• MALSORI
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• no.43
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• pp.45-55
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• 2002

High reliability of automatic speaker recognition regardless of voice conditions is necessary for forensic application. Audio recordings in real cases are not consistent in voice conditions, such as duration, time interval of recording, given text or conversational speech, transmission channel, etc. In this study the variation of verification error rate of ASR system with the voice conditions was investigated. As a result in order to decrease both false rejection rate and false acception rate, the various voices should be used for training and the duration of train voices should be longer than the test voices.

• Proceedings of the KSPS conference
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• 1996.02a
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• pp.125-135
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• 1996

본 논문에서는 자연스러운 음성을 합성하기 위하여, 한국어 음운지속시간의 변화에 있어서 문절과 구내의 음절수와 음절의 위치에 의한 영향과 인접하는 음운의 영향에 대하여 통계적으로 분석하였고, 분석된 시간 특징을 제어 요소로 하는 회귀트리를 생성하여 음운 지속시간을 모델 화하였다. 또한, 제안된 음운 지속시간 모델에 의해 예측실험을 행하여, 측정치와 예측치간의 다중 상관계수가 0.74정도이고, 각 음운의 예측오차의 75%이상이 25ms이내로 제안된 모델의 타당성이 입증되었다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.115-118
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• 1995

무제한 음성합성기를 구현하기 위하여 꼭 필요한 음성분석장치의 개발에 대하여 논하엿다. 이 분석장치는 신호처리 보드를 사용하여 PC에서 사용할 수 있도록 되어 있으며, 음성의 A/D, D/A 및 spectrogram display는 물론 pitch pulse 위치를 Glottal instint closure에 맞추어 삽입할 수 있어 linear prediction base의 무제한 합성기에서 필요한 음성 data base를 구축하기 용이하도록 개발하였다. 또한 음성인식을 위한 음성 DB나 현재 사용중인 ARS를 구축하고자 할 때에도 적은 노력과 시간이 소요되도록 하였다.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.663-667
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• 1999

In this paper, we review the conventional speaker verification algorithm and present the text-dependent speaker verification system for application over telephone lines and its result of experiments. We apply blind-segmentation algorithm which segments speech into sub-word unit without linguistic information to the speaker verification system for training speaker model effectively with limited enrollment data. And the World-mode] that is created from PBW DB for score normalization is used. The experiments are presented in implemented system using database, which were constructed to simulate field test, and are shown 3.3% EER.

• The Journal of the Acoustical Society of Korea
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• v.19 no.6
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• pp.41-46
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• 2000

In this paper, we implement the speech recognition system based on the continuous distribution HMM and Browser-embedded model using the Java. That is developed for the speech analysis, processing and recognition on the Web. Client sends server through the socket to the speech informations that extracting of end-point detection, MFCC, energy and delta coefficients using the Java Applet. The sewer consists of the HMM recognizer and trained DB which recognizes the speech and display the recognized text back to the client. Because of speech recognition system using the java is high error rate, the platform is independent of system on the network. But the meaning of implemented system is merged into multi-media parts and shows new information and communication service possibility in the future.

• MALSORI
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• no.35_36
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• pp.189-209
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• 1998

This study discusses Korean speech database that has been designed and constructed for common use, especially focusing on designing a list of words or sentences that covers various phonological environments. As the results, PBW(Phonetically Balanced words) and PBS(Phonetically Balanced Sentences) was selected from balanced text corpus using maximum entropy method. And, implemented workbench for spoken language data acquisition is presented in this paper. The workbench consists of grapheme to phoneme converter, utterance list selection module, speech data editing module, multi-layer labelling module, and phoneme context search module.

• Proceedings of the KSPS conference
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• 2005.11a
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• pp.159-162
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• 2005

Identification is the process automatically identify who is speaking on the basis of information obtained from speech waves. In training phase, each speaker models are trained using each speaker's speech data. GMMs (Gaussian Mixture Models), which have been successfully applied to speaker modeling in text-independent speaker identification, are not efficient in insufficient training data environment. This paper proposes speaker modeling method using MLLR (Maximum Likelihood Linear Regression) method which is used for speaker adaptation in speech recognition. We make SD-like model using MLLR adaptation method instead of speaker dependent model (SD). Proposed system outperforms the GMMs in small training data environment.

