• The Bulletin of The Korean Astronomical Society
• /
• v.37 no.2
• /
• pp.168.2-168.2
• /
• 2012

위성체가 발사체에 실려 발사될 때에 매우 높은 가속도에 의한 정적, 동적 하중 및 공기의 저항에 의한 하중, 연소 가스 분출시 발생하는 음향에 의한 하중, 발사체로부터 분리될 때 발생하는 충격 하중 등 여러 가지의 극심한 하중을 겪게 된다. 이러한 발사 환경에 대한 안정성을 검토하기 위해 발사체 업체에서 제공하는 매뉴얼 상의 설계 조건을 이용하여 설계하고 해석하여 검증한다. 천리안 위성의 후속 위성으로, 해상도 및 채널 성능 향상된 차세대 기상탑재체를 탑재하고 현재 개발 중인 정지궤도 복합위성에 대해 발사환경을 고려한 개념 설계 및 초기 해석을 수행하였고, 개발 가능성 분석을 그 목적으로 한다.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.4
• /
• pp.401-406
• /
• 2002

Each Korean digit is composed of only a syllable, so recognizers as well as Korean often have difficulty in recognizing it. When digit strings are pronounced, the original pronunciation of each digit is largely changed due to the co-articulation effect. In addition to these problems, the distortion caused by various channels and noises degrades the recognition performance of Korean connected digit string. This paper dealt with some techniques to improve recognition performance of it, which include defining a set of PLUs by considering phonemic variations in Korean digit and constructing a recognizer to handle speakers various speaking styles. In the speaker-independent connected digit recognition experiments using telephone speech, the proposed techniques with 1-Gaussian/state gave string accuracy of 83.2%, i. e., 7.2% error rate reduction relative to baseline system. With 11-Gaussians/state, we achieved the highest string accuracy of 91.8%, i. e., 4.7% error rate reduction.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.2
• /
• pp.120-127
• /
• 2002

As a new type of a linear decorrelating receiver, the Pseudo-Decorrelator was presented for asynchronous code division multiple access systems by the author. In this paper, the concept of the Pseudo-Decorrelator is extended to derive a multiuser receiver for WCDMA uplink systems over a Rayleigh fading multipath channel. Starting with the analysis of the multiple access components of the decision statistics, a non-square cross-correlation matrix for each bit is obtained. This cross-correlation matrix is then inverted, and the inverted matrix is applied to the decision statistics obtained from a conventional receiver. This receiver is near-far resistant and outperforms conventional receivers even for the cases in which synchronization errors, such as time delay errors and phase errors exist.

• The Journal of the Acoustical Society of Korea
• /
• v.41 no.2
• /
• pp.192-198
• /
• 2022

As an effective weapon system for anti-submarine warfare, DIrectional Frequency Analysis and Recording (DIFAR) sonobuoy detects underwater targets via beamforming with three channels composed of an omni-direcitonal and two directional channels. However, ambient noise degrades the detection performance of DIFAR sonobouy in specific direction (0°, 90°, 180°, 270°). Thus, an ambient noise redcution technique is proposed for performance improvement of acoustic target detection of DIFAR sonobuoy. The proposed method is based on OTA (Order Truncate Average), which is widely used in sonar signal processing area, for ambient noise estimation and Wiener filtering, which is widely used in speech signal processing area, for noise reduction. For evaluation, we compare mean square errors of target bearing estmation results of conventional and proposed methods and we confirmed that the proposed method is effective under 0 dB signal-to-noise ratio.

• Proceedings of the Korean Information Science Society Conference
• /
• 1999.10b
• /
• pp.307-309
• /
• 1999

입체음향이란 음원의 위치에 따라 두 귀에 입력되는 신호를 제어함으로써 시각정보 없이 음상의 위치를 파악할 수 있는 음이다. 헤드폰을 이용하면 음장이 머리 내에 위치하게 됨으로써 거리를 파악하는 것이 매우 힘들다. 본 논문에서는 모노음을 이용하여, 2채널 헤드폰에서 재생할 수 있는 3차원 음을 만들기 위하여 Interaural Time Difference(ITD)와 Interaural Intensity Difference(IID)를 이용한 머리 전달함수(Head Related Transfer Function:HRTF)를 만든 결과와 측정 HRTF 자료인 KEMAR Data를 이용한 결과를 비교하였으며, 거리 효과를 효과적으로 구현하기 위하여 잔향효과를 추가하여 음장을 머리밖으로 꺼냄으로써, 보다 향상된 3차원 음상 제어 시스템을 제안하고 실험하였다.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.13 no.5
• /
• pp.344-352
• /
• 2020

Covertness and performance are very important design goals in the underwater communications. To satisfy both of them, we proposed efficient underwater communication model which combined multiband and direct sequence spread spectrum method in order to improve performance and covertness simultaneously. Turbo coding method with 1/3 coding rates is used for channel coding algorithm, and turbo equalization method which iterately exchange probabilistic information between equalizer and decoder is used for receiver side. After optimal threshold value was set in Rake processing, this paper analyzed the performance by varying the number of chips were 8, 16, 32 and the number of bands were from 1 to 4. Through the simulation results, we confirmed that the performance improvement was obtained by increasing the number of bands and chips. 2~3 dB of performance gain was obtained when the number of chips were increased in same number of bands.

