• Transactions of the Korean Society of Automotive Engineers
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• v.3 no.6
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• pp.23-29
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• 1995

In this study, Active noise controlis implemented with single channel and multi-channel adaptive algorithm in 3 dimensional reverberant enclosure sound. field, which occurrs in complicated acoustic mode. First, for the one case excited with the resonant frequency of an enclosure, a target of control and the other cases excited with band-pass filtered random noise(100~400Hz), it is implemented to control reducing interior noise of enclosure with single channel and realtime multi-channel adaptive algorithm for global noise reduction in enclosure.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.8
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• pp.3498-3511
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• 2016

We investigate the channel state information (CSI) in multi-input multi-output (MIMO) cooperative networks that employ the amplify-and-forward transmission scheme. Least squares and expectation conditional maximization have been proposed in the system. However, neither of these two approaches takes advantage of channel sparsity, and they cause estimation performance loss. Unlike linear channel estimation methods, several compressed channel estimation methods are proposed in this study to exploit the sparsity of the MIMO cooperative channels based on the theory of compressed sensing. First, the channel estimation problem is formulated as a compressed sensing problem by using sparse decomposition theory. Second, the lower bound is derived for the estimation, and the MIMO relay channel is reconstructed via compressive sampling matching pursuit algorithms. Finally, based on this model, we propose a novel algorithm so called sparsity adaptive expectation maximization (SAEM) by using Kalman filter and expectation maximization algorithm so that it can exploit channel sparsity alternatively and also track the true support set of time-varying channel. Kalman filter is used to provide soft information of transmitted signals to the EM-based algorithm. Various numerical simulation results indicate that the proposed sparse channel estimation technique outperforms the previous estimation schemes.

• Journal of KIISE:Information Networking
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• v.30 no.3
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• pp.316-328
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• 2003

In this paper, we propose and analyze two adaptive channel allocation schemes for supporting multimedia traffics in hierarchical cellular systems. It is guaranteed to satisfy the required quality of service of multimedia traffics according to their characteristics such as a mobile velocity for voice calls and a delay tolerance for multimedia calls. In the scheme 1, only slow-speed voice calls are allowed to overflow from macrocell to microcell and only adaptive multimedia calls can overflow from microcell to macrocell after reducing its bandwidth to the minimum channel bandwidth. In the scheme II, in addition to the first scheme, non-adaptive multimedia calls can occupy the required channel bandwidth through reducing the channel bandwidth of adaptive multimedia calls. The proposed scheme I is analyzed using 2-dimensional Markov model. Through computer simulations, the analysis model and the proposed schemes are compared with the fixed system and two previous studies. In the simulation result, it is shown that the proposed schemes yield a significant improvement in terms of the forced termination probability of handoff calls and the efficiency of channel usage.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.29 no.8
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• pp.608-614
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• 2018

This paper describes the synthesis method for an adaptive SLB channel, which is robust to interference in the ULA radar system. The SLB channel based on the spatial DLC can be synthesized simply and is effective in blanking the signal coming from the sidelobe. We combined it with adaptive beamforming, which removes the strong interference using its correlation matrix. The adaptive SLB channel would suppress the interference below the noise, so it has good performance in an interference environment. This research will be applicable to planar array systems.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.517-522
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• 2004

In this paper, we have analyzed the performance enhancement of Orthogonal Frequency Division Multiplexing/Quadrature Phase Shift Keying Modulation-Digital Microwave Radio(OFDM/QPSK-DMR) system using Band Limited-Pulse Shaping Filter(BL-PSF) over microwave channel environments. For performance enhancement, the one-tap adaptive equalizer is adopted in the OFDM/QPSK-DMR system and than both BER and signature curve performance are compared with those of single carrier DMR system. Computer simulations confirm that the OFDM/QPSK-DMR system using 16 sub-carrier increase the fade margin about 2 dB over microwave channel environments and that of performance using one-tap adaptive equalizer is highly increased the fade margin as the number of sub-carriers is larger.

• 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.

• Journal of Communications and Networks
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• v.5 no.1
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• pp.25-32
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• 2003

4G mobile communication system requires the throughput of 10-100Mbps. Adaptive modulated OFDM system is promising technique for increasing the throughput. In the pilot symbol assisted high-speed packet transmission system, the data symbol duration is generally considered to be small compared to the coherence time. However, OFDM symbol duration is longer than the symbol duration of a single carrier system, so that the packet duration of the pilot symbol assisted high speed packet transmission system is long. In this case, the change of channel conditions is too fast to be accurately estimated by channel estimator at the receiver in high Doppler frequency, so that many errors occur during demodulation, especially with the data symbols at the end of each packet. In this paper, we consider the BER at various instantaneous $E_b/N_o$ that includes the demodulation errors in high Doppler frequency. When the coherence time is ten times longer than the duration of a single packet, the channel can be closely approximated as an AWGN channel. Otherwise, the approximation breaks down and the above-mentioned errors that occur during demodulation must be taken into consideration. In this paper, we propose the pilot symbol assisted high speed packet transmission system based on adaptive OFDM using a novel lookup table to consider the demodulated errors and evaluate the throughput performance.

• ETRI Journal
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• v.38 no.6
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• pp.1190-1196
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• 2016

We propose a new bandpass filter (BPF)-based online channel normalization method to dynamically suppress channel distortion when the speech and channel noise components are unknown. In this method, an adaptive modulation frequency filter is used to perform channel normalization, whereas conventional modulation filtering methods apply the same filter form to each utterance. In this paper, we only normalize the two mel frequency cepstral coefficients (C0 and C1) with large dynamic ranges; the computational complexity is thus decreased, and channel normalization accuracy is improved. Additionally, to update the filter weights dynamically, we normalize the learning rates using the dimensional power of each frame. Our speech recognition experiments using the proposed BPF-based blind channel normalization method show that this approach effectively removes channel distortion and results in only a minor decline in accuracy when online channel normalization processing is used instead of batch processing

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.3
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• pp.1047-1062
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• 2018

In this paper the problem of massive multiple input multiple output (MIMO) channel estimation with sparsity aware adaptive algorithms for $5^{th}$ generation mobile systems is investigated. These channels are shown to be non-stationary along with being sparse. Non-stationarity is a feature that implies channel taps change with time. Up until now most of the adaptive algorithms that have been presented for channel estimation, have only considered sparsity and very few of them have been tested in non-stationary conditions. Therefore we investigate the performance of several newly proposed sparsity aware algorithms in these conditions and finally propose an enhanced version of RZA-LMS/F algorithm with variable threshold namely VT-RZA-LMS/F. The results show that this algorithm has better performance than all other algorithms for the next generation channel estimation problems, especially when the non-stationarity gets high. Overall, in this paper for the first time, we estimate a non-stationary Rayleigh fading channel with sparsity aware algorithms and show that by increasing non-stationarity, the estimation performance declines.

• Journal of Korea Multimedia Society
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• v.10 no.12
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• pp.1585-1590
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• 2007

This paper introduces adaptive PRML (Partial Response Maximum Likelihood) architecture with PR (a,b,c,d,e) channel type for the improved readability of high-density optical discs with capacity greater than 30GB. The proposed PRML architecture consists of an adaptive equalizer, a Viterbi detector and a channel identifier. Detailed description for each component is included. The architecture is implemented in chip and also confirmed its performance on the test board mounting the chip. Test results show that the proposed 5-tap PRML architecture is well operated, and less than $2{\times}10^{-4}$ of BER (Bit Error Rate) is achieved with radial and tangential tilt margin of ${\pm}0.6^{\circ}$ on self-made 30GB BD at 1x speed.