• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.11
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• pp.21-27
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• 2002

When decision-directed channel estimation is used for QAM-OFDM systems, the optimal shape of the two-dimensional filter depends on the amplitudes of the modulated symbols as well as the channel characteristics such as delay spread, Doppler frequency, and signal-to-noise ratio. While most conventional channel estimation methods did not consider the amplitudes of the modulated symbols because of the large computational complexity, we propose a simple channel estimation method for multi-level-amplitude-modulated systems. The proposed method can effectively reduce the noise variance of the estimates with small-sized filtering and there is a possibility of reducing the implementation cost and producing better results by avoiding the bias due to large filter sizes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7A
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• pp.1028-1035
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• 1999

To increase the throughput of data transmission in burst-mode TDMA communication systems and also to get a good BER performance at the same time, it is essential to rapidly acquire the carrier while keeping the desirable tracking performance. To achieve this goal, in this paper, a new decision-directed carrier recovery algorithm is presented. The proposed scheme does not incorporate the PLL and suppress the Gaussian random process of input noise by the pre-stage low pass filter so as to get both the fast acquisition and a good performance. Through computer simulations, the performance of the scheme is analyzed with respect to the acquisition time and bit error rate. The cycle slip in the proposed scheme is seldom observed at very low SNR environment in contrast to the previous proposed one. Because of this merit, it is not required to do the differential encoding and decoding in the proposed scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.82-91
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• 2005

This paper introduces an efficient affine projection algorithm(APA) using iterative hyperplane projection. Among various fast converging adaptation algorithms, APA has been preferred to be employed for various applications due to its inherent effectiveness against the rank deficient problem. However, the amount of complexity of the conventional APA could not be negligible because of the accomplishment of sample matrix inversion(SMI). Moreover, the 'shifting invariance property' usually exploited in single channel case does not hold for the application of space-time decision-directed equalizer(STDE) deployed in single-input-multi-output(SIMO) systems. Thus, it is impossible to utilize the fast adaptation schemes such as fast transversal filter(FlF) having low-complexity. To accomplish such tasks, this paper introduces the low-complexity APA by employing hyperplane projection algorithm, which shows the excellent tracking capability as well as the fast convergence. In order to confirm th validity of the proposed method, its performance is evaluated under wireless SIMO channel in respect to bit error rate(BER) behavior and computational complexity.

• The Journal of the Acoustical Society of Korea
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• v.25 no.8
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• pp.383-388
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• 2006

This Paper presents a novel approach to single channel microphone speech enhancement in noisy environments. Widely used noise reduction techniques based on the spectral subtraction are generally expressed as a spectral gam depending on the signal-to-noise ratio (SNR). The well-known decision-directed(DD) estimator of Ephraim and Malah efficiently reduces musical noise under the background noise conditions, but generates the delay of the a prioiri SNR because the DD weights the speech spectrum component of the Previous frame in the speech signal. Therefore, the noise suppression gain which is affected by the delay of the a priori SNR, which is estimated by the DD matches the previous frame rather than the current one, so after noise suppression. this degrades the noise reduction performance during speech transient periods. We propose a computationally simple but effective speech enhancement technique based on the sigmoid type function for the weight Parameter of the DD. The proposed approach solves the delay problem about the main parameter, the a priori SNR of the DD while maintaining the benefits of the DD. Performances of the proposed enhancement algorithm are evaluated by ITU-T p.862 Perceptual Evaluation of Speech duality (PESQ). the Mean Opinion Score (MOS) and the speech spectrogram under various noise environments and yields better results compared with the fixed weight parameter of the DD.

