• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2027-2035
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• 2001

In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.556-562
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• 2013

This paper is aimed to improve performance of Band-Selection speech/audio Coder reconstucted band spectrum that is not sent by the comfort noise. To improve the performance, we use the Spectral Band Replication(SBR) technique instead of substitution of Comfort noise. To synthesize SBR signal, the SBR algorithm is referenced in selected signals and the spectrum synthesized by SBR is injected to non-selected band. Each sub-band spectrum has been energy-weighted by real audio signal. We propose the enhanced the Band-Selection Coder that utilizes synthesized SBR signal from selected signal instead of comfort noise.

• The Journal of the Acoustical Society of Korea
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• v.34 no.1
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• pp.82-91
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• 2015

In this paper, a novel codebook design method for data modem over voice channel is presented. Proposed method searches the symbols which have the maximum probability distribution overlap in the symbol space and minimizes the overlap to improve the symbol error rate via minimax optimization. We present numerical simulations and an example implementation. We also give the results of the experiment tests.

• Journal of the Korean Institute of Intelligent Systems
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• v.20 no.2
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• pp.208-213
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• 2010

Generally, the mal factor for speech recognition is the background noise in speech recognition. The noise is the reason to reduce the speech recognition performance. Owing to the fact, the place to recognize is very important. To improve the recognition performance from the sound having noise, we implemented the noise filtered Wiener filter at the signal process step which adopted the FIR filter. In FIR filter, it deal with the filtered speech signal which is appropriate frequency range of human speech frequency range. Therefore, we make the recognition system distinguish between noise and speech sound from the incoming speech signal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.12C
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• pp.1262-1267
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• 2005

Elliptical basis function (EBF) neural network works stably under high-level background noise environment and makes the nonlinear processing possible. It can be adapted real time VAD with simple design. This paper introduces VAD implementation using EBF and the experimental results show that EBF VAD outperforms G729 Annex B and RBF neural networks. The best error rates achieved by the EBF networks were improved more than $70\%$ in speech and $50\%$ in silence while that achieved by G.729 Annex B and RBF networks respectively.

• The Journal of the Acoustical Society of Korea
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• v.31 no.7
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• pp.472-479
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• 2012

This paper is aimed to design 9.2 kbps low bits late transform excitation coder that target to voice and audio signal. To set up low bit rate, we used Band-selection in frequency domain and gain-shape quantization and AbS structure. To decrease lots of calculation from ABS structure, we used each band IDFT and synthesis. And we designed non-transfer band for performance by inserting comfort noise. We propose coder that has low bit rate and similar performance comparing with original 10.4 kbps AMR-WB+ TCX mode.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.05a
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• pp.639-642
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• 2015

As attack to routing function on MANET(Mobile Ad-Hoc Network), hole attack make cause some critical effects. MANET is easily influenced with hole attack and can be critically effected on transmission performance, because it is configured with terminal device as temporary network and dose not have effective means for malicious attack. In this paper, effects of grayhole attack to network performance on MANTE is analyzed with computer simulation. Voice traffic is used in simulation, effects of grayhole attack is compaerd with blackhole attack. The method and result of this paper can be used for data to study grayhoke attack.

• The Journal of Natural Sciences
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• v.8 no.2
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• pp.93-99
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• 1996

In this study, we compare performance of priority schemes of FDDI and Expressnet for integrated voice and data traffic through simulation. The voice capacity of FDDI is higher than that of Expressnet for all cases considered. When compared to Expressnet, FDDI achieves a higher data throughput for file transfer traffic but it incurs a longer delay for interactive traffic.

• The Journal of the Acoustical Society of Korea
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• v.17 no.1
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• pp.9-16
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• 1998

본 논문에서는 낮은 비트율 CELP 음성 부호화기의 장구간 예측기의 성능 향상 방 법을 제안한다. 비트율을 낮추기 위해서는 분석 구간의 길이가 길어져야하며 이에 따라 장 구간 예측기의 성능이 저하되어 장구간 예측 후에도 준 주기성 성분이 상당량 존재하므로 백색 잡음으로 구성된 통계 코드북만으로는 이를 모델링하기 어려워진다. 제안 방법에서는 다중 대역 필터와 다중 펄스열을 이용하여 한 번 더 필터링(이차 장구간 예측)함으로써 장 구간 예측 후의 신호가 통계 코드북에 적합한 백색 잡음 형태로 되도록 모델링한다. 제안된 방법의 성능을 평가하기 위해 4.8kbps 비트율로 양자화한 후, 기존에 제안된 같은 전송률의 MBCELP와 DoD-CELP와 비교하였다. 실험 결과 제안된 방법이 기존 부호화기들에 비해 주/객관적인 음질에서 우수한 성능을 보여준다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.6B
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• pp.709-716
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• 2001

반복 복호 알고리듬에 의해 복호화된 터보 코드는 가산성 백색 가우시안 잡음(AWGN) 채널 환경에서 이론적으로 Shannon의 한계에 근접한 뛰어난 코딩 이득을 나타내는 것으로 보여지고 있다. 그러나, 터보 코드의 성능은 터보 부호화기에서 프레임의 크기 즉, 인터리버의 크기에 의존한다. IMT-2000과 같은 이동 통신 채널 환경에서 음성을 전송하는 경우에는 터보 코드의 프레임 크기는 매우 작다. 그리고, 그것은 터보 코드의 성능을 떨어뜨리는 직접적인 원인이 된다. 본 논문에서는 차세대 이동 통신 시스템에서 프레임 크기가 작은 음성 프레임을 이용하여 터보 코드의 성능을 검증하며, 작은 프레임 크기에 알맞은 3개의 직렬 MAP(Maximum A Posteriori probability) 복호기를 이용한 반복 복호의 터보 코드를 제안하고 부호율 1/3, 구속장의 길이 3 또는 4, 프레임 크기 24, 192 비트에 대하여 컴퓨터 모의실험을 통해 터보 코드의 성능을 분석한다.