• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2415-2418
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• 2003

In this paper, we present a new filter to adjust formant information. Spectral envelope in speech analysis shows information about characteristics of speech and formant information determines speech timbre. So, if formant position is adjusted, we can verify adjusted speech timbre. A presented filter is to adjust this formant. This filter is composed of triangular filters. Using this filter we could locate the formant frequency at target position.

• 음성과학
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• 제10권2호
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• pp.85-95
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• 2003

In speech signal processing, it is very important to detect the pitch exactly in speech recognition, synthesis and analysis. If we exactly pitch detect in speech signal, in the analysis, we can use the pitch to obtain properly the vocal tract parameter. It can be used to easily change or to maintain the naturalness and intelligibility of quality in speech synthesis and to eliminate the personality for speaker-independence in speech recognition. In this paper, we proposed a new pitch detection algorithm. First, positive center clipping is process by using the incline of speech in order to emphasize pitch period with a glottal component of removed vocal tract characteristic in time domain. And rough formant envelope is computed through peak-fitting spectrum of original speech signal infrequence domain. Using the roughed formant envelope, obtain the smoothed formant envelope through calculate the linear interpolation. As well get the flattened harmonics waveform with the algebra difference between spectrum of original speech signal and smoothed formant envelope. Inverse fast fourier transform (IFFT) compute this flattened harmonics. After all, we obtain Residual signal which is removed vocal tract element. The performance was compared with LPC and Cepstrum, ACF. Owing 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.

• 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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• 제28권6호
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• pp.526-533
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• 2009

본 논문에서는 전통 현악기 해금의 스펙트럼 모델링을 위해 캡스트럼 포락선을 이용한 포만트 합성법을 제안한다. 스펙트럼 모델링은 입력 신호를 정현파 성분과 노이즈 성분의 합으로 해석하여 음을 합성하는 기술로 주기성이 있는 현악기나 관악기의 음 합성에 효과적이다. 캡스트럼 포락선의 포만트는 정현파 성분 합성을 위한 파라미터로 활용하였다. 정현파 성분을 합성하기 위해 기존의 가산합성 방식과는 달리 IIT (Impulse Invariant Transform)로 공명기를 설계하였으며 배음간 크기 보완을 위해 대역 통과 필터를 추가하였다. 원음과 합성된 정현파 성분의 차로 구해진 노이즈 성분에 포함된 일부 유효배음을 제거하면 완전한 노이즈 성분을 구할 수 있으며 선형 보간법 (linear interpolation)에 기초하여 그 주파수 특성을 파라미터화 하였다. 최종적으로 합성된 노이즈 성분과 정현파 성분을 더하여 해금 단위음을 합성하였고 합성음은 원음과 매우 유사하였다.

• 말소리와 음성과학
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• 제7권2호
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• pp.157-162
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• 2015

This study reports on the effective way of displaying dynamic formant information on F1-F2 space. Conventional ways of F1-F2 space (different name of vowel triangle or vowel rectangle) have been used for investigating vowel characteristics of a speaker or a language based on statistics of the F1 and F2 values, which were computed by spectral envelope search method. Those methods were dealing mainly with the static information of the formants, not the changes of the formant values (i.e. dynamic information). So a better way of investigating dynamic informations from the formant values of speech signal is suggested so that more convenient and detailed investigation of the dynamic changes can be achieved on F1-F2 space. Suggested method used visualization of static and dynamic information in overlapped way to be able to observe the change of the formant information easily. Finally some examples of the implemented display on some cases of the continuous vowels are shown to prove the usefulness of suggested method.

• 대한후두음성언어의학회지
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• 제15권2호
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• pp.98-111
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• 2004

Background and Objectives : The purpose of this study is to analyze the difference of Fundamental Frequency(Hz), Closed Quotient(Qx ; %), Intensity(dB), Vocal tract length and width(cm), formant frequency(Hz), level of formant frequency(dB) depending on the larynx position. Materials and Methods : One professional male singer(career : 28 years) produced sustained vowel /a/,/e/,/i/,/o/,/u/ in two larynx position (higher, lower) with Dr. Speech and video fluoroscopy was used to quantify the vocal tract morphology. Results : In lower larynx position, CQ is increased 9.8% and Intensity is increased about 10% and level of Formant Frequency is increased. And also Vocal tract length is longer 2.4cm, Vocal tract width(Anterior width : 0.4cm, lateral width : 0.2cm) is wider than in higher larynx position. Conclusions : Singer's formant has a prominent spectrum envelope peak near 2400-2600Hz by clustering of F3, F4 and F5 near 3400Hz in lower larynx position.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제2권2호
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• pp.122-126
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• 2002

