• 말소리와 음성과학
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• 제7권1호
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• pp.97-105
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• 2015

This research mainly focuses on intonational characteristics of synchronous English speech. Intonation contours were extracted from 1,848 utterances produced in two different speaking modes (solo vs. synchronous) by 28 (12 women and 16 men) native speakers of English. Synchronous speech is found to be slower than solo speech. Women are found to speak slower than men. The effect size of speech rate caused by different speaking modes is greater than gender differences. However, there is no interaction between the two factors (speaking modes vs. gender differences) in terms of speech rate. Analysis of pitch point features has it that synchronous speech has smaller Pt (pitch point movement time), Pr (pitch point pitch range), Ps (pitch point slope) and Pd (pitch point distance) than solo speech. There is no interaction between the two factors (speaking modes vs. gender differences) in terms of pitch point features. Analysis of sentence level features reveals that synchronous speech has smaller Sr (sentence level pitch range), Ss (sentence slope), MaxNr (normalized maximum pitch) and MinNr (normalized minimum pitch) but greater Min (minimum pitch) and Sd (sentence duration) than solo speech. It is also shown that the higher the Mid (median pitch), the MaxNr and the MinNr in solo speaking mode, the more they are reduced in synchronous speaking mode. Max, Min and Mid show greater speaker discriminability than other features.

• 말소리와 음성과학
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• 제4권4호
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• pp.19-27
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• 2012

When two speakers read a text together, the produced speech has been shown to reduce a high degree of variability (e.g., pause duration and placement, and speech rate). This paper provides a quantitative analysis of speech rate variation exhibited in synchronous speech by examining the global and local patterns in two dialects of Mandarin Chinese (Taiwan and Shanghai). We analyzed the speech data in terms of mean speech rate and the reference of "Just Noticeable difference (JND)" within a subject and across subjects. Our findings show that speakers show lower and less variable speech rates when they read a text synchronously than when they read alone. This global pattern is observed consistently across speakers and dialects maintaining the unique local variation patterns of speech rate for each dialect. We conclude that paired speakers lower their speech rates and decrease the variability in order to ensure the synchrony of their speech.

• The Journal of the Acoustical Society of Korea
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• 제19권2E호
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• pp.8-13
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• 2000

In this paper, we present an adaptive formant tracking algorithm for speech using closed phase WRLS-VFF-VT method. The pitch synchronous closed phase methods is known to give more accurate estimates of the vocal tract parameters than the pitch asynchronous method. However the use of a pitch-synchronous closed phase analysis method has been limited due to difficulties associated with the task of accurately isolating the closed phase region in successive periods of speech. Therefore we have implemented the pitch synchronous closed phase WRLS-VFF-VT algorithm for speech analysis, especially for formant tracking. The proposed algorithm with the variable threshold(VT) can provide a superior performance in the boundary of phone and voiced/unvoiced sound. The proposed method is experimentally compared with the other method such as two channel CPC method by using synthetic waveform and real speech data. From the experimental results, we found that the block data processing techniques, such as the two-channel CPC, gave reasonable estimates of the formant/antiformant. However, the data windows used by these methods included the effects of the periodic excitation pulses, which affected the accuracy of the estimated formants. On the other hand the proposed WRLS-VFF-VT method, which eliminated the influence of the pulse excitation by using an input estimation as part of the algorithm, gave very accurate formant/bandwidth estimates and good spectral matching.

• 대한의용생체공학회:의공학회지
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• 제20권4호
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• pp.495-502
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• 1999

본 연구에서는 내이의 손상에 의한 감각성 난청환자들의 청력회복을 위한 청각보철장치내의 가장 중요한 부분인 어음발췌기의 음성신호처리 알고리즘 및 하드웨어를 개발하였다. 증폭, 저역통과 필터, AGC의 역할을 수행하는 외이 및 중이는 아날로그 시스템으로 모델링하였고, 시간 지연된 다중 필터 및 변환기의 역할을 수행하는 내이는 실시간 처리가 가능한 고속 DSP 회로로 구현되었다. 특히 내이의 기저막특성은 비선형 자중 필터뱅크로 모델링한후, 피치와 동기화된 다중 스펙트럼을 출력할 수 있는 (pitch-synchronous multi-spectra : PSMS) 전략을 이용함으로서 청각계의 tonotopy와 periodicity를 만족시킬 수 있었다. 또한 주요, 음성신호처리의 대부분이 S/W로 수행되므로 다양한 실험을 위한 시스템 수정이 용이하며, C 언어로 프로그램이 개발되었기 때문에 다른 프로세스를 사용하는 H/W에도 쉽게 이식될 수 있다는 장점을 가진다.

