• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.5
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• pp.112-120
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• 2013

In this paper, we propose high-performance H.264/AVC CAVLC encoder for UD video real time processing. Statistical values are obtained in one cycle through the parallel arithmetic and logical operations, using non-zero bit stream which represents zero coefficient or non-zero coefficient. To encode codeword per one cycle, we remove recursive operation in level encoding through parallel comparison for coefficient and escape value. In oder to implement high-speed circuit, proposed CAVLC encoder is designed in two-stage {statical scan, codeword encoding} pipeline. Reducing the encoding table, the arithmetic unit is used to encode non-coefficient and to calculate the codeword. The proposed architecture was simulated in 0.13um standard cell library. The gate count is 33.4Kgates. The architecture can support Ultra Definition Video ($3840{\times}2160$) at 100 frames per second by running at 100MHz.

• Journal of IKEEE
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• v.17 no.3
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• pp.352-359
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• 2013

This paper proposes an entropy coding unit for H.264/AVC baseline profile. Entropy coding requires code tables for macroblock encoding. There are patterns in codewords of each code tables. In this paper, the patterns between codewords are analyzed to reduce the hardware cost. The entropy coding unit consists of Exp-Golomb unit and CAVLC unit. The Exp-Golomb unit can process five code types in a single unit. It can perform Exp-Golomb processing using only two adders. While typical CAVLC units use various code tables which require large amounts of resources, the sizes of the tables are reduced to about 40% or less of typical CAVLC units using relationships between table elements in the proposed CAVLC unit. After the Exp-Golomb unit and the CAVLC unit generate code values, the entropy unit uses a small size shifter for bit-stream generation while typical methods are barrel shifters.

• The Journal of the Acoustical Society of Korea
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• v.15 no.5
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• pp.107-112
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• 1996

In this paper we propose a new fast codebook training algorithm for reducing the searching time of LBG algorithm. For each training data, the proposed algorithm stores the indexes of codewords that are close to that training data in the first iteration. It reduces computation time by searching only those codewords, the indexes of which are stored for each training data. Compared to one of the previous fast training algorithm, FSLBG, it obtains a better codebook with less exccution time. In our experiment, the performance of the codebook generated by the proposed algorithm in terms of peak signal-to-noise ratio(TSNR) is very close to that of LBG algorithm. However, the codewords to be searched for each training data of the proposed algorithm is only about 6%, for a codebook size of 256 and 1.6%, for a codebook size of 1.24, of LBG algorithm.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.95-100
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• 1997

In this paper, we present a new fast VQ encoding algorithm. The proposed algorithm facilitates two characteristics of a vector, i.e., mean and variance to reject many unlikely codewords and save a lot of computation time. Since the proposed algorithm, which is based upon geometric considerations, rejects those codewords that are impossible to be the closest codeword, it provides the same results as a conventional exhaustive(or full) search algorithm. The simulation results confirm the effectiveness of the proposed algorithm.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.9
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• pp.13-18
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• 2015

The mutilevel holographic data storage offers considerable advantage for capacity, since it can store more than one bit per pixel. In this paper, we search the number of codewords for each code depending on three conditions: (1) the number of levels, (2) the number of pixels in a codeword, and (3) the minimum Euclidean distance of a code. Increasing the number of levels per pixel creates more capacity, while causing more errors, by reducing the noise margin. Increasing the number of pixels in a codeword can increase the code rate, which means more capacity, but increases the complexity of the encoder/decoder of the code. Increasing the minimum distance of a code reduces the detection error, while reducing the code rate of the code. In such a fashion, a system design will always have pros and cons, but our task is to find out an effective one under the given conditions for the system requirements. Therefore, the numbers we searched can provide some guidelines for effective code design.

• The Transactions of the Korea Information Processing Society
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• v.7 no.10
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• pp.3272-3279
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• 2000

In this paper, we propose two fast VQ coclebook search algorithms efficient to the Wavelet/ VQ coding schemes. It is well known that the probability having large values in wavelet coefficient blocks is very low. In order to apply this property to codebook search, the index tables of the reordered codebook in each wavelet subband ae used. The exil condition in PDE can be satisfied in an earlystage by comparing the large coefficients of the codeword with their corresponding elements of input vector using the index tbles. As a result, search time can be reduced.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.209-214
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• 2003

Multi-mode coding method is a reliable DC-suppression method. There are two ways to improve the DC-suppression performance. One is improving scrambler's performance, and the other is improving selection criteria. The latter uses the MRDS(minimum running digital sum) criterion. It is easy to calculate, but its performance goes down when the length of codeword is getting longer. The MSW(mean squared weight) criterion that is known as the best so far regardless of the length of codeword has the high complexity. In this paper, we present the new selection criteria, MPRDS(minimum peak RDS) and A BSRDS(absolute RDS). Their performance are close to the MSW, implementation is simple. And also we present the SC(sign change) that has a subsidiary role with the original selection criteria and improve the capacity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1A
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• pp.13-17
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• 2002

For elimination DC component, the optical recording systems generally exploit WRDS (word-end running digital sum) criterion. However, its performance is degraded if the lengths of the control bits and codewords are increased. Another criterion is MSW (mean-square weight) method. Whereas MSW criterion has the complexity, this hs optimal performance. In this paper, we have proposed a new dc-component suppression method (MPRDS, minimum peak RDS) of the modulation codes for high density optical recording system. This method requires less complexity than MSW's, and the performance is near to MSW's even if the lengths of the control bits and codewords are increased.

• The Journal of the Acoustical Society of Korea
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• v.13 no.3
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• pp.63-70
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• 1994

In this Paper, we propose a new codeword tying algorithm based on a tree structured classfier. The proposed algorithm which can be viewed as a kind of soft decision using statistical properties between codewords and states has an advantage of fast construction, and guarantees a unique optimal solution. Also, it can easily be applied to any speech recognition system based on discrete hidden Markov model (HMM). Experimental results on speaker-independent isolated word recognition show error reduction of $6\%$ for the codebook of size 256 and $9\%$ for 512 size and also HMM parameter reduction of about $20\%$.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.293-298
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• 2003

In this paper. we propose a fast search algorithm for nearest neighbor vector quantization (NNVQ). The proposed algorithm rejects those codewords which can not be the nearest codeword and reduces the search range of codebook. Hence it reduces computational time and complexity in encoding process, while it provides the same SD performance as the conventional full search algorithm. We apply the proposed algorithm to the adaptive multi-rate (AMR) speech coder and a general vector quantizer designed by LBG. algorithm. Simulation results show effectiveness of the proposed algorithm.