• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.937-942
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• 1994

Text-to-Speech(TTS) conversion system can convert any words or sentences into speech. To synthesize the speech like human beings do, careful prosody control including intonation, duration, accent, and pause is required. It helps listeners to understand the speech clearly and makes the speech sound more natural. In this paper, a prosody control scheme which makes use of the information of the function word is proposed. Among many factors of prosody, intonation, duration, and pause are closely related to syntactic structure, and their relations have been formalized and embodied in TTS. To evaluate the synthesized speech with the proposed prosody control, one of the subjective evaluation methods-MOS(Mean Opinion Score) method has been used. Synthesized speech has been tested on 10 listeners and each listener scored the speech between 1 and 5. Through the evaluation experiments, it is observed that the proposed prosody control helps TTS system synthesize the more natural speech.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.35C no.7
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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.

• Speech Sciences
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• v.2
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• pp.45-55
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• 1997

This paper describes the procedure of implementing an articulatory speech simulator, in order to model the human articulatory organs and to synthesize speech from this model after. Images required to construct the vocal tract model were obtained from MRI, they were then used to construct 2D and 3D vocal tract shapes. In this paper 3D vocal tract shapes were constructed by spatially concatenating and interpolating sectional MRI images. 2D vocal tract shapes were constructed and analyzed automatically into a digital filter model. Following this speech sounds corresponding to the model were then synthesized from the filter. All procedures in this study were using MATLAB.

• Speech Sciences
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• v.5 no.1
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• pp.35-47
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• 1999

This paper describes procedures to model and synthesize emotional speech in a multimedia environment. At first, procedures to model the visual representation of emotional speech are proposed. To display the sequences of the images in synchronized form with speech, MSF(Multimedia Speech File) format is proposed and the display software is implemented. Then the emotional speech sinal is collected and analysed to obtain the prosodic characteristics of the emotional speech in limited domain. Multi-emotional sentences are spoken by actors. From the emotional speech signals, prosodic structures are compared in terms of the pseudo-syntactic structure. Based on the analyzed result, neutral speech is transformed into a specific emotinal state by modifying the prosodic structures.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3C
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• pp.234-239
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• 2008

For the efficient coding and transmission, the speech signal can be classified into three distinctive classes: voiced, unvoiced, and transition classes. At low bit rate coding below 4 kbit/s, conventional sinusoidal transform coders synthesize speech of high quality for the purely voiced and unvoiced classes, whereas not for the transition class. The transition class including plosive sound and abrupt voiced-onset has the lack of periodicity, thus it is often classified and synthesized as the unvoiced class. In this paper, the efficient algorithm for the transition class detection is proposed, which demonstrates superior detection performance not only for clean speech but for noisy speech. For the detected transition frame, phase information is transmitted instead of magnitude information for speech synthesis. From the listening test, it was shown that the proposed algorithm produces better speech quality than the conventional one.

• Speech Sciences
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• v.10 no.1
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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.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 1993.06a
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• pp.1321-1325
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• 1993

This paper investigates the use of Fuzzy vector quantizer(FVQ) in speech synthesis. To compress speech data, we employ K-means algorithm to design codebook and then FVQ technique is used to analysize input speech vectors based on the codebook in an analysis part. In FVQ synthesis part, analysis data vectors generated in FVQ analysis is used to synthesize the speech. We have fined that synthesized speech quality depends on Fuzziness values in FVQ, and the optimum fuzziness values maximized synthesized speech SQNR are related with variance values of input speech vectors. This approach is tested on a sentence, and we compare synthesized speech by a convensional VQ with synthesized speech by a FVQ with optimum Fuzziness values.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.28B no.10
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• pp.755-760
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• 1991

In this paper, we propose a new formant locus overlapping method which can effectively enhance a naturalness of synthetic speech produced by ddemisyllable based Korean text-to-speech system. At first, Korean demisyllables are divided into several number of segments which have linear formant transition characteristics. Then, database, which is composed of start point and length of each formant segments, is provided. When we synthesize speech with these demisyllable database, we concatenate each formant locus by using a proposed overlapping method which can closely simulate haman articulation mechanism. We have implemented a Korean text-to-speech system by using this method and proved that the formant loci of synthetic speech are similar to those of the natural speech. Finally, we could illustrate that the resulting spectrograms of proposed method are more similar to natural speech than those of conventional method.

• Phonetics and Speech Sciences
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• v.10 no.4
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• pp.121-125
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• 2018

This paper describes procedures of changing and measuring voice quality in terms of jitter. Jitter synthesis method was applied to the TD-PSOLA analysis system of the Praat software. The jitter component is synthesized based on a Gaussian random noise model. The TD-PSOLA re-synthesize process is used to synthesize the modified voice with artificial jitter. Various vocal jitter parameters are used to measure the change in quality caused by artificial systematic jitter change. Synthetic vowels, natural vowels and short sentences are used to check the change in voice quality through the synthesizer model. The results shows that the suggested method is useful for voice quality control in a limited way and can be used to alter the jitter component of voice.

• Proceedings of the KSPS conference
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• 1996.10a
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• pp.435-447
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• 1996

In this paper, we propose the method that extracts the speech feature using the hearing model through signal processing techniques. The proposed method includes the following procedure ; normalization of the short-time speech block by its maximum value, multi-resolution analysis using the discrete wavelet transformation and re-synthesize using the discrete inverse wavelet transformation, differentiation after analysis and synthesis, full wave rectification and integration. In order to verify the performance of the proposed speech feature in the speech recognition task, korean digit recognition experiments were carried out using both the DTW and the VQ-HMM. The results showed that, in the case of using DTW, the recognition rates were 99.79% and 90.33% for speaker-dependent and speaker-independent task respectively and, in the case of using VQ-HMM, the rate were 96.5% and 81.5% respectively. And it indicates that the proposed speech feature has the potential for use as a simple and efficient feature for recognition task