• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2007.02a
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• pp.13-17
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• 2007

스케일러블 비디오 코딩(SVC, Scalable Video Coding)은 MPEG(Moving Picture Expert Group)과 VCEG (Video Coding Expert Group)의 JVT(Joint VIdeo Team)에 의해 현재 표준화 되고 있는 새로운 압축 표준 기술이며 시간, 공간 및 화질의 스케일러빌리티를 지원하기 위해 계층 구조를 가지고 있다. 공간적 스케일러빌리티를 위해 기본 계층으로부터 텍스처, 움직임 그리고 잔차신호 정보를 예측하여 사용한다. 그러나 고효율의 압축효과를 얻기 위해 기존의 방식에서는 기본계층에서 얻은 세가지 정보이외에 현재 향상 계층에서 자체적으로 얻은 부호화 정보를 비교하여 최소의 RD(Rate Distortion) 비용을 가지는 정보를 이용하여 부호화 하도록 되어 있다. 하지만 이러한 방식은 향상 계층에서 인터 모드 결정 시 $16\times16,\;16\times8,\;8\times16,\;8\times8,\;4\times4,\;4\times8,\;4\times4$ 블록 모드에 대한 움직임 벡터 예측 및 보상 과정을 거쳐야 하기 때문에 향상 계층에서의 부호화 복잡도는 기본 계층에 비해 상당히 증가하게 된다. 본 논문에서는 기본계층에서 예측한 움직임 벡터 정보를 이용하여 항상 계층에서 모드 결정을 고속화하는 방법에 대해 소개한다. 제안된 방법은 기본 계층에서 예측한 블록모드 중에서 큰 블록인 $16\times16$ 블록에서 움직임 벡터가 (0, 0) 일 경우에 대하여 향상 계층에서는 $16\times16$매크로 블록에 대해서만 움직임 예측 및 보상을 수행함으로써 향상 계층에서 움직임 모드 결정을 조기에 완료하게 된다. 이것은 하위 공간 계층에서 예측한 움직임 벡터 정보가 아주 작을 때는 큰 블록 크기로 모드로 결정되는 일반적인 원리를 이용한 것이고 이 제안 방법을 이용하였을 경우 향상계층에의 모드 결정과정을 고속화함으로써 전체 스케일러빌 비디오 부호하기의 연산량 및 복잡도를 최대 70%까지 감소 시켰다. 그러나 연산량 감소에 따른 비트율의 증가와 화질 열화는 각각 최대 1.32%와 최대 0.11dB로 무시할 수 있을 정도로 작음을 확인 하였다.

• Journal of the Institute of Convergence Signal Processing
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• v.15 no.2
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• pp.48-54
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• 2014

For the element ${\alpha}{\in}GF(p^n)$, two kinds of bases are known. One is a conventional polynomial basis of the form $\{1,{\alpha},{\alpha}^2,{\cdots},{\alpha}^{n-1}\}$, and the other is a normal basis of the form $\{{\alpha},{\alpha}^p,{\alpha}^{p^2},{\cdots},{\alpha}^{p^{n-1}}\}$. In this paper we consider the method of generating normal bases which construct the finite field $GF(p^n)$, as an n-dimensional extension of the finite field GF(p). And we analyze the code sequence generating algorithm and derive the implementation functions of code sequence generator based on the normal bases. We find the normal polynomials of degrees, n=5 and n=7, which can generate normal bases respectively, design, and construct the code sequence generators based on these normal bases. Finally, we produce two code sequence groups(n=5, n=7) by using Simulink, and analyze the characteristics of the autocorrelation function, $R_{i,i}(\tau)$, and crosscorrelation function, $R_{i,j}(\tau)$, $i{\neq}j$ between two different code sequences. Based on these results, we confirm that the analysis of generating algorithms and the design and implementation of the code sequence generators based on normal bases are correct.

• Journal of Korea Multimedia Society
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• v.11 no.10
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• pp.1359-1365
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• 2008

The portion of header in H.264 gets higher than those of previous standards instead of its better compression efficiency. Therefore, this paper proposes a new technique to compress the header of H.264. Unifying a sentence elementary in H.264, H.264 does not consider the distribution of element which be encoded and uses existing Exp-Golomb method, but it is uneffective for variable length coding. Most of the header are block type(s) and motion vector difference(s), and there are redundancies in the header of H.264. The redundancies in the header of H.264 which are analyzed in this paper are three. There are frequently appearing symbols and non-frequently appearing symbols in block types. And when mode 8 is selected in macroblock, all of four sub-macroblock types are transferred. At last, same values come in motion vector difference, especially '0.' This paper proposes the algorithm using type code and quadtree, and with them presents the redundant information of header in H.264. The type code indicates shape of the macroblock and the quadtree does the tree structured motion compensation. Experimental results show that proposed algorithm achieves lower total number of encoded bits over JM12.4 up to 32.51% bit reduction.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.4
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• pp.286-291
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• 2003

