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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
The Journal of the Acoustical Society of Korea
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Journal DOI :
The Acoustical Society of Korea
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Volume & Issues
Volume 31, Issue 8 - Nov 2012
Volume 31, Issue 7 - Oct 2012
Volume 31, Issue 6 - Aug 2012
Volume 31, Issue 5 - Jul 2012
Volume 31, Issue 4 - May 2012
Volume 31, Issue 3 - Apr 2012
Volume 31, Issue 2 - Feb 2012
Volume 31, Issue 1 - Jan 2012
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An Experimental Study on the Fitting of 64 Channel Digital Hearing Aid by In-situ Method
Jarng, Soon-Suck ;
The Journal of the Acoustical Society of Korea, volume 31, issue 5, 2012, Pages 273~279
DOI : 10.7776/ASK.2012.31.5.273
In this thesis, a nonlinear compression fitting method was studied for each frequency channel of a 64 channel digital hearing aid. Unlike conventional fitting formula method done from the result of the hearing loss test, the present fitting method uses the auditory threshold of sound pressure measured near the tympanic membrane while ITE (In-The-Ear) hearing aid is fitted into the user's ear canal. Also, the spectral distribution of the voice sound pressure was used for realizing of output sound pressure compression curves against input sound pressure level. Theoretical research results of FFT-iFFT compression algorithm has been evaluated by experimental gain measurements at each different input sound pressure level 50 dB, 70 dB, 90 dB respectively.
A Study on Effective Output Control Technique for Rotational Transmission Beam Drive of Sonar Transmitter
Lee, Byung-Hwa ;
The Journal of the Acoustical Society of Korea, volume 31, issue 5, 2012, Pages 280~287
DOI : 10.7776/ASK.2012.31.5.280
This paper presents a study on the experimental analysis of the impedance characteristics according to the rotational direction of the transmission beam of a cylindrical sensor array. Besides, this suggests a real time control technique of the transmitter output for the effective maximum power transmission, in order to drive efficiently the rotational transmission beam of the active sonar transmitter. The output characteristics of the transmitter and the real-time impedance variations of the sensor array are analyzed under the overload conditions. They are caused by electric and acoustic boundary conditions when the rotational transmission beam is operated. From these results, a new output control method of the transmitter is proposed to protect the transmitter and its loads. It can maximize the output power without the transmission pause even if the transient phenomena occur. The proposed technique is verified from the experiment.
Design of a Variable Resonator for the Sacred Bell of the Great King Seongdeok
Kim, Seock-Hyun ; Jeong, Won-Tae ; Kang, Yun-June ;
The Journal of the Acoustical Society of Korea, volume 31, issue 5, 2012, Pages 288~297
DOI : 10.7776/ASK.2012.31.5.288
This study proposes a design model of the variable type resonator which corrects the temperature variance according to the season, in order to maximize the resonance effect in the Sacred bell of the Great King Seongdeok. In the bell, the 1st natural frequency (64 Hz) and the 2nd natural frequency (168 Hz) are the most important partial tones. Resonance conditions of the two components are determined for the internal acoustic cavity system, which consists of bell body cavity, gap and the resonator. Acoustic frequency response characteristics of the internal cavity are determined by the boundary element analysis using SYSNOISE. As an external factor, temperature variance according to the season largely influences the resonance condition and the length of the resonator should be controlled to maximize the resonance effect. As a measure, this study proposes a design model of the variable type resonator for the Sacred Bell of the Great King Seongdeok, which can control the length at the belfry according to the season.
A Relationship of Tone, Consonant, and Speech Perception in Audiological Diagnosis
Han, Woo-Jae ; Allen, Jont B. ;
The Journal of the Acoustical Society of Korea, volume 31, issue 5, 2012, Pages 298~308
DOI : 10.7776/ASK.2012.31.5.298
This study was designed to examine the phoneme recognition errors of hearing-impaired (HI) listeners on a consonant-by-consonant basis, to show (1) how each HI ear perceives individual consonants differently and (2) how standard clinical measurements (i.e., using a tone and word) fail to predict these differences. Sixteen English consonant-vowel (CV) syllables of six signal-to-noise ratios in speech-weighted noise were presented at the most comfortable level for ears with mild-to-moderate sensorineural hearing loss. The findings were as follows: (1) individual HI listeners with a symmetrical pure-tone threshold showed different consonant-loss profiles (CLPs) (i.e., over a set of the 16 English consonants, the likelihood of misperceiving each consonant) in right and left ears. (2) A similar result was found across subjects. Paired ears of different HI individuals with identical pure-tone threshold presented different CLPs in one ear to the other. (3) Paired HI ears having the same averaged consonant score demonstrated completely different CLPs. We conclude that the standard clinical measurements are limited in their ability to predict the extent to which speech perception is degraded in HI ears, and thus they are a necessary, but not a sufficient measurement for HI speech perception. This suggests that the CV measurement would be a useful clinical tool.
