• Proceedings of the Korea Committee for Ocean Resources and Engineering Conference
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• 2004.05a
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• pp.386-389
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• 2004

Beamforming for underwater acoustic communication is affected by the broadband feature of underwater acoustic communication signal, which has the low center frequency compared to the signal bandwidth. In this paper, the baseband equivalent array signal model is derived and we present computer simulation results for the broadband finite impulse response (FIR) beamformer performance according to the FIR filter order and the tap spacing. If the FIR filter order is increased above the optimum value, the beamformer performance is degraded. Also the tap spacing is related to the optimum FIR filter order.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2151-2156
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• 2006

Beamforming for underwater acoustic communication (UAC) is affected by the broadband feature of UAC signal, which has relatively low currier frequency as compared to the signal bandwidth. The narrow-band assumption does not hold good in UAC. In this paper, we discuss a broadband FIR beamformer for UAC using the baseband equivalent way signal model. We consider the broadband FIR beamformer for QPSK UAC with carrier frequency 25kHz and symbol rate 5kHz. Array geometry is a uniform linear way with 8 omni-directional elements and sensor spacing is the half of the carrier wavelength. The simulation results show that the broadband n beamformer achieves nearly optimum signal to interference and noise ratio (SINR) and outperforms the conventional narrowband beamformer by SINR 0.5dB when two-tap FIR filter is employed at each sensor and the inter-tap delay is a quarter of the symbol interval. The broadband FIR beamformer performance is more degraded as the FIR filter length is increased above a certain value. If the inter-tap delay is not greater than half of the symbol period, SINR performance does not depend on the inter-tap delay. More training period is required when the inter-tap delay is same as the symbol period.

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.520-525
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• 2009

The conventional studies about an adaptive beamformer assumed that the interference signals are stationary, so they used time-average of signals or Least Mean Squares. However, these methods showed low performance of canceling the non-stationary interferences. In this paper, the MAFF-RLS algorithm is developed in order to cancel non-stationary interferences, and the GSC structure using this algorithm is proposed. Furthermore, the performance of the MAFF-RLS beamformer is verified by simulation using MATLAB. This simulation results show the performance of the proposed beamformer is better than that of the SMI and the conventional RLS beamformer.

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.548-556
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• 2009

This paper proposes an efficient speaker array system to spotlight a sound into a target position in a reverberant environment. The proposed method introduces a criterion of maximizing the energy of system response from each channel to the target position. Simulation results with a sixteen channel speaker array system prove that the performance of the proposed method improves that of the conventional method that uses a broadband beamformer in a reverberant environment.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2E
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• pp.14-19
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• 1994

In this paper, we derive expressions for the output signal-to-interference plus noise ratio(SINR) of the linearly constrained broadband beamformer in noncoherent situations using a vector approach. The incoming broadband signals are assumed to have flat spectra.

• Journal of Biomedical Engineering Research
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• v.32 no.3
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• pp.198-206
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• 2011

Hearing devices like cochlear implant, vibrant soundbridge, etc. try to offer better sound for people. In hearing devices, several beamformers including conventional directional microphone are applicable to noise reduction. Each beamformer has different directional response and it could change sound intelligibility or quality for listeners. Therefore, we investigated the performance of three beamformers, which are first and second order directional microphone, and broadband beamformer(BBF) with a computer simulation assuming hearing device microphone configuration. We also calculated objective measurements which have been used to evaluate speech enhancement algorithms. In the simulation, a single speech and a single babble noisewere propagated from the front and $135^{\circ}$ azimuth degrees respectively. Microphones were configured in an end-fire array and the spacing was varied in comparison. With 3 cm spacing, BBF had about 3 dB higher enhanced SNR than that of directional microphones. However, enhancement of segmental SNR and frequency weighted segmental SNR were similar between the first order directional microphone and broadband beamformer. In addition when steady state noise was used, broadband beamformer showed the increased performance and had the highest enhanced SNR, and segmental SNR.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.880-882
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• 2004

This paper presents the design and simulation results of 7 waveguide dielectric rod array antennas for 7 channel DBF(Digital beamformer) receivers for stratospheric system. A waveguide. dielectric antenna type, which has FTEP(Flat-toped element pattern) to manufacture easily. Also, the calculated element patterns according to the permittivity are compared. The designed antenna will be used for generating multibeam radiation pattern by means DBF.

• 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.

• 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.