• 음성과학
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• 제10권1호
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• pp.85-94
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• 2003

In this paper an algorithm to reduce the size of Text-to-Speech database is proposed. The algorithm is based on the characteristics of Korean phonemic units. From the initial database, a reduced phoneme unit set is induced by articulatory similarity of concatenating phonemes. Speech data is read by one female announcer for 1000 phonetically balanced sentences. All the recorded speech is then segmented by phoneticians. Total size of the original speech data is about 640 MB including laryngograph signal. To synthesize wave, RE-PSOLA (Residual-Excited Pitch Synchronous Overlap and Add Method) was used. The voice quality of synthesized speech was compared with original speech in terms of spectrographic informations and objective tests. The quality of the synthesized speech is not much degraded when the size of synthesis DB was reduced from 320 MB to 82 MB.

• 음성과학
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• 제7권3호
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• pp.153-165
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• 2000

During the phonation of voiced sounds, instants exist where the glottis is opened or closed, due to the periodic vibration of the vocal cord. When closed, this is called the glottal closure instant(GCI) or epoch.. The correct detection of the GCI is one of the important problems in speech processing for pitch detection, pitch synchronous analysis, and so on. Recently, it has been shown that the local maxima points of the wavelet transformed speech signal correspond to the GCIs of speech signal. In this paper, we investigate the accuracy of Gels estimated from this wavelet transformed speech signal. For this purpose we compare them with the negative peak points of the differentiated EGG signal that represents the actual GCIs of speech signal.

• 한국음향학회지
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• 제23권1호
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• pp.61-74
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• 2004

본 논문에서는 채널 환경에 강인한 화자 인식 시스템을 위하여 문맥과 화자에 종속적인 켑스트럼 추출 방법과 추출된 켑스트럼에서 화자 정보의 손실을 최소화하는 채널 정규화 방법을 제안하였다. 제안된 추출 방법은 화자의 고유한 피치를 이용한 피치 동기 분석 방법에 기반을 두어 켑스트럼을 추출한다. 따라서 일명 피치 동기 켑스트럼 (PSC)은 유성음 구간에서 성도의 임펄스 응답을 보다 정확하게 표현할 수 있다. 또한 피치는 채널 환경에서 스펙트럼에 비해 강인하므로 피치 동기 켑스트럼은 채널에 의한 스펙트럼의 왜곡을 보상할 수 있다. 제안된 채널 정규화방법인 포먼트 평활화 피치 동기 켑스트랄 평균 차감법 (FBPSCMS)은 포먼트 평활화 켑스트랄 평균 차감법을 PSC에 적용하여 프레임 내 처리의 정확도를 개선시킨다. 제안된 방법들의 화자 인식 성능을 비교하기 위해 남자 112명과 여자 56명에 대해 WMIT과 전화선 환경의 NTIMIT을 이용한 화자 식별을 수행하였다. 실험 결과 피치 동기 LPCC는 기존 단구간 켑스트럼과 비교하여 에러 감소율을 최대 7.7%까지 향상시켰고, FBPSCMS는 극점 필터링 CMS에 비해 보다 안정되고 낮은 에러율을 나타내었다.

• 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.

• 전자공학회논문지C
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• 제35C권7호
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• pp.59-68
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• 1998

This paper is to develop an algorithm for the unlimited korean speech synthesis. We present the results controlled of prosodic factors with isolated words as aynthesis basis unit int he time domain. With a new pitch-synchronous and parametric speech synthesis mehtod in the time domain here we mainly present the results of controlled prosody factors such a spitch periods, energy envelops and durations and the evaluaton of synthetic speech qualities. In the case of synthesis, it is possible ot synthesize connected words by controlling of a continuous unified prosody that makes to improve the naturalities. In the results of experiment, it also has been to be improved uncontinuities of pitch and zeroing of energy in the junction parts of speech waveforms. Specially it has been to be possible to synthesize speeches with unlimitted durations and tones. So on it makes the noisiness and the clearness better by improving the degradation effects from the phase distortion due to the discontinuities in the waveform connection parts.

• 한국음향학회지
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• 제20권6호
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• pp.46-50
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• 2001

본 논문은 HMM (Hidden Markov Model) 음성 인식 시스템에 적용할 수 있는 새로운 인식 시간 알고리즘인 스코아 캐쉬기법을 제안한다. 다른 많은 기법들이 인식 시간을 줄이면서 계산량을 줄이기 위하여 어느 정도의 인식율 저하를 감수하는 반면에 제안하는 스코아 캐쉬기법은 인식율 저하를 전혀 일으키지 않으면서 인식 시간을 상당량 줄일 수 있는 기법이다. 단독어 인식 시스템에 적용 가능할 뿐 아니라 연속어 인식에도 적용이 가능하며, 기존에 이미 설계된 인식 시스템의 구조를 전혀 흩트리지 않고 간단히 하나의 함수만 대치함으로서 인식시간을 크게 감축할 수 있다 또한 기존의 계산량 감축 알고리즘과 함께 적용 가능하므로 추가의 계산량 감소를 얻을 수 있다. 스코아 캐쉬 기법을 적용한 결과 최대 54% 만큼 계산량을 줄일 수 있었다.