• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.293-296
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• 1998

본 논문에서는 다양한 멀티미디어 서비스를 위해 가변율 half rate 음성 부호화기를 설계하였다. 유, 무성음과 묵음의 구분을 위해 본 논문에서는 프레임 에너지와 음성 파라메터들을 이용한 효과적인 voicing 결정 알고리즘을 사용하였다. 유성음을 위한 half rate 음성 부호화기는 저속에서 좋은 특성을 보이는 generalized AbS구조를 이용하였다. LPC 계수는 LSP 계수로 변환한 후 predictive 2-stage VQ를 통해서 양자화하며, 여기 신호는 음질저하를 최소화하며 복잡도를 감소시킨 shift 방식의 대수적 고정 코드북 구조를 사용하고, 적응코드북과 여기코드북의 이득은 VQ로 양자화 하였다. 무성음을 위한 부호화기는 대부분이 유성음을 위한 부호화기와 동일하지만, 무성음에서는 피치간 상관도가 매우 낮으므로 피치 보간 방법을 사용하지 않고 개루프로 피치 lag를 찾은 후 전체 프레임에 사용한다. 1 kb/s 부호화기는 묵음 구간과 주변소음 구간에 사용되며 이 구간의 신호를 피치 성분이 미약한 주변소음들로 제한하고 이에 최적인 부음성 부호화기를 설계하였다. 최종적으로 완성된 가변율 half rate 부호화기는 voice activity factor(VAF)가 0.47인 시험음성에서 약 2.6 kb/s의 평균 전송률을 보였다. 주관적 음질 평가의 일환으로 IS-96 표준 코덱인 가변율 8 kb/s QCELP와 A-B preference 시험을 실시하였다. 시험 결과 평균전송률이 약 2배인 가변율 8 kb/s QCELP 보다 우수한 음질 성능을 보였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.133-134
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• 2011

본 논문에서는 소스코딩에 활용되고 있는 LDPCA 부호를 가변 부호율을 얻는 채널코딩 목적으로써의 사용 가능성을 타진한다. 컴퓨터 성능 실험의 결과로 가변 부호율과 어느 정도의 BER 성능을 확인하였지만, 체크노드 머징기법이 여타의 LDPC 부호에 비해 성능을 크게 떨어뜨리는 것으로 나타났다. 향후 체크노드 머징이 가능하면서도 좀 더 개선된 성능을 가지는 LDPCA 를 설계하는 일이 중요할 것으로 생각된다.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.48 no.2
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• pp.35-39
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• 2011

Error correcting codes with easy variability in code rate and codeword length in addition to powerful error correcting capability are required for present and future mobile multimedia communication systems. And low complexity is also needed for the compact mobile terminals. In general, the irregular random LDPC(low-density parity-check) code is known to have the superior performance among various LDPC codes. But it has inefficiency since the various parity check matrices for various services should be stored for encoding and decoding. The structured LDPC codes which can easily provide various rates and lengths are studied recently. Therefore, the flexibility, memory size, and error performance of various structured LDPC codes are compared and analyzed in this paper. And the most appropriate structured LDPC code is also suggested.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2005.11a
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• pp.310-313
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• 2005

5GHz를 사용하고 최소6Mbps에서 최대 54Mbps까지 지원하는 IEEE 802.11a 무선 랜 에서는 데이터의 전송중에 발생하는 랜덤오류 및 연집오류의 정정을 위해서 길쌈부호기와 인터리버를 규정하고 있다. 길쌈부호기에서 다양하고 높은 데이터 전송율을 확보하기 위해서 변조방식과 전송율에 따라 높은 부호율을 얻기 위해 여러 개의 펑쳐링 기법을 사용해서 2/3와 3/4과 같은 높은 부호율이 얻어지도록 하고 있다. 펑쳐링을 거친 데이터는 Coding rate만큼의 비율로 데이터가 많아지고 이 데이터를 처리하기 위해서 가변클럭을 사용해야 한다. 가변클럭의 사용은 동기화 및 back-end 작업 시 여러 가지 문제를 발생시킨다. 본 논문에서 펑쳐링의 출력 비트를 바꾸고 인터리버에서 사용되는 메모리를 8x1의 메모리로 세분화 함으로써 Code rate에 상관없이 하나의 클럭으로 데이터의 병목현상을 처리 할 수 있다.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.8 no.4
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• pp.332-341
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• 1997

