• 음성과학
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• 제10권1호
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• pp.117-129
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• 2003

Many people suffer from various laryngeal diseases. Since we can notice voice change easily, acoustic analysis can be helpful to diagnose the diseases. Several attempts have been made to clarify the relation between the parameters and the state of sick vocal folds but any decisive parameters are not found yet. The purpose of this study was to select and develop those parameters useful for diagnosing and differentiating laryngeal diseases. We examined eight MDVP parameters, and two additional MFCC and LPC parameters obtained from the production of an open vowel by 252 subjects with or without laryngeal diseases. Using a statistical procedure through the artificial neural networks, we attempted to differentiate laryngeal disease groups. Results showed that the LPC parameters indicated the highest differentiating rate by the networks followed by the MFCC and the MDVP parameters. In addition, Jita, Shim and NHR among the MDVP parameters came out better parameters in diagnosing laryngeal diseases.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2013년도 추계학술대회 논문집
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• pp.346-353
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• 2013

Industrial processes need to be monitored in real-time based on the input-output data observed during their operation. Abnormalities in an induction motor should be detected early in order to avoid costly breakdowns. To early identify induction motor faults, this paper effectively estimates spectral envelopes of each induction motor fault by utilizing a linear prediction coding (LPC) analysis technique and an expectation maximization (EM) algorithm. Moreover, this paper classifies induction motor faults into their corresponding categories by calculating Mahalanobis distance using the estimated spectral envelopes and finding the minimum distance. Experimental results shows that the proposed approach yields higher classification accuracies than the state-of-the-art approach for both noiseless and noisy environments for identifying the induction motor faults.

• 한국음향학회지
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• 제14권3호
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• pp.49-60
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• 1995

본 논문은 소규모 시스템에 적용 가능한 한국어 문자-음성 변환 시스템의 설계 및 구현에 대한 연구를 목적으로 한다. 본 논문에서 채택한 음성합성 방법은 파라메터 합성법으로서 LPC(linear Predictive Coding)계열의 PARCOR(PARtial autoCORrelation) 계수를 음향 파라메터로 사용하였으며, 음성합성 단위로는 가장 기본적인 단위인 음소를 채택하였다. 합성 파라메터로는 유성음의 경우 PARCOR계수, 피치, 진폭을 무성음의 경우 잔차신호와 PARCOR계수를 사용하였다. 특히 무성음의 경우 LPC합성시 음질이 떨어진다는 단점이 있었으나, 본 논문에서는 LPC분석시 얻어지는 잔차신호를 무성음의 여기신호로 사용하여 단어 단위의 합성에서 60%의 이해도를 얻을 수 있었다. 합성결과 단어 단위의 합성에 적용 가능하였고, 문장단위의 합성을 위해서는 음소 지속시간 조절에 대한 연구가 진행되어야 할것이다. 본 논문의 구현환경으로는 486 PC상에서 음성의 입,출력을 위해 70[Hz]-4.5[KHz] 대역통과 필터와 증폭기, 그리고 TMS320C30 디지털 신호처리 프로세서를 장착한 DSP 보드를 사용하였다.

• 전자공학회논문지B
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• 제30B권10호
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• pp.42-49
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• 1993

In this paper, we evaluate the quality of synthesized speech by rule using the LPC CD as a objective measure, and then compare the test result with the subjective one. Speech used for the test consists of 108 words which are selected by word construction method using Korean attribute and frequency distribution, synthesized by demi-syllable rule. By evaluating the quality of synthesized speech by reule objectively, we have tried to resolve the problems such as lots of evaluation time, expansion of test scale, and variables of analysis result arised by subjective measure. We have, also, proved the validity of the objective test using the LPC CD, by comparing intelligibility which is the index for the subjective quality evaluation of synthesized speech by rule with MOS. From this results, we can provide a guide for quality assessment that would be useful in the R&D of synthesis method and the commercial products using synthesized speech.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 제11회 음성통신 및 신호처리 워크샵 논문집 (SCAS 11권 1호)
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• pp.91-94
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• 1994

