• Journal of the Society of Disaster Information
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• v.13 no.1
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• pp.51-58
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• 2017

This paper proposes a direction estimation scheme with directional speaker and smart device for evacuation guidance. When there is worst disaster environment filled with smoke and noisy sound, evacuee can not get any information about evacuation routes. The proposed scheme can be used for detecting evacuation routes with audible and inaudible signal from directional speaker. At this point, evacuee can get evacuee guidance by using smartphone application that the proposed scheme is applied. The performance of the proposed scheme is evaluated by experiment with three different types of smart devices in large indoor environment. The purpose of experiment is to detect the direction of transmitted signal from directional speaker. Therefore, The experiment is conducted by analyzing the strength of transmitted signal by distance. The experimental results show that even if the smart device is located up to 20m away from the speaker, it is possible to detect the sending direction of the signal. We confirmed the possibility of the proposed technology in 8kHz and 20kHz signal detection by smart device.

• The Journal of the Acoustical Society of Korea
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• v.18 no.2E
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• pp.10-17
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• 1999

The aim of the work described in this paper is to develop a complex underground acoustic system which detects and localizes the origin of an underground hammering sound using an array of hydrophones located about 100m underground. Three different methods for the sound localization will be presented, a time-delay method, a power-attenuation method and a hybrid method. In the time-delay method, the cross correlation of the signals received from the array of sensors is used to calculate the time delays between those signals. In the power-attenuation method, the powers of the received signals provide a measure of the distances of the source from the sensors. In the hybrid method, both informations of time-delays and power-ratios are coupled together to produce better performance of position estimation. A new acoustic imaging technique has been developed for improving the hybrid method. This new acoustic imaging method shows the multi-dimensional distribution of the normalized cost function, so as to indicate the trend of the minimizing direction toward the source location. For each method the sound localization is carried out in three dimensions underground. The distance between the true and estimated origins of the source is 28m for a search area of radius 250m.

• The Journal of the Acoustical Society of Korea
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• v.36 no.5
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• pp.345-352
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• 2017

We propose a methods which is reducing direction estimation error of sound source in the reverberant and noisy environments. The proposed algorithm divides speech signal into voice and unvoice using VAD. We estimate the direction of source when current frame is voiced. TDOA (Time-Difference of Arrival) between microphone array using the GCC-PHAT (Generalized Cross Correlation with Phase Transform) method will be estimated in that frame. Then, we compare the peak value of cross-correlation of two signals applied to estimated time-delay with other time-delay in time-table in order to improve the accuracy of source location. If the angle of current frame is far different from before and after frame in successive voiced frame, the angle of current frame is replaced with mean value of the estimated angle in before and after frames.

• Advances in aircraft and spacecraft science
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• v.3 no.3
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• pp.331-349
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• 2016

Stiff and light fibre reinforced composites as used in air- and space-craft applications tend to high sound emission. Therefore, the damping properties are essential for the entire structural and acoustic engineering. Viscous damping is an established and reasonably linear model of the dissipation behaviour. Commonly, it is assumed to be isotropic and constant over all modes. For anisotropic materials it depends on the fibre orientation as well as the elastic and thermal material properties. To portray the orthogonal anisotropic behaviour, a model for unidirectional fibre reinforced plastics (frp) has been developed based on the classical laminate theory by ADAMS and BACON starting in 1973. Their approach includes three damping coefficients - for longitudinal damping in fibre direction, damping transversal to the fibres and shear based dissipation. The damping of a laminate is then accumulated layer wise including the anisotropic stiffness. So far, the model has been applied mainly to thermoset matrix materials. In this study, an experimental parameter estimation for different thermoplastic frp with angle ply and cross ply layups was carried out by measuring free vibrations of cantilever beams. The results show potential and limits of the ADAMS/BACON damping criterion. In addition, a possibility of modelling the anisotropic damping is shown. The implementation in standard FEA software is used to study the influence of boundary conditions on the damping properties and numerically estimate the radiated sound power of thin-walled frp parts.

• Korean Journal of Fisheries and Aquatic Sciences
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• v.54 no.5
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• pp.798-807
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• 2021

The directivity characteristics of acoustic transducers for conventional single-beam echo sounders considerably limit the detection of fish-size information in acoustic field surveys. To overcome this limitation, using the split-aperture technique to estimate the direction of arrival of single-echo signals from individual fish distributed within the sound beam represents the most reliable method for fish-size classification. For this purpose, we design and develop a split-beam data acquisition and processing system to obtain fish-size information in conjunction with a 50-kHz single-beam echo sounder. This split-beam data acquisition and processing system consists of a notebook PC, a field-programmable gate array board, an external single-transmitter module with a matching network, and four-channel receiver modules operating at a frequency of 50-kHz. The functionality of the developed split-beam data processor is tested and evaluated. Acoustic measurements in an experimental water tank showed that the developed data acquisition and processing system can be used as a fish-sizing echo sounder to estimate the size distribution of individual fish, although an external single-transmitter module with a matching network is required.