In this paper, we propose an adaptive trellis-coded Multipe Quadrature Amplitude Modulation (ATCMQAM) for slowly varying Rayleigh fading channels. The proposed system adaptively controls the coding rate combined with modulation level of pargmatic approach to trellis-coded modulation according to the instantaneous fading channel conditions, and employs MQAM as modulation scheme. Results by computer simulation show that the proposed adaptive model using MQAM can realize higher quality transmission with the improvement more than 0.5~1 bit in average bit rate, and there is a coding gain of 2~5 dB, depending on the high SNR value, compared with the conventional adaptive model employing MPSK.

• 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 the Korea Institute of Information and Communication Engineering
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• v.12 no.1
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• pp.189-195
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• 2008

The OFDMA Physical layer in the WiMAX standard of IEEE 802.16e adopts 114 LDPC codes with various code rates and block sizes as a channel coding scheme to meet varying channel environments and different requirements for transmission performance. In this paper, the performances of the LDPC codes are evaluated according to various code rates and block-lengths throueh simulation studies using min-sum decoding algorithm in AWGN chamois. As the block-length increases and the code rate decreases, the BER performance improves. In the cases with code rates of 2/3 and 3/4, where two different codes ate specified for each code rate, the codes with code rates of 2/3A and 3/4B outperform those of 2/3B and 3/4A, respectively. Through the statistical analyses of the number of decoding iterations the decoding complexity and the word error rates of LDPC codes are estimated. The results can be used to trade-off between the performance and the complexity in designs of LDPC decoders.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.1
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• pp.33-38
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• 2012

As underwater channel is very complex and time-varying, don't supports good-quality for communication service. In this paper, a multi-carrier CDMA(MC-CDMA) system for the reliability and robust service in the underwater acoustic channel is proposed and analyzed for its performance. Applied variable rate algorithm to the proposed system gets a channel state information from relationship between SINR and user data-rate. Using channel state information make spectrum usage more efficient and overall system performance improved. In this paper, the performance of proposed system analyzed by simulation. And Pseudo-Random spread codes used in the system are discussed.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.4
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• pp.803-812
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• 2018

The Morse code was firstly used for a wire telegraph service on May 1884 in USA. After about 10 years Korean researchers developed the Morse code in 1884 and the Morse code was firstly introduced in Korea for the wire telegraph service in September 1885. For about 130 years later, the Hangul Jamo Morse code has been used without any serious modification up to now. The Morse code is the variable length code system which each codes have a different code length. The related reference shows that English alphabet Morse code was applied in the coding theory. However, a result of analysis showed that the Hangul Jamo Morse code was not applied in the coding theory as the english alphabet code system. This paper presents the analysis result of Hangul Jamo Morse code according to the variable length coding theory. As a result of the analysis, when Hangul Jamo are coded with application of the variable length coding theory, the transmission data rate could be increased up to 25% compare to the existing code system. This paper shows a new Hangul Jamo Morse code system based on the analysis result.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.7A
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• pp.994-1004
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• 2000

In this paper, bitstreams are composed of using H.263 for a moving picture coding in the band-limited and error-prone environment such as wireless environment. EREC sub-frames are implemented by applying the proposed EREC algorithm in order to be UEP for the real data parts of implemented bitstreams. Because those are able to do resynchronization with a block unit, propagation of the error can be minimized, and the position of the important bits such as INTRADC and MVD can be known. Class is separated using the position of these important bits, and variable puncturing tables are designed by the class informations and the code rates of turbo codes are differently designed in according to the class. Channel coding used the turbo codes, and an interleaver to be designed in the turbo codes does not eliminate redundancy bits of the important bits in applying variable code rates of EREC sub-frames unit and is always the same at the transmitter and the receiver although being variable frame size. As a result of simulation, UEP with the code rate similar to EEP is obtained a improved result in the side of bit error probability. And the result of applying it to image knows that the subjective and objective quality have been improved by the protection of important bits.