본 연구는 piezo-electric transducer 등가회로의 모델링에 관한 연구로서 진동시에 발생하는 공진주파수와 harmonics까지 고려한 등가모델을 제시하였다. 실험결과 공진주파수만을 고려한 모델의 경우 harmonics를 제외한 부분에서 평균 6.5%의 오차가 발생하였으며 harmonics까지 고려한 모델에서는 평균오차가 5.3%로서 그 차이점을 줄일 수 있어 실제등가회로에 접근함을 알 수 있었다. 본 연구의 결과는 음향변환기의 전기적 특성을 파악함으로써 전기-음향 상사관계의 제반응용에 이용될 것이 기대된다.

When chopped light inpinges on some condenced matters such as HgS, HgI2 and GaSe semiconductors, in an enclosed cell, the acoustic signals are produced within the cell. These acoustic signals were detected by using a gas-phase microphone in order to investigate the physical properties of the samples. In order to carry out investigation, PA-cell was first designed and made so as to produce higher sensitivity to acoustic signals. Second, an analysis of the photoacoustic spectrum of the various compounds was carried out to obtain the intensity of the PA-signal in terms of light wavelength and to calculate the energy band gaps occuring according to energy transitions. The agreement between the results obtained by this conventional PAS technique and the results obtained by the optical spectrum method was good. In additional analysis conducted on the basis of the R-G theory and the Sze theory are capable of determining the characteristics of energy transition of semiconductors.

A new type single beam photoacoustic spectrometer suitable for measuring optical absorption of condensed powder matter with the automatic calibration capability of a source power spectrum is introduced. The signal processing until of this spectrometer consists of a photoacoustic cell a lock-in amp., a switching circuit and a personal computer. The measured optical absorption spectra of a few material by this method are good agreement with the results obtained by the double-beam photoacoustic spectrometer.

The limitation and validity of Landau-Rumer theory and Akhieser method for the sound attenuation in crystalline quartz is discussed by comparing with some experimental facts. It is turned out that landau-Rumer theory is most likely satisfied at T<40。K and Akhieser method at T>100。K.

本 硏究는 아직까지 國內에서는 航空機騷音 評價方法이 定立되지 않은 段階에서 外國의 航空機騷音 評價方法을 比較해 보고 合理的인 國內의 評價方法을 찾아보며 新說될 國際空港에 활주로 豫定地域인 A, B 두 地久의 타당성을 검토해 보고 또 各國에서 重要時 여기고 있지 않은 高騷音機種이 混入해 들어올 때 얼마만큼 全 騷音에 영향을 미치는가 하는 混入率 補正을 數式化해 봄으로써 앞으로 어떠한 機種이 出現되더라도 騷音度의 增減效果를 豫測해 볼수 있었고 각 混入率別로 補正한 콘라의 比較로서 보다 정확한 騷音 被害 豫想 地域을 推定해보며 그 地域의 騷音評價 및 對策을 알아 보았다. 따라서 本 硏究의 結果를 살펴보면, 첫째, 現在 最少 6dB(A)에서 最大 8dB(A)의 차이가 있는 低騷音被害地域의 차이있음을 알 수 있었다. 둘째, 淸源 新國際空港이 서울과의 近接性(124Km)이 좋지 못하나 長期的인 眼目에서 國土의 均衡發展을 爲한 中部園 空港의 槪念으로 볼 때 바람직한 計劃이라고 본다. 셋째, 新國際空港이 들어설 地域이 A地久일 때는 淸州 地方과 西淸州地域이 騷音被害地域으로 豫想되고 B地久일 때는 증평읍 一帶가 被害地域으로 豫想됨으로 모든 여건을 考慮 淸州地域의 騷音被害를 最少限으로 줄이기 위해서 B地久가 가장 適合하다. 넷째, 航空機騷音 評價는 dB(J) 試算式에 의해 算出하여 騷音 영향 評價를 해야 한다고 생각된다.

파동 이론에 의한 수중에서의 저주파 음파전달은 음향학적 경계 조건에 의해 결정되는 Normal Mode로 특징지어 진다. Normal Mode는 수층(Water Layer)뿐만 아니라 수직적으로 층상 구조인 해저 퇴적층(Subatrate)의 음향 특성을 포함하여 결정되는 파동 방정식의 해로서 이에 의해 수층 및 해저 퇴적층에서의 음압 분포와 감쇠를 계산할 수 있다. 본 논문은 저주파 음파 전달에 관한 Normal Mode 이론에 의하여 음원의 주파수와 해저 퇴적층에서의 음속 분포등에 따른 각 Mode의 음압 분포, 감쇠등에 관한 음향학적 해석으로 원거리까지 진행하는 수중 음파의 해저 퇴적층 투과 심도를 추출하였다.

In this study, the system is proposed which can be used for recognition of Koean single vowles "ㅏ, ㅓ, ㅗ, ㅜ, ㅡ, ㅣ, ㅐ, ㅔ, ㅚ,", and automatic recognition is processed using $\mu$-computer. 3 men of not-being-studied are participated in this experiment. Using the period of vowels, one part of the steady state is selected for high speed recognition, and amplitude comparison method, LPC, PARCOR, and Formant are used for parameter of recognition. Formant is obtained by peak picking method using LPC, and then vowels are recognized by amplitude comparison method, LPC, PARCOR, and Formant. As a result, Recognition rates are 90.1% for amplitude comparison method, 93.1% for LPC, 100% for PARCOR, 88.8% for using formant.

