• Journal of the Institute of Electronics Engineers of Korea TC
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• v.49 no.5
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• pp.7-14
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• 2012

The quantized beamforming systems always need the channel state information, which must be quantized into a finite set of vectors (named codebook), and feedback only sends the index representing the desired vector. Thereby it minimized the impact of feedback errors, caused by feedback overhead and delay. In this regard, index assignment (IA) methods, an exhaustive-search and group-based schemes, have been presented for minimizes the performance degradation without additional feedback bits. In this paper, we proposed enhanced group-based IA method, which used the optimal codebook design with chordal distance, having the adaptive properties in application of the existing IA methods. When the number of transmit antennas is 4 and LTE codebook is used, Monte-Carlo simulations verify that the proposed scheme has a power advantage of 0.5~1dB to obtain the same bit error rate than methods without IA, and it has 0.1~0.2 dB better performance compared with the existing IA methods over same environment.

• 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.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.55 no.3
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• pp.119-123
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• 2006

The orthogonal frequency-division multiplexing (OFDM) technique is well known to be robust against the frequency-selective fading in wireless channels. It is due to the exploitation of a guard interval that is inserted at beginning of each OFDM symbol. Based on the conventional OFDM and a polyphase filtered orthogonal frequency division multiplexing (PF-OFDM) technique, we developed an adaptive beamforming algorithm for antenna arrays. The proposed algorithm would lead to an efficient use of channel, since it is possible to eliminate a guard interval and also easily suppress interchannel interference at the same time. In this paper, a series of computer simulations have been provided to show the performance of the proposed system.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.384-388
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• 2008

Multiple antenna technology, such as multiple input multiple output (MIMO), beamforming is one of the most promising technology for broadband wireless communication. In OFDM environment, we found that as the number of paths is increased, smart antenna system cannot fully exploit beamforming gain. In this paper, we propose a beamforming scheme and analyze the performance of the proposed beamforming scheme in WiBro smart antenna system. WiBro is an OFDMA-based multiple access service for wireless broadband multimedia environment. The proposed beamforming technique for WiBro smart antenna system can effectively improve the performance of the systems in multipath fading environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.8
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• pp.890-895
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• 2016

As the wireless communication technique is developing rapidly, the use of smart devices is increasing. Due to gradually increasing data traffic, a new area, more than 6GHz of bandwidth to increase capacity of the network, has been studied. Millimeter Wave(MmWave) communications utilizes the bandwidth above 6GHz, which makes it possible to achieve one gigabit per second data rate. To overcome the path loss due to the smaller wavelength, the mass of the antenna arrangement is used. This paper presents an algorithm that maximizes the spectral efficiency of the system in the pre-coding process using a hybrid beamforming. Also it is suggested with the optimization of the number of beams that maximizes the spectral efficiency was maximized by the propose method.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.20 no.3
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• pp.248-256
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• 2009

EB(Eigen-Beamforming) has widely been applied to MIMO(Multiple-Input Multiple-Output) systems to form beams which maximize the effective signal-to-interference plus noise ratio(SINR) of the receiver using the singular value decomposition(SVD) of the MIMO channel. However, the signal detection performance for the mobile station near the cell boundary is severely degraded and the transmission efficiency decreases due to the influence of the interference signal from the adjacent cells. In this paper, we propose an adaptive interference mitigation method for the EB transmission, and evaluate the reception performance. In particular, a reception strategy which adaptively utilizes optimal combining(OC) and minimum mean-squared error for Intercell spatial demultiplexing(MMSE-lSD) is proposed, and the reception performance is investigated in terms of the effective SINR and system capacity. For the average system capacity, the proposed adaptive reception demonstrates the performance enhancement compared to the conventional EB reception using the receiver beamforming vector, and up to 2 bps/Hz performance gain is achieved for mobile station located at the cell edge.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.10a
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• pp.347-350
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• 2007

Adaptive arrays have been proposed as an effective means of mitigating Multiple Access Interference (MAI) and improving the performance of existing and future wireless communication systems. In this paper, we apply the analytical method proposed to analyse the theoretical Mean Bit Error Rate (BER) of an uncoded IS-95 based CDMA system with an array antenna at the BS in a Ricean fading environment. We present a modified expression for the Signal to Interference plus Noise (SINR) ratio as a function of the number of users, number of antennas and noise levels. We also verify the analytical results by means of simulations by considering different user and channel scenarios.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.7
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• pp.945-951
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• 2007

Noise cancellation is essential for the devices which use speech as an interface. In real environments, speech quality and recognition rates are degraded by the auditive noises coming near the microphone. In this paper, we propose a noise cancellation algorithm using stereo microphones basically. The advantage of the use of multiple microphones is that the direction information of the target source could be applied. The proposed noise canceller is based on the Wiener filter. To estimate the filter, noise and target speech frequency responses should be known and they are estimated by the spectral classification in the frequency domain. The performance of the proposed algorithm is compared with that of the well-known Frost algorithm and the generalized sidelobe canceller (GSC) with an adaptation mode controller (AMC). As performance measures, the perceptual evaluation of speech quality (PESQ), which is the most widely used among various objective speech quality methods, and speech recognition rates are adopted.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.4
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• pp.624-633
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• 2018

While the multimedia lecture system has been getting smart using immerging technology, the microphone still relies on the classical approach such as holding in hand or attaching on the body. In this paper, we propose a ceiling mounted array microphone system that allows a wide reception coverage and instructors to move freely without attaching microphone. The proposed system adopts cell and handover of mobile communication instead of a complicated beamforming method and implements a wide range microphone over several cells with low cost. Since the characteristics of unvoiced speech is similar to Pseudo Noise it is shown that soft handover are possible with 3 microphones connected to delay-sum multipath receiver. The proposed system is tested in $6.3{\times}1.5m$ area. For real-time processing the correlation range can be reduced by 82% or more, and the output latency delay can be improved by using the delay adaptive filter.