• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.4
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• pp.685-690
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• 1994

A problem associated with vector quantization(VQ) is the computational complexity incurred in searching for a codevector with the closet to a given input vector, where the complexity increases exponentionally with proportion to codebook size and then limits practical application. In this paper, a simple and fast, but efficient, VQ encoding algorithm is presented using a reference codevector as start codevector of premature exit condition, which eliminates distance claculation of unlikely codevectors. The algorithm is to find reference codevector having the possibility to be the nearest vector to input vector first and then to incorporate premature exit condition. The proposed algorithm needs only 10~15% of mathematical operations compared with the conventional full search VQ. Algorithm the number of additions and comparsions of the proposed algorithm is not reduced greatly, the number of multiplication is reduced up to 70~80% compared with other fast VQ encoding methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.1C
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• pp.12-18
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• 2008

This paper This paper proposes a new inverse-quantization(IQ) table interpolation algorithm, specialized Digital Signal Processor(DSP) instructions and hardware architecture for digital audio codecs. Non-linear inverse quantization algorithm is representatively used in both MPEG-1 Layer-3 and MPEG-2/4 Advanced Audio Coding(AAC). The proposed instructions are optimized for the non-linear inverse quantization. The proposed algorithm can minimize operational complexity which reduces total computational load. Performance comparisons show a significant improvement of average error. The proposed instructions and hardware architecture can reduce 20% of the instruction counts and minimize computational loads of IQ algorithms effectively compared with existing IQ table interpolation algorithms. Proposed algorithm can implement commercial DSPs.

• Journal of the Korean Institute of Intelligent Systems
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• v.25 no.2
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• pp.133-138
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• 2015

This paper presents a performance improvement of gray level co-occurrence matrix(GLCM) based on the nonuniform quantization, which is generally used to analyze the texture of images. The nonuniform quantization is given by Lloyd algorithm of recursive technique by minimizing the mean square error. The nonlinear intensity levels by performing nonuniformly the quantization of image have been used to decrease the dimension of GLCM, that is applied to reduce the computation loads as a results of generating the GLCM and calculating the texture parameters by using GLCM. The proposed method has been applied to 30 images of $120{\times}120$ pixels with 256-gray level for analyzing the texture by calculating the 6 parameters, such as angular second moment, contrast, variance, entropy, correlation, inverse difference moment. The experimental results show that the proposed method has a superior computation time and memory to the conventional 256-level GLCM method without performing the quantization. Especially, 16-gray level by using the nonuniform quantization has the superior performance for analyzing textures to another levels of 48, 32, 12, and 8 levels.

• Journal of Korea Society of Industrial Information Systems
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• v.5 no.2
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• pp.16-21
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• 2000

Vector quantization(VQ) is widely used in many high-quality and high-rate data compression applications such as speech coding, audio coding, image coding and video coding. When the size of a VQ codebook is large, the computational complexity for the full codeword search method is a significant problem for many applications. A number of complexity reduction algorithms have been proposed and investigated using such properties of the codebook as the triangle inequality. This paper proposes a new fast VQ search algorithm that is based on a multi-stage structure for searching for the best codeword. Even using only two stages, a significant complexity reduction can be obtained without any loss of quality.

• Journal of Korea Multimedia Society
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• v.5 no.3
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• pp.249-254
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• 2002

We consider a buffer-constrained adaptive quantization algorithm for image compression. Buffer control algorithm was considered with source coding scheme by some researchers and recently a formal description of the algorithm in terms of rate-distortion has been developed. We propose a buffer control algorithm that incorporates the buffer occupancy into the Lagrange multiplier form in a rate-distortion cost measure. Although the proposed algorithm provides the suboptimal performance as opposed to the optimal Vieterbi algorithm, it can be implemented with very low computaional complexity. In addition stability of this buffer control algorithm has been mentioned briefly using Liapnov stability theory.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.4
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• pp.387-397
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• 1992

