• Title/Summary/Keyword: Unit concatenation

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A Spectral Smoothing Algorithm for Unit Concatenating Speech Synthesis (코퍼스 기반 음성합성기를 위한 합성단위 경계 스펙트럼 평탄화 알고리즘)

  • Kim Sang-Jin;Jang Kyung Ae;Hahn Minsoo
    • MALSORI
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    • no.56
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    • pp.225-235
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    • 2005
  • Speech unit concatenation with a large database is presently the most popular method for speech synthesis. In this approach, the mismatches at the unit boundaries are unavoidable and become one of the reasons for quality degradation. This paper proposes an algorithm to reduce undesired discontinuities between the subsequent units. Optimal matching points are calculated in two steps. Firstly, the fullback-Leibler distance measurement is utilized for the spectral matching, then the unit sliding and the overlap windowing are used for the waveform matching. The proposed algorithm is implemented for the corpus-based unit concatenating Korean text-to-speech system that has an automatically labeled database. Experimental results show that our algorithm is fairly better than the raw concatenation or the overlap smoothing method.

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The Study on the Expential Smoothing Method of the Concatenation Parts in the Speech Waveform (음성 파형분절의 지수함수 스므딩 기법에 관한 연구)

  • 박찬수
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1991.06a
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    • pp.7-10
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    • 1991
  • In a text-to-speech system, sound units (phonemes, words, or phrases, etc.) can be concatenated together to produce required utterance. The quality of the resulting speech is dependent on factors including the phonological/prosodic contour, the quality of basic concatenation units, and how well the units join together. Thus although the quality of each basic sound unit is high, if occur the discontinuity in the concatenation part then the quality of synthesis speech is decrease. To solve this problem, a smoothing operation should be carried out in concatenation parts. But a major problem is that, as yet, no method of parameter smoothing is available for joining the segment together. Thus in this paper, we proposed a new aigorithm that smoothing the unnatural discountinuous parts which can be occured in speech waveform editing. This algorithm used the exponential smoothing method.

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The Smoothing Method of the Concatenation Parts in Speech Waveform by using the Forward/Backward LPC Technique (전, 후방향 LPC법에 의한 음성 파형분절의 연결부분 스므딩법)

  • 이미숙
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1991.06a
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    • pp.15-20
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    • 1991
  • In a text-to-speech system, sound units (e. q., phonemes, words, or phrases) can be concatenated together to produce required utterance. The quality of the resulting speech is dependent on factors including the phonological/prosodic contour, the quality of basic concatenation units, and how well the units join together. Thus although the quality of each basic sound unit is high, if occur the discontinuity in the concatenation part then the quality of synthesis speech is decrease. To solve this problem, a smoothing operation should be carried out in concatenation parts. But a major problem is that, as yet, no method of parameter smoothing is availalbe for joining the segment together.

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On the Control of Energy Flow between the Connection Parts of Syllables for the Korean Multi-Syllabic Speech Synthesis in the Time Domain Using Mono-syllables as a Synthesis Unit (단음절 합성단위음을 사용한 시간영역에서의 한국어 다음절어 규칙합성을 위한 음절간 접속구간에서의 에너지 흐름 제어에 관한 연구)

  • 강찬희;김윤석
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.24 no.9B
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    • pp.1767-1774
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    • 1999
  • This paper is to synthesize Korean multi-syllabic speeches in the time domain using mono-syllables as a synthesis unit. Specially it is to control the shape forms of speech energy flows between the connection parts of syllables in the case of concatenation mono-syllables. For this it is controlled with the prosody parameters1) extracted from speech waveforms in the time domains and presented the experimental results controlled the energy flows by using the induced concatenation rules from the korean syllable shapeforms in connetion parts of syllables. In the results of experiments, it is removed the incontinuities of energy follows in the connection parts produced by concatenating the mono-syllables in the time domain and also improved the qualities and naturalites of synthesized speeches.

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A Study on Realization of Continuous Speech Recognition System of Speaker Adaptation (화자적응화 연속음성 인식 시스템의 구현에 관한 연구)

  • 김상범;김수훈;허강인;고시영
    • The Journal of the Acoustical Society of Korea
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    • v.18 no.3
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    • pp.10-16
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    • 1999
  • In this paper, we have studied Continuous Speech Recognition System of Speaker Adaptation using MAPE (Maximum A Posteriori Probability Estimation) which can adapt any small amount of adaptation speech data. Speaker adaptation is performed by the method of MAPB after Concatenation training which is making sentence unit HMM linked by syllable unit HMM and Viterbi segmentation classifies speech data to be adaptation into segmentation of syllable unit data automatically without hand labelling. For car control speech the recognition rates of adaptation of HMM was 77.18% which is approximately 6% improvement over that of unadapted HMM.(in case of O(n)DP)

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A Study on the Speaker Adaptation of a Continuous Speech Recognition using HMM (HMM을 이용한 연속 음성 인식의 화자적응화에 관한 연구)