• The Journal of the Acoustical Society of Korea
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• v.33 no.5
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• pp.326-334
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• 2014

This paper proposes a novel covariance matrix estimation scheme for minimum variance distortionless response (MVDR) beamforming. By accurately tracking direction-of-sound source arrival (DoA) information using audio-visual sensors, the covariance matrix is efficiently estimated by adopting a variable forgetting factor. The variable forgetting factor is determined by considering signal-to-interference ratio (SIR). Experimental results verify that the performance of the proposed method is superior to that of the conventional one in terms of interference/noise reduction and speech distortion.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2C
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• pp.246-253
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• 2010

In this paper, a method of removing the space skew and cross-talk cancellation is proposed where the self-generated signals from the subject are used to obtain the subject's location. In the proposed method, the good spatial sound image is maintained even when the listener moves from the sweet spot. Two major parts of the proposed method are as follows: listener position tracking using the stimuli from the subject and removal of the space skew and cross-talk signals. Listener position tracking is achieved by estimation of the time difference of arrival (TDoA). The position of the listener is then computed using the Talyer-series estimation method. The head-related transfer functions (HRTF) are used to remove the space skew and cross-talk signals, where the direction of the HRTF is given by the one estimated from the listener position tracking. The performance evaluation is carried out on the signals from the 100 subjects that are composed of the 50 female and 50 male subjects. The positioning accuracy is achieved by 70%~90%, under the condition that the mean squared positioning error is less than $0.07m^2$. The subjective listening test is also conducted where the 27 out of the 30 subjects are participated. According to the results, 70% of the subjects indicates that the overall quality of the reproduced sound from the proposed method are improved, regardless of the subject's position.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2015.11a
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• pp.16-17
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• 2015

본 논문에서는 실내 환경 중 TV 시청환경에서 마이크로폰 어레이를 이용하여 다양한 다중 음원 방향을 추정하는 기법을 제안한다. 제안된 기법은 기존의 하나의 음원에 특화되어 있는 GCC-PHAT 기반의 방법을 GCC-PHAT 버퍼와 NMF를 도입하여 다중음원의 방향 추정을 가능하게 만들었다. 제안된 기법의 성능을 평가하기 위해서 실 거주 환경에서 발생하는 소음원과 TV 소리 방향 추정 결과에 대한 실측치와 추정치 간의 오차인 절대 평균오차를 측정하였으며, 실험 결과 제안한 기법이 기존의 방법인 GCC-PHAT보다 우수한 추정 성능을 보임을 확인하였다.

• The Journal of the Acoustical Society of Korea
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• v.39 no.3
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• pp.179-190
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• 2020

In this paper, we estimated the Direction of Arrival (DOA) using Conventional BeamForming (CBF), adaptive beamforming and compressive beamforming. Minimum Variance Distortionless Response (MVDR) and Multiple Signal Classification (MUSIC) are used as the adaptive beamforming, and grid-free compressive sensing is applied for the compressive sensing beamforming. Theoretical background and limitations of each technique are introduced, and the performance of each technique is compared through simulation and real experiments. The real experiments are conducted in the presence of reflected signal, transmitting a sound using two speakers and receiving acoustic data through a linear array consisting of eight microphones. Simulation and experimental results show that the adaptive beamforming and the grid-free compressive beamforming have a higher resolution than conventional beamforming when there are uncorrelated signals. On the other hand, the performance of the adaptive beamforming is degraded by the reflected signals whereas the grid-free compressive beamforming still improves the conventional beamforming resolution regardless of reflected signal presence.

• The Journal of the Acoustical Society of Korea
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• v.29 no.7
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• pp.458-467
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• 2010

In this paper, a new audio reproduction system was developed in which the cross-talk signals would be reasonably cancelled at an arbitrary listener position. To adaptively remove the cross-talk signals according to the listener's position, a method of tracking the listener position was employed. This was achieved using the two microphones, where the listener direction was estimated using the time-delay between the two signals from the two microphones, respectively. Moreover, room reverberation effects were taken into consideration where linear prediction analysis was involved. To remove the cross-talk signals at the left-and right-ears, the paths between the sources and the ears were represented using the KEMAR head-related transfer functions (HRTFs) which were measured from the artificial dummy head. To evaluate the usefulness of the proposed listener tracking system, the performance of cross-talk cancellation was evaluated at the estimated listener positions. The performance was evaluated in terms of the channel separation ration (CSR), a -10 dB of CSR was experimentally achieved although the listener positions were more or less deviated. A real-time system was implemented using a floating-point digital signal processor (DSP). It was confirmed that the average errors of the listener direction was 5 degree and the subjects indicated that 80 % of the stimuli was perceived as the correct directions.