• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.12C
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• pp.1601-1605
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• 2004

When codeword sequence has two or less successive transitions, the performance of 4th-order partial response maximum-likelihood (PRML) detector can be improved. However, the code leads to an unacceptable loss of performance due to the low code rate. For a rate 718 code that MTR constraint (i) of each codeword is limited to 2, and j is allowed to be 3 when codewords are connected, we modified the trellis of PRML detector to combine j=2 with J=3. We confirmed that the rate 718 coded 4th-order PRML detection with combined trellis achieves the SNR gain more than 2dB compared to the rate 819 coded 4u_order PRML detection at 10-s BER in high-density longitudinal or perpendicular magnetic recording systems.

• The Journal of the Acoustical Society of Korea
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• v.24 no.4E
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• pp.123-127
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• 2005

In this paper, we propose a new method for reducing the number of distance calculations in the LBG (Linde, Buzo, Gray) algorithm, which is widely used method to construct a codebook in vector quantization of speech recognition system. The proposed algorithm can reduce the distance calculation between input vector and codeword by utilizing the observation that codewords are quickly stabilized as the number of iteration increases. From the simulation results, it is shown that we can reduce the running times over $43.77\%$ on average in comparison with current LBG algorithm without sacrificing the performance of codebook.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.8 no.3
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• pp.57-70
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• 2008

This paper introduces a new paradigm of physical layer security through channel coding for wireless networks. The well known spread spectrum based physical layer security in wireless network is applicable when code division multiple access (CDMA) is used as wireless air link interface. In our proposal, we incorporate the proposed security protocol within channel coding as channel coding is an essential part of all kind of wireless communications. Channel coding has a built-in security in the sense of encoding and decoding algorithm. Decoding of a particular codeword is possible only when the encoding procedure is exactly known. This point is the key of our proposed security protocol. The common parameter that required for both encoder and decoder is generally a generator matrix. We proposed a random selection of generators according to a security key to ensure the secrecy of the networks against unauthorized access. Therefore, the conventional channel coding technique is used as a security controller of the network along with its error correcting purpose.

• ETRI Journal
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• v.30 no.1
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• pp.167-169
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• 2008

In this paper, we propose hardware architecture for a high-speed context-adaptive variable length coding (CAVLC) decoder in H.264. In the CAVLC decoder, the codeword length of the current decoding block is used to determine the next input bitstreams (valid bits). Since the computation of valid bits increases the total processing time of CAVLC, we propose two techniques to reduce processing time: one is to reduce the number of decoding steps by introducing a lookup table, and the other is to reduce cycles for calculating the valid bits. The proposed CAVLC decoder can decode $1920{\times}1088$ 30 fps video in real time at a 30.8 MHz clock.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.51-54
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• 2004

In this paper, we analyze the performance of LDPC decoder for DVB-S2 system. The performance analysis is performed by computer simulations based on AWGN channel and high order modulation technique including 16APSK and 32APSK. For normal frame codeword length N = 64800, the performance of LDPC decoder is only away 0.7dB to 1dB from Shannon limit with respect to each modulation. The constructions and encoding process of LDPC codes which are used for DVB-S2 system are also presented and described.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.679-682
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• 1998

This paper presents the VLSI implementation of RS(reed-solomon) decoder using the Modified Euclidean Algorithm(hereafter MEA) for DVD(Digital Versatile Disc) and CD(Compact Disc). The decoder has a capability of correcting 8-error or 16-erasure for DVD and 2-error or 4-erasure for CD. The technique of polynomial evaluation is introduced to realize syndrome calculation and a polynomial expansion circuit is developed to calculate the Forney syndrome polynomial and the erasure locator polynomial. Due to the property of our system with buffer memory, the MEA architecture can have a recursive structure which the number of basic operating cells can be reduced to one. We also proposed five criteria to determine an uncorrectable codeword in using the MEA. The overall architecture is a simple and regular and has a 4-stage pipelined structure.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.4 no.1
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• pp.34-39
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• 2004

Huffman coding which has been used in many data compression algorithms is a popular data compression technique used to reduce statistical redundancy of a signal. It has been proposed that the Huffman algorithm can decode efficiently using characteristics of the Huffman tables and patterns of the Huffman codeword. We propose a new Huffman decoding algorithm which used a multi way tree search and present an efficient hardware implementation method. This algorithm has a small logic area and memory space and is optimized for high speed decoding. The proposed Huffman decoding algorithm can be applied for many multimedia systems such as MPEG audio decoder.

• The Journal of the Acoustical Society of Korea
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• v.12 no.1
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• pp.55-60
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• 1993

본 논문은 불특정 화자의 단모음 인식에 관한 연구로써, VQ(Vectro Quantization)와 MLP(multi-layer perceptron)에 의한 음성 인식 방법을 제안한다. 이 방법은 VQ codebook을 구하고 이를 이용해서 관측열(observation sequence)을 구해각 codeword가 데이터로부터 가질 수 있는 확률값을 계산하여 이 값을 신경 회로망의 입력으로 사용하는 방법이다. 인식 대상으로는 한국어 단모음을 선정하였으며 10명의 남성 화자가 8개의 단모음을 10번씩 발음한 것으로 시스템의 효율성을 알아보기 위해 VQ/HMM(hidden markov model)에 의한 인식과 비교 실험한다. 실험 결과에 의하면, 시스템의 단순성에도 불구하고 학습능력애 뛰어난 관계로 VQ/HMM보다 VQ와 MLP에 의한 음성 인식률이 향상됨을 보여준다.

• Proceedings of the KIEE Conference
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• 1987.07b
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• pp.1434-1437
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• 1987

We present a very simple algorithm for reducing the encoding (codebook search) complexity of vector quantization (VQ), exploiting some features of a vector currently being encoded. A proposed VQ of 16 (=$4{\times}4$) vector dimension and 256 codewords shows a slight performance degradation of about 0.1-0.9 dB, however, with only 16 or 32 among 256 codeword searches, i.e., with just 1/16 or 1/8 search complexity compared to a full-search VQ. And the proposed VQ scheme is also compared to and shown to be a bit superior to tree-search VQ with regard to their SNR performance and memory requirement.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1E
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• pp.35-40
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• 1997

The LZW algorithm has some redundancies for voice traffic. Thus, the V-LZW(Voice LZW) algorithm that decreases the redundancies of the LZW algorithm is suggested for efficent compression of voice. The V-LZW algorithm uses the differential method that reduces the codeword size as well as the length of repeated string. According to the simulation results, it could be seen teat the performance of V-LZW algorithm is better by 35% to 44% in voice compression ratio than that of the conventional modified LZW algorithm. Compared with conventional 32 kbps ADPCM coding with 26 dB SNR, V-LZW has 21 kbps transmission rate, no additional quantization error and simple hardware complexity.