• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.23 no.4
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• pp.491-498
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• 2012

Recently, many papers have been proposed in order to improve the OFDM system performance in T-DMB DOCR (Digital On Channel Repeater), by using removing the feedback signal so that the transmitter power can be increased or by using the equalizer to remove ICI. Despite these efforts, however, signal quality at the receiving terminal has not been improved because of constellation smearing in T-DMB DOCR. In this paper, in order to suppress constellation smearing, we propose an effective equalizer algorithm that can improve system performance. We perform adaptive channel estimation and non-coherent decision directed noise cancellation method that can estimate the channel subsequently during data symbols period in the frequency domain. So we can obtain better quality of the signal at the receiving terminal. In order to secure QoS(Quality of Service) required in T-DMB handsets, we evaluate SNR and BER in T-DMB DOCR(Digital On Channel Repeater) and verified by simulation. In this simulation results, this system is satisfied the performance of BER=$10^{-5}$ at less than SNR=14 dB at the receiver after compensation of phase noise -18 dBc.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1
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• pp.77-83
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• 2009

This paper proposes a new noise-biased compensation of minimum statistics(MS) method using a nonlinear function and a priori speech absence probability(SAP) for speech enhancement in non-stationary noisy environments. The minimum statistics(MS) method is well known technique for noise power estimation in non-stationary noisy environments. It tends to bias the noise estimate below that of true noise level. The proposed method is combined with an adaptive parameter based on a sigmoid function and a priori speech absence probability (SAP) for biased compensation. Specifically. we apply the adaptive parameter according to the a posteriori SNR. In addition, when the a priori SAP equals unity, the adaptive biased compensation factor separately increases ${\delta}_{max}$ each frequency bin, and vice versa. We evaluate the estimation of noise power capability in highly non-stationary and various noise environments, the improvement in the segmental signal-to-noise ratio (SNR), and the Itakura-Saito Distortion Measure (ISDM) integrated into a spectral subtraction (SS). The results shows that our proposed method is superior to the conventional MS approach.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2E
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• pp.71-80
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• 2002

This paper suggests the Cardioid beamforming algorithm of the doublet sensors employing DIFAR (directional frequency analysis and recording) sensor signals in the frequency domain. The algorithm enables target bearing estimation using the signals from directional sensors. The algorithm verifies its applicability by successfully estimating bearings of a target projecting ten narrow-band signals in shallow water. The estimated bearings agree very well with those from GPS (global positioning system) data. Assuming the bearings from GPS data to be real values, the estimation errors are analyzed statistically. The histogram of estimation errors in each frequency have Gaussian shape, the mean and standard deviation dropping in the ranges -1.1°∼ 6.7°and 13.3∼43.6°, respectively. Estimation errors are caused by SNR (signal to noise ratio) degradation due to propagation loss between the source and receiver, daily fluctuating geo-magnetic fields, and non-stationary background noises. If multiple DIFAR systems are employed, in addition to bearing, range information could be estimated and finally localization or tracking of a target is possible.

• The Journal of the Acoustical Society of Korea
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• v.25 no.4E
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• pp.154-158
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• 2006

In this paper, we suggest a noise estimation method for speech enhancement in nonstationary noisy environments. The proposed method consists of the following two main processes. First, in order to receive fewer affect of variable signals, a best fitting regression line is used, which is obtained by applying a least squares method to coefficient magnitudes in a node with a uniform wavelet packet transform. Next, in order to update the noise estimation efficiently, a differential forgetting factor and a correlation coefficient per subband are used, where subband is employed for applying the weighted value according to the change of signals. In particular, this method has the ability to update the noise estimation by using the estimated noise at the previous frame only, without utilizing the statistical information of long past frames and explicit nonspeech frames by voice activity detector. In objective assessments, it was observed that the performance of the proposed method was better than that of the compared (minima controlled recursive averaging, weighted average) methods. Furthermore, the method showed a reliable result even at low SNR.

