• Electronics and Telecommunications Trends
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• v.14 no.5 s.59
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• pp.147-148
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• 1999

이 시스템은 MPEG-4 비디오 부호화 방식, G.723.1/MPEG-4 음성부호화 방식, 실시간 통신용 인터넷 프로토콜 방식을 이용하여 인터넷/인트라넷 환경에서 상대방의 모습을 보면서 통화할 수 있도록 구현된 것으로서 비디오 캡처 보드를 장착한 PC에서 소프트웨어만으로 실시간 동작하는 것이 가능하다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.687-695
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• 2002

Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.443-451
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• 2003

Since a codebook-based CELP coder models its excitation signal according to one of several bit rates pre-assigned to codebooks and synthesizes speech signal using codebooks, it can not support encoding of speech signal at an arbitrary bit rate in one encoder. The proposed variable bit rate speech coder encodes the excitation signal based on the bit rate assigned to a present frame of speech using one-dimensional SPIHT and wavelet transform. Also it does't need to model excitation signal (or codebook) to some types as CELP coder, and can encode excitation signal at various bit rates without exact pitch information according to user requirement. As a result, since the coder doesn't have a codebook structure, it has relatively low coder complexity and provides equal or better speech quality compared to G.729 and G.723.1 coder.

• Journal of Physiology & Pathology in Korean Medicine
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• v.20 no.3
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• pp.719-723
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• 2006

Chlororinated bibenzyl compounds (1 and 2) inhibited the growth of the Gram positive bacterium Bacillus subtilis ATCC 19659, (2 mm inhibition zone and 2 mm inhibition zone at $30{\;}{\mu}g/disc$), Candida albicans ATCC 14053, (2 mm inhibition zone and 2 mm inhibition zone at $30{\;}{\mu}g/disc$), and the dermatophytic fungi Trichophyton mentagrophytes ATCC 28185, (3 mm inhibition zone and 7 mm inhibition zone at 30 Ug/disc) and Cladosporium resl'nae ATCC 52833 (1 mm inhibition zone at $30{\;}{\mu}g/disc$).

• Journal of Korean Society on Water Environment
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• v.28 no.5
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• pp.723-728
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• 2012

In this study, zeolites were synthesized by a fusion and a hydrothermal methods using a coal fly ash and a waste catalyst. The recovery performance of metal ions on the structure property of synthetic zeolites was evaluated as comparing the adsorption kinetics (Lagergen 2nd order model) and isotherm (Langmuir model) of $Co^{2+},\;Ni^{2+}$, and $Cu^{2+}$ ions. The synthetic zeolites (Z-C1 and Z-W5) were similarly assigned to XRD peaks in a reagent grade Na-A zeolite (Z-WK : $Na_{12}Al_{12}Si_{12}O_{48}\;27.4H_2O$). Adsorption rates of Z-W5 and Z-C1 were in the order of $Cu^{2+}\;>\;Co^{2+}\;>\;Ni^{2+}\;and\;Ni^{2+}\;>\;Cu^{2+}\;>\;Co^{2+}$, respectively. They had influenced upon structure properties of zeolite. Selectivities of metal ions and maximum equilibrium adsorption capacities, $q_{max}$, in Z-C1 and Z-W5 were in the order of $Ni^{2+}$ (127.9 mg/g) > $Cu^{2+}$ (94.7 mg/g) > $Co^{2+}$ (82.6 mg/g) and $Cu^{2+}$ (141.3 mg/g) > $Co^{2+}$ (122.2 mg/g) > $Ni^{2+}$ (87.6 mg/g), respectively. The results show that the synthetic zeolites, Z-C1 and Z-W5, are able to recover metal ions selectively in wastewater.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.4
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• pp.723-730
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• 2015

In this paper, a location tracking compensation algorithm based on the Least-Squares Method ($LCA_{LSM}$) was proposed to improve the autonomous tracking efficiency for the docent robot in exhibition hall, and the performance of the $LCA_{LSM}$ is analyzed by several practical experiments. The proposed $LCA_{LSM}$ compensates the collected location coordinates for the robot using the Least-Squares Method (LSM) in order to reduce the cumulated errors that occur in the Encoder/Giro sensor (E/G) and to enhance the measured tracking accuracy rates in the autonomous tracking of the robot in exhibition hall. By experiments, it was confirmed that the average error reduction rates of the $LCA_{LSM}$ are higher as 4.85% than that of the $LCA_{KF}$ in Scenario 1 (S1) and Scenario 2 (S2), respectively on the location tracking. In addition, it was also confirmed that the standard deviation in the measured errors of the $LCA_{LSM}$ are much more low and constant compared to that of the E/G sensor and the $LCA_{KF}$ in S1 and S2 respectively. Finally, we see that the suggested $LCA_{LSM}$ can execute more the stabilized location tracking than the E/G sensors and the $LCA_{KF}$ on the straight lines of S1 and S2 for the docent robot.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.289-291
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• 2001

본 논문에서는 실시간처리 리눅스에 기반한 VoIP 단말기를 설계 구현한다. 설계 구현된 하드웨어 시스템은 i386 프로세서를 기반으로 설계되며, 실시간음성처리 지원을 위해 음성코덱 칩과 실시간처리 리눅스인 RTLinux를 사용하여 실시간 음성처리 모듈을 구현한다. 설계 구현된 시스템의 테스트 및 타당성 검증을 위해 LAN환경에서의 음성채팅 프로그램에 적용하여 통화품질을 시험하였다. 음성처리에 사용한 음성처리 모듈은 ITU-T 음성 코덱인 G.723.1 사용하여 30ms 내에 24 바이트로 인코딩/디코딩된 음성 데이타를 전송하도록 구현하였다.

• Proceedings of the KAIS Fall Conference
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• 2010.05a
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• pp.269-272
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• 2010

In this paper, we propose the computation reduction methods of delta pitch search that is used in G.723.1 vocoder. In order to decrease the computational amount in delta pitch search the characteristic of proposed algorithms is as the following. First, scheme to reduce the computation amount in delta pitch search uses NAMDF. Developed the second scheme is the skipping technique of lags in pitch searching by using the threshold value. By doing so, we can reduce the computational amount of pitch searching more than 64% with negligible quality degradation.

• MALSORI
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• no.64
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• pp.105-119
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• 2007

Noise reduction technologies are indispensable to achieve acceptable speech quality in VoIP systems. This paper proposes a Wiener filter optimized to the estimated SNR of noisy speech for the noise reduction in VoIP environments. The proposed noise canceller is applied as a pre-processor before speech encoding. The performance of the proposed method is evaluated by the PESQ in various noisy conditions. In this paper, the proposed algorithm is applied to G.711, G.723.1, and G.729A which are all VoIP speech codecs. The PESQ results show that the performance of our proposed noise reduction scheme outperforms those of the noise suppression in the IS-127 EVRC and the ETSI standard for the advanced distributed speech recognition front-end.

• Proceedings of the IEEK Conference
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• 2000.11c
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• pp.161-164
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• 2000

This paper presents an implementation of internet gateway system, named AX-2000 and call simulator. AX-2000 plays a important role in internet telephony technology and is composed of various board such as MPU, AVU. DVU, AMU. FXS, FXO, EM. Also AX-2000 supports G.729.a, G.723.1 for voice compression, G3 FAX Relay(T.38) and H.323. A capability of AX-2000 is 8 analog voice channel or 30 digital voice channel. For functional verification of AX-2000 voice interface, call simulator is designed. The call simulator makes actual call path between SUT(system under test) and reference AX-2000 system, then through call path examines functions of voice interface.