• Title/Summary/Keyword: Line Coder

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Study on the ASCII Code in the side of the Transmission Efficiency in Data Communications (데이터통신 전송효율과 ASCII 부호체계 고찰)

  • Hong, Wan-Pyo
    • The Journal of the Korea institute of electronic communication sciences
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    • v.6 no.5
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    • pp.657-664
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    • 2011
  • This paper proposes the revised ASCII code. The study started with consideration whether the ASCII code is proper or not in the side of the transmission efficiency in data communications. In data communications, when the consecutive "0" bits from the information devices input into the line coder, its consecutive "0" bits are scrambled to the predetermined patterns not to the consecutive "0" signal. The paper used to study with the statistical data for the frequency of the letters of the alphabets and the proposed rule of characters coding in reference. As a result of the study, when the proposed ASCII code is applied, the operation efficiency of the scrambler in the line coder is improved upto average 30%.

Enhancement of Super-wideband Coder by Considering Audio Feature in MDCT Domain (MDCT 도메인에서 오디오 신호 특징을 고려한 초광대역 코덱 개선)

  • Hong, Ki-Bong;Jeong, Gyu-Hyeok;Lee, In-Sung
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.48 no.5
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    • pp.129-136
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    • 2011
  • This paper presents the coding method that have multi-mode and efficiency of audio codecs using the feature of audio signal. Recently, the developed extension super-wideband codec based on G.718 wideband divides two mode between Generic and Sinusiodal. So codec efficently encode audio signal exist in super-wideband. But the codec is not as efficent coding for harmonic component of wind instrument and string instrument and individual-Line component of percussion instrument. The proposed method are modeling and encoding multiple pitch and individual-line feature using multi mode coding. For the performance evaluation, we used SNR in MDCT domain for objective test and MUSHRA test for subjective test. As a result, the performance of SNR and MUSHRA test of the proposed method have better performance than the G.718 super-wideband codec.

Transcoding Algorithm for SMV and AMR Speech Coder (SMV와 AMR 음성부호화기를 위한 상호부호화 알고리즘)

  • Lee, Duck-Jong;Jeong, Gyu-Hyeok;Lee, In-Sung
    • The Journal of the Acoustical Society of Korea
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    • v.27 no.8
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    • pp.427-434
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    • 2008
  • In this paper, a transcoding algorithm for SMV and AMR speech coder is proposed. In the application requiring the interoperability of different networks, two speech coders must work together with the structure of cascaded connection, tandem. The tandem which is one of the simplest methods has several problems such as long delay, high complexity and the quality degradation due to twice complete encoding/decoding process. These problems can be solved by using transcoding algorithm. The proposed algorithm consists of LSP (Line Spectral Pair) conversion, pitch delay conversion, and fast fixed codebook search. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem with reduced computational complexity and delay.

On the Research of a Speech Coder Using a Multi-Level Amplitude Codebook (다중레벨 진폭 코드북을 이용한 음성 부호화기에 관한 연구)

  • 홍성훈;김정진박영호배명진
    • Proceedings of the IEEK Conference
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    • 1998.10a
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    • pp.1219-1222
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    • 1998
  • This paper analyzes the dynamic spars algebraic codebook used to model a residual signal and proposes a new algebraic codebook structure as well as a searching process with improved performance. The proposed algorithm improves the disadvantage of algebraic codebook without increased computation. First, this paper makes it possibel to select various pulse amplitudes differently from the conventional method which looks up the sign bit simply. In addition, two pulses are made to be selected on the same track. For speech quality on the telephone line 5.6kbps speech coder using the proposed algorithm was equivalent to the 6.3kbps MP-MLQ in the viewpoint of subjective speech quality. However, speech degradation was caused a little compared to the MP-MLQ where MNRU 1=15dB.

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Transcoding Algorithm for SMV and G.723.1 Vocoders via Direct Parameter Transformation (SMV와 G.723.1 음성부호화기를 위한 파라미터 직접 변환 방식의 상호부호화 알고리듬)

  • 서성호;장달원;이선일;유창동
    • Proceedings of the IEEK Conference
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    • 2003.07e
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    • pp.2228-2231
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    • 2003
  • In this paper, a transcoding algorithm for the Selectable Mode Vocoder (SMV) and the G.723.1 speech coder via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the Other Without going through the decoding md encoding process. The proposed algorithm is composed of four parts: the parameter decoding, line spectral pair (LSP) conversion, pitch period conversion, excitation conversion and rate selection. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem transcoding with reduced computational complexity and delay.

