• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.603-611
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• 2003

This paper proposes an effective feature compensation scheme based on speech model for achieving robust speech recognition. Conventional model-based method requires off-line training with noisy speech database and is not suitable for online adaptation. In the proposed scheme, we can relax the off-line training with noisy speech database by employing the parallel model combination technique for estimation of correction factors. Applying the model combination process over to the mixture model alone as opposed to entire HMM makes the online model combination possible. Exploiting the availability of noise model from off-line sources, we accomplish the online adaptation via MAP (Maximum A Posteriori) estimation. In addition, the online channel estimation procedure is induced within the proposed framework. For more efficient implementation, we propose a selective model combination which leads to reduction or the computational complexities. The representative experimental results indicate that the suggested algorithm is effective in realizing robust speech recognition under the combined adverse conditions of additive background noise and channel distortion.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.358-367
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• 2016

SSL (Sound Source Localization) has been applied to several applications such as man-machine interface, video conference system, smart car and so on. But in the process of sound source localization, angle estimation error is occurred mainly due to the non-linear characteristics of the sine inverse function. So an approach was proposed to decrease the effect of this non-linear characteristics, which divides the microphone's covering space into narrow regions. In this paper, we proposed an optimal space dividing way according to the pattern of microphone array. In addition, sound source's 2-dimensional position is estimated in order to evaluate the performance of this dividing method. In the experiment, GCC-PHAT (Generalized Cross Correlation PHAse Transform) method that is known to be robust with noisy environments is adopted and triangular pattern of 3 microphones and rectangular pattern of 4 microphones are tested with 100 speech data respectively. The experimental results show that triangular pattern can't estimate the correct position due to the lower space area resolution, but performance of rectangular pattern is dramatically improved with correct estimation rate of 67 %.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.3 s.108
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• pp.291-297
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• 2006

The BVI(blade vortex interaction) noise Prediction has been one of the most challenging acoustic analyses in helicopter aeromechanical Phenomenon. It is well known high resolution airloads data with accurate tip vortex positions are necessary for the accurate prediction of this phenomenon. The truly unsteady time-marching free-wake method, which is able to capture the tip vortices instability in hover and axial flights, is expanded with the rotor flapping motion and trim routine to predict unsteady airloads in forward and descent flights. And Farassat formulation 1-A based on the FW-H equation is applied for the noise prediction considering the blade flapping motion. Main objective of this study is to validate the newly developed prediction code. To achieve the objective, the descent flight condition of AH-1 OLS(operational loads survey) configuration is analyzed using present code. The predicted sectional thrust distribution and sectional airloads time histories show the present scheme is able to capture well the unsteady airloads caused by a parallel BVI. Finally, the predicted noise data, observed in two different positions where are 3.44 times of rotor radius far from the hub center, are quite reasonable agreements with the experimental data compared to the other analysis results.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.10
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• pp.116-122
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• 2016

This paper proposes a practical object shape reconstruction method using an underwater imaging sonar. In order to reconstruct the object shape, three methods are utilized. Firstly, the vertical field of view of imaging sonar is modified to narrow angle to reduce an uncertainty of estimated 3D position. The wide vertical field of view makes the incorrect estimation result about the 3D position of the underwater object. Secondly, simple noise filtering and range detection methods are designed to extract a distance from the sonar image. Lastly, a low pass filter is adopted to estimate a probability of voxel occupancy. To demonstrate the proposed methods, object shape reconstruction for three sample objects was performed in a basin and results are explained.

• The Journal of the Acoustical Society of Korea
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• v.37 no.5
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• pp.276-283
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• 2018

The purpose of the study is to identify a sound generating organ that has a major influence on the central frequency of the cicadas with the Helmholtz resonator structure for the first time. The sound of cicadas Cryptotympana atrata and Hyalessa fuscata were recorded and analyzed, then the motion of the tymbals was analyzed with a high-speed camera to compare the relationship between the frequency of sound and the motion of the tymbals. As a result, there was little difference in the frequency distribution of calling song and scream for two species. The tymbals of C. atrata oscillated in three vibration modes, while those of H. fuscata oscillated in one mode. There was no difference in the frequency of both tymbals of both cicadas, and three vibration modes of C. atrata generated sound with different frequency bands. The frequency band of tymbals and the central frequency band of calling song were very similar. In conclusion, it is presumed that the frequency of the cicadas with the Helmholtz resonator structure was determined by mode frequency of the tymbals than resonance condition of the abdomen.

