Go to the main menu
Skip to content
Go to bottom
REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
The Journal of the Acoustical Society of Korea
Journal Basic Information
Journal DOI :
The Acoustical Society of Korea
Editor in Chief :
Volume & Issues
Volume 19, Issue 8 - Nov 2000
Volume 19, Issue 7 - Oct 2000
Volume 19, Issue 6 - Aug 2000
Volume 19, Issue 5 - Jul 2000
Volume 19, Issue 4 - May 2000
Volume 19, Issue 3 - Apr 2000
Volume 19, Issue 2 - Feb 2000
Volume 19, Issue 1 - Jan 2000
Volume 19, Issue 3E - 00 2000
Volume 19, Issue 2E - 00 2000
Volume 19, Issue 1E - 00 2000
Selecting the target year
Influences of Air Cavity on the Sensitivity of a Mandrel Type fiber Optic Acoustic Sensor
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 3~7
This paper is on the sensitivity characteristics of a concentric composite mandrel type fiber optic acoustic sensor with inclusion of an air cavity With the finite element method, we have analyzed sensitivity variation of the sensor in relation to its geometrical factors such as thickness of the air cavity, thickness of the foaming layer, and the ratio of inner diameter/outer diameter of the mandrel. Results of the analysis suggest a thicker air cavity, a thinner foaming layer, and a smaller ratio of the inner diameter/outer diameter of the mandrel to be desirable for higher sensitivity. The sensor structure designed with the above rules provides the sensitivity of about 0.8dB higher than that of a normal concentric composite mandrel sensor without the inherent air cavity.
Influence of Environmental Conditions on the Sensitivity of a Mandrel Type Fiber Optic Acoustic Sensor
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 8~12
This paper describes the sensitivity stability of a mandrel type fiber optic acoustic sensor with respect to its environmental conditions such as hydrostatic pressure and underwater temperature. The sensors under consideration have various mandrel structures such as a cylindrical mandrel, a concentric composite mandrel, and an air-backed concentric composite mandrel. The analysis results show that the sensors have such good robustness, less than 0.15dB, in its sensitivity with respect to the variation in hydrostatic pressure. Further, the nylon concentric composite mandrel type sensor including an air cavity turns out to have the most superior stability than others to the underwater temperature variations.
A Study on the Enhanced Time Domain Aliasing Cancellation Transform of the AC-3 Algorithm
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 13~18
This paper presents the result of a technique to enhance TDAC in the AC-3 algorithm. To reduce block boundary noise without decreasing the performance of transform coding, We propose new special windows which improve the defect of the AC-3 algorithm that could not properly cancel aliasing in the transient period. In addition, a fast MDCT calculation algorithm based on a fast Fourier transform, is adopted.
한국어 억양의 트리 기반 모델링
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 19~32
A Study on the fabrication of Bandpass filter Using a Simulator
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 33~39
We have studied to obtain the frequency characteristics of the Surface Acoustic Wave(SAW) bandpass filter, having low shape factor, it's interdigital transducer(IDT) was formed on the 35° Y-cut X-propagation Quartz substrate and was evaporated by Aluminium. And then, we performed computer-simulation by a simulator. And, we can design that the apodization weighted type IDT as an input transducer of the filter and the withdrawal weighted type IDT as an output transducer of the filter from the results of our computer-simulation. Also, we have employed that the number of pairs of the input and output IDT are 2200 pairs and 1000pairs, respectively and used the Kaiser-Bessel window function in order to minimize the effect of ripple. And, while the width and the space of IDT's finger are 6㎛ m and 5.75㎛, respectively and we could obtain the resonable results when the IDT thickness was 6000Å in consideration of the ratio of SAW's wavelength, and IDT aperture is 2mm. Frequency response of the fabricated SAW bandpass filter has the property that the center frequency is about 70MHz, shape factor is less than 1.3, bandwidth at the 1.5dB is probably 1.3MHz, out-band attenuation is almost -45dB, insertion loss is 19dB and ripple in the width of bandpass is 1dB approximately. Therefore, these frequency characteristics of the fabricated SAW bandpass filter are agreed well with the designed values.
