• 한국융합학회논문지
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• 제10권9호
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• pp.193-198
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• 2019

본 연구는 근위축성측삭경화증(amyotrophic lateral sclerosis, ALS)으로 진단된 여성을 대상으로 음향음성학적 스펙트로그램 분석을 이용하여 11개월 동안 모음과 이중모음의 포먼트 변화(vowel formant variation)를 분석하였다. 검사어는 단모음 /a, i, u/와 이중모음 /h + ja + da/, /h + wi + da/, /h +ɰi+ da/를 이용하였다. 발화자료는 'Alvin' 프로그램을 이용하여 모니터에 제시된 단어읽기과제를 통해 수집되었고, 녹음환경은 nyquist frequency는 5,500Hz, sampling rate는 11,000Hz으로 설정하였다. 녹음자료는 스펙트로그램을 이용하여 강도, 음도와 이중모음의 포먼트를 분석하였다. 분석결과, ALS의 진행과정에서 기본주파수와 강도가 저하되었고, 단모음에서의 포먼트 변화보다는 이중모음의 포먼트 기울기의 감소가 특징으로 확인되었다. 이 결과는 병의 진행에 따른 ALS의 모음왜곡이 혀와 턱의 협응력 감소에 기인함을 시사한다.

• 한국지능시스템학회:학술대회논문집
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• 한국퍼지및지능시스템학회 2002년도 추계학술대회 및 정기총회
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• pp.341-344
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• 2002

음성은 화자들의 특징을 가지고 있다. 이 논문에서는 신경망에 기초한 Incremental Learning을 이용하여 화자인식시스템을 소개한다. 컴퓨터를 통하여 녹음된 문장들은 FFT를 거치면서 Frequency 영역으로 바뀌고, 모음들의 특징을 가지고 있는 Formant를 이용하여 모음들을 추출한다. 추출된 모음들은 LPC처리를 통하여 화자의 특성을 가지고 있는 Coefficient값들을 얻는다. LPCC과정과 Vector Quantization을 통해 10개의 특징 점들은 학습을 위한 Input으로 들어가고 화자 수에 따라 증가되는 Hidden Layer와 Output Layer들을 가지고 있는 신경망을 통해 화자인식을 수행한다.

• 대한음성학회지:말소리
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• 제45호
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• pp.63-77
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• 2003

The purpose of voice color conversion is to change the speaker identity perceived from the speech signal. In this paper, we propose a new voice color conversion algorithm through the formant shifting and the spectrum-tilt modification in the frequency domain. The basic idea of this technique is to convert the positions of source formants into those of target speaker's formants through interpolation and decimation and to modify the spectrum-tilt by utilizing the information of both speakers' spectrum envelops. The LPC spectrum is adopted to evaluate the position of formant and the information of spectrum-tilt. Our algorithm enables us to convert the speaker identity rather successfully while maintaining good speech quality, since it modifies speech waveforms directly in the frequency domain.

• 음성과학
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• 제3권
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• pp.83-92
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• 1998

Vowels are classified by vocal tract shapes. These shapes form constriction points along the tract, which have an influence on such vocal tract resonance as $F_l,\;F_2,\;F_3$, and so on. This study reviews the perturbation theory of the tract and determines the corresponding formant frequencies from modified vocal tracts using vocal tract area function. Then, formant variation is observed from the theory. Finally, each set of $F_l,\;F_2,\;and\;F_3$ frequency is input to a speech synthesis software to make a vowel sound. Auditory impression of each sound without any modification of its vocal tract shape is almost the same as the corresponding phonetic symbol. Formant frequencies of $F_l,\;F_2,\;F_3$ vary according to the perturbation theory. Generally, constriction along the node causes formant values to decrease while constriction along the anti-node cause it to increase. Vocal tracts modified by more than $3\;cm^2$ change vowel qualities of /a/ and /i/ into those of f /v/ and /$\varepsilon$/, respectively. This study will be helpful in simulating sounds from modified vocal tracts before any operation. Further studies are desirable to compare vocal tract shapes of various languages and their sounds together.

• 대한음성학회지:말소리
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• 제58호
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• pp.19-34
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• 2006

The present study investigates the vowels of Bahasa Malaysia and Bahasa Indonesia in terms of the first two formant frequencies and provides a three-dimensional formant chart of vowels by plotting F1, F2, and the frequency of datapoints on 4 different scales: Hz, mel, bark, and the number of ERB. For this study, we recruited 30 male native speakers of Bahasa Malaysia and Bahasa Indonesia (15 each) which include 6 vowels (i, e, a, o, u, e) in various contexts. The three-dimensional formant chart showed geophysics of vowel space, such that mountain peaks stand in particular locations with a higher frequency of occurrence of datapoints. The geophysics of vowel space may shed lights on the perceptual structure of vowel space. The results also showed that vowels in utterance-final positions have a significantly higher F1 and a significantly lower F2 than those in utterance-medial or utterance-initial positions, which means that vowels in utterance-final positions are more back and lower in vowel space than those in utterance-medial or utterance-initial positions.

