• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2001.05a
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• pp.1003-1008
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• 2001

In beamforming method, source positions are predicted by MUSIC (Multiple Signal Classification) power method and composite sound fields can then be decomposed into each partial field by beamforming, detenninistically without restriction of the distance between reference microphones and sources. However, reference microphone array shape is important in both MUSIC and beamforming method. Thus the present paper describes the effect of the reference microphone array shape.

• The Journal of the Acoustical Society of Korea
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• v.26 no.5
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• pp.188-193
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• 2007

In general, analog amplifiers for 5.1 or more channels have been used to configure home theater systems. In order to make high-performance systems, it is desirable to process audio signals in digital techniques in consideration of output and efficiency of speakers. Especially we need 7.1-channel system to separate audio signals efficiently. In this paper we implemented the architecture of S/PDIF 7.1-channel digital amplifier for home theater systems. The amplifier shows good performance with less loss of original sounds because of both strong characteristics against noises and direct processing of input data.

• Journal of Broadcast Engineering
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• v.23 no.4
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• pp.519-525
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• 2018

This paper deals with noise suppression of multi-channel data captured by microphone array using multi-channel Wiener filter. Multi-channel Wiener filter does not rely on information about the direction of the target speech and can be partitioned into an MVDR (Minimum Variance Distortionless Response) spatial filter and a single channel spectral filter. The acoustic transfer function between the single speech source and microphones can be estimated by subspace decomposition of multi-channel Wiener filter. The errors are incurred in the estimation of the acoustic transfer function due to the errors in the estimation of correlation matrices, which in turn results in speech distortion in the MVDR filter. To alleviate the speech distortion in the MVDR filter, diagonal loading is applied. In the experiments, database with seven microphones was used and MFCC distance was measured to demonstrate the effectiveness of the diagonal loading.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.5-10
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• 2002

This paper proposes new 16 kbps wideband speech codec with bandwidth of 7 kHz. The proposed codec decomposes the input speech signal into low-band and high-band signals using QMF (Quadrature Mirror Filter), then AMR (Adaptive Multi Rate) speech codec processes the low-band signal and new transform-domain codec based on G.722.1 wideband cosec compresses the high-band signal. The proposed codec allocates different number of bits to each band in an adaptive way according to the property of input signal, which provides better performance than the codec with the fixed bit allocation scheme. In addition, the proposed cosec processes high-band signal using wavelet transform for better performance. The performance of proposed codec is measured in a subjective method. and the simulations with various speech data show that the proposed coders has better performance than G.722 48 kbps SB-ADPCM.

• Composites Research
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• v.14 no.5
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• pp.46-53
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• 2001

In this research, damage behavior of singly oriented ply (SOP) fiber metal laminate (FML) subjected to concentrated load was studied. The static indentation tests were conducted to study fiber orientation effect on damage behavior of FML. During the static indentation tests, acoustic emission technique (AE) was adopted to study damage characteristics of FML. AE signals were obtained by using AE sensor with 150kHz resonance frequency and the signals were compared with indentation curves of FML. The damage process of SOP FML was divided by three parts, i.e., crack initiation, crack propagation, and penetration. The AE characteristics during crack initiation show that the micro crack is initiated at lower ply of the plate, then propagate along the thickness of the plate with creating tiber debonding. The crack grow along the fiber direction with occurring 60∼80dB AE signal. During the penetration, the fiber breakage was observed. As fiber orientation increases, talc fiber breakage occurs more frequently. The AE signal behaviors support these results. Cumulative AE counts could well predict crack initiation and crack propagation and AE amplitude were useful for the prediction of damage failure mode.

• The Journal of the Acoustical Society of Korea
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• v.30 no.5
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• pp.239-248
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• 2011

In order to improve the quality of strain images in medical ultrasound imaging, displacements need to be accurately estimated. In this paper, in order to apply one-dimensional displacement estimation methods to two-dimensional motion estimation, the axial and lateral displacements are separately estimated. In order to estimate lateral displacements, one-dimensional signals aligned in the lateral direction are converted to analytic signals, which are then crosscorrelated. Strain images are produced by first compensating two-dimensional displacements for lateral motion with lateral motion displacement estimates obtained from the proposed lateral displacement estimation algorithm and then estimating axial displacements. Both phantom and human data experiments show that the proposed method provides better signal-to-noise ratio and contrast-to-noise ratio characteristics than a conventional strain imaging method that utilizes axial displacement estimates only.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.21-29
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• 2001

In order to detect the precise port/starboard direction of arrival of target signal in real noisy ocean environments, Inverse beamforming (IBF) algorithm is surveyed theoretically and the detection performances of IBF are analyzed with simulations. Cardioid Inverse beamforming algorithm was proposed for port/starboard discrimination and the performance was studied with simulations. It is shown that IBF has a 3dB array gain advantage over Conventional beamforming (CBF) under ideal conditions. This 3 dB advantage is proven theoretically and illustrated with simulations. The fact that the IBF beamwidth is narrower than the CBF beamwidth by a factor of 0.68 proves the performance of defection and spatial resolution improvement. Comparing the simulation results of Cardioid Inverse beamforming and Conventional Cardioid beamforming, it is shown that Cardioid Inverse beamformer has enhanced performance in minimum detection level, detection accuracy and resolution. Due to the results of moving target bearing detection test in endfire, it is shown that Cardioid Inverse beamformer has better performance, comparing the Conventional Cardioid beamformer.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.481-487
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• 2004

In this paper. the structure of bandwidth scalable wideband speech codec and its high-band codec are proposed. In the high-band codec. the signal is divided into frequency bands. and each band is quantized in DCT domain. The DCT coefficients are splitted into magnitude and sign, and each is quantized independently by a specialized method based on its characteristics. In addition. the quantized gain parameter in the low-band codec is utilized in the high-band codec for an enhanced performance. The bandwidth scalable wideband speech codec using G.729E for low-band and the proposed codec for high-band is developed, and it is confirmed that the proposed codec has better subjective performance than 24kbps G.722.1.

• The Journal of the Acoustical Society of Korea
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• v.36 no.1
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• pp.64-69
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• 2017

In this paper, we propose a single-channel speech enhancement method based on restoration of both spectral amplitudes and phases for PTT (Push-To-Talk) communication. The proposed method combines the spectral amplitude and phase enhancement to provide high-quality speech unlike other single-channel speech enhancement methods which only use spectral amplitudes. We carried out side-by-side comparison experiment in various non-stationary noise environments in order to evaluate the performance of the proposed method. The experimental results show that the proposed method provides high quality speech better than other methods under different noise conditions.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.26-30
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• 1997

In this paper, the thickness measurement method of adhesive layers of multilayers using power cepstrum signal processing technique has been proposed. The peak values for reflected signal from each layer have been separated by power cepstrum technique. Therefore, thickness of adhesive layers have been measured by the intervals of peak signal. In the experiment, the adhesive layers of 0.5mm-0.75mm thickness using epoxy(2-Ton and Plastic Steel Putty(A)) between the aluminum and the brass were formed. The adhesive layer thickness which is calculated with data of reflected signal by ultrasonic pulse-echo method was within error 1.34% of the measured values.