• Proceedings of the KIEE Conference
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• 2011.07a
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• pp.1642-1643
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• 2011

본 논문은 적응형 알고리즘(Adaptive Algorithm)을 이용하여 재머(Jammer)를 억제하는 것을 기존의 알고리즘을 시뮬레이션으로 확인한다. 본 논문에서는 원하는 방향으로 빔을 형성하는 동시에 특정방향 주로 부엽(sidelobe)으로 들어오는 재머의 신호를 억제시키는데 있어 여러 가지 적응형 알고리즘을 확인하고, 알고리즘을 이용하여 부엽으로 들어오는 재머신호를 어떻게 억제하는지 시뮬레이션을 통해 결과를 확인하도록 할 것이다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.341-344
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• 2001

We propose a novel method to simplify the computational load of ILSP algorithm for CDMA environment. Since this method processes the block matrix by a vector sequentially, the complex matrix computation ran be avoided. The performance of the algorithm is verified by computer simulations.

• Journal of Communications and Networks
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• v.15 no.4
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• pp.370-382
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• 2013

Distributed massive MIMO systems, which have high bandwidth efficiency and can accommodate a tremendous amount of traffic using algorithms such as zero-forcing beam forming (ZFBF), may be deployed in large public venues with the antennas mounted under-floor. In this case the channel gain matrix H can be modeled as a multi-banded matrix, in which off-diagonal entries decay both exponentially due to heavy human penetration loss and polynomially due to free space propagation loss. To enable practical implementation of such systems, we present a multi-banded matrix inversion algorithm that substantially reduces the complexity of ZFBF by keeping the most significant entries in H and the precoding matrix W. We introduce a parameter p to control the sparsity of H and W and thus achieve the tradeoff between the computational complexity and the system throughput. The proposed algorithm includes dense and sparse precoding versions, providing quadratic and linear complexity, respectively, relative to the number of antennas. We present analysis and numerical evaluations to show that the signal-to-interference ratio (SIR) increases linearly with p in dense precoding. In sparse precoding, we demonstrate the necessity of using directional antennas by both analysis and simulations. When the directional antenna gain increases, the resulting SIR increment in sparse precoding increases linearly with p, while the SIR of dense precoding is much less sensitive to changes in p.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.332-339
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• 2012

Beamspace transform algorithm transforms spatial-domain data - such as x, y, z dimension - into incidence-angle-domain data, which is called beamspace-domain data. The beamspace transform method is generally used in source localization and tracking, and adaptive beamforming problem. When the beamspace transform method is used in multichannel audio source separation, the inverse beamspace transform is also important because the source image have to be reconstructed. This paper studies the beamspace transform and inverse transform algorithms for multichannel audio source separation system, especially for the beamspace-domain multichannel NMF algorithm.

• The Journal of the Acoustical Society of Korea
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• v.25 no.1
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• pp.30-35
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• 2006

In this paper. we Propose a frequency domain beamforming algorithm composed of inverse-filtering stages followed by a MVDR (Minimum-Variance Distortionless Response) beamformer or a GSC (Generalized Sidelobe Canceller). The proposed method is shown to require less complexity than the conventional RTF-MVDR and TF-GSC. respectively, and it is shown that the Proposed method is equivalent to the conventional RTF-MVDR and TF-GSC in optimum solution. In order to evaluate the performance of the Proposed method. speech recognition experiments are performed using the speech database recorded in a car. The Proposed method shows equal or slightly degraded Performance comparing to the conventional methods in terms of the speech recognition rate.

• Journal of Korea Society of Digital Industry and Information Management
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• v.9 no.1
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• pp.95-102
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• 2013

This paper covers nulling algorithm. In this algorithm, we assume that nulling points are already known. In general, nulling algorithm using matrix equation was utilized. But, this algorithm is pointed out that computational complexity is disadvantage. So, we choose gradient method to reduce the computational complexity. In order to further reduce the computational complexity, we propose approximate gradient method using characteristic of trigonometric functions. The proposed method has same performance compared with conventional method while having half the amount of computation when the number of antenna and nulling point are 20 and 1, respectively. In addition, we could virtually eliminate the trigonometric functions arithmetic. Trigonometric functions arithmetic cause a big problem in actual implementation like FPGA processor(Field Programmable gate array). By utilizing the above algorithm in a multi-cell environment, beamforming gain can be obtained and interference can be reduced at same time. By the above results, the algorithm can show excellent performance in the cell boundary.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.784-788
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• 2008

In this research, we propose a remote speech input unit, a new method of user-friendly speech input in speech recognition system. We focused the user friendliness on hands-free and microphone independence in speech recognition applications. Our module adopts two algorithms, the automatic speech detection and speech enhancement based on the microphone array-based beamforming method. In the performance evaluation of speech detection, within-200msec accuracy with respect to the manually detected positions is about 97percent under the noise environments of 25dB of the SNR. The microphone array-based speech enhancement using the delay-and-sum beamforming algorithm shows about 6dB of maximum SNR gain over a single microphone and more than 12% of error reduction rate in speech recognition.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.7
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• pp.46-52
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• 1998

This paper presents an experimenal proof for criteria of selecting an optimum adaptive beamforming (ABF) algorithm for a large distorted phased array. A single point target embedded in clutter model is suggested to compare four well-known ABF algorithms. These algorithms are tested to low variance and high variance real data for self-calibrating a large distored phased array. It is shown that these algorithms require at least one dominant scatterer with large radar cross section (RCS) or multiple scatterers with moderate RCS in the field of view. Experimental results are provided to demonstrate the comparisons of the four algorithems in terms of gain loss and image correlaion coefficient, along with corresponding reconstructed cross-range images and range-azimuth images.

• Journal of Communications and Networks
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• v.18 no.3
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• pp.320-326
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• 2016

In existing literature on multiple-input multiple-output (MIMO) relaying communication systems, antenna selection is often implemented by maximizing the channel capacity or the output single-to-noise ratio (SNR). In this paper, we propose an energy-efficient low-complexity antenna selection scheme for MIMO relaying communication systems. The proposed algorithm is based on beamforming and maximizing the Frobenius norm to jointly optimize the transmit power, number of active antennas, and antenna subsets at the source, relaying and destination. We maximize the energy efficiency between the link of source to relay and the link of relay to destination to obtain the maximum energy efficiency of the system, subject to the SNR constraint. Compared to existing antenna selection methods forMIMO relaying communication systems, simulation results demonstrate that the proposed method can save more power in term of energy efficiency, while having lower computational complexity.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.409-413
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• 2008

In this paper, we introduce a platform of advanced multiple antenna system based on orthogonal frequency-division multiplexing (OFDM). The advanced multiple antennas have beamforming gain using array antenna. In array antenna systems, received signal has phase delay caused distance of each antennas, therefore it should compensate with optimum weight vector which calculated by Lagrange algorithm. To implement the presented above procedures using Digital Signal Processor (DSP), we should fixed-point design. The performance of implemented platform is verified through MATLAB$^{(R)}$ simulations with various signal environments.