• 대한후두음성언어의학회지
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• 제10권2호
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• pp.119-123
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• 1999

Background and Objectives : Conditions such as muscle atrophy, stretching of strap muscles, and continued craniofacial growth factors have been cited as contributing to the changes observed in the vocal tract structure and function in elderly speakers. The purpose of the present study is to compare F$_1$ and F$_2$ frequency levels in elderly and young adult male and female speakers producing a series of vowels ranging from high-front to low-back placement. Material and Methods : The subjects were two groups of young adults(10 males, 10 females, mean age 21 years old range 19-24 years) and two groups of elderly speakers(10 males, 10 females, mean age 67 years : range 60-84 years). Each subject participated in speech pathologist to be a speaker of unimpared standard Korean. The headphone was positioned 2 cm from the speakers lips. Each speaker sustained the five vowels for 5 s. Formant frequency measures were obtained from an analysis of linear predictive coding in CSL model 4300B(Kay co). Results : Repeated measure AVOVA procedures were completed on the $F_1$ and $F_2$ data for the male and female speakers. $F_2$ formant frequency levels were proven to be significantly lower fir elderly speakers. Conclusions : We presume $F_2$ vocal cavity(from the point of tongue constriction to lip) lengthening in elderly speakers. The research designed to observe dynamic speech production more directly will be needed.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(4)
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• pp.381-384
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• 2002

In speech signal processing, it Is very important to detect the pitch exactly in speech recognition, synthesis and analysis. but, it is very difficult to pitch detection from speech signal because of formant and transition amplitude affect. therefore, in this paper, we proposed a pitch detection using the spectrum flattening techniques. Spectrum flattening is to eliminate the formant and transition amplitude affect. In time domain, positive center clipping is process in order to emphasize pitch period with a glottal component of removed vocal tract characteristic. And rough formant envelope is computed through peak-fitting spectrum of original speech signal in frequency domain. As a results, well get the flattened harmonics waveform with the algebra difference between spectrum of original speech signal and smoothed formant envelope. After all, we obtain residual signal which is removed vocal tract element The performance was compared with LPC and Cepstrum, ACF 0wing to this algorithm, we have obtained the pitch information improved the accuracy of pitch detection and gross error rate is reduced in voice speech region and in transition region of changing the phoneme.

• 한국음향학회지
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• 제10권4호
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• pp.12-18
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• 1991

음성 신호처리 분야에서 기본주파수를 정확히 검출하는 것이 아주 중요하다. 주파수 영역에서 피치검출 방법의 문제점은 대체로 배경잡음이나 제 1 포먼트에 의하여 발생한다. 그러므로, 본 논문에서는 스펙트럼 AMDF 함수를 이용하여 잡음의 영향이나 제 1 포먼트의 영향을 줄이는 주파수영역 피치검출 앨고리즘을 제안하였다. 여러 가지 컴퓨터 시뮬레이션 결과 제안한 앨고리즘이 기본주파수 검출에 효과적으로 나타났다.

• 대한음성학회지:말소리
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• 제67호
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• pp.33-60
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• 2008

The aim of this study is to show the differences in acoustic parameters between a pathological voice /a/ caused by vocal polyp and a normal voice /a/ produced after LMS (Laryngeal Microscopic Surgery). It was expected that voices of two kinds could be analyzed effectively in terms of HNR in specific frequency bands than in all frequency bands. For this study, 10 patients' voice were recorded before and after LMS and then were manipulated in terms of four acoustic parameter. It was found out that (a) frequency bands of 500Hz in the range of 1,000Hz to 4,000Hz were very useful to obtain HNR values; (b) frequency bands in the range of 1,248Hz to 5,500Hz on a log scale were very useful to obtain HNR values; (c) F0 dropped after LMS but not significantly; (d) the bandwidth of the second formant (B2) decreased significantly after LMS, while that of the first formant (B1) decreased after LMS but not significantly.

• 음성과학
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• 제12권4호
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• pp.43-52
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• 2005

The common form of pre-emphasis filter is $H(z)\;=\;1\;- az^{-1}$, where a typically lies between 0.9 and 1.0 in voiced signal. Also, this value reflects the degree of filter and equals R(1)/R(0) in Auto-correlation method. This paper proposes a new flattening algorithm to compensate the weaked high frequency components that occur by vocal cord characteristic. We used interval information of LSP to estimate formant frequency. After obtaining the value of slope and inverse slope using linear interpolation among formant frequency, flattening process is followed. Experimental results show that the proposed algorithm flattened the weaked high frequency components effectively. That is, we could improve the flattened characteristics by using interval information of LSP as flattening factor at the process that compensates weaked high frequency components.