Segmentation of Continuous Speech based on PCA of Feature Vectors
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 40~45
In speech corpus generation and speech recognition, it is sometimes needed to segment the input speech data without any prior knowledge. A method to accomplish this kind of segmentation, often called as blind segmentation, or acoustic segmentation, is to find boundaries which minimize the Euclidean distances among the feature vectors of each segments. However, the use of this metric alone is prone to errors because of the fluctuations or variations of the feature vectors within a segment. In this paper, we introduce the principal component analysis method to take the trend of feature vectors into consideration, so that the proposed distance measure be the distance between feature vectors and their projected points on the principal components. The proposed distance measure is applied in the LBDP(level building dynamic programming) algorithm for an experimentation of continuous speech segmentation. The result was rather promising, resulting in 3-6% reduction in deletion rate compared to the pure Euclidean measure.
Characteristics of Surface Backscattering Signal in the Coastal Bay
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 46~53
In coastal bay waters, bubbles are generated by relatively heavy ship-traffic, breaking waves due to man-made structures and biological activities. Therefore, the bubble-generating mechanism as well as the bubble density distribution in the bay are quite different from the open ocean where breaking waves are major contributor for bubble density distribution. High frequency surface-backscattered signals were obtained in the coastal bay waters and they were analyzed to compare with those from the open waters in terms of the sea-surface backscattering strength at various grazing angles, the reverberation characteristics in the sub-surface layer and spectral spreading of the scattered signals. The results show that, the surface scattered signals have an irregular distribution of amplitude in time and the width of the spectral spreading is wider than that of the open sea with rough surface. Furthermore, the amplitude distribution of the reverberation signals is not following the Rayleigh distribution, that is eon to be a typical pattern for the open ocean. The results of our analysis imply that the bubble size and the bubble density in the bay are quite different from those observed in the open waters.
Acoustic Source Models for MUSIC to Identifying Near Field Source
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 54~60
Acoustic source localization using MUSIC etc. utilizes the propagation model in the medium. A plane wave model is a well-known source model for the identification of distant sources in the SONAR and a monopole source model becomes the one for the identification of rather near range sources. This paper introduces a dipole source model and a tripole source model consisting of one monopole and one dipole source. The simplifying procedures provide the simplified factors rather than the superposition of the relating monopole sources. The simulations show that the tripole model is useful in the general case including pure monopole, pure dipole, or pure quadrupole source identification.
Performance Improvement on Hearing Aids Via Environmental Noise Reduction
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 61~67
Recent progress in digital and VLSI technology has offered new possibility fer noticeable advance of hearing aids. Yet, environmental noise remains one of the major problems to hearing aid users. This paper describes results which speech recognition performance and speech discrimination performance was measured for listeners with sensorineural hearing loss, while listeners in speech-band noise. In addition, to ameliorate hearing-aided environments of hearing impaired listeners, environmental noise reduction using speech enhancement techniques are investigated as a front-end of conventional hearing aids. Speech enhancement techniques are implemented in a realtime system equipped with DSP board. The clinical test results suggest that the speech enhancement technique may work in synergy with gain functions fer the greater SNR improvement as the preprocessing algorithm of digital hearing aids.
Improved Minimum Variance Matched field Processing Technique for Underwater Acoustic Source Localization
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 68~72
Matched field processing technique is performed by considering complex underwater environments. Specially, the performance of minimum variance processor is greatly degraded by eigenvalue problem. In this paper, we propose the minimum variance matched field processor using shaping matrix. This shaping matrix makes that the input covariance matrix is invertible and enhances the desired acoustic source component. It was proved effectively range/depth localization of the proposed method with simulated data and vertical array data collected by NATO SACLANT Center north of the island of Elba off the Italian west coast.
