• ETRI Journal
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• v.43 no.6
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• pp.966-977
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• 2021

Filterbank multicarrier with offset quadrature amplitude modulation (FBMC-OQAM) is an attractive alternative to the orthogonal frequency division multiplexing (OFDM) modulation technique. In comparison with OFDM, the FBMC-OQAM signal has better spectral confinement and higher spectral efficiency and tolerance to synchronization errors, primarily due to per-subcarrier filtering using a frequency-time localized prototype filter. However, the filtering process introduces intrinsic interference among the symbols and complicates channel estimation (CE). An efficient way to improve the CE in FBMC-OQAM is using a technique known as windowed frequency domain averaging (FDA); however, it requires a priori knowledge of the window length parameter which is set based on the channel's frequency selectivity (FS). As the channel's FS is not fixed and not a priori known, we propose a k-nearest neighbor-based machine learning algorithm to classify the FS and decide on the FDA's window length. A comparative theoretical analysis of the mean-squared error (MSE) is performed to prove the proposed CE scheme's effectiveness, validated through extensive simulations. The adaptive CE scheme is shown to yield a reduction in CE-MSE and improved bit error rates compared with the popular preamble-based CE schemes for FBMC-OQAM, without a priori knowledge of channel's frequency selectivity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.11C
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• pp.1111-1117
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• 2003

The maximum Doppler frequency, or equivalently, the mobile speed is very useful information to optimize the performance of many wireless communication systems. However, the performance of a maximum Doppler frequency estimator is limited since it requires an estimate of the signal-to-noise ratio (SNR) of the channel environment. In this paper, the improved method for the maximum Doppler frequency estimations based on the decision feedback Doppler adaptive band-limit (DF-DABL) method is proposed. To reduce the effect of additive noise, the proposed algorithm uses a novel Doppler adaptive band-limit (DABL) technique. The distortion due to the additive noise is drastically removed by the proposed DF-DABL method. Especially, the DF-DABL method does not need any other channel information such as SNR.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.10C
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• pp.976-982
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• 2005

In this paper, a channel-adaptive beamforming method is proposed for OFDMA (Orthogonal Frequency Division Multilexing Access) systems with smart antenna, in which the size of a cluster is determined adaptively depending on the frequency selectivity of the channel. The proposed method consists of 4 steps: initial channel estimation, refinement of channel estimates, region-splitting, and computation of weight vector for each region. In the proposed method, the size of a cluster for resource unit is determined adaptively according to a region-splitting criterion. It is shown by simulation that the proposed method shows good performances in both frequency-flat and frequency-selective channels.

• Speech Sciences
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• v.8 no.2
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• pp.73-81
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• 2001

In this paper, we focus on the real time implementation of the single channel adaptive noise canceller(ANC) by using TMS320C30 EVM board. The implemented single channel adaptive noise canceller is based on a reference paper [1] in which it is simulated by using the recursive average magnitude difference function(AMDF) to get a properly delayed input speech on a sample basis as a reference signal and normalized least mean square(NLMS) algorithm. To certify results of the real time implementation, we measured the processing time of the ANC and enhancement ratio according to various signalto-noise ratios(SNRs). Experimental results demonstrate that the processing time of the speech signal of 32ms length with delay estimation of every 10 samples is about 26.3 ms, and almost the same performance as given in [1] is obtained with the implemented system.

• Journal of Electrical Engineering and information Science
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• v.1 no.1
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• pp.17-22
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• 1996

Adaptive code rate change schemes in DS-SSMA systems are proposed. In the proposed schemes, the error correcting code rate is changed according to the channel states. Two channel states having significant effects on the bit error probability are considered: one is the effective number of users, and the other is the fading environment. These channel states are estimated based on retransmission requests. The criterion for the change of the code rate is to maximize the throughput under given error bound.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.56 no.7
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• pp.1331-1338
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• 2007