In this paper, the implementation of the real time LPC synthesizer using NEC 77p20, the DSP (Digital Signal Processor) chip which facilitates and simplifies the digital hardware, is considered. This method shows the good quality with the low bit rate below 9.6kbps and has the advantage of the flexibility and the simplicity.

This paper is a study on the Endpoint Detection for Korean Speech Recognition. In speech signal process, analysis parameter was classification from Zero Crossing Rate(Z.C.R), Log Energy(L.E), Energy in the predictive error(Ep) and fundamental Korean Speech digits, /영/-/구/ are selected as date for the Recognition of Speech. The main goal of this paper is to develop techniques and system for Speech input ot machine. In order to detect the Endpoint, this paper makes choice of Log Energy(L.E) from various parameters analysis, and the Log Energy is very effective parameter in classifying speech and nonspeech segments. The error rate of 1.43% result from the analysis.

본 논문에서는 음성신호에 섞인 잡음을 효과적으로 제거하기 위해 Two-sided Transversal Filter를 이용하는 방법을 제시하였다. 이 방법은 기존의 방법과는 달리 신호의 단기자기상관관계(short-term autocorrelation)을 충분히 이용하기 위해 적응 필터의 입력으로, 지연된 신호뿐만 아니라 미래의 신호도 이용하도록 하였다. 이 결과 음성신호에 섞인 잡음(백색잡음, 유색잡음)을 아주 효과적으로 제거함을 알 수 있다.

This paper reports on a Voiced-Unvoiced-Silence Classification of speech for Korean Speech Recognition. In this paper, it is describe a method which uses a Pattern Recognition Technique for classifying a given speech segment into the three classes. Best result is obtained with the combination using ZCR, P1, Ep and classification error rate is less than 1%.

The purposed of this paper is to realize two dimensional acoustic image using focusing method. In the experiment, it consists of phase delay part, 14 analog multiplexers, multi-amplifiers, A/D converter, DMA interfacing parts, two linear arrays(resonance frequency 25 KHz, radius 0.75 Cm diac type). Finally in this experimental result, both electrical deflection and dynamic focusing are realized and gets two dimensional acoustic image using focusing method.

In this paper, a method measuring the reverberation time simultaneously on all of the audio frequency range was studied. The system developed in this study is composed of a pink noise generator, filter bank, and microcomputer with graphic display. In the experiment, reverbration time data measured by existing analog method and by the system were compared, and were almost same through the audio frequency. Based on the conclusion, the subject method has more convenience and accuracy with algorithm program development without existing problems and it was found out that this kind of method is widely available for many branches of room accoustics and architectual acoustics.

Because of layer and scattering in the ocean, there are some problem in algorithm currently used for the recognition of targets. Those are time delay of processing and circuit design. The simple method of detecting direct sound wave in noise caused by time delay is proposed-recognized, estimated, and then direcxt sound wave is reconstructed by the AMDF and $\mu$-processor. 2KHz, 4KHz, 8KHz, 12KHz, 16KHz sound waves are used in experiment. To obtain a reference signal, anechoic water tank is used is processing and aluminium water tank used instead of real ocean. As a result, there are a few errors which caused by anechoic water tank error, decreasing of frequency make errors. Possibility of application to Sonar Signal Processing is proved.

In this study, the underwater acoustic images were obtained by ultrasonicwave. The experiment was performed in the anechoic watertank, using a passive sonar array for one and two sound source respectively by X-Y scanning technique. The receiving array was consist of 8 disc type transducers with 1.5cm diameter at 25KHz resonance frequency. The scanned data were processed by the FORTRAN IV algorithm for the reconstruction of image, and the image had some noise due to the surface reflected waves. As the result, it was found that the acoustic imaging by electrical deflection and dynamic focusing technique is applicable to SONAR with the suppression of surface reflected wave.

Traditional methods for estimating the location of underwater target, i.e. the triangulation method and the wavefront curvature method, have been utilized. The location of a target is defined by the range and the bearing, which estimates can be obtained by evaluating the time delay between neighboring sensors. Many components of error occur in estimating the target range, among which the error due to the fluctuation of heading angle is outstanding. In this paper, the wavefront curvature method was used. We considered the error due to the heading fluctuation as the $\beta$-density process, from which we analized the range estimates with $\beta$-density function exist in some finite limits, and its mean value and variation are depicted as a function of true range and heading fluctuation. Given heading angles and sensor separation, maximum estimated heading errors are presented as a function of true range.

수동 Wave-front curvature ranging 방법의 통계적성능에 대한 일반적 결과가 유도된다. 센서들의 비선형 및 부동간격 센서들의 운동효과로 인한 표적거리의 편의(bias) 및 편차9variance)가 센서간의 운동이 상관관계를 가지고 있을 때 유도된다. 이 논문은 센서운동이 정규분포(normal distribution)를 가지며 평균이 0, 같은 편차 및 투시할 수 있는 higher moment를 가지고 있다고 가정한다. 유도된 일반적 결과를 이용하여 거리편의 및 편차는 표적거리, 방위, 센서운동의 상관계수 및 방향각(heading)표준편차 변화에 대해 그림으로 나타나 있다. 표적거리가 센서간격보다 아주 큰 경우 센서간격이 감소할수록, 표적거리가 증가할수록, 거리편의 및 편차는 증가한다. 상관계수가 증가할수록 거리편의는 직선적으로 감소하며, 거리편차는 센서 방향각 오차가 클 때 최소가 되는 상관계수가 존재하며 이는 표적거리, 방위 및 센서간격의 복합함수이다.