• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.713-717
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• 2014

This paper presents a beamforming technique for locating impulsive sound source. The conventional frequency-domain beamformer is advantageous for localizing noise sources for a certain frequency band of concern, but the existence of many frequency components in the wide-band spectrum of impulsive noise makes the beamforming image less clear. In contrast to a frequency-domain beamformer, it has been reported that a time-domain beamformer can be better suited for transient signals. Although both frequency- and time-domain beamformers produce the same result for the beamforming power, which is defined as the RMS value of its output, we can use alternative directional estimators such as the peak value and crest factor to enhance the performance of a time-domain beamformer. In this study, the performance of three different directional estimators, the peak, crest factor and RMS output values, are investigated and compared with the incoherent interfering noise embedded in multiple microphone signals. The proposed formula is verified via experiments in an anechoic chamber using a uniformly spaced linear array. The results show that the peak estimation of beamformer output determines the location with better spatial resolution and a lower side lobe level than crest factor and RMS estimation in noise free condition, but it is possible to accurately estimate the direction of the impulsive sound source using crest factor estimation in noisy environment with stationary interfering noise.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.4
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• pp.160-168
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• 2011

A Frequency domain beamforming technique is widely used in sonar systems with a large number of beams and sensors. In the battlefield environment requiring real-time signal processing, it is needed to optimize the computational complexity of the spectrum computation to implement an efficient and fast frequency domain beamformer. So, in this paper, we proposed the pruned-GSFFT (pruned generalized sliding fast Fourier transform) as a new spectrum computation method. The proposed method help to reduce the computational complexity of the real-time partial spectrum computation by eliminating the redundancy between consecutive input samples and skipping the regardless frequency bands. Also the characteristics of the proposed pruned-GSFFT method and its computational complexity are compared to those of previous FFT algorithms.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.10a
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• pp.543-543
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• 2010

The objective of this research is estimating the location of interested sound source by using the similarity between a beamformer output in time domain and the candidate signal. The waveform of beamformer output at the location of sound source is similar with the waveform emitted by that source. To estimate the location of sound source by using this feature, we define quantified similarity between candidate signal and beamformer output. The candidate signal describes the signal which is generated by interested source. In this paper, similarity is defined by four methods. The two methods use time vector comparison, and the other two methods use time-frequency map or linear prediction coefficients. To figure out the results and performance of localization by using similarities, we demonstrate two conditions. The one is when two pure tone sources exist and the other condition is when several bird sounds exist. As a consequence, inner product with two time-vectors and structural similarity with spectrograms can estimate the locations of interest sound source.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.199-202
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• 2002

This paper introduces the near field shading beamformer using widely known Chebyshev and Hanning window in the field of digital signal processing. The proposed shading beamformer improves the estimation of range as well as azimuth angle of targe residing in near field. A series of sensor weighting values are calculated from the FFT operation of given shading functions in time domain. This paper verifies the performance of the focused beamformer having the proposed shading sensor weights which are used to detect the range of target. Throughout computer simulations this paper exploits the performance improvement of the proposed shading beamformer as varying the frequency band of the received radiated signal along the non-uniform array.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.83-84
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• 2008

It is efficient to use the time-domain beamforming to operate the various pulse with the different pulse length, frequency, bandwidth in active sonar system. In this paper, we propose a time-domain beamformer with the cumulative processing in the decomposed channel using the polynomial interpolation to solve the problem of the computational cost, high transmission data rate, and the lack of internal memory.

• International Journal of Internet, Broadcasting and Communication
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• v.11 no.2
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• pp.76-83
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• 2019

In this paper, we propose the beamforming algorithm for the performance improvement of MIMO MC-CDMA system. The proposed multibeamforming of MIMO MC-CDMA structure having the same number of beamformer as the number of transmit antenna is derived by calculating the error signals between the coded pilot symbols and the corresponding received signals from the multiple transmitters of the desired user in the frequency domain, transforming the frequency-domain error signals into time-domain error signals, and updating the weights of the multibeamformer in the time-domain in the direction minimizing the mean squared error (MSE). The proposed approach can track each direction of arrival (DOA) of the signals from multi-antennas of a desired user. The performance improvement is investigated through computer simulation by applying the proposed approach to MIMO MC-CDMA system in a multipath fading channel with multiusers.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2009.04a
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• pp.490-495
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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, two 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 show the most accurate determination of multiple sources' positions.

• 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.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.26 no.5
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• pp.501-510
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• 2016

Capabilities of several beamforming techniques are compared for estimating the position of a moving source. Beamforming has enabled to widen our perspective of aeroacoustics in wind tunnel experiments and has provided useful approach in array measurements. Meanwhile beamforming techniques have been developed in a way to improve estimation accuracy and to save ing effort at the same time. In order to achieve reasonable outcome from aeroacoustic measurement, it is important to identify the spectral characteristics of source and to select an appropriate beamformer. Though aeroacoustic sources normally generates broadband noises, many array signal processing have been focused on narrowband processing which makes calculation numerically efficient. However, calculation in frequency-domain requires selection of single frequency of interest which affects spatial resolution and sidelobe level as a consequence. To be able to localize broadband noise source, it is proposed to use broadband beamforming. The formulas implements the deletion of diagonal term from cross spectral matrix. In this study, trajectory of flying source emitting broadband noise was simulated and several beamformers are applied.

• The Journal of the Korea institute of electronic communication sciences
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• v.12 no.3
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• pp.433-438
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• 2017

Containing an Angle-of-Arrival(: AOA) estimation and interference suppression techniques, an adaptive beamformer is one of core techniques for the Signal Intelligence(: SIGINT) which collect various intelligence utilizing cutting edge devices including the radar and satellite. It generates a beam with the directivity in a corresponding direction, to efficiently receive a signal from the specific direction, using antenna array. In this paper, we present the received signal model including interference signals and noise, which can be applied to an input of the signal intelligence satellite system equipped with the AOA estimation and the interference cancellation techniques, and analysis the characteristics of various signals, which can be included in the proposed received signal model. This proposed signal model can be directly applied to the performance evaluation for a variety of beamforming techniques. Also, we verify the spectrum characteristic of the presented received signal model in the frequency domain through computer simulation examples.