• Journal of IKEEE
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• v.19 no.3
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• pp.311-318
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• 2015

This dissertation proposes method for development technology of voice and SMS(Short Message Service) alarming system based on modbus protocol. The proposed technology is composed of the following 3 stages; hardware development based on microprocessor, development of input and output driver for modem, mp3 decoder, making modbus protocol stack. In the stage of hardware development based on microprocessor, we develop hardware which receives alarm from modbus master and transmit sms message, play mp3. In the stage of development of input / ouput device driver such as modem, mp3 decoder, we develop program which control each devices. In the stage of making modbus protocol stack, voice and sms alarm system is made for receiving alarm via modbus protocol. To evaluate performance of proposed technology, we issued alarm to voice and sms alarming system on purpose. As a result, response speed of detecting alarm was 10.7ms, communication distance was 1.2Km, operating temperature was from $-25^{\circ}C$ to $70^{\circ}C$, we confirmed supporting modbus protocol. And we verified that proposed voice and sms alarming system in the thesis has a performance to be used as an industrial building alarming system.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1
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• pp.93-101
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• 1996

In this paper, we implement the system for speech processing to provide the subscribers who are using the telephone network with better speech services. We develop the specified board which is processing speech signal and devise the system which carries out storing and replaying the speech signal under the condition that one master board controls multiple DSP(Digital Signal Processing) boards using VME bus. We use CPU30 board as a maste board and develop SPM(Signal Processing Module) board as a DSP board and then evaluate performance of the system.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.23 no.5
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• pp.963-968
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• 2013

In this paper, a major factor in determining voice quality that corresponds to the jitter and dejitter algorithm for removing jitter will be described. We analyze legacy dejitter algorithm and propose the study applying Time Series Model to improve performance of the dejitter algorithm.

• Smart Media Journal
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• v.9 no.3
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• pp.116-121
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• 2020

Digital Information Display and KIOSK have a problem that initial introduction cost and maintenance cost due to the development cost of dedicated contents and installation cost are high due to the characteristics of the product. In order to solve these problems, We designed and implemented of voice transmission system using Open Source Hardware and Event based Non-Blocking I/O Algorithm.

• The Journal of the Acoustical Society of Korea
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• v.18 no.4
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• pp.17-22
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• 1999

It is difficult to integrate the speech processing systems and machine translation system in the spoken language translation system by reason that each system uses its own data and basic processing unit. So, we need a common I/O unit which is used in the whole system. In this paper, we propose a Pscudo-Morpheme as the interface between speech processing systems and language translation system. We implement a morphological analysis system for Pseudo-morpheme. The speech processing system using this pseudo-morpheme can get better result than other systems using the phrase or the general morpheme. So, the quality of the whole spoken language translation system can be improved. The analysis-ratio of our implemented system is 98.9%. This is similar to the common morphological analysis systems.

• The Journal of the Acoustical Society of Korea
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• v.20 no.8
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• pp.19-23
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• 2001

DSP-based implementation is pervasive in wireless communication parts for systems and handsets according to developing high-speed and low-power programmable Digital Signal Processor (DSP). In this paper, we present a real-time implementation of multi-channel Adaptive Multi-rate (AMR) speech coder. The real-time implementation of an AMR algorithm is achieved using 32-bit fixed-point TMS320C6202 DSP chip that operates at 250 MHz. We performed cross compile, linear assembly optimization and TMS320C62xx assembly optimization for real-time implementation. Furthermore, speech data input/output function and communication function with external CPU is included in an AMR speech coder. The AMR Speech coder developed using DSP EVM board was evaluated in ETRI IMT-2000 Test-bed system.

• KIPS Transactions on Software and Data Engineering
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• v.2 no.1
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• pp.71-80
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• 2013

Due to the increased use of mobile devices, there is a lot of discussion on mobile accessibility. Mobile accessibility means that everyone, who includes the disabled, the elderly people, can easily use the functions of mobile devices. In this paper, we presented and implemented a mobile interface using a speech I/O APIs to improve the accessibility. The proposed interfaces are implemented on Android platforms and they used speech recognition and text-to-speech APIs supported as built-in services. In addition, to facilitate the internet access for visually impaired or blind people, we also implemented the web browsing application (web reader).

• Smart Media Journal
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• v.11 no.10
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• pp.22-29
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• 2022

Voice interactions are particularly effective in applications targeting the digital underprivileged who are not proficient in the use of smart devices. However, applications based on open APIs are using voice signals only for short, fragmentary input and output due to the limitations of existing touchscreen-oriented UI and API provided. In this paper, we design a conversational voice interaction model for interactions between users and intelligent mobile/IoT applications and propose a keyword detection algorithm based on the edit distance. The proposed model and scheme were implemented in an Android environment, and the edit distance-based keyword detection algorithm showed a higher recognition rate than the existing algorithm for keywords that were incorrectly recognized through speech recognition.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.97-100
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• 2004

유비쿼터스 컴퓨팅 환경에서의 차세대 PC는 다양한 입출력 장치를 이용하여 사용자에게 효과적으로 실제와 같은 정보를 제공하며, 사용자들의 편의를 고려해 웨어러블 형태의 플랫폼으로 발전하고 있다. 이러한 사용자 편의를 고려한 기술개발 동향(소형화, 경량화, 착용화)에 발맞추어 웨어러블 컴퓨팅 환경에서의 HCI 방안으로 음성 인식과 합성은 주요한 자리매김을 하고 있다. 본 논문에서는, 현재 정부에서 국가적인 차원으로 연구 개발 중인 차세대 PC 플랫폼 기반에서 음성합성 엔진을 구현하며, 구현상의 문제점 파악 및 개선사항에 대해 제안한다. 또한, 실질적인 구현 결과를 토대로 사용자 편의성 및 S/W 개발 환경을 고러한 차세대 PC플랫폼의 개선사항에 대해 제안을 한다.

• Electronics and Telecommunications Trends
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• v.5 no.2
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• pp.152-155
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• 1990

지능형 컴퓨터 개발 과제에서 사용자 인터페이스 역할을 담당할 멀티미디어 I/O 인터페이스에서 다루어야 할 멀티미디어의 종류와 이를 위한 디바이스에 대해 알아본다. 미디어의 종류로서 문장, 그래픽스, 음성, 영상 등의 미디어를 선정하고 이들과 관련된 처리 기술들을 살펴본다. 그리고 이들의 입출력 및 처리를 담당할 디바이스들의 종류를 살펴보고 이들이 갖추어야 할 최소한의 요건들을 제시한다.