelet multitone(DWMT) data transmission is developed. In this algorithm, the time domain equalizer(TEQ) consists of two stages, i.e., the channel impulse response shortening equalizer(TEQ-S) in the first stage and the channel frequency flattening equalizer(TEQ-F) in the second stage. TEQ-S reduces the length of transmission channel impulse response to decrease intersymbol interference(ISI) followed by TEQ-F that enhances the channel frequency response characteristics to the level of an ideal channel, hence diminishes the bit error rate. TEQ-S is implemented using the least-squares(LS) method, while TEQ-F is designed by using the least mean-square(LMS) algorithm. Since DWMT system also requires of the frequency domain equalizer in order to further reduce ICI and ISI the hardware complexity is an another concern. However, by adopting an well designed and trained TEQ, the hardware complexity of the whole DWMT system can be greatly reduced.

• ETRI Journal
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• v.31 no.2
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• pp.162-172
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• 2009

A linear-prediction-based blind equalization algorithm for single-input single-output (SISO) finite impulse response/infinite impulse response (FIR/IIR) channels is proposed. The new algorithm is based on second-order statistics, and it does not require channel order estimation. By oversampling the channel output, the SISO channel model is converted to a special single-input multiple-output (SIMO) model. Two forward linear predictors with consecutive prediction delays are applied to the subchannel outputs of the SIMO model. It is demonstrated that the partial parameters of the SIMO model can be estimated from the difference between the prediction errors when the length of the predictors is sufficiently large. The sufficient filter length for achieving the optimal prediction is also derived. Based on the estimated parameters, both batch and adaptive minimum-mean-square-error equalizers are developed. The performance of the proposed equalizers is evaluated by computer simulations and compared with existing algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9A
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• pp.755-761
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• 2005

In this paper, we proposed a channel estimation method by impulse signal train in OFDM. In order to estimate the channel response, 4 impulse signals are generated and transmitted during one OFDM (Orthogonal Frequency Division Multiplexing) symbol. The intervals between the impulse signals are all equal in time domain. At the receiver, the impulse response signals are summed and averaged. And then, the averaged impulse response signal is zero padded and fast Fourier transformed to obtain the channel estimation. The BER performance of the proposed method is compared with those of conventional estimation method using the long training sequence in fast fading environments. The simulation results show that the proposed method improves by 3 dB in terms of Eb/No, compared with the conventional method.

• Journal of Broadcast Engineering
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• v.16 no.2
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• pp.207-215
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• 2011

A signal passed through multi-path channel suffers ISI(Inter-Symbol Interference) and severe distortions caused by channel delay spread and noise components at the SC-FDE(Single Carrier with Frequency Domain Equalizer) transmission. Conventional UW(Unique-Word) based SC-FDE iterative channel estimation improves channel estimation performance by smoothing estimated CIR(Channel Impulse Response) of the noise components outside the channel length at time domain and restoring the broken cyclic property through UW reconstruction. In this paper, we propose channel estimation scheme through noise suppression within channel length. To suppress the noise, we estimate noise standard deviation as estimated CIR of the noise components outside the channel length and make criteria of the noise standard deviation gain that doesn't affect the original signal samples. When estimated CIR samples within channel length are less than the criteria value using the noise standard deviation and gain, the noise components are removed. Simulation results show that the proposed channel estimation scheme brings good channel MSE(Mean Square Error) and good BER(Bit Error Rate) performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1B
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• pp.66-78
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• 2002

In this paper, we propose a new symbol offset estimation technique for an orthogonal frequency-division multiplexing (OFDM)-based wireless LAN. When both inter-symbol interference (ISI) and inter-channel interference (ICI) do not exist in an OFDM symbol, symbol offsets cause circular shifts in the estimated channel impulse response (CIR) by the amount of symbol offset. Also, the power delay profile of a typical multipath wireless channel can be modeled by exponentially decaying function, and most energy of multipath channel is concentrated at the beginning part of the CIR. Based on these properties, the proposed symbol offset estimation technique estimates the CIR, which is circularly shifted by the amount of symbol offset, and then calculates the partial mean power from the estimated impulse response by using a moving window with a finite length. And, symbol offset can be estimated from the index of a moving window having the maximal partial mean power. The proposed technique can reduce noise effect in the process of the CIR estimation, and remove ISI and ICI using repetitive training symbol structure in time-domain for minimum training overhead. The performances of the proposed symbol offset estimation technique in typical indoor channels are demonstrated by computer simulation.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.339-346
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• 2010

For the implementation of the crosstalk canceller, the filters with large length are needed, which is because that the length of the filters greatly depends on the length of the head-related impulse responses. In order to reduce the length of the crosstalk cancellation filters, many methods such as frequency warping, common acoustical pole and zero (CAPZ) modeling have been researched. In this paper, we propose a new method combining these two methods. To accomplish this, we design the filters using the CAPZ modeling on the warped domain, and then, we implement the filters using the poles and zeros de-warped to the linear domain. The proposed method provides improved channel separation performance through the frequency warping and significant reduction of the complexity through the CAPZ modeling. These are confirmed through various computer simulations.

• ETRI Journal
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• v.34 no.4
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• pp.511-517
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• 2012

The best linear unbiased estimator (BLUE) is most suitable for practical application and can be determined with knowledge of only the first and second moments of the probability density function. Although the BLUE is an existing algorithm, it is still largely unexplored and has not yet been applied to channel estimation in amplify and forward (AF)-based wireless relay networks (WRNs). In this paper, a BLUE-based algorithm is proposed to estimate the overall channel impulse response between the source and destination of AF strategy-based WRNs. Theoretical mean square error (MSE) performance for the BLUE is derived to show the accuracy of the proposed channel estimation algorithm. In addition, the Cram$\acute{e}$r-Rao lower bound (CRLB) is derived to validate the MSE performance. The proposed BLUE channel estimation algorithm approaches the CRLB as the length of the training sequence and number of relays increases. Further, the BLUE performs better than the linear minimum MSE estimator due to the minimum variance characteristic exhibited by the BLUE, which happens to be a function of signal-to-noise ratio.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.1978-1982
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• 2001

In a Multi-carrier modulation system, CP (Cyclic prefix) is inserted in the transmit tame in order to eliminate the ISI (Intersymbol Interference) and ICI (Interchannel Interference) caused by delay spread of a received signal, which in rum degrades the throughput of the system. TEQ (Time-domain equalizer) improves the system throughput by shortening the CIR (Channel Impulse Response) time and maintaining the CP length to the minimum regardless of the channel condition. In this paper, a new MMSE (Minimum Mean Square Error) TEQ algorithm is proposed and its performance is analyzed in order to speed up computing the optimum tap coefficients of the equalizer by employing Newton method.

• Journal of the Korean Institute of Telematics and Electronics
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• v.19 no.4
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• pp.38-48
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• 1982

Performance and realization of finite impulse response (FIR) digital filters that use an adaptive delta modulator (ADM) as an analog/digital converter have been studied. This filter requires no multiplication and offers many advantages over conventional PCM filters in low power consumption, small size, and cost effectiveness. Analytical formulas have been derived for the expected mean-squared errors and also for the word length necessary to achieve the desired performance. Computer simultation has been done to optimize the parameter values and to verify the results of performance analysis. In addition, design of FIR ADM digital filters for processing single and multi-channel signals has been considered.