• International Journal of Naval Architecture and Ocean Engineering
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• v.7 no.2
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• pp.426-434
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• 2015

In this paper we proposed a dada-aided Doppler estimation method for underwater acoustic communication. The training sequence is non-dedicate, hence it can be designed for Doppler estimation as well as channel equalization. We assume the channel has been equalized and consider only flat-fading channel. First, based on the training symbols the theoretical received sequence is composed. Next the least square principle is applied to build the objective function, which minimizes the error between the composed and the actual received signal. Then an iterative approach is applied to solve the least square problem. The proposed approach involves an outer loop and inner loop, which resolve the channel gain and Doppler coefficient, respectively. The theoretical performance bound, i.e. the Cramer-Rao Lower Bound (CRLB) of estimation is also derived. Computer simulations results show that the proposed algorithm achieves the CRLB in medium to high SNR cases.

• The Bulletin of The Korean Astronomical Society
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• v.37 no.2
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• pp.157.2-157.2
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• 2012

Direct-conversion receivers often suffer from a DC-offset that is a by-product of the direct conversion process to baseband. In general, a basic approach to reduce the DC-offset is to do simple average of the baseband signal and remove the DC by subtracting the average. However, this gives rise to a residual DC offset which degrades the performance when the receiver adopts the coding schemes with high coding rates such as 8-PSK. Therefore, more advanced methods should be additionally required for better performance. While the training sequences are basically designed to have good auto-correlation properties to facilitate the channel estimation, they may be not good for the simultaneous estimation of the channel response and the DC-offset. Also the DC offset compensation under a bad condition does not give good results due to the estimation error. Correspondingly, the proposed scheme employs the two important points. First, the training sequence codes are divided into two groups by MSE(Mean Squared Errors) for estimating the channel taps and then SNR calculated from each group is compared to predefined threshold to do fine DC-offset estimation. Next, ON/OFF module is applied for preventing performance degradation by large estimation error under severe channel conditions. The simulation results of the proposed scheme shows good performances compared to the existing algorithm. As a result, this scheme is surely applicable to the receiver design in many communications systems.

• The Journal of the Korea Contents Association
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• v.9 no.5
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• pp.84-90
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• 2009

In this paper, the new channel estimation was proposed that have the low complexity and high performance using Fuzzy inference method uses recently from various field for estimation about uncertainty in channel estimation of OFDM. Proposed method is channel estimation performance improve, calculation and interpolation for statistics character of channel using the pilot before LS channel estimation by Fuzzy inference method. Simulation result in QPSK proposed channel estimation method shows the enhancement of 5.5dB compared to the LS channel estimation and the deterioration of 1.3dB compared to the MMSE channel estimation in mean square error point $10^{-3}$. symbol error rate shows similarity performance the MMSE $10^{-1.96}$, proposed channel estimation $10^{-1.93}$ and enhancement of $10^{-0.35}$ compared to the LS channel estimation in signal to noise ratio point 20dB.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.199-200
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• 2008

In this paper, efficient time synchronization scheme for OFDM based WLAN system and its performance simulation results are presented. Assuming AGC and packet detection is done within 7 short training symbols. This scheme consists of coarse and fine estimation, and exhibits robustness over fading and AWGN channel. The presented synchronization scheme achieves the success rate of about 96% over the SNR of 5 dB.

• The Journal of the Acoustical Society of Korea
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• v.17 no.1
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• pp.55-59
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• 1998

본 논문에서는 잡음에 강한 음성 인식기를 위한 음성의 특징 추출에 관해서 살펴 보았다. 지금까지의 음성 인식기는 조용한 실험실 환경하에서 학습이 이루어지나 실제 테스 트는 여러 가지 환경에서 이루어지므로, 이러한 환경 변화에 따라 음성인식 시스템의 성능 이 감소함을 보여왔다. 이를 보완하기 위해 여러 가지 연구가 진행되고 있으나 본 연구에서 는 음성의 특징 추출 부분에서 순시적인 신호대 잡음비 예측과 잡음에 강한(RASTA)처리를 하므로써 인식율을 향상시켰다.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1983.10a
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• pp.85-87
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• 1983

In this paper, a improved whitening filter for narrowband interference rejetion in PN spread spectrum communication systems is proposed. This whitening filter has the feedback from the output to the input of adaptive transversal filter and is superior to conventional LMS whitening filter in respect to SNR and power spectrum since the calculation of tap weight is based on maximum likelihood estimation