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Performance evaluation and Modeling for 500Kbps Digital Power Line Communication System (500Kbps급 디지털 전력선 통신 시스템의 모델링과 성능분석)

  • Kim, Bum Gyu;Choi, Sung Hwan;Kwon, Ho Yeol
    • Journal of Industrial Technology
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    • v.18
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    • pp.431-437
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    • 1998
  • In this paper, we presented a modeling of the power line channel and a new digital communication system over the channel. Firstly, we proposed a new tranceiver structure with DS-CDMA spread spectrum technique and convolutional coder and block interleaver against severe noisy power line environment. Also, QPSK modulation technique was used to get bandwidth efficiency. And then we performed a simulative evaluation of the system using MATLAB communication/simulink toolbox. According to the simulation results, the proposed system gives $10^{-6}$ BER at 20dB SNR.

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Design and Implementation of a Bluetooth LAN access system for VoIP phone (Bluetooth를 이용한 VOIP Phone 의 Wireless LAN Access System 개발)

  • 김정근;김영덕;장태규
    • Proceedings of the IEEK Conference
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    • 2002.06a
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    • pp.343-346
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    • 2002
  • This paper presents a Prototype system developed for a Bluetooth interfaced VoIP system. The VoIP phone is developed based on tile implementation of a CELP coder on the TI 16bit DSP Processor A PC interfaced with Bluetooth module is used to designing a access point system. Host controller protocol stack is implemented to realize gateway between the wireless and wired line networks. A server application program for user management and call processing, which is based on TCP/IP peer to peer connection, is implemented for tile evaluation of overall interface system.

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A New Vocoder based on AMR 7.4Kbit/s Mode for Speaker Dependent System (화자 의존 환경의 AMR 7.4Kbit/s모드에 기반한 보코더)

  • Min, Byung-Jae;Park, Dong-Chul
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.9C
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    • pp.691-696
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    • 2008
  • A new vocoder of Code Excited Linear Predictive (CELP) based on Adaptive Multi Rate (AMR) 7.4kbit/s mode is proposed in this paper. The proposed vocoder achieves a better compression rate in an environment of Speaker Dependent Coding System (SDSC) and is efficiently used for systems, such as OGM(Outgoing message) and TTS(Text To Speech), which needs only one person's speech. In order to enhance the compression rate of a coder, a new Line Spectral Pairs(LSP) code-book is employed by using Centroid Neural Network (CNN) algorithm. In comparison with original(traditional) AMR 7.4 Kbit/s coder, the new coder shows 27% higher compression rate while preserving synthesized speech quality in terms of Mean Opinion Score(MOS).

An Efficient Transcoding Algorithm For G.723.1 and EVRC Speech Coders (G.723.1 음성부호화기와 EVRC 음성부호화기의 상호 부호화 알고리듬)

  • 김경태;정성교;윤성완;박영철;윤대희;최용수;강태익
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.5C
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    • pp.548-554
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    • 2003
  • Interoperability is ole the most important factors for a successful integration of the speech network. To accomplish communication between endpoints employing different speech coders, decoder and encoder of each endpoint coder should be placed in tandem. However, tandem coder often produces problems such as poor speech quality, high computational load, and additional transmission delay. In this paper, we propose an efficient transcoding algorithm that can provide interoperability to the networks employing ITU-T G.723.1[1]and TIA IS-127 EVRC[2]speech coders. The proposed transcoding algorithm is composed of four parts: LSP conversion, open-loop pitch conversion, fast adaptive codebook search, and fast fixed codebook search. Subjective and objective quality evaluation confirmed that the speech quality produced by the proposed transcoding algorithm was equivalent to, or better than the tandem coding, while it had shorter processing delay and less computational complexity, which is certified implementing on TMS320C62x.

Transcoding Algorithm for SMV and G.723.1 Vocoders via Direct Parameter Transformation (SMV와 G.723.1 음성부호화기를 위한 파라미터 직접 변환 방식의 상호부호화 알고리듬)

  • 서성호;장달원;이선일;유창동
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.40 no.6
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    • pp.61-70
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    • 2003
  • In this paper, a transcoding algorithm for the Selectable Mode Vocoder (SMV) and the G.723.1 speech coder via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding process. The proposed algorithm is composed of four parts: the parameter decoding, line spectral pair (LSP) conversion, pitch period conversion, excitation conversion and rate selection. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem transcoding with reduced computational complexity and delay.