• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.61-64
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• 2004

최근 PDA(Personal Digital Assistants)와 같은 휴대형 단말기들은 다양한 멀티미디어 기술과 무선 인터넷 기술의 영향으로 정보단말기로서 각광을 받고 있다. 그러나 현재의 단말기는 프로세서와 메모리의 한계로 인하여 원활한 음성인식 시스템을 구축하기에는 한계가 있다. 이를 보완하는 방법으로 본 논문에서는 Client/server로 분리된 음성 인식 시스템을 구축하였다. 구축한 시스템은 무선 네트워크 환경을 이용하여 PDA(Personal Digital Assistants)에서 음성 파일 또는 특징 파라미터를 Serve 측으로 전송하여 Server측에서 음성 인식을 수행한 후 그 결과를 모바일 단말기로 되돌려 주는 시스템이다. 구성된 시스템을 평가하기 위해서는 국어 공학센터의 음성 DB(KLE 452DB)를 이용하여 음향 모델을 생성한 후 다양한 환경(연구실, 복도, 주차장 도서관 로비)에서 발성한 후 이를 교내 무선 인터넷망(Nespot)을 통하여 송신하여 실시간 인식하였다. 실험 결과, 각각 $84.04\%\;72.28\%\;69.47\%\;67.61\%$의 평균 인식률을 얻을 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.21 no.7
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• pp.606-613
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• 2002

In this paper we applied a transmit beamforming technique to the underwater acoustic communication system for high rate data transmission. A prototype transmit system was designed and implemented with the general purpose DSP processor and multiple digital-to-analog converters. The performances of the implemented system were evaluated by the experiment in water tank. In order to simplify the procedure the channel coding and equalizer were omitted. And the simplest OOK (On-Off Keying) technique in digital communication methods was applied. The experimental result shows that the transmission data rate is higher about 3 times in the case of 5 hydrophone transmitting may than 1 hydrophone transmitter at bit error rate 10/sup -2/. We verified that the maximum data rate was 400 bps for speech signal transmission in water tank.

• Journal of Digital Contents Society
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• v.8 no.4
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• pp.467-474
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• 2007

This paper documents the design and implementation of an automatic creating web-site tool for the blind to build their own homepage by using both voice recognition and voice mixed technology with equal ease as the non-disabled. The blind can make voice mails, schedules, address lists and bookmarks by making use of the tool. It also facilitates communication between the non-disabled with the help of their information management system. This tool converts basic commands into voice recognition, also making an offer of text-to-speech which supports voice output. In the end, the tool will remove the blind's social isolation, allowing them to enjoy the information age like the non-disabled.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.63-66
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• 2000

고품질의 디지털 이동 통신 서비스를 제공하기 위해서는 다양한 통신 상황 및 통신 경로에 대한 음질의 검증이 필요하다. 그러나 다양한 환경에서의 음질 측정을 위하여 실제 통신 시스템 환경을 구축하는 것은 시간과 비용이 많이 드는 문제점을 가진다. 이러한 문제점들을 해결하기 위하여 본 논문에서는 디지털 이동 통신시스템의 End-to-End 음성 전달 과정 전체를 Software로 Simulation하는 Simulator를 개발하였으며. 이를 통하여 통신 환경과 경로에 따른 통신 시스템의 통화 품질을 매우 간단하게 측정할 수 있다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1992.06a
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• pp.155-159
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• 1992

본 논문은 디지틀 음성통신 서비스의 품질기준 (통화품질, 접속품질)을 설정하기 위한 실험용 통화모델 시스템의 사용자 접속 규격을 구현한 것이다. 본 논문에서 구현한 사용자 접속 규격은 디지틀 음성통신망에서 사용하려는 접속 protocol을 그대로 적용하여 실제의 상황과 같도록 하기 위하여 종합정보통신망 (ISDN)의 표준규격인 사용자 - 망 접속 규격을 이용하여 구현하였다. CCITT 권고안 I series의 규격을 만족하는 Protocol Simulator 와의 연동실험을 통하여 구현한 모델 시스템에 대한 사용자 접속 규격의 처리 및 성능을 확인하였다.

• Speech Sciences
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• v.11 no.3
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• pp.175-180
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• 2004

In case of the hearing impairment with tinnitus and increased sound sensitivity, it is known that the patients tend to appeal the psychologically oriented social handicap rather than communication disability. The audiologist who is responsible for such patients in aural rehabilitation should pay special attention to the counseling techniques including tinnitus retain therapy (TRT), ear protector, noise generator, or specific acoustic training based on close cooperation and rapport. And then the audiologist should try to lessen their reaction to the tinnitus by using a hearing aid. This therapies tries to focus not a. total approach but a treatment to lessen the severity of tinnitus. This paper as a case report that a unilateral sharply slopped sensorineural hearing impaired person with tinnitus and increased sound sensitivity by using four channel digital signal processing (DSP) hearing aid with programming increment at low level (PILL).

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.2 no.2
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• pp.37-43
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• 2009

In this study, multi-channel digital hearing aid algorithm for low power system is proposed. First, MDCT(Modified Discrete Cosine Transform) method converts time domain of input speech signal into frequency domain of it. Output signal from MDCT makes a group about each channel, and then each channel signal adjusts a gain using LCF(Loudness Compensation Function) table depending on hearing loss of an auditory person. Finally, compensation signal is composed by TDAC and IMDCT. Its all of process make progress 16-bit fixed-point operation. We use fast-MDCT instead of MDCT for reducing system complexity and previously computed tables instead of log computation for estimating a gain. This algorithm evaluate through computer simulation.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.925-928
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• 2008

With basis of the development of information communication techniques in recent year, the digital processing techniquy also has been growed fast. The digital processing technique have used signals - speech and image processing- for processing of transmission and analysis. After we get and save the signals. Effective signal processing techniques have varied filters and typical digital filters are FIR filter and IIR filter. The FIR digital filter is more secure because phase response characteristics have linear phase. But, FIR digital filters have a problem to product the Gibbs phenomenon generating around a discontinuous point. A propose of filer is to remove the problem. Therefore, in this paper I was proposed a method using FIR digital filter applied a modified window function and the method was compared with conventional methods.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.6
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• pp.990-998
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• 2008

In this paper we achieve experimental data evaluation using SSB(Single-side band) modulation in the ocean. Present research in underwater communication is applying digital modulation, OFDM and MIMO system. However, Commercial modems using analog modulation techniques in oceans. So, we achieved experimental for modem appliance development of correct high quality in South Korea sea characteristics. This experimets achievd useing SSB analog modulation in Jin-hae shore of shallow water condition. Used data are tonal and LFM signal for getting underwater channel characterisitcs and female Korean speech for speech communications.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.167-172
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• 2002

The EVRC (Enhanced Variable Rate Codec) is currently in service as a speech cosec in digital cellular systems in North America and Korea. In the EVRC, the LSP (Line Spectral Pairs) related to energy distribution of speech signals in the frequency domain are coded by weighted split vector quantization. Considering that the LSP codebooks might be trained with the language of the develop country of the codebooks or English, it is expected that codebooks trained with Korean provide the performance improvements in the communication in Korean. In this paper, the EVRC LSP codebooks are designed with korean adopting the LBG algorithm based vector quantization, and the performance improvement of the vector quantization and the accompanying speech quality improvement are demonstrated by spectral distortion, SNR and SegSNR measurements, respectively.