• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.05a
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• pp.67-70
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• 2015

This paper describes a one of HRD Program focus on ICT technology called TOPCIT as that companies and higher education providers voiced the need for a standardized, objective competency index that can reinforce the on-site competency of college students majoring in ICT/SW. And reduce the gap between the viewpoints of industrial and academic circles regarding the qualifications of a competent specialist in ICT field. For this reason, T.OPCIT developed and evaluated a performance-evaluation-centered test designed to diagnose and assess the competency of ICT specialists and SW developers critically needed to perform jobs on the professional frontier. This TOPCIT concept has been promoted not just in KOREA but in many ASEAN countries, e.g Thailand, Nepal, Bhutan, Philippines, Mongolia and Cambodia during the ICTD-USO Forum organised by ITU-ASP.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.35-47
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• 2001

This paper describes an efficient harmonic-CELP speech coder by taking advantages of harmonic and CELP coders into account. According to frame voicing decision, the proposed harmonic-CELP coder adopts the RP-VSELP coder as a fast CELP in case of an unvoiced frame, or an improved harmonic coder in case of a voiced frame. The proposed coder has main features as follows: simple pitch detection, fast harmonic estimation, variable dimension harmonic vector quantization, perceptual weighting reflecting frequency resolution, fast harmonic synthesis, naturalness control using band voicing, and multi-mode. These features make the proposed coder require very low complexity, compared with HVXC coder To demonstrate the performance of the proposed coder, a 2.4 kbps coder has been implemented and compared with reference coders. From results of informal listening tests, the proposed coder showed good quality while requiring low delay and complexity.

• Speech Sciences
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• v.13 no.2
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• pp.133-145
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• 2006

In order to understand the various groups' perceptual pattern in both VCV trochee and iambus, this study examined the identification correctness and cue robustness for the unit intervals in light of background language, age, and English listening ability. The 4 groups of Native Speakers of English, Korean College Students of High Listening Achievement, Korean College Students of Low Listening Achievement, and Korean Elementary Students took part in the experiments. Tokens of $/d{\ae}per,\;d{\ae}per,\;d{\ae}per,\;d{\ae}per,\;d{\ae}per,\;d{\ae}per$ in trochee and of $/{\eth}{\partial}\;p{\ae}d,\;{\eth}{\partial}\;b{\ae}d,\;{\eth}{\partial}\;t{\ae}d,\;{\eth}{\partial}\;d{\ae}d,\;{\eth}{\partial}\;k{\ae}d,\;{\eth}{\partial}\;g{\ae}d/$ in iambus were extracted and modified into experimental signals composed of two digits(voiced-1, voiceless-0) by following the temporal intervals, in which the signals consisted of preceding vowel, closure, VOT, and post-vowel. In the first experiment of identification correctness in VCV iambus environment, all groups showed almost 100% correctness rate, while in trochee environment all groups were different(native speaker 87%, college high 74%, college low 70%, elementary 65%). In the second experiment of cue robustness, all groups showed the similar perceptual pattern in both environments. There was the order of robustness cues in VCV trochee: pre-vowel ${\gg}$ closure ${\gg}$ VOT ${\gg}$ post-vowel, while the order in VCV iambus: VOT ${\gg}$ post-vowel ${\gg}$ closure ${\gg}$ pre-vowel. In some condition, however, we found moderately different perceptual pattern depending on language, age and listening level.

• Journal of Broadcast Engineering
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• v.18 no.6
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• pp.919-926
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• 2013

The acoustic signal from a distance sound source in an enclosed space often produces reverberant sound that varies depending on room impulse response. The estimation of the level of reverberation or the quality of the observed signal is important because it provides valuable information on the condition of system operating environment. It is also useful for designing a dereverberation system. This paper proposes a speech quality estimation method based on the harmonicity of received signal, a unique characteristic of voiced speech. At first, we show that the harmonic signal modeling to a reverberant signal is reasonable. Then, the ratio between the harmonically modeled signal and the estimated non-harmonic signal is used as a measure of standard room acoustical parameter, which is related to speech clarity. Experimental results show that the proposed method successfully estimates speech quality when the reverberation time varies from 0.2s to 1.0s. Finally, we confirm the superiority of the proposed method in both background noise and reverberant environments.

• The Korean Journal of Food And Nutrition
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• v.31 no.3
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• pp.355-366
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• 2018

The aim of this study is to investigate the awareness and satisfaction level of school meal services by elementary school students and their parents. Approximately 97.2% of student-subjects have agreed on the necessity of a free meal service for school lunch; 44.3% of student-subjects voiced the need to provide free meal services in order to eliminate discrimination of low-income students. Over one-third of student-subjects (36.7%) cited nutrition as the main benefit of providing a free meal service. The majority of parent-subjects (95.1%) have recognized the need for a free meal service in school; approximately 37.3% of parent-subjects responded to need the free meal service in order to eliminate the discrimination of impoverished students. Both student- and parent-subjects expressed a high level of satisfaction with the quality of ingredients and the type of soup/nutrition provided. Student-subjects insisted on better food hygiene and a new menu, but cited the noisy cafeteria as a problem associated with school meal services. In addition, approximately 56.5% of student-subjects responded to the need for nutritional education in school. Parent-subjects were primarily concerned with hygiene regarding the preparation of school meal services, noting the temperature of foods as the biggest problem in school meal services. The majority of parent-subjects (88.1%) responded to the need for the nutritional education in school. Results of this survey indicate that school meal services can be improved by increasing menu options and increasing food hygiene.