Low-noise Design Method of Small-size Underwater Acoustic System and Module
Lee, Dong-Ho ; Shin, Young-San ; Song, Jin-Ho ; Wee, Jae-Kyung ; Lee, Jeong-Min ; Seol, Jae-Soo ;
The Journal of the Acoustical Society of Korea, volume 31, issue 5, 2012, Pages 309~316
DOI : 10.7776/ASK.2012.31.5.309
This paper suggest configuration method of small-size system and design method of module using multi power. In particular module designed to focus on PDN(Power Distribution Network) and filter configuration method to improve the characteristics of noise sensitive analog IC. For the prevention of high-voltage noise, manufactured module is used the ground-isolated technique and via stitching, and is connected grounds with a series of filters. In this paper, so we analyzed PDN structure through the simulation using lumped modeling and confirmed through measurement. Simulation results, when connecting 4.7uH inductor, we made certain that noise of -7dB decreases much more than when it did not. And it was confirmed 12% less than Background Noise.
Audio Source Separation Method Based on Beamspace-domain Multichannel Non-negative Matrix Factorization, Part I: Beamspace-domain Multichannel Non-negative Matrix Factorization system
Lee, Seok-Jin ; Park, Sang-Ha ; Sung, Koeng-Mo ;
The Journal of the Acoustical Society of Korea, volume 31, issue 5, 2012, Pages 317~331
DOI : 10.7776/ASK.2012.31.5.317
In this paper, we develop a multichannel blind source separation algorithm based on a beamspace transform and the multichannel non-negative matrix factorization (NMF) method. The NMF algorithm is a famous algorithm which is used to solve the source separation problems. In this paper, we consider a beamspace-time-frequency domain data model for multichannel NMF method, and enhance the conventional method using a beamspace transform. Our decomposition algorithm is applied to audio source separation, using a dataset from the international Signal Separation Evaluation Campaign 2010 (SiSEC 2010) for evaluation.
Audio Source Separation Method based on Beamspace-domain Multichannel Non-negative Matrix Factorization, Part II: A Study on the Beamspace Transform Algorithms
Lee, Seok-Jin ; Park, Sang-Ha ; Sung, Koeng-Mo ;
The Journal of the Acoustical Society of Korea, volume 31, issue 5, 2012, Pages 332~339
DOI : 10.7776/ASK.2012.31.5.332
Beamspace transform algorithm transforms spatial-domain data - such as x, y, z dimension - into incidence-angle-domain data, which is called beamspace-domain data. The beamspace transform method is generally used in source localization and tracking, and adaptive beamforming problem. When the beamspace transform method is used in multichannel audio source separation, the inverse beamspace transform is also important because the source image have to be reconstructed. This paper studies the beamspace transform and inverse transform algorithms for multichannel audio source separation system, especially for the beamspace-domain multichannel NMF algorithm.
Stereo-10.2Channel Blind Upmix Technique for the Enhanced 3D Sound
Choi, Sun-Woong ; Hyun, Dong-Il ; Lee, Suk-Pil ; Park, Young-Cheol ; Youn, Dae-Hee ;
The Journal of the Acoustical Society of Korea, volume 31, issue 5, 2012, Pages 340~351
DOI : 10.7776/ASK.2012.31.5.340
In this paper, we proposed the stereo-10.2channel blind upmix algorithm for the enhanced 3D sound. Recently, consumers want to enjoy better sound and the use of a various of multichannel configuration has been steadily improved. Thus, upmix algorithms have been researched. However, conventional upmix algorithms have the problem that distorts the spatial information of original source. To solve this problem and enhance the spatial sound quality, we proposed front and rear channel gain adjustment and 10.2 channel upmix algorithm for each additional channel. The listening test results show that it maintains spatial information of stereo input and enhances 3D sound effects unlike other conventional upmix algorithms.