• The Journal of the Acoustical Society of Korea
• /
• v.32 no.5
• /
• pp.393-401
• /
• 2013

In this paper, a $48{\times}64$ channel 2D array ultrasonic transducer with piezoelectric single crystals was designed, fabricated, and evaluated. Structure of the transducer was chosen to facilitate the electric connection on the planar array, and then components were fabricated in accordance with the structure. Detailed structure of the transducer was designed through finite element analyses. In order to improve the performance of the transducer, the crosstalk between adjacent elements was reduced through the control of kerf width and material, and the target frequency bandwidth was achieved through optimal design of the thickness of the single crystal and matching layers. After fabricating a prototype of the transducer according to the design and measuring its characteristics, the results were compared with those of finite element analyses to evaluate the performance of the developed transducer.

• Journal of Navigation and Port Research
• /
• v.39 no.4
• /
• pp.293-297
• /
• 2015

Underwater acoustic communications has been limited use for military purposes in the past. However, the fields of underwater applications expend to detection, submarine and communication in recent. The excessive multipath encountered in underwater acoustic communication channel is creating inter symbol interference, which is limiting factor to achieve a high data rate and bit error rate performance. To improve the performance of a received signal in underwater communication, many researchers have been studied for channel coding scheme with excellent performance at low SNR. In this paper, we applied BCJR decoder based ( 2,1,7 ) convolution codes and to compensate for the distorted data induced by the multipath, we applying the turbo equalization method. Through the underwater experiment on the Gyeungcheun lake located in Mungyeng city, we confirmed that turbo equalization structure of BCJR has better performance than hard decision and soft decision of Viterbi decoding. We also confirmed that the error rate of decoder input is less than error rate of $10^{-1}$, all the data is decoded. We achieved sucess rate of 83% through the experiment.

• The Journal of the Acoustical Society of Korea
• /
• v.29 no.2E
• /
• pp.86-99
• /
• 2010

This paper analyzes the performance of various single channel speech enhancement algorithms when they are applied to automatic speech recognition (ASR) systems as a preprocessor. The functional modules of speech enhancement systems are first divided into four major modules such as a gain estimator, a noise power spectrum estimator, a priori signal to noise ratio (SNR) estimator, and a speech absence probability (SAP) estimator. We investigate the relationship between speech recognition accuracy and the roles of each module. Simulation results show that the Wiener filter outperforms other gain functions such as minimum mean square error-short time spectral amplitude (MMSE-STSA) and minimum mean square error-log spectral amplitude (MMSE-LSA) estimators when a perfect noise estimator is applied. When the performance of the noise estimator degrades, however, MMSE methods including the decision directed module to estimate a priori SNR and the SAP estimation module helps to improve the performance of the enhancement algorithm for speech recognition systems.

• The Journal of the Acoustical Society of Korea
• /
• v.23 no.8
• /
• pp.554-561
• /
• 2004

As the technologies of semiconductor and multimedia communication have been improved. the high-quality video and the multi-channel audio have been highlighted. MPEG Audio Layer 3 decoder has been implemented as a Processor using a standard. Since the synthesis filter of MPEG-1 Audio Layer 3 decoder requires the most outstanding operation in the entire decoder. the synthesis filter that can reduce the amount of operation is needed for the design of the high-speed processor. Therefore, in this paper, the synthesis filter. the most important part of MPEG Audio, is materialized in FPGA using the method of DAULT (distributed arithemetic look-up table). For the design of high-speed synthesis filter, the DAULT method is used instead of a multiplier and a Pipeline structure is used. The Performance improvement by 30% is obtained by additionally making the result of multiplication of data with cosine function into the table. All hardware design of this Paper are described using VHDL (VHIC Hardware Description Language) Active-HDL 6.1 of ALDEC is used for VHDL simulation and Synplify Pro 7.2V is used for Model-sim and synthesis. The corresponding library is materialized by XC4013E and XC4020EX. XC4052XL of XILINX and XACT M1.4 is used for P&R tool. The materialized processor operates from 20MHz to 70MHz.