• IEIE Transactions on Smart Processing and Computing
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• v.2 no.1
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• pp.11-19
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• 2013

A video-zoom driven audio-zoom algorithm is proposed to provide audio zooming effects according to the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone array in conjunction with a soft masking process that uses the phase differences between microphones. The audio-zoom processed signal is obtained by multiplying the audio gain derived from the video-zoom level by the masked signal. The proposed algorithm is then implemented on a portable digital imaging device with a clock speed of 600 MHz after different levels of optimization, such as algorithmic level, C-code and memory optimization. As a result, the processing time of the proposed audio-zoom algorithm occupies 14.6% or less of the clock speed of the device. The performance evaluation conducted in a semi-anechoic chamber shows that the signals from the front direction can be amplified by approximately 10 dB compared to the other directions.

• Proceedings of the Korea Electromagnetic Engineering Society Conference
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• 2003.11a
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• pp.501-505
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• 2003

There are many antenna array errors. They will distort the array beam pattern and result in an increased sidelobe level. A calibration technique is proposed for correcting the antenna array errors such as mutual coupling and unequal feeder characteristics. These are modeled as a matrix representing the interaction between the radiating elements. The matrix is estimated from the measured array response vectors. The antenna array errors are corrected by modifying the beamforming weight vector. It is verified by the electromagnetic simulation and experiment that the proposed technique reduces the sidelobe level and increases the antenna gain.

• Proceedings of the KOSOMBE Conference
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• v.1997 no.11
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• pp.563-566
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• 1997

In this paper, several bandwidth sampling methods were compared in which contains "multi-order sampling", which was proposed or envelope detections in RF ultrasonic signals by a computer simulation and on experiment quadrature sampling method. The second-order sampling method were compared with it. As a conclusion, a multi-order sampling method, especially 5th order sampling method showed quite good envelope detection property. This means that more economical and quite good performance digital beamforming system can be built by adopting this new bandwidth sampling method.

• ETRI Journal
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• v.24 no.3
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• pp.197-204
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• 2002

This paper presents the results of the numerical simulation of a multibeam active phased array antenna for a High Altitude Platform Station (HAPS). The simulation takes into account the random errors caused by the non-identity of the array elements and the inaccuracy of the antenna calibration. The results of our statistical simulation show that the strict requirements on the sidelobe envelope for HAPSs can be met when the amplitude and phase distribution errors are minor, a condition which may be achieved by using digital beamforming.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.28 no.6
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• pp.478-486
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• 2017

In this paper, we propose an algorithm for the wideband beamforming in a conformal phased array antenna by compensating the errors. For the wideband beamforming, we used the True Time Delay(TTD), which was fabricated on the RF circuit board to obtain long delay lines. Beamforming errors in the conformal array antenna are the mutual coupling between the array elements, the dispersive error in the TTD circuit, and the quantization error by the digital control. We apply the compensation algorithm to the conformal phased array antenna of wideband 2~4 GHz, and verify the usefulness by comparing the results with the experiment results.

• Journal of the Korea Society of Computer and Information
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• v.13 no.5
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• pp.111-116
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• 2008

Radar system must estimate exactly quickness and target in interference channel. Because interference of radio channel is multipath channel by artificial structure and nature structure. signal estimation is difficult. As long as, get rid of interference signal have been study digital beamforming, adaptive array antenna and so on. In this paper, proposed SPT-SALCMV beamforming algorithm get rid of coherent interference algorithm and adaptive array antenna. Adaptive array forms null pattern and reduces gains for direction of interference signal. And estimate signal that want by keeping gains of beam pattern changelessly to target signal direction. In this paper, proposed SPT-SALCMV algorithm was exactly received position of target. But general SPT-LCMV algorithm resulted beam error about 30degrees. Therefore, proved that SPT-SALCMV algerian that propose in this paper is more excellent than genaral SPT-LCMV algorithm.

• Journal of Biomedical Engineering Research
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• v.34 no.1
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

• 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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.2 s.332
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• pp.49-59
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• 2005

An adaptive hybrid beamformer is proposed to improve the reception performance of the advanced television system committee (ATSC) digital television (DTV) in a mobile environment. Dynamic multipaths and Doppler shifts severely degrade the reception performance of the ATSC DTV receiver. Accordingly, a hybrid beamformer, called a Capon and least mean square (CLMS) beamformer, is presented that uses direction of arrival (DOA) information and the least mean square (LMS) beamforming algorithm. The proposed CLMS algerian efficiently removes dynamic multipaths and compensates for the phase distortion caused by Doppler shifts in mobile receivers. After the CLMS beamformer is operated, the subsequent use of an equalizer removes any residual multipath effects, thereby significantly improving the performance of DTV receivers. The performances of the proposed CLMS, Capon, and LMS beamformersare compared based on computer simulations. In addition, the overall performance of the CLMS beamformer followed by an equalizer is also considered.

• 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.