This paper is a study on the design and implementation of the Korean Tet-to-Speech system. In this paper, parameter symthesis method is chosen for speech symthesis method and PARCOR coeffient, one of the LPC analysis, is used as acoustic parameter, We use a diphone as synthesis unit, it include a basic naturalness of human speech. Diphone DB is consisted of 1228 PCM files. LPC synthesis method has defect that decline clearness of synthesis speech, during synthesizing unvoiced sound In this paper, we improve clearness of synthesized speech, using residual signal as ecitation signal of unvoiced sound. Besides, to improve a naturalness, we control the prosody of synthesized speech through controlling the energy and pitch pattern. Synthesis system is implemented at PC/486 and use a 70Hz-4.5KHz band pass filter for speech imput/output, amplifier and TMS320c30 DSP board.

• 대한전자공학회논문지
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• 제25권4호
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• pp.430-436
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• 1988

LSF(Line Spectral Frequency) derived from LPC has known as a very useful transmission parameter of speech signals, for it has a good linear interpolation characteristics and a low spectrum distortion at low bit rates coding. This paper presents that it is possible to extract directly the formant frequencies of speech signals from LSF parameter without application of FFT algorithm by comparing the distribution of LSF parameter with the frequency distribution of analysis filter. This paper suggests the advanced algorithm that results in improving the speed of convergence at analytic solution method. Also, for the flexibility of parameters, the process that transforms from LSF to LPC is presented.

• 한국음향학회지
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• 제1권1호
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• pp.45-53
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• 1982

A robust linear predictive coding method that can be used in noisy as well as quiet environment has been studied. In this method, noise autocorrelation coeffieients are first obtained and updated during nonspeech periods. Then, the effect of additive noise in the input speech is removed by subtracting values of the noise autocorrelation coefficients of corrupted speech in the course of computation of linear prediction coefficients. When signal-to-noise ratio of the input speech ranges from 0 to 10 dB, a performance improvement of about 5 dB can be gained by using this method. The proposed method is computationally very efficient and requires a small storage area.

• 대한의용생체공학회:학술대회논문집
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• 대한의용생체공학회 1990년도 추계학술대회
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• pp.96-98
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• 1990

• 대한전기학회:학술대회논문집
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• 대한전기학회 2004년도 학술대회 논문집 정보 및 제어부문
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• pp.310-312
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• 2004

The aim of this study was to analyze the characteristics of knee joint sound in frequency domain and classify the knee joint diseases. The spectral analysis of knee joint sounds was performed using LPC(Linear Predictive Coding) and Wigner-Ville distribution. Ten normal subjects and 5 patients with meniscal tearing were enrolled. Each subject was seated on a chair and underwent active knee flexion and extension for 60 seconds. Sampling frequency was 10kHz and electronic stethoscope and electro-goniometer were applied during the knee motion for data collection. The spectral analysis showed 3 peaks in both groups and the difference energy distribution in time-frequency domain. These results suggest that the diagnosis of knee joint pathology using the auscultation could be easier and more correct.

• 한국전자통신학회논문지
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• 제8권9호
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• pp.1293-1298
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• 2013

본 논문에서는 음성신호를 개선할 목적으로 잡음으로 오염된 음성신호로부터 잡음성분을 제거하기 위한 위너 필터를 사용한 잡음제거 알고리즘을 제안한다. 제안한 알고리즘은 먼저 잡음 복원 및 제거 방법에 기초하여 잡음으로 오염된 신호로부터 각 프레임에서 백색잡음의 잡음 스펙트럼을 제거한다. 또한 본 알고리즘은 선형예측 분석 방법에 기초한 위너 필터를 사용하여 음성신호를 강조한다. 본 실험에서는 일본 남성화자에 의한 음성과 잡음데이터를 사용하여 본 알고리즘의 실험 결과를 나타낸다. 백색잡음에 의하여 오염된 음성신호에 대하여 스펙트럼 왜곡률 척도를 사용하여 본 알고리즘이 유효하다는 것을 확인한다. 실험으로부터 백색잡음에 대하여 이전의 위너 필터와 비교하여 최대 4.94 dB의 출력 스펙트럼 왜곡률이 개선된 것을 확인할 수 있었다.