• Industrial Engineering and Management Systems
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• v.13 no.3
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• pp.267-277
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• 2014

In this paper, we propose an educational system that involves a business game simulator and related curriculum. To develop these two elements, we examined the decision-making process related to business management and identified some significant skills thereby. In addition, we created an original simulator, named BizLator (http://bizlator.com), to help students develop these skills efficiently. Next, we developed a curriculum suitable for the simulator. We confirmed the effectiveness of the simulator and curriculum in a business-game-based class at Aoyama Gakuin University in Tokyo. On the basis of this, we compared our education system with a conventional system. This allowed us to identify advantages of and issues with our proposed system. Furthermore, we proposed a speech recognition support system named BizVoice in order to provide the teachers with more meaningful feedback, such as level of students' understanding. Concretely, BizVocie fetches students' speech of discussion during the game and converts the voice data to text data with speech recognition technology. Finally, teachers can grasp students' parameters of understanding, and thereby, the students also can take more effective class using BizLator. We also confirmed the effectiveness of the system in the class of Aoyama Gakuin Universiry.

• Journal of Satellite, Information and Communications
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• v.12 no.1
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• pp.101-106
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• 2017

In this paper, we present Emergency exit signs are installed to provide escape routes or ways in buildings like shopping malls, hospitals, industry, and government complex, etc. and various other places for safety purpose to aid people to escape easily during emergency situations. In case of an emergency situation like smoke, fire, bad lightings and crowded stamped condition at emergency situations, it's difficult for people to recognize the emergency exit signs and emergency doors to exit from the emergency building areas. This paper propose an automatic emergency exit sing recognition to find exit direction using a smart device. The proposed approach aims to develop an computer vision based smart phone application to detect emergency exit signs using the smart device camera and guide the direction to escape in the visible and audible output format. In this research, a CAMShift object tracking approach is used to detect the emergency exit sign and the direction information extracted using template matching method. The direction information of the exit sign is stored in a text format and then using text-to-speech the text synthesized to audible acoustic signal. The synthesized acoustic signal render on smart device speaker as an escape guide information to the user. This research result is analyzed and concluded from the views of visual elements selecting, EXIT appearance design and EXIT's placement in the building, which is very valuable and can be commonly referred in wayfinder system.

• Korean Journal of Otorhinolaryngology-Head and Neck Surgery
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• v.55 no.8
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• pp.498-507
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• 2012

Background and Objectives We aimed to develop a Korean version of the velopharyngeal insufficiency (VPI) speech corpus system. Subjects and Method After developing a 3-channel simultaneous speech recording device capable of recording nasal/oral and normal compound speech separately, voice data were collected from VPI patients aged more than 10 years with/without the history of operation or prior speech therapy. This was compared to a control group for which VPI was simulated by using a french-3 nelaton tube inserted via both nostril through nasopharynx and pulling the soft palate anteriorly in varying degrees. The study consisted of three transcriptors: a speech therapist transcribed the voice file into text, a second transcriptor graded speech intelligibility and severity and the third tagged the types and onset times of misarticulation. The database were composed of three main tables regarding (1) speaker's demographics, (2) condition of the recording system and (3) transcripts. All of these were interfaced with the Praat voice analysis program, which enables the user to extract exact transcribed phrases for analysis. Results In the simulated VPI group, the higher the severity of VPI, the higher the nasalance score was obtained. In addition, we could verify the vocal energy that characterizes hypernasality and compensation in nasal/oral and compound sounds spoken by VPI patients as opposed to that characgerizes the normal control group. Conclusion With the Korean version of VPI speech corpus system, patients' common difficulties and speech tendencies in articulation can be objectively evaluated. Comparing these data with those of the normal voice, mispronunciation and dysarticulation of patients with VPI can be corrected.