  • Kim, Sang-Bum;Lee, Young-Jae;Koh, Si-Young;Hur, Kang-In
    • The Journal of the Acoustical Society of Korea
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    • v.15 no.4
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    • pp.5-11
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    • 1996
  • In this study, the method of speaker adaptation for uttered sentence using syllable unit hmm is proposed. Segmentation of syllable unit for sentence is performed automatically by concatenation of syllable unit hmm and viterbi segmentation. Speaker adaptation is performed using MAPE(Maximum A Posteriori Probabillity Estimation) which can adapt any small amount of adaptation speech data and add one sequentially. For newspaper editorial continuous speech, the recognition rates of adaptation of HMM was 71.8% which is approximately 37% improvement over that of unadapted HMM

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Speech Synthesis Based on CVC Speech Segments Extracted from Continuous Speech (연속 음성으로부터 추출한 CVC 음성세그먼트 기반의 음성합성)

  • 김재홍;조관선;이철희
    • The Journal of the Acoustical Society of Korea
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    • v.18 no.7
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    • pp.10-16
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    • 1999
  • In this paper, we propose a concatenation-based speech synthesizer using CVC(consonant-vowel-consonant) speech segments extracted from an undesigned continuous speech corpus. Natural synthetic speech can be generated by a proper modelling of coarticulation effects between phonemes and the use of natural prosodic variations. In general, CVC synthesis unit shows smaller acoustic degradation of speech quality since concatenation points are located in the consonant region and it can properly model the coarticulation of vowels that are effected by surrounding consonants. In this paper, we analyze the characteristics and the number of required synthesis units of 4 types of speech synthesis methods that use CVC synthesis units. Furthermore, we compare the speech quality of the 4 types and propose a new synthesis method based on the most promising type in terms of speech quality and implementability. Then we implement the method using the speech corpus and synthesize various examples. The CVC speech segments that are not in the speech corpus are substituted by demonstrate speech segments. Experiments demonstrate that CVC speech segments extracted from about 100 Mbytes continuous speech corpus can produce high quality synthetic speech.

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A Study on the Voice Conversion with HMM-based Korean Speech Synthesis (HMM 기반의 한국어 음성합성에서 음색변환에 관한 연구)

  • Kim, Il-Hwan;Bae, Keun-Sung
    • MALSORI
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    • v.68
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    • pp.65-74
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    • 2008
  • A statistical parametric speech synthesis system based on the hidden Markov models (HMMs) has grown in popularity over the last few years, because it needs less memory and low computation complexity and is suitable for the embedded system in comparison with a corpus-based unit concatenation text-to-speech (TTS) system. It also has the advantage that voice characteristics of the synthetic speech can be modified easily by transforming HMM parameters appropriately. In this paper, we present experimental results of voice characteristics conversion using the HMM-based Korean speech synthesis system. The results have shown that conversion of voice characteristics could be achieved using a few sentences uttered by a target speaker. Synthetic speech generated from adapted models with only ten sentences was very close to that from the speaker dependent models trained using 646 sentences.

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Speech Unit Concatenation by Phase Succession in an ABS/OLA Sinusoidal Model (ABS/OLA Sinusoidal 모델에서 위상계승을 이용한 단위음성의 연결)

  • Bae Jae-Hyun;Byeon Heo-Jin;Oh Yung-Hwan
    • Proceedings of the Acoustical Society of Korea Conference
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    • spring
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    • pp.11-14
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    • 1999
  • 본 논문에서는 중첩가산 Sinusoidal 합성방식에서 매칭된 정현파별로 위상을 계승하는 단위음성 연결방법을 제안한다. 선행 단위음의 마지막 프레임, 후행 단위음의 첫 프레임, 후행 단위음의 나머지 프레임의 단계로 나누어 각 단계마다 제안한 방식으로 선행 프레임의 위상을 계승하였다. 실험결과 후행 단위음의 연결 위치를 이동하는 기존의 방식을 사용한 연결음에 비해 연결부분에서 음성파형의 급격한 변화가 줄었다.

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Control of Duration Model Parameters in HMM-based Korean Speech Synthesis (HMM 기반의 한국어 음성합성에서 지속시간 모델 파라미터 제어)

  • Kim, Il-Hwan;Bae, Keun-Sung
    • Speech Sciences
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    • v.15 no.4
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    • pp.97-105
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    • 2008
  • Nowadays an HMM-based text-to-speech system (HTS) has been very widely studied because it needs less memory and low computation complexity and is suitable for embedded systems in comparison with a corpus-based unit concatenation text-to-speech one. It also has the advantage that voice characteristics and the speaking rate of the synthetic speech can be converted easily by modifying HMM parameters appropriately. We implemented an HMM-based Korean text-to-speech system using a small size Korean speech DB and proposes a method to increase the naturalness of the synthetic speech by controlling duration model parameters in the HMM-based Korean text-to speech system. We performed a paired comparison test to verify that theses techniques are effective. The test result with the preference scores of 73.8% has shown the improvement of the naturalness of the synthetic speech through controlling the duration model parameters.

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