• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.7
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• pp.1203-1212
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• 1994

This paper proposes a variable block size vector quantization based on a biorthogonal wavelet transform for image compression. An image is first decomposed with the biorthogonal wavelet transform into multiresolution image and the wavelet coefficients of the middle frequency bands are segmented using the quadtree sturcture to extract the perceptually important regions in the middle frequency bands. A sedges of middle frequency bands exist the corresponding position of high frequency bands, the complicated quadtree structure of middle frequency bands is equally applied to the high frequency bands. Therefore the overhaed information of the quadtree codes needed to segment the high frequency bands can be reduced. The segmented subblocks are encoded with the codebook designed at the each scales and directions. The simulation results showed that the proposed methods could reproduce higher quality image with bit rate reduced about 20(%) than of the preceding VQ method and sufficiently reduce the bolck effect and the edge degradation.

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.190-196
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• 2011

Bandwidth extension is a technique to improve speech quality and intelligibility, extending from 300-3400 Hz narrowband speech to 50-7000 Hz wideband speech. This paper designs an artificial bandwidth extension (ABE) module embedded in the AMR (adaptive multi-rate) decoder, reducing LPC/LSP analysis and algorithm delay of the ABE module. We also introduce a fast search codebook mapping method for ABE, and design a low power BWE technique based on the AMR decoder. The proposed ABE method reduces the computational complexity and the algorithm delay, respectively, by 28 % and 20 msec, compared to the traditional DTE (decode then extend) method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal.

• The KIPS Transactions:PartB
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• v.16B no.4
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• pp.281-288
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• 2009

This paper proposes a novel algorithm for the efficient classification and annotation of medical images, especially X-ray images. Since X-ray images have a bright foreground against a dark background, we need to extract the different visual descriptors compare with general nature images. In this paper, a Color Structure Descriptor (CSD) based on Harris Corner Detector is only extracted from salient points, and an Edge Histogram Descriptor (EHD) used for a textual feature of image. These two feature vectors are then applied to a multi-class Support Vector Machine (SVM), respectively, to classify images into one of 20 categories. Finally, an image has the Annotation Code Array based on the pre-defined hierarchical relations of categories and priority code order, which is given the several optimal keywords by the Annotation Code Array. Our experiments show that our annotation results have better annotation performance when compared to other method.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.696-708
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• 2002

In this paper, a novel transcoding algorithm for the Adaptive Multi Rate(AMR) and the Enhanced Variable Rate Codec(EVRC) vocoders via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding processes. The proposed algorithm consists of the parameter decoding, frame classification, mode decision, and transcoders for two frame types. The transcoders convert the parameters such as LSP, frame energy, pitch delay for the adaptive codebook, fixed codebook vector, and codebook gains. Evaluation results show that while exhibiting better computational and delay characteristics, the proposed algorithm produces equivalent speech quality to that produced by the tandem transcoding algorithm.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.49 no.2
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• pp.9-16
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• 2012

In this paper, the beamforming and user scheduling scheme for device-to-device (D2D) underlaying celullar networks with partial channel state information is investigated. In the proposed scheme, cellular users whose channels are nearly orthogonal to the channel between base station and D2D receiver are selected for unitary beamforming so as to minimize the interference to the D2D receiver. The interference caused by underlaying D2D communication can be minimized by reducing the D2D transmitter's transmission power. To enhance the cellular users' throughput further, we also investigate codebook generation method in which each code vector belongs to multiple unitary code matrices. From simulation results, we show that the proposed beamforming and user scheduling scheme mitigates D2D interference to the cellular networks and the codebook generation method improves cellular users' throughput.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.37 no.3
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• pp.1-10
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• 2000