The spatial domain design of 2-dimensional separable denominator digital filters(SDDF) based on the reduced dimensional decomposition can be realized when the given 2-D impulse response specifications are decomposed into a pair of 1-D specifications via singular value decompositions(SVD). Because of use of the balaned approximation and equivalent transform as 1-D design algorithm, 2-D design algorithm retains the advantage that is numerically stable and can minimize quantization errors. In this paper in order to analyze and reduce these errors, minimum comfficient quantization realization is directly derived from impulse response specification. And using the equivalent trans form relation between mininum coefficient quantization error and minimum roundoff error realizations, we optimally realize a SDDF. This algorithm is analyzed by the simulation, which shows that it is superior to direct or balanced realization in quantization errors.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.54-57
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• 1991

본 논무에서는 HMM과 VQ를 이용한 고립단어에 대한 화자종속 및 화자독립 음성인식시스템을 만들고 여기에 화자적응을 하는 방법에 대한 연구를 했다. 화자적응방법에는 크게 VQ코드북을 적응시키는 방법과 HMM패러미터블 적응시키는 방법이 있다. 코드북적응을 하는 방법으로서 기존코드북에 대해 새로운화자의 적응음성을 양자화한 뒤 각 코드벡터에 해당하는 적응음성의 평균을 구해서 새로운 화자의 코드북을 구해주는 방법과 기준코드북에 대해 새로운화자의 적응음성을 양자화할 때 HMM의 각 상태에서 각각의 코드벡터를 발생할 확률을 거리오차의 계산에서 고려해 비록 거리오차는 크지만 그 코드벡터를 발생할 확률이 매우 높으면 적응음성이 그 코드벡터에 index되게해서 각 코드벡터에 해당하는 모든 적응음성데이타의 평균을 새로운 코드북으로 하는 두가지 알고리즘을 제안한다. 이렇게 함으로써 기존의 기준코드북을 초기 코드북으로해서 LBG알고리즘을 사용해서 적응음성데이타에 대한 새로운 코드북을 만드는 방법에 비해 5-10배의 계산시간을 감소하게 된다. 이 새로운 코드북으로 적응음성데이타를 다시 index해서 이 index된 음성렬로 HMM패러미터를 적응했다. 제안된 알고리즘이 코드북적응을 하는 경우에 기존의 적응방법에 비해 5-10배의 계산 시간을 단축하면서 인식률에서는 더 나은결과를 얻었다. 또 같은 적응방법에 대해서 화자종속모델 보다는 화자독립모델에 대해서 화자적응하는 것이 더 나은 인식결과를 보여주었다.

• The Journal of the Acoustical Society of Korea
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• v.30 no.5
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• pp.265-272
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• 2011

In most of the ISF (or LSF) based real time speech codec, SVQ (split vector quantization) method is used to decrease the quantizer complexity and memory size of codebook. However, it produces drawback that the level of correlation between code vectors can not be used during vector splits. This paper presents a new method of adaptive ISF vector quantization, which compensates the drawbacks of SVQ structured quantizer for wideband speech codec. In each different frame, the proposed method makes use of the correlation between splitted vectors by adaptively changing codebook distribution according to ordering property of ISF. The algorithm is evaluated in AMR-WB, and shows about 1.5 bit per frame improvement.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.579-582
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• 2001

Vector Transformation is a new method unified vector quantization and coding. So far, codebook generation applied to coding was LBG algorithm. But using the advantage of SOFM(Self-Organizing Feature Map) based on neural network can improve a system's performance. In this paper, we generated VTC(Vector Transformation Coding) codebook applied with SOFM algorithm and compare the result for several coding rates with LBG algorithm. The problem of Vector quantization is complicated calculation and codebook generation. So, to solve this problem, we used neural network approach method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.3C
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• pp.250-257
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• 2006

In this paper, we propose an efficient algorithm for the RDO mode decision in H.264/AVC encoder. Based on the properties of DCT coefficients and the RDO mode decision processing, we derive a new condition for detecting an error block having all-zero DCT coefficient (AZCB). (I)DCT, (I)Q, and entropy coding are skipped for AZCBs in the proposed algorithm. It makes the reduction of the computational complexity for the RDO mode decision. Simulation results show that the proposed algorithm achieves computational saving over 40% compared to the conventional method.