• Journal of the Korea Society of Computer and Information
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• v.21 no.4
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• pp.47-54
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• 2016

Cellular formation and layout problem has been known as a NP-hard problem. Because of the algorithm that can be solved exact solution within polynomial time has been unknown yet. This paper suggests a systematic method to be obtain of 2-degree partial directed path from the frequency of consecutive forward order. We apply the modified Kruskal algorithm of minimum spanning tree to be obtain the partial directed path. the proposed reverse constructive algorithm can be solved for this problem with O(mn) time complexity. This algorithm performs same as best known result of heuristic and metaheuristic methods for 4 experimental data.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.1 no.2
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• pp.199-207
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• 1997

The Maximum Likelihood Sequence Estimation scheme is modified to improve the error performance of the correlative coding in the Tamed FM. To remove intersymbol interference, the Decision Feedback Equalization scheme with the delayed LMS algorithm and the Viterbi algorithm(10-symbol delay) in the delayed adaptive equalization are proposed for the performance of decision-directed adaptive equalization under the High Frequency channels, and the condition of convergence is analyzed.

• ETRI Journal
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• v.36 no.5
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• pp.721-729
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• 2014

We propose a novel phase-based method for single-channel speech enhancement to extract and enhance the desired signals in noisy environments by utilizing the phase information. In the method, a phase-dependent a priori signal-to-noise ratio (SNR) is estimated in the log-mel spectral domain to utilize both the magnitude and phase information of input speech signals. The phase-dependent estimator is incorporated into the conventional magnitude-based decision-directed approach that recursively computes the a priori SNR from noisy speech. Additionally, we reduce the performance degradation owing to the one-frame delay of the estimated phase-dependent a priori SNR by using a minimum mean square error (MMSE)-based and maximum a posteriori (MAP)-based estimator. In our speech enhancement experiments, the proposed phase-dependent a priori SNR estimator is shown to improve the output SNR by 2.6 dB for both the MMSE-based and MAP-based estimator cases as compared to a conventional magnitude-based estimator.

• Journal of Satellite, Information and Communications
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• v.10 no.1
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• pp.29-32
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• 2015

It has been a big trend of the convergence technologies about communication systems and vehicular industry to improve safety and convenience. To achieve a number of infotainment vehicular applications, vehicles should transmit information with high reliability. A robust and accurate channel estimation scheme is of great importance to achieve the goal. In this paper, we present a novel enhanced decision-directed channel estimation scheme called FADP (Frequency Averaging Data Pilot) for dynamic time-varying vehicular channels in IEEE 802.11p. We use linear averaging filtering in frequency domain, and utilize the correlation characteristic of the channels between the adjacent two data symbols, update the CR in time domain to get more accuracy. Finally, analysis and simulation results reveal that compared with exist schemes, the proposed scheme has a good performance in mean square error (MSE) and bit error rate (BER).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6A
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• pp.660-666
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• 2004

Transmitted data may be compensated by using estimated channel values that are obtained with pilot symbols in OFDM-CDMA systems. Generally, a USE (Minimum Mean-Squared Error) estimator using correlations between pilot symbols gives good results, but its structure is so complicated. Starting with a modification of PA (Pilot-Aided) algorithm using pilot symbols and PADD (Pilot-Aided Decision-Directed) algorithm using both pilot and data symbols, a new channel estimation algorithm with more simpler structure is proposed. The performance of this algorithm is evaluated with varying mobile speed in a Ralyleigh multipath fading environment through computer simulations. The simulation results show that the proposed channel estimation algorithm outperforms a conventional PA algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2397-2404
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• 2012

Underwater acoustic communication has multipath error because of reflection by sea-level and sea-bottom. The multipath of underwater channel causes signal distortion and error floor. In order to improve the performance, it is necessary to employ an iterative coding scheme. Among the iterative coding scheme, turbo codes and LDPC codes are dominant channel coding schemes in recent. This paper concluded that turbo coding scheme is optimal for underwater communications system in aspect to performance, coded word length, and equalizer combining. Also, decision directed phase recovery was used for correcting phase offset induced by multipath. Based on these algorithms, we confirmed the performance in the environment of oceanic experimentation.