In this Paper, we designed an LSP(Line Spectral Pairs) parameter quantizer with cascaded structure of vector quantizer and scalar quantizer for the wideband speech coder. We have chosen the 16th-order of the LP coefficients. These coefficients are then transformed into the LSP parameters which have the excellent properties for quantization and easy stability checking condition of synthesis filter. In the first stage of quantization, input LSP parameters are split-vector-quantized using two 8-th order codebooks. In the second stage, the components of residual vector are individually quantized by the scalar quantizer utilizing the ordering property of LSP parameters. The designed adaptive VQ-SQ quantizer using 35 bits/frame shows the wideband transparency that the average spectral distortion should be less than 1.6 ㏈ and less than 4% of the frames should have SD above 3 ㏈. The simulation results show that the designed quantizer provides a 2-3 bits/frame saving over the typical vector-scalar quantizer.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.1
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• pp.124-130
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• 2013

In this paper, a new motion estimation algorithm with low-computational complexity is proposed to improve the performance of H.264/AVC. The proposed architecture uses the directions of the median motion vector which is computed by the motion vectors of the three neighbor macroblocks in Integer Motion Estimation. By using the directions of the vector, the proposed architecture has a single computational level instead of multi-computational levels in Integer Motion Estimation. The proposed motion estimation is synthesized using the TSMC 0.18um standard cell library. The synthesis result shows that the gate count is about 217.92K at 166MHz and it was improved about 69% compared with previous one.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.6
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• pp.127-136
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• 1998

This paper proposes a new MVQMultistage Vector Quantizer) using a hybrid structure. While in a conventional MVQ, the quantizers of all stages perform the encoding procedure for input signals, we introduce a quantizer that performs selectively. The proposed quantizer with a hybrid structure is composed of a FSVQ(Finite-State Vector Quantizer) for the first stage and a ordinary VQ(Vector Quantizer) for the second stage. A input block is firstly encoded by the FSVQ of the first stage. If the Euclidean distortion measure between original signals and the codevector selected from the state codebook of the FSVQ is less than a prespecified value, only the FSVQ is used for image coding. Otherwise, both the FSVQ of the first stage and the ordinary VQ of the second stage are used for image coding. While the conventional MVQ has an advantage that can achieve low encoding complexity in comparison to the ordinary VQ, but has a disadvantage that is suboptimal with respect to the performance measure and can not achieve the bit rate reduction, the proposed method achieve not only the bit rate reduction but also the performance improvement.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.9 no.2
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• pp.131-140
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• 1998

MAI(Multi-Access Interference) and fast channel variation due to the fading environment are the major problems in the mobile CDMA communication systems. Recently, interest has been increasing in applying the Adaptive Linear MMSE Detector to MAI cancellation in the CDMA reverse link. In this paper, we propose a modified Adaptive Linear MMSE Detector structure which can be used in Long-duration code CDMA system in the presence of independent Rayleigh fading. We use independent multiple tap-weight vector structure to cope with the variation of spreading sequence pattern between neighbor symbols because of the Long-duration code. In this case, more exact channel parameter estimation is required. To solve this problem, we use coherent CDMA structure which can track the channel parameters like amplitude and phase by employing the low power pilot channel in the CDMA reverse link.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.48 no.5
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• pp.88-98
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• 2011

Recently, according as the importance of GIS(geography information system) database security is embossed, much researches had been achieved about GIS network security. But most such researches are weak against sourceful illegal reproductions and distributions of GIS vector data map. In this paper, we proposed an efficient layer unit contents based partial encryption technique in the vector map compression domain to prevent illegal distributions and unauthorized accesses. This method achieves a partial encryption about each central coordinate and directional parameters of a MCA(minimum coding attribute) that is created at the vector map compression processing in the vector space. First, the position encryption is applied as permutating randomly the center coordinate of each record that is minimum unit of vector map shape. And second, the direction encryption that changing shapes of vector map topography is applied as encrypting the direction of vertices's coordinates of each record. In experimental results, we confirmed that our proposed method can encipher the large volumed vector map data effectively in low computational complexity. Also, we could minimize the decline of compression efficiency that occurred by conventional contents based encryption schemes using AES or DES algorithms.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.5
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• pp.1187-1193
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• 2014

This paper presents an effective motion estimation to improve the performance of the motion compensated temporal filtering (MCTF) which is a core part of the wavelet-based scalable video coding. The proposed scheme makes the motion vector field uniform by the modified median operation and the search strategies using adjacent motion vectors, in order to enhance the pixel connectivity which is significantly relevant to the performance of the MCTF. Moreover, the motion estimation with variable block sizes that reflects the features of frames is introduced for further correlation improvement of the motion vector field. Experimental results illustrate that the proposed method reduces the decomposed energy on the temporal high frequency subband frame up to 30.33% in terms of variance compared to the case of the full search with fixed block sizes.

• Journal of Broadcast Engineering
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• v.23 no.1
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• pp.86-92
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• 2018

In this paper, we propose a method for coding history detection of digital speech signal. In digital speech communication and storage, the signal is encoded to reduce the number of bits. Therefore, when a speech signal waveform is given, we need to detect its coding history so that we can determine whether the signal is an original or an coded one, and if coded, determine the number of times of coding. In this paper, we propose a coding history detection method for 12.2kbps AMR codec in terms of original, single coding, and double coding. The proposed method extracts a speech-specific feature vector from the given speech, and models the feature vector using a deep neural network. We confirm that the proposed feature vector provides better performance in coding history detection than the feature vector computed from the general spectrogram.