In this study, we find features of 3 emotions (Happiness, Angry, Surprise) as the fundamental research of emotion recognition. Speech signal with emotion has several elements. That is, voice quality, pitch, formant, speech speed, etc. Until now, most researchers have used the change of pitch or Short-time average power envelope or Mel based speech power coefficients. Of course, pitch is very efficient and informative feature. Thus we used it in this study. As pitch is very sensitive to a delicate emotion, it changes easily whenever a man is at different emotional state. Therefore, we can find the pitch is changed steeply or changed with gentle slope or not changed. And, this paper extracts formant features from speech signal with emotion. Each vowels show that each formant has similar position without big difference. Based on this fact, in the pleasure case, we extract features of laughter. And, with that, we separate laughing for easy work. Also, we find those far the angry and surprise.

• 한국통신학회논문지
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• 제16권12호
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• pp.1324-1335
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• 1991

본 논문은 음절을 음소로 분리하고 각각의 특징을 추출하여 음소를 판별할 수 있는 패턴을 설정하는 실험에 관한 연구이다. 음소분리는 피치검출, 각 성문피크펄스의 폭, 포락실, 진폭의 바이어스, 발성시간 등의 파라메타를 적용한다. 최초의 피치는 성문펄스의 폭, 에너지, 정규화와 성문피크의 가변바이어스 등의 변화에 따라 모음 포라선의 봉우리 부분에서 검출, 이를 기준으로 전체의 피치구간을 추적한다. 모음은 포만트 패턴의 유동을 감소시키는 방법과 제2포만트만으로 모음의 판별이 가능함을 제시하며, 피치 고저에 무관한 압축파형을 추정한다. 자음은 포락실, 스펙트럼, 압추파형, 분석방법 등을 발음방법과 음소 상호 영향에서 패턴을 추출한다. 실험결과 모음음소 90%, 초성자음80%, 종성자음 60% 판별된다.

• 한국통신학회논문지
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• 제27권3A호
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• pp.196-204
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• 2002

STC(Sinusoidal Transform Coding) 방식은 주파수 영역에서 음성신호의 스펙트럼 피크치들을 정현파로 모델링하여 합성하는 음성부호화 방식을 말한다. 저전송률 STC 방식에서는 스펙트럼의 모든 피크를 이용하는 대신, 기본 주파수와 고조파에 해당하는 스펙트럼 포락선에서의 크기와 그때의 위상을 이용하여 음성을 합성한다. 본 논문에서는 정현파 모델에 기반한 2.4kbps 음성부호화 알고리즘을 제안한다. 피치정보는 모든 스펙트럼 피크를 사용한 합성음과 선택된 주파수와 고조파를 이용한 합성음과의 평균자승에러를 이용하여 추정하고, 위상정보는 여기신호 펄스의 시작시기를 나타내는 onset time과 성도 모델 전달함수의 위상을 이용하여 얻는다. 크기정보는 SEEVOC 알고리즘과 선형예측계수를 이용하여 추정한다. 실험결과, 합성음의 스펙트럼 특성은 원음성의 포만트 정보를 대부분 가지고 있으며, 위상정보도 원음성의 위상을 잘 따라감을 확인하였다. 합성음의 음질평가를 위해서 informal한 MOS(Mean Opinion Score) 테스트를 시행하였으며, 2.0kbps의 HVXC와 비교하여 대체적으로 MOS 3.1 이상의 음질을 얻을 수 있었다.

• 한국농업기계학회:학술대회논문집
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• 한국농업기계학회 2000년도 THE THIRD INTERNATIONAL CONFERENCE ON AGRICULTURAL MACHINERY ENGINEERING. V.II
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• pp.196-203
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• 2000

On the assumption that the voices of the cows are produced by the linear prediction filter, we characterized the cows' voices. The order of this filter is determined by examining the voices characteristics both in time and frequency domains. The proposed order of the linear prediction filter is 15 for modeling voice production of the cow. The combination of the two parameters of the fundamental frequency, the slope of the straight line regressed from the log-log spectra of the amplitude-envelope and the only one coefficient involved in the linear prediction filter can differentiate the two cows.