• The Journal of the Acoustical Society of Korea
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• v.36 no.4
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• pp.238-246
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• 2017

High frequency ultrasound imaging typically suffers from low sensitivity due to the small aperture of high frequency transducers and shallow imaging depth due to the frequency-dependent attenuation of ultrasound. These limitations should be overcome to obtain high-frequency, high- resolution ultrasound images. One practical solution to the problems is a high-performance signal receiver capable of detecting a very small signal and amplifying the signal with minimal electronic noise addition. This paper reports a recently developed low-noise, wideband ultrasound receiver for high-frequency, high-resolution ultrasound imaging. The developed receiver has an amplification gain of up to 73 dB and a variable amplification gain range of 48 dB over an operating frequency of 80 MHz. Also, it has an amplification gain flatness of ${\pm}1dB$. Due to these high performances, the developed receiver has a signal-to-noise ratio of at least 8.4 dB and a contrast-to-noise ratio of at least 3.7 dB higher than commercial receivers.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.41-48
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• 1997

This paper proposes the methods for improving the recognition rate of theARS, especially equipped with the speech recognition capability. Telephone speech, which is the input to the ARS, is usually affected by the announcements from the system, channel noise, and channel distortion, thus directly applying the recognition algorithm developed for clean speech to the noisy telephone speech will bring the significant performance degradation. To cope with this problem, this paper proposes three methods: 1)the accurate detection of the inputting instant of the speech in order to immediately turn off the announcements from the system at that instant, 2)the effective end-point detection of the noisy telephone speech on the basis of Teager energy, and 3)the SDCN-based compensation of the channel distortion. Experiments on speaker-independent, noisy telephone speech reveal that the combination of the above three proposed methods provides great improvements on the recognition rate over the conventional method, showing about 77% in contrast to only 23%.

• The Journal of the Acoustical Society of Korea
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• v.40 no.3
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• pp.247-253
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• 2021

Directional ultrasonic speakers that are used to hear sound only in a specific area have been continuously researched on various improvements in terms of sound quality and cost compared to general speakers. In this paper, we propose a DSP based hybrid-type ultrasonic speaker that can be heard at the same time as a general speaker in order to compensate for the sound in the low-band range, considering that it is difficult to hear the low-band sound below 500 Hz due to the sensor characteristics of the ultrasonic speaker. In the case of the system that is implemented by simply connecting a general speaker and an ultrasonic speaker, there are issues of high cost and difficulties of control as two amplifiers are used to playback ultrasonic and general sound sources. In addition, sound quality deteriorates due to the difference in playback time between ultrasonic and general sound sources. In order to improve issues of cost, control and sound quality, we developed hybrid-type ultrasonic speaker with a DSP based amplifier that can simultaneously playback by synchronizing the general sound source with the regenerated ultrasonic sound source, in addition to implement the existing CODEC functions such as Dynamic Range Control (DRC) and Equalizer (EQ).

• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.167-174
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• 2016

Underwater communication is difficult to increase the communication capacity because the carrier frequency is lower than that of radio communications on land. This is limited to the bandwidth of the signal under the influence of the characteristics of an ocean medium. As the high transmission speed and large transmission capacity have become necessary in the limited frequency range, the studies on MIMO (Multiple Input Multiple Output) communication have been actively carried out. The performance of the MIMO communication is lower than that of the SIMO (Single Input Multiple Output) communication because cross-talk occurs due to multiusers along with inter symbol interference resulting from the channel characteristics such as delay spread and doppler spread. Although the adaptive equalizer considering multi-channels is used to mitigate the influence of the cross-talk, the algorithm is normally complicated. In this paper, time reversal mirror technique with the characteristic of a self-equalization will be applied to simplify the compensation algorithm and relieve the cross-talk in order to improve the communication performance when the signal transmitted from two channels is received over interference on one channel in the same time. In addition, the performance of the MIMO communication based on the time reversal mirror is verified using data from the SAVEX15(Shallow-water Acoustic Variability Experiment 2015) conducted at the northern area of East China Sea in May 2015.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.39 no.3
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• pp.239-248
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• 2003

This paper was conducted as an attempt in order to construct the data bank of target strength for acoustic estimation of fish length in the coastal waters of Korea. The fish length dependence of acoustic target strength for 13 large yellow croakers (Pseudosciaena crocea) at 75 kHz was investigated and the prediction of the target strength by using the Kirchhoff-Ray Mode model (KRM model) was compared with target strength measurements. The results obtained are summarized as follows; 1. In the averaged target strength pattern for 13 large yellow croakers the maximum target strength was -35.13 dB at $-13.35^{\circ}$ on a tilted angle. 2. The relationship between fork length(L, cm) and averaged target strength(TS, dB) was expressed as follows; TS=23. 76log (L) -73.45 (r=0.47) TS=20log(L) -67.35 From this result, the conversion coefficient was -73.45 dB and 6.1 dB lower than the coefficient -67.35 dB where the value of the slope of the regression equation is forced to be 20. 3. Averaged target strength and a length conversion coefficient derived from a target strength histogram for 13 large yellow croakers of mean length 25.59 cm were -41.23 dB, -69.72 dB, respectively. 4. In the range of $$2;{\ll} L (fish length /{\lambda}(wave length);{\ll}40$$, the prediction of the averaged target strength by the KRM model increased gradually with the increasing of $L/{\lambda}$ and was lower than the measured target strength.