A new method for detecting and eliminating the Electrocardiogram(ECG) artifact from the scalp Electroencephalogram(EEG) is proposed. Based on the single channel EEG, the proposed method consists of 4 procedures: emphasizing the R-wave of ECG artifact from EEG using the least squares acceleration(LSA) filter, detecting the R-wave from the LSA filtered EEG using the phase space method and R-R interval, generating the delayed impulse synchronized to the R-wave and elimination of the ECG artifacts based on the adaptive digital filter using the impulse and raw EEG. The performance of the proposed method was evaluated in the two separating parts of R-wave detection and, ECG estimation and elimination from EEG. In the R-wave detection, the proposed method showed the mean error rate of 6.285(%). In the ECG estimation and elimination using simulated and/or real EEG recordings, we found that the ECG artifacts were successfully estimated and eliminated in comparison with the conventional multi-channel techniques, in which independent component analysis and ensemble average method are used. From this we can conclude that the proposed method is useful for the detecting and eliminating the ECG artifact from single channel EEG and simple for ambulatory/portable EEG monitoring system.

• International journal of advanced smart convergence
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• v.9 no.1
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• pp.177-184
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• 2020

In this paper, we studied to direction of arrival (DoA) estimation to use DoA and optimum weight algorithms in coherent interference channels. The DoA algorithm have been considerable attention in signal processing with coherent signals and a limited number of snapshots in a noise and an interference environment. This paper is a proposed method for the desired signal estimation using MUSIC algorithm and adaptive beamforming to compare classical subspace techniques. Also, the proposed method is combined the updated weight value with LCMV beamforming algorithm in adaptive antenna array system for direction of arrival estimation of desired signal. The proposed algorithm can be used with combination to MUSIC algorithm, linearly constrained minimum variance beamforming (LCMV) and the weight value method to accurately desired signal estimation. Through simulation, we compare the proposed method with classical direction of in order to desired signals estimation. We show that the propose method has achieved good resolution performance better that classical direction arrival estimation algorithm. The simulation results show the effectiveness of the proposed method.

• Journal of Communications and Networks
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• v.14 no.5
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• pp.501-509
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• 2012

This paper proposes a joint timing synchronization, channel estimation, and data detection for the impulse radio ultra-wideband systems. The proposed timing synchronizer consists of coarse and fine timing estimation. The synchronizer discovers synchronization points in two stages and performs adaptive threshold based on the maximum pulse averaging and maximum (MAX-PA) method for more precise synchronization. Then, iterative channel estimation is performed based on the discovered synchronization points, and data are detected using the selective rake (S-RAKE) detector employing maximal ratio combining. The proposed synchronizer produces two signals-the start signal for channel estimation and the start signal for start frame delimiter (SFD) detection that detects the packet synchronization signal. With the proposed synchronization, channel estimation, and SFD detection, an S-RAKE receiver with binary pulse position modulation binary phase-shift keying modulation was constructed. In addition, an IEEE 802.15.4a channel model was used for performance comparison. The comparison results show that the constructed receiver yields high performance close to perfect synchronization.

• Proceedings of the IEEK Conference
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• 2003.07a
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• pp.234-237
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• 2003

The accuracy of channel estimation significantly affects the performance of coherent OFDM receiver. It is desirable to employ a good channel estimator while requiring low implementation complexity. In this paper, we propose a channel estimator that employs a simple two-dimensional (2-D) moving average (MA) filter as the channel estimation filter. The optimum tap size of the 2-D MA FIR filter is analytically designed in the time and frequency domain in association with the channel condition and pilot signal to interference power ratio. The analytic results can be applied to the design of adaptive channel estimator. Finally, the performance of the proposed channel estimator is verified by computer simulation.

• The Journal of the Acoustical Society of Korea
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• v.18 no.1E
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• pp.3-6
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• 1999

In digital mobile communication system, the transmission errors affect the quality of output speech seriously. There are many error concealment techniques using a posteriori probability which provides information about any transmitted parameter. They need knowledge about channel transition probability as well as the 1st order Markov transition probability of codec parameters for estimation of transmitted parameters. However, in applications of mobile communication systems, the channel transition probability varies depending on nonstationary channel characteristics. The mismatch of designed channel transition probability of the estimator to actual channel transition probability degrades the performance of the estimator. In this paper, we proposed a new parameter estimator which adapts to the channel characteristics using short time average of maximum a posteriori probabilities(MAPs). The proposed scheme, when applied to the LSP parameter estimation, performed better than the conventional estimator which do not adapt to the channel characteristics.