• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.5-11
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• 1995

In this paper, we design the new high-speed seek algorithm which can access the data of disk (in 3.5-inch optical disk drive). When seek the target track on optical disk in high-speed using the ref-T. E(Reference Tracking Error) and sample pulse signal, it's always possible high-speed seek without re-check and compensation for error by the ref_T. E and sample pulse signal. The performance test result is represented as follow ; it is satify the average seek time of 28mses.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.275-280
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• 1999

Deep-water sonar transducers of FFR (Free Flooded Ring) type have been designed using a coupled FE-BEM. The proposed sonar transducers are composed of piezoelectric ceramic tubes and structural steel materials for simple fabrication. In order to have an omnidirectional beam pattern around 30 kHz, a conic steel is placed below a piezoelectric tube or a steel disc is placed between two piezoelectric tubes. The dynamics of the sonar transducer is modelled in three dimensions and is analyzed with external electrical excitation conditions. Various results are available such as directivity patterns and transmitting voltage responses. The most optimal structure and dimensions of the steel material were calculated, so that the beam patterns of the sonar transducers had +/- 3 dB omnidirectivity at 30 kHz.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.27-36
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• 2002

VoiceXML is a standard dialog mark-up language for the neat generation voice applications. The current VoiceXML 1.0 specification is silent on who place outbound calls for push-based voice applications. The push-barred voice applications become very important in modern information systems such as CRM. In this paper, we design and implement an extended VoiceXML platform that supports both inbound and outbound voice information services. We also extend the VoiceXML DTD so as to be able to inbound/outbound fax based on Call Control Requirements of W3C.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.183-186
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• 2004

본 논문에서는 수중 초음파 통신용 QPSK 버스트 수신기를 DSP시스템을 이용하여 구현하기위한 시스템 설계에 대하여 논한다. 본 논문에서 고려하는 시스템은 25kHz의 반송주파수를 사용하고, 심벌율은 5kHz이며, 데이터 전송율은 10,000bps이다. 송신기에서 심벌정보를 전송하기 위해 펄스성형필터를 거친 신호를 디지털 믹서기를 이용하여 디지털 영역에서 반송주파수 대역으로 신호를 변조한 후 200kHz로 샘플링하는 D/A변환기를 이용하여 전송 아날로그 신호를 생성한다. 수신기에서는 수신 신호를 디지털로 처리하기 위하여 100kHz로 free running하는 A/D 변환기를 이용하여 수신 데이터를 얻는다. 수신기에서는 32심벌 길이의 프리앰블을 이용하여 프레임 동기를 찾음과 동시에 개략적인 심벌시간 동기와 위상편이를 추정한다. 추정한 위상편이값은 2차 PLL (phase-looked loop)의 초기값으로 사용하여 위상 추적을 수행하는 전송 시스템이다. 또한, 된 논문에서는 실해역 전송 시험 테이터를 통하여 조류의 변화에 의해 발생하는 Doppler 편이를 보상하기 위하여 PLL이 필수적으로 필요함을 보인다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.221-224
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• 2004

A theoretical model has been studied to describe the sound radiation analysis for structure vibration noise of vehicle tires under the action of random moving line forces. When a tire is analyzed, it had been modeled as curved beams with distributed springs and dash pots that represent the radial , tangential stiffness and damping of tire, respectively. The reaction due to fluid loading on the vibratory response of the curved beam is taken into account. The curved beam is assumed to occupy the plane y=0 and to be axially infinite. The curved beam material and elastic foundation are assumed to be lossless Bernoulli-Euler beam theory including a tension force, damping coefficient and stiffness of foundation will be employed. The expression for sound power is integrated numerically and the results examined as a function of Mach number, wave-number ratio and stiffness factor. The experimental investigation for structure vibration noise of vehicle tire under the action of random moving line forces has been made. Based on the Spatial Transformation of Sound Field techniques, the sound power and sound radiation are measured. Results strongly suggest that operation condition in the tire material properties and design factors of the tire govern the sound power and sound radiation characteristics.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.293-296
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• 1998

본 논문에서는 다양한 멀티미디어 서비스를 위해 가변율 half rate 음성 부호화기를 설계하였다. 유, 무성음과 묵음의 구분을 위해 본 논문에서는 프레임 에너지와 음성 파라메터들을 이용한 효과적인 voicing 결정 알고리즘을 사용하였다. 유성음을 위한 half rate 음성 부호화기는 저속에서 좋은 특성을 보이는 generalized AbS구조를 이용하였다. LPC 계수는 LSP 계수로 변환한 후 predictive 2-stage VQ를 통해서 양자화하며, 여기 신호는 음질저하를 최소화하며 복잡도를 감소시킨 shift 방식의 대수적 고정 코드북 구조를 사용하고, 적응코드북과 여기코드북의 이득은 VQ로 양자화 하였다. 무성음을 위한 부호화기는 대부분이 유성음을 위한 부호화기와 동일하지만, 무성음에서는 피치간 상관도가 매우 낮으므로 피치 보간 방법을 사용하지 않고 개루프로 피치 lag를 찾은 후 전체 프레임에 사용한다. 1 kb/s 부호화기는 묵음 구간과 주변소음 구간에 사용되며 이 구간의 신호를 피치 성분이 미약한 주변소음들로 제한하고 이에 최적인 부음성 부호화기를 설계하였다. 최종적으로 완성된 가변율 half rate 부호화기는 voice activity factor(VAF)가 0.47인 시험음성에서 약 2.6 kb/s의 평균 전송률을 보였다. 주관적 음질 평가의 일환으로 IS-96 표준 코덱인 가변율 8 kb/s QCELP와 A-B preference 시험을 실시하였다. 시험 결과 평균전송률이 약 2배인 가변율 8 kb/s QCELP 보다 우수한 음질 성능을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.9 no.3
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• pp.24-30
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• 1990

In this paper, meander line type EMAT(Electro-Magnetic Acoustic Transducer) has been designed and fabricated with effective properties for detecting flaw existing within one wavelength in depth, and its characteristics have been analyzed. For the purpose of getting effective dynamic and static magnetic intensity, the coil has been arrayed using wire with interval of 0.75 mm and width of 0.65 mm and permanent magnets with 1500 Gauss have been constructed respectively. The center frequency and fractional bandwidth of the fabricated EMAT was 2 MHz and 36% respectively and its impulse response has been measured by non-contacting technique(the distance between the conducting media and the coil was 0.15mm). In the measuring results, it has been shown that Insertion Loss(IL) was 45.46dB and it was good agreement with theoretical result.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4
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• pp.42-47
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• 2001

In this paper, we propose an interpreter, which recognizes the VoiceXML markups, verifies the validation of the document, and interprets the VoiceXML documents using DI parser and the generated AST by the parser. The VoiceXML interpreter consists of DI parser and executor, and the DI parser uses recursive descent parsing technology, and the executor uses FIA (Form Interpretation Algorithm) proposed by VXML forum. This system uses the Java language in order to develop the runtime environment for VoiceXML efficiently, thus this system has portability.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.89-93
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• 2001

In recent years Fiber optic hydrophone systems have been the focus of much attention in the sonar world. For sonar arrays, a fiber optic approach offers the major benefit of passive multiplexing of large numbers of hydrophones without underwater electronics. This paper describes recent development work covering array construction, opto-electronics development, hydrohpone design and sea trials. And the development of an interferometric mult-channel fiber optic hydrophone system which uses time division multiplexing capable of driving in excess of 32 channel is described. For this, a 12 channel time division multiplexing array has been constructed, and the performance of this system is demonstrated by sea trial.

• 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.