• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.8B
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• pp.771-779
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• 2002

A new signal model suitable for indoor wireless environments with large angle spread is proposed to improve the performance of indoor wireless communication systems. Antenna array techniques adopted for this model are discussed. It is based on the far-field signal assumption. The optimal beamforming weight vector is obtained by applying the antenna algorithm based on the maximum signal to interference noise ratio criterion to the model. The proposed model is verified using a mathematical analysis and computer simulations.

• 한국지구물리탐사학회:학술대회논문집
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• 2007.06a
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• pp.279-284
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• 2007

The cultural noise sources in magnetotellurics were localized using the source localization method. Conventional beamforming techniques are not applicable for electromagnetic source localization. In this study, the matched field processing and genetic algorithm are used to localize an electromagnetic source and estimate the polarization direction. The source localization using MT field data shows the characteristics of estimated source distribution related to the strength of cultural noise.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1E
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• pp.45-53
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• 2010

In this paper, we present the use of the Features Weighted Mahalanobis Distance (FWMD) in improving the performance of Likelihood Maximizing Beamforming (Limabeam) algorithm in speech recognition for microphone array. The proposed approach is based on the replacement of the traditional distance measure in a Gaussian classifier with adding weight for different features in the Mahalanobis distance according to their distances after the variance normalization. By using Features Weighted Mahalanobis Distance for Limabeam algorithm (FWMD-Limabeam), we obtained correct word recognition rate of 90.26% for calibrate Limabeam and 87.23% for unsupervised Limabeam, resulting in a higher rate of 3% and 6% respectively than those produced by the original Limabearn. By implementing a HM-Net speech recognition strategy alternatively, we could save memory and reduce computation complexity.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.53 no.6
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• pp.446-452
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• 2004

Under the condition of a common narrowband target signal and interference signals from several directions, the linearly constrained minimum variance (LCMV) method using the generalized sidelobe canceller (GSC) for adaptive beamforming has been exploited successfully However, in the case of wideband signals, the length of the adaptive filter must be extended. As a result, the complexity of the beamformer increases, which makes real-time implementation difficult. In this paper, we improve the convergence characteristics of the adaptive filter using the transform domain normalized least mean square (NLMS) approach based on the subband GSC structure without the increase of complexity. Besides, the M-MAX algorithm, which is one of various selective coefficient updating methods, is employed in order to remarkably reduce the computational cost without decreasing the convergence quality. With the combination of these methods, we propose a computationally efficient wideband adaptive beamformer and verify its efficiency through a series of simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6A
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• pp.494-499
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• 2009

In this paper, we provide an efficient method of user selection for achieving the maximum system throughput in downlink multiuser Multiple-Input Multiple-Output (MIMO) systems. A proposed method is for selecting a fine user set only with powers of each user channel and angles between them. This algorithm is simpler than SUS because there is no considering about the optimal value of correlation. The proposed method finds the user set toward maximizing system throughput, so it has high performance.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.10a
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• pp.890-893
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• 2014

In this paper, we present a signal processing algorithm for a marine radar system, in which the evaluation of probability of collision as well as target detection and tracking are performed. Moreover, the digital signal processor that implements the algorithm is proposed. As simulation environment, a mechanically scanning antenna utilizing FMCW signal is used, conducting the beamforming operation with 1 degrees intervals. Test board consists of DSP chips and FPGA, which enable the implemented system to operate in real-time.

• International Journal of Internet, Broadcasting and Communication
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• v.14 no.3
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• pp.69-78
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• 2022

In this paper, we propose the new technique adopting power control to MIMO(multi-input multi-output)-OFDMA(orthogonal frequency division multiplexing Access) system with multi-beamformer. The proposed power controlling algorithm for MIMO-OFDMA allocates the transmitting power of each subcarrier based on the CSI(channel state information) and the interference signal. CSI is feedback from base station to mobile station to decide the transmitting power of each subcarrier. Through the proposed technique, we can control iteratively the transmitting power and update the weight of beamformer simultaneously. Therefore, the SNIR of each subcarrier become to converge the target SNIR and the beam is formed toward the desired direction. And the performance of MIMO-OFDMA system with the proposed approach is very improved. The improvement in bit error rate is investigated through computer simulation of a MIMO-OFDMA system with the proposed approach.

• Journal of Electrical Engineering and information Science
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• v.1 no.1
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• pp.156-159
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• 1996

Element failure occurs with high probability for every array used in the real world ; that is, it is a common phenomenon that there are one or more elements with no output. Element failure produces an elevated sidelobe level and fails to reject the interference signals in an adaptive beamformer. In this paper, we present the adaptive beamforming algorithm for array with element failure. The presented method minimizes the array output power subject to a set of linear constraints which maintain the frequency response in the look direction and force the weights of the inoperative elements to zero. Numerical results have been included.

• The Journal of the Acoustical Society of Korea
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• v.17 no.2E
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• pp.38-46
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• 1998

In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.

• The Journal of the Acoustical Society of Korea
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• v.18 no.3
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• pp.27-34
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• 1999

This paper proposes an algorithm to design the non-linear array so as to form efficiently the one-shot beam with relatively less sensors for acoustic measurement. In this algorithm, according to the spatial sampling theory the part for high frequency(HF) band has equispaced sensor array and the sensor distances below the HF band are decided as a function of number of HF sensors. As the results of the simulations, the mean and variances of directivity index(DI) of non-linear array which has less sensors are similar to those of linear array. and the DI variation for beam steering angle is very small. And the beam width at -2dB point is 6.8°. Thus it is confirmed that the design algorithm for non-linear array which is proposed to have less sensors can be efficiently used in acoustic measurement.