• Proceedings of the KSPS conference
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• 2006.11a
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• pp.71-74
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• 2006

Many works have been done in the field of retrieving audio segments that contain human speeches without captions. To retrieve newly coined words and proper nouns, subwords were commonly used as indexing units in conjunction with query or document expansion. Among them, document expansion with subwords has serious drawback of large computation overhead. Therefore, in this paper, we propose Expected Matching Score based document expansion that effectively reduces computational overhead without much loss in retrieval precisions. Experiments have shown 13.9 times of speed up at the loss of 0.2% in the retrieval precision.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1B
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• pp.35-41
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• 2010

In this paper, we propose a 3D view design scheme which arranges 2D information in a 3D virtual space with a flexible interface and voice information. The scheme allows the user interface of the 2D image in 3D virtual space anytime from any view point. Voice information can be easily attached. It is this simple and efficient image and voice information arrangement in 3D virtual space that improves information transfer.

• Speech Sciences
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• v.9 no.3
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• pp.47-59
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• 2002

A new brilliant service using human-voice and DTMF (Dual Tone Multi Frequency) technique is expected nowadays in order to obtain valuable information on the internet more easily. VoiceXML (Voice eXtensible Markup Language) is the right choice that makes the new service possible. In this paper, the design and implementation of IVR (Interactive Voice Response) server using VoiceXML is described, where it connects with internet and IVR server efficiently. IVR server using VoiceXML is composed of two groups: VoiceXML document handling and VoiceXML execution. Scenario part of IVR server corresponds to VoiceXML document, the execution is performed by VoiceXML execution.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.451-453
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• 2002

현재의 웹 환경은 HTML로 구성이 되어있고 이로인해 하이퍼링크를 따라가기 위해 마우스 클릭을 통해 작업하는 GUI환경이 주를 이룬다. 하지만 이러한 방법은 인간이 가장 손쉽게 사용하는 음성과 비교해 볼때 상당히 불편한 축에 속한다. 이를 해결하기 위해 현재 무르익은 음성인식 기술과 전화기를 통해 정보를 제공하고자 하는 XML의 파생인 VoiceXML을 이용하여 현재 HTML이 주류를 이루는 웹 환경을 VoiceXML을 이용한 음성인터페이스 환경을 마련하고자 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1447-1450
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• 2002

현재의 웹 환경은 HTML로 구성이 되어있고 이로인해 하이퍼링크를 따라가기 위해 마우스 클릭을 통해 작업하는 GUI환경이 주를 이룬다. 하지만 이러한 방법은 인간이 가장 손쉽게 사용하는 음성과 비교해 볼 때 상당히 불편한 축에 속한다. 이를 해결하기 위해 현재 무르익은 음성인식 기술과 전화기를 통해 정보를 제공하고자 하는 XML의 파생인 VoiceXML을 이용하여 현재 HTML이 주류를 이루는 웹 환경을 VoiceXML을 이용한 음성인터페이스 환경을 마련하고자 한다.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.51 no.10
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• pp.439-445
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• 2002

In this paper force feedback control system is investigated for improving the quality of the haptic feedback in virtual reality applications. We suggested the method of controlling the haptic device and modelling the virtual environment. Haptic device is composed of five bar link structure, voice coil motor, control board, and virtual environment modeling program. We applied voice coil motor in the actuating system for simple structure and easy control. Virtual environment modelling is constructed in PC, and the control signals of the actuators and the encoder data are transferred to the control system through USB. Experiment is performed to evaluate the characteristics of the haptic device.

• Journal of Software Assessment and Valuation
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• v.15 no.2
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• pp.73-83
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• 2019

Voice Recognition has recently been improved with AI(Artificial Intelligence) and has greatly improved the false recognition rate and provides an effective and efficient Human Machine Interface (HMI). This trend has also been applied in the defense industry, particularly in the aviation, F-35. However, for the quality evaluation of Voice Recognition, the defense industry, especially the aircraft, requires measurable quantitative models. In this paper, the quantitative evaluation model is proposed for applying Voice Recognition to aircraft. For the proposal, the evaluation items are identified from the Voice Recognition technology and ISO/IEC 25000(SQuaRE) quality attributes. Using these two perspectives, the quantitative evaluation model is proposed under aircraft operation condition and confirms the evaluation results.

• The Journal of Information Systems
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• v.31 no.2
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• pp.137-158
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• 2022

Purpose As artificial intelligence voice agents (AIVA) have been widely adopted in services, diverse forms of their voices, which are the main interface with users, have been experimented. The purpose of this study is to examine how users evaluate vocal characteristics (gender, voice pitch, and voice pace) of AIVA, depending on prior research on human voice attractiveness. Design/methodology/approach This study employed an experimental survey which 516 participated in. Each participant was randomly assigned into one of eight situations (e.g., male - higher pitch - faster pace) and listened a AIVA voice sample, which introduce weather information. Next, a participant answered three consequence factors (attractiveness, trust, and anthropomorphism). Findings The results reveal that female voices of AIVA were perceived as more attractive and trustworthy than male voices. As far as voice pitch goes, while lower-pitch voices were preferred in female voices, higher-pitch voices were preferred in male voices. Finally, faster voices of AIVA were more attractive than slower voices.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.11C
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• pp.1152-1157
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• 2002

In this paper, a detailed procedure of development for an AAL2 processor widely used in voice gateway application is introduced. The processor supports CPS and SSCS with voice service and framed mode data service. It provides 4 ATM virtual connections, which include 1020 AAL2 channels. The processor has one UTOPIA Level 1 interface for an ATM cell interface and 4 TDM ports for a voice channel interface. The TDM ports carry PCM/ADPCM voice streams. Most AAL2 processors are implemented as software, or hardware and software, so its latency is large. But this processor has very low latency as to CPS and SSCS because all of them are implemented in hardware. Also, it allows not only loopback and switching of CPS packets, but loopback and switching of TDM channels. The key feature is that the internal structure of the CPS and SSCS in this processor seems like as each software function, so they are called whenever they are required. In addition, they are reusable for another design and are scalable for more channels.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1988.10a
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• pp.12-16
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• 1988

본 논문에서는 안내원의 작업처리시간을 줄이기 위한 방법으로 DDX-1A를 이용한 자동호 분배 장치에 자도음성안내 기능을 구현하기 위한 T-level Prncessor 인 PCP (Protocol Convert Processor) VCP(Voice Contro Processor)와 B-level Processor Avru(voice Response)와 B-level Processor AVRU(Automatic Voice Response Unit)의 H/W 기능 및 상호 interface 에 관하여 고찰한다.