• 음성과학
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• 제1권
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• pp.295-314
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• 1997

In this study, we will implement a flexible formant analysis and synthesis system. In the analysis part, the two-channel (i.e., speech & EGG signals) approach is investigated for accurate estimation of formant information. The EGG signal is used for extracting exact pitch information that is needed for the pitch synchronous LPC analysis and closed phase LPC analysis. In the synthesis part, Klatt formant synthesizer is modified so that the user can change synthesis parameters arbitarily. Experimental results demonstrate the superiority of the two-channel analysis method over the one-channel(speech signal only) method in analysis as well as in synthesis. The implemented system is expected to be very helpful for studing the effects of synthesis parameters on the quality of synthetic speech and for the development of Korean text-to-speech(TTS) system with the formant synthesis method.

• 음성과학
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• 제2권
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• pp.9-23
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• 1997

Through iterative analysis and synthesis experiments on Korean monosyllables, the Korean text-to-speech system was implemented using the phoneme-based formant synthesis method. Since the formants of initial and final consonants in this system showed many variations depending on the medial vowels, the database for each phoneme was made up of formants depending on the medial vowels as well as duration information of transition region. These techniques were needed to improve the intelligibility of synthetic speech. This paper investigates also methods of concatenating the synthesis units to improve the quality of synthetic speech.

• 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.

• 한국음향학회지
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• 제15권4호
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• pp.98-104
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• 1996

본 연구에서는 반음절단위 규칙합성에서 연속음성을 합성할 때 조음결합에 의한 천이구간이 없는 반음절의 연결로 접속되어 부자연스러운 합성음이 되는 것을 개선하기 위하여 연쇄모음의 천이구간을 보상하는 방법으로 포만트천이의 변경규칙을 제안하였다. 반음절 단위만으로는 포만트천이가 발생하는 부분을 채울 수 없기 때문에 반음절단위의 음성데이타와 모음의 반음절 단위의 정상부위로부터 세그멘트한 정상모음 42개를 추가하여 데이터베이스를 구축하였으며 포만트를 변경하는 방법으로 포만트합성에서의 공진회로를 이용하였다. 제안한 방법의 타당성을 확인하기 위하여 음성합성시 연쇄모음 부분에 포만트천이의 변경규칙을 적용하여 원음성 및 변경규칙을 적용하지 않은 반음절단위 음성합성방식에 의한 합성음성의 스펙트로그램과 비교하고 MOS 테스트를 실시한 결과 보다 자연스러운 합성음성을 얻을 수 있음을 확인하였다.

• 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.

• 한국컴퓨터정보학회:학술대회논문집
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• 한국컴퓨터정보학회 2014년도 제49차 동계학술대회논문집 22권1호
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• pp.5-8
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• 2014

This paper presents a parallel approach of formant synthesis method for haegeum on graphics processing units (GPU) using spectral modeling. Spectral modeling synthesis (SMS) is a technique that models time-varying spectra as a combination of sinusoids and a time-varying filtered noise component. A second-order digital resonator by the impulse-invariant transform (IIT) is applied to generate deterministic components and the results are band-pass filtered to adjust magnitude. The noise is calculated by first generating the sinusoids with formant synthesis, subtracting them from the original sound, and then removing some harmonics remained. The synthesized sounds are consequently by adding sinusoids, which are shown to be similar to the original Haegeum sounds. Furthermore, GPU accelerates the synthesis process enabling- real time music synthesis system development, supporting more sound effect, and multiple musical sound compositions.

• 음성과학
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• 제10권4호
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• pp.137-147
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• 2003

Speech synthesis can classify by synthesis way, that is waveform coding, source coding and mixture coding. Specially, waveform coding is suitable for high quality synthesis. However, it is not desirable by synthesis techniques of syllable or phoneme unit because it do not separate and handles excitation and formant part. Therefore, there is a need for pitch alteration method applied in synthesis by the rule in waveform coding. This study propose about pitch alteration method that use spectrum scaling after do to flatten spectra by subband linear approximation to minimize spectrum distortion. This paper show evaluation whether show excellency of some measure compared with LPC, Cepstrum, lifter function and method that propose. estimation method seeks distribution of each flattened signal and measured degree of flattened spectra Signal flattened is normalized, So that highest point amounts to zero, and distribution of signal ,whose average is zero, is calculated. this show result that measure the spectrum distortion rate to estimate performance of method that propose. The average spectrum distortion rate was kept below the average 2.12%, so the method that propose is superiors than existent method.

• 음성과학
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• 제8권1호
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• pp.7-19
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• 2001

The acoustic parameters of nine Korean numbers were analyzed by Praat, a speech analysis software, and synthesized by SenSynPPC, a Klatt formant synthesizer. The overall intensity, pitch and formant values of the numbers were modified dynamically by a step of 1 dB, 1 Hz and 2.5% respectively. The study explored the sensitivity of listeners to changes in the three acoustic parameters. Twelve subjects (male and female) listened to 390 pairs of synthesized numbers and judged whether the given pair sounded the same or different. Results showed that subjects perceived the same sound quality within the range of 6.6 dB of intensity variation, 10.5 Hz of pitch variation and 5.9% of the first three formant variations. The male and female groups showed almost the same perceptual ranges. Also, an asymmetrical structure of high and low boundary was observed. The ranges may be applicable to the development of a speaker identification system while the method of synthesis modification may apply to its evaluation data.

• 음성과학
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• 제13권1호
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• pp.7-27
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• 2006

This study explored the discrimination of synthesized English vowel pairs by twenty-seven American and Korean, male and female listeners. The average formant values of nine monophthongs produced by ten American English male speakers were employed to synthesize the vowels. Then, subjects were instructed explicitly to respond to AX discrimination tasks in which the standard vowel was followed by another one with the increment or decrement of the original formant values. The highest and lowest formant values of the same vowel quality were collected and compared to examine patterns of vowel discrimination. Results showed that the American and Korean groups discriminated the vowel pairs almost identically and their center formant frequency values of the high and low boundary fell almost exactly on those of the standards. In addition, the acceptable range of the same vowel quality was similar among the language and gender groups. The acceptable thresholds of each vowel formed oval to maintain perceptual contrast from adjacent vowels. The results suggested that nonnative speakers with high English proficiency could match native speakers' performance in discriminating vowel pairs with a shorter inter-stimulus interval. Pedagogical implications of those findings are discussed.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1998년도 학술발표대회 논문집 제17권 1호
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• pp.159-162
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• 1998

Speech synthesis coding is classified into three categories: waveform coding, source coding and hybrid coding. To obtain the synthetic speech with high quality, the synthesis by waveform coding is desired. However, it is difficult to apply waveform coding to synthesis by syllable or phoneme unit, because it does not divide the speech into excitation and formant component. Thus it is required to alter the excitation in waveform coding for applying waveform coding to synthesis by rule. In this paper we propose a new pitch alteration method that minimizes the spectrum distortion by using the behavior of cepstrum. This method splits the spectrum of speech signal into excitation spectrum and formant spectrum and transforms the excitation spectrum into cepstrum domain. The pitch of excitation cepstrum is altered by zero insertion or zero deletion and the pitch altered spectrum is reconstructed in spectrum domain. As a result of performance test, the average spectrum distortion was below 2.29%.