• The Journal of the Acoustical Society of Korea
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• v.21 no.7
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• pp.665-670
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• 2002

The most common form of pre-emphasis is y(n)=s(n)-As(n-1), where A typically lies between 0.9 and 1.0 in voiced signal. Also, this value reflects the degree of pre-emphasis and equals R(1)/R(0) in conventional method. This paper proposes a new flattening method to compensate the weaked high frequency components that occur by vocal cord characteristic. We used interval information of LSP to estimate formant frequency, After obtaining the value of slope and inverse slope using linear interpolation among formant frequency, flattening process is followed. Experimental results show that the proposed method flattened the weaked high frequency components effectively. That is, we could improve the flattening characteristics by using interval information of LSP as flattening factor at the process that compensates weaked high frequency components.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.7
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• pp.957-963
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• 2007

In this paper, Nonlinear Autoregressive (NAR) method based on Least Square-Support Vector Regression (LS-SVR) is introduced and tested for nonlinear sustained vowel modeling. In the database of total 43 sustained vowel of Benign Vocal Fold Lesions having aperiodic waveform, this nonlinear synthesizer near perfectly reproduced chaotic sustained vowels, and also conserved the naturalness of sound such as jitter, compared to Linear Predictive Coding does not keep these naturalness. However, the results of some phonation are quite different from the original sounds. These results are assumed that single-band model can not afford to control and decompose the high frequency components. Therefore multi-band model with wavelet filterbank is adopted for substituting single band model. As a results, multi-band model results in improved stability. Finally, nonlinear sustained vowel modeling using NAR based on LS-SVR can successfully reconstruct synthesized sounds nearly similar to original voiced sounds.

• ETRI Journal
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• v.38 no.6
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• pp.1064-1073
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• 2016

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.

• Koreanishche Zeitschrift fur Deutsche Sprachwissenschaft
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• v.5
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• pp.67-89
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• 2002

Die meisten $W\"{o}rter$ im Deutschen, die zur lexikalichen Hauptkategorie $geh\"{o}ren,\;erf\"{u}llen$ die prosodischen Bedingungen, class sie ein phonologisches Wort bilden und class ein phonologisches Wort zumindest aus zwei Moren besteht. In dieser Arbeit wird gezeigt, welche Konsequenzen diese Constraints $f\"{u}r$ die prosodische Gestalt der deutschen $W\"{o}rter$ haben. Eine davon bezieht sich auf das $Ph\"{a}nomen$, das in der Literatur als 'minimales Wort' bekannt ist. Die distributionellen $Beschr\"{a}nkungen$ eines ungespannten kurzen Vokals im Deutschen sind darauf $Zur\"{u}ckzuf\"{u}hren$, class ein prosodisches Wort mindestens zwei Moren enthalten muss. Die Forderung nach einem minimalen Wort wirft aber die Frage, warum ein Stamm wie feige eine zweisilbige Struktur CVCV mit einer finalen Schwasilbe aufweisen, ein Stamm wie reif dagegen eine einsilbige Struktur eve. Allein die Forderung nach einem zweimorigen prosodischen Wort wurde auch eine ungrammatische Form wie feig $erf\"{u}llen$. Bei Formen wie feige ist festzustellen, dass das Constraint IDENT-IO [voiced] wichtiger als das Constraint ist, das einen einsilbigen Stamm verlangt. Eine Analyse, in der die finale Schwa-Silbe in einem CVCV-Stamm als ein stammbildendes Element oder Pseudosuffix aufgefasst wird, kann diese Interaktion zwischen Constraint nicht erfassen. Im Vergleich dazu zeigen die zweisilben Flexionsformen, bei denen Schwa-Silben als ein echtes Suffix fungieren, dass das Constraint 'Realisiere Morphem' nur dann verletzt werden kann, wenn es zur $Erf\"{u}llung\;des\;h\"{o}her$ rangierten Constraints OCP(nucleus) dient. Dieses Constraint ist seinerseits nur dann verletzbar, wenn damit das $h\"{o}here$ Constraint Coda-Cond erfullt werden kann.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.303-304
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• 2007

In this paper, we implement an automatic distance speech recognition system for voiced-enabled services. We first construct a baseline automatic speech recognition (ASR) system, where acoustic models are trained from speech utterances spoken by using a cross-talking microphone. In order to improve the performance of the baseline ASR using distance speech, the acoustic models are adapted to adjust the spectral characteristics of speech according to different microphones and the environmental mismatches between cross-talking and distance speech. Next we develop a voice activity detection algorithm for distance speech. We compare the performance of the base-line system and the developed ASR system on a task of PBW (Phonetically Balanced Word) 452. As a result it is shown that the developed ASR system provides the average word error rate (WER) reduction of 30.6 % compared to the baseline ASR system.