• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.17-18
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• 2006

In this paper, we propose a robust detection scheme of OSTBCs with channel estimation errors over time-selective fading channels. Channel estimation errors are inevitable over time-selective fading channels and even small channel estimation errors dramatically degrade the performance of space-time block coding schemes. Therefore, it is desired to investigate the effect of channel estimation errors on the performance of the proposed detection scheme compared with the existing detection scheme. The proposed detection scheme minimizes noise enhancement and impact of channel estimation errors which occur in an existing detection scheme. It is shown by simulations that the proposed detection scheme performs better than the existing detection scheme over time-selective fading channels.

• Transactions of the Korean Society of Automotive Engineers
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• v.2 no.6
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• pp.29-37
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• 1994

Active noise control of random noise which propatate in the vehicle cabin as a form of spherical wave is the target of this study. In the previous study, the adaptive algorithm for adaptive controller is presented for the application in active noise control system. And for the preliminary study of adaptive active noise control in vehicle cabin as a real system, a computer simulation is performed on the effectiveness of the adaptive algorithm in the amplitude of the pressure fluctuation. This work studies the implementation of multi-channel feedforward adaptive algorithm for the reduction of the noise inside a vehicle cabin using a number of secondary sources derived by adaptive filtering of reference noise source. Multi-channel adaptive feedforward algorithm are verified in numerical simulation and semi-experimental justification of developed system is made on a domestic passenger car. In the results of semi-experimental study, the noise of specific region in the interior of automobile are reduced for the appreciabe sound pressure level in the operating engine rpm and finally this study suggests the capabilities of the real time active noise control in 3 dimensional acoustic fields.

• Proceedings of the KIEE Conference
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• 2004.05a
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• pp.65-67
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• 2004

Change of state of Channel between two wireless terminals which is caused by noise and multiple environmental conditions for happens frequently from the Wireles Network. So, When it is like that planning a wireless network protocol or performance analysis, it follows to change of state of time-varying channel and packet the analysis against a transmission efficiency is necessary. In this paper, analyzes transmission time of a packet and a packet in a time-varying and packet based Wireless Network. To reflecte the feature of the time-varying channel, we use a Signal Flow Graph model. From the model the mean of transmission time and the mean of queue length of the packet are analyzed in terms of the packet distribution function, the packet transmission service time, and the PER of the time-varying channel.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.11a
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• pp.455-461
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• 2000

A multichannel signal processing algorithm for generating real time virtual sound field was proposed. Evaluation of the system performance was done by an objective function that minimizes the difference between the real and generated signals at each control point. Since impulse responses at the surface of a rigid sphere show characteristics similar to those of real HRTF, a rigid sphere model was adequate to simulate the multichannel sound system. A two-channel system and two four-channel systems were studied with various combinations of source locations and speaker positions. The results show that a two-channel system has its best configuration when the angle spanned by the loudspeakers is less than $60^{\circ}$. In the case of four-channel systems, the overall performance was highly improved with one pair of speakers fixed at an optimal position. Left/right symmetry was a reasonable choice, but the additional front/back symmetry degraded the performance of system.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.289-290
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• 2007

In order to improve the performance of a speech recognition system in the presence of noise, we propose a noise robust front-end using two-channel speech signals by separating speech from noise based on the computational auditory scene analysis (CASA). The main cues for the separation are interaural time difference (ITD) and interaural level difference (ILD) between two-channel signal. As a result, we can extract 39 cepstral coefficients are extracted from separated speech components. It is shown from speech recognition experiments that proposed front-end has outperforms the ETSI front-end with single-channel speech.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.1
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• pp.44-53
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• 1999

