• The Transactions of the Korea Information Processing Society
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• v.7 no.8
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• pp.2341-2348
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• 2000

The special effects on the compressed :vII'EG domain become one of the interesting problems. In this paper, we developed minimal deCilder to effect in DCT compressed domain, proposed the video effects including wipe, dissolve, and zooming. To increase the expandability and portabilitv, the minimal decoder and the effects arc implemented to filters of COM based DirectShow.

• 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.

• Information and Communications Magazine
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• v.14 no.8
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• pp.25-40
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• 1997

본고는 LMDS기술을 이용한 디지틀 무선CATV 전송시스팀 구성에 관한 것으로써 시스팀 구성의 특징과 시스팀 구성요소별 기능, 가입자 접속방법의 특징을 제시하고, 향후 시스팀 진화를 통하여 양방향으로 통신케함으로써 다양한 부가 통신서비스를 26GHz를 사용하는 LMDS로 가입자를 무선으로 접속시키고,효과적으로 정보를 전송할 수 있도록 MPEG-II 기술을 적용하며, 강우감쇄, 주파수 간섭등의 환경조건에서도 안정된 시스팀 가용율을 확보할 수 있는 구조로 되어있다.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.782-792
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• 2011

In this paper, we address a new fast DCT-II/DFT/HWT hybrid transform architecture for digital video and fusion mobile handsets based on Jacket-like sparse matrix decomposition. This fast hybrid architecture is consist of source coding standard as MPEG-4, JPEG 2000 and digital filtering discrete Fourier transform, and has two operations: one is block-wise inverse Jacket matrix (BIJM) for DCT-II, and the other is element-wise inverse Jacket matrix (EIJM) for DFT/HWT. They have similar recursive computational fashion, which mean all of them can be decomposed to Kronecker products of an identity Hadamard matrix and a successively lower order sparse matrix. Based on this trait, we can develop a single chip of fast hybrid algorithm architecture for intelligent mobile handsets.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.10
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• pp.13-26
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• 1993

Motion picture algorithms are realized on the multiprocessor system presented in the Study I. For the most efficient processing of the algorithms, pipelining and geometrical parallel processing methods are employed, and processing time, communication load and efficiency of each algorithm are compared. The performance of the implemented system is compared and analysed with reference to MPEG coding algorithm. Theoretical calculations and experimental results both shows that geometrical partitioning is a more suitable parallel processing algorithm for moving picture coding having the advantage of easy algorithm modification and expansion, and the overall efficiency is higher than pipelining.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2005.11a
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• pp.335-338
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• 2005

In this paper, we propose a downlink power control scheme to apply in the unequal error protection multi-code CDMA mobile medicine system. The mobile medicine system contains (i) blood pressure and body temperature measurement value, (ii) ECG medical signals measured by the electrocardiogram device, (iii) mobile patient's history, (iv) G.729 audio signal, MPEG-4 CCD sensor video signal, and JPEG2000 medical image. By the help of the multi-code CDMA spread spectrum communication system with downlink power control scheme and unequal error protection strategy, it is possible to transmit mobile medicine media and meet the quality of service. Numerical analysis and simulation results show that the system is a well transmission platform in mobile medicine.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2471-2474
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• 2003

The modified discrete cosine transform (MDCT) and its inverse transform (IMDCT) are employed in subband/transform coding schemes as the analysis/synthesis filter bank based on time domain aliasing cancellation (TDAC). And they are the most computational intensive operations in layer III of the MPEG audio coding standard. In this paper, we propose a new efficient algorithm for the MDCT/IMDCT computation. It is based on the MDCT/IMDCT computation algorithm using the discrete cosine transforms (DCTs), and it employs two discrete cosine transform of type II(DCT-II) to compute the MDCT/IMDCT. In addition to, it takes advantage of ability in calculating the MDCT/IMDCT computation, where the length of a data block is divisible by 4. The proposed algorithm in this paper requires less calculation complexity than the existing methods. Also, it can be implemented by the parallel structure,, and its structure is particularly suitable for VLSI realization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.770-776
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• 2004

The goal of perceptual audio coder is to reduce redundancy and irrelevancy of audio signal based on the concept of masking. Several studies on masking effect reveal that the masking threshold varies as a function of the noise-like or tone-like nature of audio signals. Therefore, tonality of audio signal influences significantly the quality and efficiency of perceptual audio coder In this paper, we propose a new effective algorithm for tonality measure using spectrum energy. Since the proposed algorithm consists of a few transcendental functions and simple operations, it has lower complexity than MPEG psychoacoustic model-II. The proposed algorithm was tested with some audio signals, and DSP implementation showed that the proposed algorithm could be implemented with 3 MIPS. These results illustrate the efficiency of proposed algorithm in both performance and complexity.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.36C no.1
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• pp.59-65
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• 1999

In this paper, we design a entorpy encoder of HDTV/DTV encoder blocks on the basis of MPEG-II. The designed entropy encoder outputs its bit stream at 9Mbps bit rate inserting zero-stepping block to protect the depletion of buffer in case that the generated bit stream is stored in buffer and uses not only PROM bit combinational circuit to solve the problem of critical path, and packer block, one of submerge, is designed to packing into 24 bit unit using barrel shifter, and it is constructed to blocks of header information encoder, input information delay, submerge, and buffer control. Designed circuits is verified by VHDL function simulation, as a result of performing P&R with Gate compiler that apply $0.8{\mu}m$ Gate Array specification, pin and gate number of total circuits has been tested to each 235 and about 120,000.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11C
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• pp.1491-1497
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• 2004

The psychoacoustic model in MPEG audio layer-III (MP3) encoder is optimized for the fixed-point processing. The optimization process consists of determining the data word length of arithmetic unit and the algorithm for transcendental functions that are often used in the psychoacoustic model. In order to determine the data word length, we defined a statistical model expressing the relation between the fixed-point operation errors of the psychoacoustic model and the probability of alteration of the allocated bits doe to these errors. Based on the simulations using this model, we chose a 24-bit data path and constructed a 24-bit fixed-point MP3 encoder. Sound quality tests using the constructed fixed-point encoder showed a mean degradation of -0.2 on ITU-R 5-point audio impairment scale.