This paper proposes a novel video coding system with rate control based on fractal algorithm To overcome the demerits of excessive amounts of coded bit generated by previous fractal coding methodology. the proposed system classifies the Image into three classes such as background, motion compensation, and fractal coding area. The motion vector for motion compensation, and the fractal offset value that is difference value between the predicted offset and the least-square approximated value are coded with variable length code The decision method which determines threshold value of partitioning quadtree is applied to the bit-rate control algorithm considering the quantity of currently generated bits and fixed channel bandwidth Experimental result shows that the proposed system enhances compression ratio 1.8 times higher than previous method for the same image quality, and performs efficient rate control for fixed channel bandwidth.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.4A
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• pp.259-270
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• 2005

This paper proposes a mathematical model to analyze call-and packet-level performance of the TD-CDMA/TDD system which could serve a flexible radio resource management against multi-type heterogeneous and asymmetrical traffic conditions. On call-level analysis, the mathematical model based on queueing theory performs multi-dimensional operations using random vectors or matrices to consider multiple types of traffic and also deal with asymmetrical up- and down-direction transmissions separately. Employing the mathematical model, we obtain rail blocking probability for each type of traffic and also the optimum switching-point with the smallest call flocking probability. And on packet-level analysis, employing a non-prioritized queueing scheme between circuit and packet calls, we solve 2-dimensional random vector problem composed of the queue length for packets and the number of circuit calls being served. Finally, packet-level performance is analyzed in terms of the packet loss probability and the buffer size required under mixed-traffic conditions of multiple types of circuit and packet calls.

• Journal of Digital Convergence
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• v.12 no.3
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• pp.269-275
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• 2014

The DAB system was designed to provide CD quality audio and data services for fixed, portable and mobile applications with the required BER below $10^{-4}$. However for the T-DMB system with the video service of MPEG-4 stream, BER should go down $10^{-8}$ by adding FEC blocks which consist of the Reed-Solomon (RS) encoder/decoder and convolutional interleaver/deinterleaver. In this paper we propose two types of turbo coded T-DMB system without altering the puncturing procedure and puncturing vectors defined in the standard T-DMB system for compatibility. One(Type 1) can replace the existing RS code, convolutional interleaver and RCPC code by a turbo code and the other one (Type 2) can substitute the existing RCPC code by a turbo code. Simulation results show that two new turbo coded systems are able to yield considerable performance gain after just 2 iterations. Type 2 system is better than type 1 but the amount of performance improvement is small.

• KIPS Transactions on Software and Data Engineering
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• v.3 no.2
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• pp.65-72
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• 2014

Change coupling is a measure to show how strongly change-related two entities are. When two source files have been frequently changed together, they are regarded as change-coupled files and they will probably be changed together in the near future. In the previous studies, the change coupling between two files is defined with the number of common changed time, that is, common commit time of the files. However, the frequency-based technique has limitations because of 'tangled changes', which frequently happens in the development environments with version control systems. The tangled change means that several code hunks have been changed at the same time, though they have no relation with each other. In this paper, the change types of the code hunks are also used to define change coupling, in addition to the common commit time of target files. First, the frequency vector based on change types are defined with the extracted change types, and then, the similarity of change patterns are calculated using the cosine similarity measure. We conducted experiments on open source project Eclipse JDT and CDT for case studies. The result shows that the applicability of the proposed method, compared to the previous studies.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.791-794
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• 2000

본 논문에서는 TMS320C549 를 사용하여 4 채널 G.729A 음성 부호화기를 실시간 구현하였으며, 실제로 음성 사서함 서비스 시스템에 응용하였다. 구현된 G.729A 는 패널 당 부호화기와 복호화기에 각각 14.5MIPS 와 3.6 MIPS 를 소요하였으며, 메모리는 코드와 데이터 부분에 각각 9.88K 워드, 1.69 K 워드를 필요로 하였다. 결과적으로 개발된 VMS 시스템에는 두 개의 DSP 를 사용하여 DSP 당 4 채널씩 총 8 채널을 수용하였다. 실험 결과, ITU-T에서 제공된 모든 테스트 벡터 결과와 비트 단위로 동일하였다.