In this paper, we propose a pilot based time-averaging channel estimation method and analyze error performances for efficient transmission of OFDM(Orthogonal Frequency Division Multiplexing) in multipath fading environment. Frequency domain channel estimations have been used in OFDM systems to compensate signal distortions due to fading on each subcarrier. The frequency domain estimation scheme uses scattered pilot to estimate channel response by simple interpolation. This implies that the estimated channel response includes signal distortions due to the noise. In this paper, we propose time-averaged channel estimation method to remove the distortion by noise. The proposed scheme can effectively remove noise components by taking time-average of the estimated channel response after estimating frequency domain channel. The computer simulations were performed to evaluate the performance of the proposed channel estimator. For the Rician channel, we compared the performance of the proposed method to that of a conventional one using channel estimation by gaussian interpolation when SER(Symbol Error Rate) = $10^{-4}$, and compared to perfect channel estimation case. The proposed method showed differences of 0.07 dB, 0.6 dB compared to perfect channel estimation and improvements of 1.7 dB, 1.9 dB for 16 QAM, 64 QAM respectively compared to conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.5C
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• pp.556-564
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• 2007

This paper proposes an efficient scheme to track a time variant channel induced by multi-path Rayleigh fading in mobile MIMO-OFDM systems with null sub-carriers. The proposed adaptive channel tracking scheme removes in the frequency domain the interfering signals of the other transmit (Tx) antennas by using a predicted channel frequency response before starting the channel estimation. Time domain channel estimation is then performed to reduce the additive white Gaussian noise (AWGN). The simulation results show that the proposed method is better than the conventional channel tracking method [3] in time varying channel environments. At a Doppler frequency of 300 Hz and bit error rates (BER) of 10-3, signal-to-noise power ratio (Eb/N0) gains of about 2.5 dB are achieved relative to the conventional channel tracking method [3]. At a Doppler frequency of 600 Hz, the performance difference between the proposed method and conventional one becomes much larger.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.9
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• pp.4390-4407
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• 2019

The paucity of pilot signals in Massive MIMO system is a vital issue. To accommodate substantial number of users, pilot signals are reused. This leads to interference, resulting in pilot contamination and degrades channel estimation at the Base Station (BS). Hence, mitigation of pilot contamination is exigency in Massive MIMO system. The proposed Time Shifted Pilot Signal Transmission with Pilot signal Hopping (TSPTPH), addresses the pilot contamination issue by transmitting pilot signals in non-overlapping time interval with hopping of pilot signals in each transmission slot. Hopping is carried by switching user to new a pilot signal in each transmission slot, resulting in random change of interfering users. This contributes to the change in channel coefficient, which leads to improved channel estimation at the BS and therefore enhances the efficiency of Massive MIMO system. In this system, Uplink Signal Power to Interference plus Noise Power Ratio (SINR) and data-rate are calculated for pilot signal reuse factor 1 and 3, by estimating the channel with Least Square estimation. The proposed system also reduces the uplink Signal power for data transmission of each User Equipment for normalized spectral efficiency with rising number of antennas at the BS and thus improves battery life.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4C
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• pp.418-424
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• 2007

This paper proposes a scheme to track channel response in OFDM systems with null sub-carriers. The proposed channel tracking scheme estimates the channel response first in the frequency domain by using the decision directed data. The time domain channel estimation is then performed to remove additive white Gaussian noise (AWGN) components further. Due to the channel estimation in the frequency domain, no inverse matrix calculation is required in the time domain channel estimation. Computational reduction in the proposed method is about 93%, compared with the conventional time domain channel estimation method. Mean square error (MSE) and bit error rate (BER) performances are evaluated by using computer simulation. The proposed method shows the same performance as that of the conventional time domain channel estimation even though the significant computational reduction.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.5
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• pp.649-655
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• 2022

In this paper, a determinant-based two-channel noise reduction method which utilizes speech presence probability (SPP) is proposed. The proposed method improves noise reduction performance from the conventional determinant-based two-channel noise reduction method in [7] by applying SPP to the Wiener filter gain. Consequently, the proposed method adaptively controls the amount of noise reduction depending on the SPP. For performance evaluation, the segmental signal-to-noise ratio (SNR), the perceptual evaluation of speech quality, the short time objective intelligibility, and the log spectral distance were measured in the simulated noisy environments considered various types of noise, reverberation, SNR, and the direction and number of noise sources. The experimental results presented that determinant-based methods outperform phase difference-based methods in most cases. In particular, the proposed method achieved the best noise reduction performance maintaining minimum speech distortion.