• Journal of Broadcast Engineering
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• v.4 no.2
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• pp.127-133
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• 1999

The conventional fractal image compression based on discrete wavelet transform uses the fixed block size in fractal coding and reduces PSNR at low bit rate. This paper proposes a fractal image coding based on discrete wavelet transform which improves PSNR by using variable block size in fractal coding. In the proposed method. the absolute values of discrete wavelet transform coefficients are computed. and the discrete wavelet transform coefficients of different highpass subbands corresponding to the same spatial block are assembled. and the fractal code for the range block of each range block level is assigned. and then a decision tree C. the set of choices among fractal coding. "0" encoding. and scalar quantization is generated and a set of scalar quantizers q is chosen. And then the wavelet coefficients. fractal codes. and the choice items in the decision tree are entropy coded by using an adaptive arithmetic coder. This proposed method improved PSNR at low bit rate and could achieve a blockless reconstructed image. As the results of experiment. the proposed method obtained better PSNR and higher compression ratio than the conventional fractal coding method and wavelet transform coding.rm coding.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.7 s.361
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• pp.45-53
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• 2007

In this paper, we propose an efficient hardware architecture for H.264/AVC CAVLC (Context-based Adaptive Variable Length Coding) encoding. Previous CAVLC architectures search all of the coefficients to find statistic characteristics in a block. However, it is unnecessary information that zero coefficients following the last position of a non-zero coefficient when CAVLC encodes residual coefficients. In order to reduce this unnecessary operation, we propose two techniques, which detect the first and last position of non-zero coefficients and arrange non-zero coefficients sequentially. By adopting these two techniques, the required processing time was reduced about 23% compared with previous architecture. It was designed in a hardware description language and total logic gate count is 16.3k using 0.18um standard cell library Simulation results show that our design is capable of real-time processing for $1920{\times}1088\;30fps$ videos at 81MHz.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.2
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• pp.56-68
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• 2002

The Universal Variable Length Coding(UVLC) scheme in H.26L has nice features such as error resiliency and two-way decodability. However, it has lower coding efficiency than the conventional Huffman coding. To improve the coding efficiency of UVLC, we Propose to use a dynamic codeword mapping that changes association between symbols and codewords in order to utilize the statistical characteristics of symbols as much as possible but without losing any features of the UVLC. Both encoder and decoder use the same re-association table, and hence the encoder need not send additional overhead for the re-mapping relationship to the decoder. Simulation results show that without significant change of the current H.26L coding scheme, the proposed method additionally attains up to about 8% and about 5% bit reductions respectively in intra and inter frames over the current H.26L encoding method.

• KIPS Transactions on Computer and Communication Systems
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• v.1 no.3
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• pp.187-192
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• 2012

This paper describes efficient H.264/AVC video scrambling methods for digital rights management. The proposed scrambling methods are to scramble level and suffix in entropy encoding and MVD in motion estimation of the H.264 video compression process. Other scrambling methods have been proposed but they degrade the compression efficiency or make it difficult to achieve real-time processing due to the large amount of computational efforts. Since the proposed scrambling methods resolve the drawbacks of other approaches, they do not cause image distortion and the original compression efficiency is maintained. We verified our scrambling methods and evaluated the performance by conducting several experiments with H.264 reference program. Finally, we implemented video player system using USB dongle in order to apply the proposed scrambling/descrambling methods to H.264 video compression.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.2
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• pp.439-446
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• 2011

In this paper, we proposed and implemented a dedicated hardware for Huffman coding which is a method of entropy coding to use compressing multimedia data with video coding. The proposed Huffman codec consists Huffman encoder and decoder. The Huffman encoder converts symbols to Huffman codes using look-up table. The Huffman code which has a variable length is packetized to a data format with 32 bits in data packeting block and then sequentially output in unit of a frame. The Huffman decoder converts serial bitstream to original symbols without buffering using FSM(finite state machine) which has a tree structure. The proposed hardware has a flexible operational property to program encoding and decoding hardware, so it can operate various Huffman coding. The implemented hardware was implemented in Cyclone III FPGA of Altera Inc., and it uses 3725 LUTs in the operational frequency of 365MHz

• Journal of Broadcast Engineering
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• v.28 no.1
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• pp.100-108
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• 2023

In hybrid block-based video coding, transform coding converts spatial domain residual signals into frequency domain data and concentrates energy in a low frequency band to achieve a high compression efficiency in entropy coding. The state-of-the-art video coding standard, VVC(Versatile Video Coding), uses DCT-2(Discrete Cosine Transform type 2), DST-7(Discrete Sine Transform type 7), and DCT-8(Discrete Cosine Transform type 8) for primary transform. In this paper, considering that DCT-2, DST-7, and DCT-8 are all linear transformations, we propose an inverse transform that reduces the number of multiplications in the inverse transform by using the linearity of the linear transform. The proposed inverse transform method reduced encoding time and decoding time by an average 26%, 15% in AI and 4%, 10% in RA without the increase of bitrate compared to VTM-8.2.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.2
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• pp.33-41
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• 2001

Currently, the compression is one of the most important technology in multimedia society. Audio files arc rapidly propagated throughout internet Among them, the most famous one is MP-3(MPEC-1 Laver3) which can obtain CD tone from 128Kbps, but tone quality is abruptly down below 64Kbps. MPEC-II AAC(Advanccd Audio Coding) is not compatible with MPEG 1, but it has high compression of 1.4 times than MP 3, has max. 7.1 and 96KHz sampling rate. In this paper, we propose an algorithm that decreased the capacity of AAC encoding computation but increased the processing speed by optimizing psychoacoustic model which has enormous amount of computation in MPEG 2 AAC encoder. The optimized psychoacoustic model algorithm was implemented by C++ language. The experiment shows that the psychoacoustic model carries out FFT(Fast Fourier Transform) computation of 3048 point with 44.1 KHz sampling rate for SMR(Signal to Masking Ratio), and each entropy value is inputted to the subband filters for the control of encoder block. The proposed psychoacoustic model is operated with high speed because of optimization of unpredictable value. Also, when we transform unpredictable value into a tonality index, the speed of operation process is increased by a tonality index optimized in high frequency range.