Study for Improving Acoustic Characteristics of the Conference Room
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 73~82
This paper presents a significant improvement achieved in the acoustic characteristics of a conference room of I-company originally designed and built without consideration for room acoustic characteristics. The reverberation time of the room was excessive, which resulted in poor 'Definition' for speech. Thus the improvement of room acoustic characteristics was required. In this study, experiments and simulations using the principles of geometrical acoustics were executed to improve the acoustic characteristics in the room. By performing a series of simulation, it has become possible to predict acoustic parameters for an improvement plan. And the improvement plan of room acoustic characteristics was established. And for the verification of the simulation results, the experimental data were compared and evaluated against those from the simulation. Additionally, the proposed method in this paper demonstrated its effectiveness by successfully improving the room acoustic characteristics resulted from repair construction.
A Study on Noisy Speech Recognition Using Discriminative Training for PMC Algorithm
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 83~89
In this paper, we proposed a discriminative adaptation method for PMC algorithm and achieved improved speech recognition rate. For the adaptation, we adopted modified PMC(MPMC) which is a variant of PMC and discriminatively adapted the association factor for each mixture of the HMM in the MPMC. From the recognition experiments, the proposed method showed better recognition rate than the conventional PMC. Also, compared with STAR algorithm which is another model parameter compensation method, the proposed method showed superior performance when the SNR is very low and the adaptation data is not sufficient.
Estimation of Critical Threshold for Rejection in HMM Based Recognition Systems
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 90~94
In this paper, we propose an efficient method of estimating a critical threshold which is used to reject unreliable patterns in a HMM based recognition system. The rejection methods based on the anti-models which are formulated as the statistical hypothesis determine whether or not to accept an input pattern by comparing the likelihood ratio of HMM and anti-models to a critical threshold. It is quite difficult to fix a threshold for the probability of a HMM because the range of such probabilities varies severely depending on the chosen class model. We estimate the critical threshold, which is very class-dependent, using the likelihood scores for the training database. In our experiments, we applied the proposed estimating method of the threshold to the HMM based 3D hand gesture recognition system. We found that this method can be used successfully for rejecting unreliable input gestures regardless of the types of anti-models.
Acoustic Characteristics and Auditory Cues for Korean Lax vs. Tense Fricative Distinction
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 95~100
The purpose of this paper is to show their distinctive auditory cues. Until now research of acoustic characteristics has been confined to simple experiments concerning the restricted conditions. Therefore this paper examines all of the acoustic characteristics of the lax and tense Fricative Consonants and shows to how acoustic characteristics can be used to differentiate lax and tense Fricative Consonants. The results of this paper are (a) auditory cues are especially important if there is a large difference between acoustic characteristics, (b) the lax and tense Fricative Consonant's distinctive auditory cues contain a hierarchy, and (c) there is a different hierarchy between CV and VCV.
On a Speech Coding Algorithm for Low Cost Implementation of Voice Telegram System
The Journal of the Acoustical Society of Korea, volume 19, issue 2, 2000, Pages 101~105
A telegram has been used to transmit the emergency news or celebration message. So, it has been very important media in our life. Although the telegram processing is more and more convenient, on the other hand, the telegram service contains only text message. The voice telegram is that delivering user's voice with text message. So, the voice telegram can be delivered sender's emotions and feelings. However, since voice information contains lots of data, large memory size and high cost processor are needed to deliver itself. In this paper, we proposed a new speech waveform coding method that has low complexity and low cost implementation for the voice telegram system. First, we fixed one basic speech waveform per pitch period and measured the waveform similarity between basic and neighbor speech waveform. Second, if the similarity satisfied threshold values, we compress the neighbor speech waveform with pitch and magnitude value per pitch period and if not, we save speech waveform. When the compression is about 45%, we obtained about 4 point in MOS.