• The Journal of the Acoustical Society of Korea
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• v.23 no.8
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• pp.570-575
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• 2004

In the underwater acoustic systems. we can receive signals and retrieve information about a target by using a beamforming method. The most important thing in the beamforming is finding the way to optimize the mainlobe beamwidth and the sidelobe level to the desired value. One of the prominent results of beamforming method. which has been studied. is Philip's weighting function method(1) . Philip's method adaptively adjusts its weights of array to meet the desired mainlobe beamwidth and sidelobe level. It is very similar to the design method in adaptive filter. However. this method cannot easily bring us to the desired sidelobe level due to complementary relation between mainlobe beamwidth and sidelobe level. In this paper, we propose a new algorithm using partial constrained adaptation. This method makes us circumvent the above problem and meet the specification of design easily. The proposed algorithm presents a Pattern synthesis that designer can easily control the mainlobe beamwidth and the sidelobe level to the desired value while calculation time to converge is decreasing.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.19 no.9
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• pp.907-914
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• 2009

An approach to 3D visualization of multiple sound sources has been developed with the application of a moving array technique. Frequency domain beamforming algorithm is used to generate a beam power map and the sound source is modeled as a point source. When a conventional delay and sum beamformer is used, it is considered that 2D distribution of sensors leads to have deficiency in spatial resolution along a measurement distance. The goal of moving an array in this study is to form 3D array aperture surrounding multiple sound sources so that the improved spatial resolution in a virtual space can be expected. Numerical simulation was made to examine source localization capabilities of various shapes of array. The 3D beam power maps of hemispherical and spherical distribution are found to have very sharp resolution. For experiments, several sound sources were placed in the middle of defined virtual space and arc-shaped line array was rotated around the sources. It is observed that spherical array shows the most accurate determination of multiple sources' positions.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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• pp.200-204
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• 2005

The use of beamforming method and de-Dopplerization technique was applied in studying the rotating sound sources. Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. Two main issues of the signal reconstruction in time domain are addressed herein: First, to remove Doppler effect from the measured data and to restore the original emission data of the moving source. The difference of the time domain beamforming from the frequency domain beamforming was mentioned. Also, the time domain beamforming method is deployed in the test and the comparisons were made to the frequency domain results. The time domain signal reconstruction was numerically simulated prior to the application. To validate the de-Dopplerization Performance, the rotating Point sources were examined and localized by the use of a phased array of microphone. The application of prop-rotor was conducted in a hovering condition. The results of reconstructing time signals of rotating sources and its locations were shown in the power distribution maps. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies of interest.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.6 s.111
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• pp.565-573
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• 2006

Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. The purpose of this research is development of beamforming technique can be applied to the rotor noise source identification. For the do-Dopplerization and reconstruction of emitted sound wave, Forward Propagation Method is applied to the time domain beamforming technique. And validation test were performed using rotating sound source constructed by bended pipe and horn driver. In the validation test using sinusoidal sound wave, sufficient performance of signal processing can be seen, and the effect of measuring duration for accuracy was compared. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies and collective pitch angle, in hover condition.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.7
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• pp.1271-1277
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• 2008

This paper proposes the code acquisition scheme on a MIMO channel for preamble search in a CDMA-MIMO uplink system. The multiple transmit/receive antennas are used for beamforming. The performance of a ML code acquisition technique based on transmit and receive beamforming is analyzed by considering rho detection probability. The acquisition performance and MAT for a MIMO code acquisition system are numerically evaluated. It is shown that multiple transmit antennas can give the code acquisition system a transmit beamforming gain and result in much better performance than a SIMO case.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2006.07a
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• pp.568-568
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• 2006

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2006.07a
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• pp.569-569
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• 2006

• 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.

• Journal of Broadcast Engineering
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• v.11 no.4 s.33
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• pp.521-532
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• 2006

Recently, many research initiatives have been underway to improve reception performance of ATSC based DTV signal in mobile channel by adopting multiple antennas. In this paper, we propose a new mobile channel model which can be applicable to any array geometry. And then we propose new channel estimation algorithm which uses PN5l1 sequence in field synch. The proposed algorithm is to estimate channel by correlating the input signal in If frequency band and finding maximum peak, which does not need complicated synchronization circuit. Finally, we propose new receiver structures which can be implemented at the front-end of the existing receiver with no modification. With computer simulation, we verify the performance of the proposed model and verify the performance of the receiver structure with computer simulation.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.212-221
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• 2010

The bubbles could be created from multiple source in the surface of ocean. The bubbles which are created like this affect to the intensity and the sound speed of acoustic wave which are used from the ocean. From the research, it observed Reverberation from the sea surface, using Vbeam transmission and reception equipment that was a cross-shaped Array, from 2008 July 2nd to July 3rd at 4Km from the East Sea Mookho port. It is difficult to analyze data because the received signal has ambient noise and occurrence ISI(Inter Symbol Interference) for multipath. This paper remove that ambient noise and ISI using the directions of arrival beamforming and the filter and sum beamforming. After beamforming, a following DFE(Decision Feedback Equalizer) removes the remaining multipath components. Experimental results show that the proposed technique reduce the errors caused by ISI.