• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.4B
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• pp.457-462
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• 2001

유/무선 통신 시스템에서는 통신망보다 서로 다른 음성 부호화기를 사용하므로 음성신호는 두 번의 부/복호화과정을 거치게 되어 음질저하, 연산량 증가, 그리고 전달 지연 증가 등의 문제가 발생된다. 본 논문에서는 위의 문제점들을 개선하기 위하여 유/무선 음성통신에 사용되는 음성 부호화기의 상호 부호화 알고리듬을 제안하고, 5.3 kbps G. 723.1의 패킷과 8 kbit/s G.729.A 패킷을 서로 변환하는 방법을 제안한다. 여러 가지 음성 상호 대해 객관적 음질 평가와 주관적 선호도 평가를 수행한 결과, 제안된 상호 부호화 알고리듬이 이중 부/복호화보다 짧은 전달 지연 시간과 26-37% 적은 연산량으로 동등한 음질의 음성신호를 복호화함을 확인하였다.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.34-40
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• 2010

As internet spread widely, various service which use the internet have been provided. One of the service is a internet phone. Its usage is increasing by the advantage of cost. But it has a falling off in quality of speech. because it use packet switching method while existing telephone use circuit switching method. Although vocoder use PLC (Packet Loss Concealment) algorithm, it has a weakness of continuous packet loss. In this paper, we propose methods to improve a lowering in quality of speech under continuous loss of packet by using PLC algorithm used in advanced G.729 and G.711. The proposed methods are LP (Linear Prediction) parameter interpolation, excitation signal reconstruction and excitation signal gain reconstruction. As a result, the proposed method shows superior performance about 11%.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.3 s.309
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• pp.54-60
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• 2006

This paper presents an effective algorithm for transcoding between the Enhanced Variable Rate Codec(EVRC) and G.729A. The simplest way to communicate between heterogeneous speech networks is the cascade connection of two different codecs, called tandem coding. However, tandem coding not only produces high computational loads, but also makes long delay, These problems can be solved by using the transcoding algorithm. The proposed algorithm consists of LSP (Line Spectral Pair) conversion, pitch delay conversion and algorithm for reduction of delay. Experimental results show the proposed algorithm produces lower computational complexity, shorter algorithm delay, and similar speech quality when compared with the tandem algorithm.

• Communications of the Korean Mathematical Society
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• v.13 no.1
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• pp.109-121
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• 1998

Let X,Y be connected, locally connected, semilocally simply connected and $f,g : X \to Y$ be a pair of maps. We find an upper bound of the Reidemeister number R(f,g) by using the regular coverig spaces.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.873-874
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• 2008

In this paper, we propose a packet loss concealment (PLC) algorithm for CELP speech coders, which is based on multiple adaptive codebooks by using comfort noise for the lost packet recovery. The multiple adaptive codebooks are composed of a conventional adaptive codebook to model periodic excitation of speech and another adaptive codebook to provide a better estimate of excitation when packets are lost in the speech onset region. The performance of the proposed PLC algorithm is evaluated by implementing it into the G.729 decoder and compared with that of the PLC algorithm employed in the G.729 decoder by means of perceptual evaluation of speech quality (PESQ). It is shown from the experiments under different burstiness of packet loss rates of 3% and 5% that the proposed PLC algorithm provides higher PESQ scores than the G.729 PLC algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.7B
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• pp.528-535
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• 2012

This paper proposes silence drop first(SDF) active buffer management algorithm to increase the voice capacity when silence suppression is used. This algorithm finds and drops silence packet rather than voice packet in the queue for resolving buffer overflow of queue. Simulations with voice codec of G.729A and G.711 are performed. By using proposed SDF algorithm, the voice capacity is increased by 84.21% with G.729A and 38.46% with G.711. Further more, SDF algorithm reduces the required link capacity and loosens the silence packet inter-arrival time limit to provide target voice quality compared with that of conventional algorithms.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.5
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• pp.129-132
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• 2009

Frame erasure is one of the most difficult problems in voice over IP (VoIP) networks and is a major source of speech quality degradation. In this paper, a frame erasure concealment algorithm based on nonlinear regression analysis is presented to minimize speech quality deterioration in code-excited linear prediction (CELP) based coders. We applied the proposed scheme to the ITU-T G.729 standard and obtained improved perceptual evaluation of speech quality (PESQ) scores compared to the conventional methods.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.84-89
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• 2003

In this paper, we real-time implemented to the TMS320C5416 the vocoder of variable bit rate applied the SOLA-B algorithm by Henja to the ITU-T G.729A vocoder of 8kbps transmission rate. This proposed method using the SOLA-B algorithm is that it is reduced the duration of the speech in encoding and is played at the speed of normal by extending the duration of the speech in decoding. At this time, we bandied that the interval of cross correlation function if skipped every 3 sample for decreasing the computational amount of SOLA-B algorithm. The real-time implemented vocoder of C.729A and SOLA-B algorithm is represented the complexity of maximum that is 10.2MIPS in encoder and 2.8MIPS in decoder of 8kbps transmission rate. Also, it is represented the complexity of maximum that is 18.5MIPS in encoder and 13.1MIPS in decoder of 6kbps, it is 18.5MIPS in encoder and 13.1MIPS in decoder of 4kbps. The used memory is about program ROM 9.7kwords, table ROM 4.5kwords, RAM 5.1 kwords. The waveform of output is showed by the result of C simulator and Bit Exact. Also, for evaluation of speech quality of the vocoder of real-time implemented variable bit rate, it is estimated the MOS score of 3.69 in 4kbps.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1211-1214
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• 1998

Real-time implementation of Conjugate-Structure Algebraic CELP(CS-ACELP) is presented. ITU-T Study Group(SG) 15 has standardized the CS-ACELP speech coding algorithm as G.729. A real-time implementation of the CS-ACELP is achieved using 16 bit fixed point DSP16210 Digital Signal Processor (DSP) of Lucent Technologies. The speech coder has been implemented in the bit-exact manner using the fixed point CS-ACELP C source which is the part of the G.729 standard. To provide a multi-channel vocoder solution to digital communication system, we try to minimize the complexity(e.g., MIPS, ROM, RAM) of CS-ACELP. Our speech coder shows 15.5 MIPS in performance which enables 4 channel CS-ACELP to be processed with one DSP16210.

• Mycobiology
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• v.42 no.4
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• pp.339-345
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• 2014

During 2011~2013, a total of 729 samples for 19 types of medicinal plant were collected from Seoulyekryungsi in Seoul, Korea, and investigated for the presence of aflatoxins. The samples were analyzed using immunoaffinity column cleanup and high-performance liquid chromatography coupled to a fluorescence detector after post-column derivatization. Aflatoxins were found in 124 out of the 729 analyzed samples: 65 containing aflatoxin B1 (AFB1), 24 with aflatoxin B2 (AFB2), 15 with aflatoxin G1 (AFG1), and 20 samples with aflatoxin G2 (AFG2). The ranges for positive samples were $0.1{\sim}404.7{\mu}g/kg$ for AFB1, $0.1{\sim}10.0{\mu}g/kg$ for AFB2, $0.1{\sim}635.3{\mu}g/kg$ for AFG1, $0.1{\sim}182.5{\mu}g/kg$ for AFG2, and $0.1{\sim}1,043.9{\mu}g/kg$ for total aflatoxins. Most of the medicinal plant samples (721, 98.9%) were below legal limits, but 8 samples exceeded the legal limits of 10 and $15{\mu}g/kg$ established by the Korean standard for AFB1 and total aflatoxins (the sum of AFB1, AFB2, AFG1 and AFG2), respectively.