Although Standard Korean is based on Seoul Korean, the phonetic realization of syllable codas in Seoul Korean has not been satisfactorily investigated. This paper aims to study how Seoul speakers pronounce syllable codas in certain phonetic contexts and what pronunciation they prefer among variants. It is noted that the realization of a syllable coda is different word by word and generation by generation. It is also noted that the syllable coda of a word is realized differently depending on the following vowel. And we discussed how the Principles of Standard Korean Pronunciation should be revised in the future, based on the results of this study.

This study investigates spectral characteristics of frication noise in Korean sibilants in terms of center of gravity and skewness. Specifically, the present study seeks to observe the two parameters with emphasis on place of articulation in different vowel environments. This study also examines whether these parameters can discriminate phonation types. The results showed that the fricatives are palatalized in front of the front vowel /i/ and the affricates are articulated at the same place of articulation regardless of the following vowels. This study also suggests that the place of articulation of the fricatives followed by /i/ is the same as those of the Korean affricates. With regard to the phonation type, there was a significant difference in the center of gravity between lax and tense series for both fricatives and affricates.

This study compared pitch range, total duration of utterances, pause, boundary tones and mean duration of intonation final syllables of 4 sentences produced by 5 transgenders (male to female) to those of twenty normal adults. Voice analysis was performed by Praat (version 4.049). The results showed that there were differences in pitch range, total duration of utterances, pause, boundary tones and mean duration of intonation final syllables among the 3 groups(transgenders, normal males and normal females).

Most existing telephone speech transmitted in current public networks is band-limited to 0.3-3.4 kHz. Compared with wideband speech(0-8 kHz), the narrowband speech lacks low-band (0-0.3 kHz) and high-band(3.4-8 kHz) components of sound. As a result, the speech is characterized by the reduced intelligibility and a muffled quality, and degraded speaker identification. Bandwidth extension is a technique to provide wideband speech quality, which means reconstruction of low-band and high-band components without any additional transmitted information. Our new approach considers to exploit harmonic synthesis method for reconstruction of low-band speech over the CELP coded speech. A spectral distortion measurement and listening test are introduced to assess the proposed method, and the improvement of synthesized speech quality was verified.

In this paper, new voice message system supporting massive outbound call is proposed. Basic idea of the proposed system is to pre-process all the text-to-speech conversion process, mixing of text and attached music file and to store the results of pre-process in the cache server which is connected to the IVR. New voice message system is optimized for the voice message system supporting massive outbound call by distributing the load of the web server caused by server-side script implementation which is accessing database and generating dynamic Voice XML document over client module and server module of web server. The proposed voice message system was test-deployed in one domestic voice message application service provider and it is shown that proposed voice message system reduced the response latency problem of test-bed voice message system.

An automatic pronunciation correction system provides learners with correction guidelines for each mispronunciation. For this purpose we develope a speech recognizer which automatically classifies pronunciation errors when Koreans speak a foreign language. In order to develope the methods for automatic assessment of pronunciation quality, we propose a language model based score as a machine score in the speech recognizer. Experimental results show that the language model based score had higher correlation with human scores than that obtained using the conventional log-likelihood based score.

In this paper, we describe a cross-morpheme pronunciation variation model which is especially useful for constructing morpheme-based pronunciation lexicon to improve the performance of a Korean LVCSR. There are a lot of pronunciation variations occurring at morpheme boundaries in continuous speech. Since phonemic context together with morphological category and morpheme boundary information affect Korean pronunciation variations, we have distinguished phonological rules that can be applied to phonemes in within-morpheme and cross-morpheme. The results of 33K-morpheme Korean CSR experiments show that an absolute reduction of 1.45% in WER from the baseline performance of 18.42% WER was achieved by modeling proposed pronunciation variations with a possible multiple context-dependent pronunciation lexicon.

An automatic pronunciation correction system provides learners with correction guidelines for each mispronunciation. In this paper we propose an HMM based speech recognizer which automatically classifies pronunciation errors when Koreans speak Japanese. We also propose two pronunciation networks for automatic detection of mispronunciation. In this paper, we evaluated performances of the networks by computing the correlation between the human ratings and the machine scores obtained from the speech recognizer.

In this paper, we make a study of classification normal from abnormal - normal, asthma through analysis of thoracic sound to take use thoracic sound detection system. Thoracic sound detection system has a function to store thoracic sound and analyze the data. The wave shape of thoracic sound is similar to noise and is systematically generated by inhalation and exhalation breathing, therefore, in this paper, to classify asthma sound in thoracic sound, we could discriminate between normal and abnormal case using level crossing rate(LCR) and spectrogram energy rate.

This paper is aiming at implementation of real-time speaker verification system using DSP board. Dialog/4, which is based on microprocessor and DSP processor, is selected to easily control telephone signals and to process audio/voice signals. Speaker verification system performs signal processing and feature extraction after receiving voice and its ID. Then through computing the likelihood ratio of claimed speaker model to the background model, it makes real-time decision on acceptance or rejection. For the verification experiments, total 15 speaker models and 6 background models are adopted. The experimental results show that verification accuracy rates are 99.5% for using telephone speech-based speaker models.

The development of the durational component of prosody modelling was overviewed and discussed in text-to-speech conversion of spoken English and Korean, showing the strengths and weaknesses of each approach. The possibility of integrating linguistic feature effects into the duration modelling of TTS systems was also investigated. This paper claims that current approaches to language timing synthesis still require an understanding of how segmental duration is affected by context. Three modelling approaches were discussed: sequential rule systems, Classification and Regression Tree (CART) models and Sums-of-Products (SoP) models. The CART and SoP models show good performance results in predicting segment duration in English, while it is not the case in the SoP modelling of spoken Korean.