• Journal of Digital Convergence
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• v.18 no.7
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• pp.369-374
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• 2020

A variety of sensory elements are used in video advertisements promoting products. Video advertising using visual and auditory elements is a representative means of marketing. The advertising video that promotes the product by using such a sensory element is imprinted on viewers by continuously or repeatedly exposing visual elements such as a logo or a specific image or phrase. Such visual images are used as an effective way to symbolize brand image effectively. If the visual elements were symbolized in the advertising images of most car brands, Kia's K5's advertising images symbolized auditory elements, or sounds, to produce K5's unique advertising images. In this paper, we compared Kia's K5 advertisement image symbolizing auditory elements with other brands' advertisement image of other companies, and studied the techniques and effects used in advertisement image production.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.10C
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• pp.951-962
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• 2002

A DMT transceiver recovers the sampling time from reserved sub-carriers, the pilots. Since the pilots are available after the FFT, the symbol synchronization must be done before sample synchronization. In DMT VDSL system, symbol synchronization is handled separately from sample synchronization, although the two processes are intimately related. The DMT symbol itself contains sufficient information, the cyclic extension, for symbol synchronization. Using only the sign bit of received signal, the Maximum Likelihood Estimation solution is derived. The Tx windowing in the transmitter of DMT VDSL system results in the blurring of MLE peaks. We propose the weighted summing MLE method using the sign bit which produces the clearly sharp top of MLE peaks. The stability of symbol synchronization is improved significantly by averaging over a few symbols. This paper presents the study of the original MLE and the weighted summing MLE using sign bit. A clock difference between transmitter and receiver destroys the oahogonality of the carriers. Therefore, a receiver using asynchronous sampling must perform timing correction in the discrete-time domain. We introduce an efficient digital sample synchronization method which is based on temporal and frequency domain digital signal processing.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2003.11a
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• pp.61-64
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• 2003

무손실 이미지 압축은 (Lossless Image Compression)은 손실이미지 압축(Lossy Image Compression)에 비해, 압축률(compression ratio)은 떨어지지만, 반면 원이미지와 복원이미지가 완전히 일치하므로, 원인이미지의 품질을 그대로 유지학 수 있다. 따라서, 이미지의 품질(Quality)과 압축효율(compression ratio)은 서로 상반된 관계에 있으며, 지금도 좀 더 놀은 압축효과를 얻으려는 여러 무손실 압축 방법이 발표되고 있다. 무손실 이미지 압축은 이미지의 정확성과 정밀성이 요구되는, 의료영양분야에서 가장 널리 쓰이고 있으며, 그밖에, 원본이미지를 기본으로 다른 이미지프로세싱이 필요한 경우, 압축 복원을 반복적으로 수행할 필요가 있을 때, 기타 사진 예술분야, 원격 영상 등 정밀성이 요구되는 분양에서 쓰이고 있다. [7]. 무손실 이미지 압축의 가장 대표적인 CALIC[3]과 JPEG_LS[2]를 들 수 있다. CALIC은 비교적 높은 압축률을 나타내지만, 3-PASS의 과정을 거치는 복잡도가 지적되고 있다. 반면 JPEG-LS는 압축률은 CALIC에 미치지 못하지만 빠른 코딩/디코딩 속도를 보인다. 본 논문에서는 여거 가지의 예측 모드를 두어, 블록단위별로 주변 CONTEXT에 따라, 최상의 예측 모드를 판단하여, 이를 적용, 픽셀의 여러 값을 최소화하였다. 그 후 적응산술 부호기(Adaptive arithmetc coder)를 이용하여, 인코딩을 하였다. 이때 최대 에러값은 64를 넘지 않게 했으며, 또한 8*8블록별로 에러의 최대값을 측정하여 그 값을 $0\~7$까지의 8개의 대표값으로 양자화하는 방법을 통하여 그에 따라 8개의 보호화 심볼 모델중 알맞은 모델에 적용하였다. 이를 통해, 그 소화값의 확률 구간을 대폭 넓힘으로써, 에러 이미지가 가지고 있는 엔트로피에 좀 근접하게 코딩을 할 수 있게 되었다. 이 방법은 실제로 Arithmetic Coder를 이용하는 다른 압축 방법에 그리고 적용할 수 있다. 실험 결과 압축효율은 JPEG-LS보다 약 $5\%$의 압축 성능 개선이 있었으며, CALIC과는 대등한 압축률을 보이며, 부호화/복호화 속도는 CALIC보다 우수한 것으로 나타났다.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.13 no.5
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• pp.344-352
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• 2020

Covertness and performance are very important design goals in the underwater communications. To satisfy both of them, we proposed efficient underwater communication model which combined multiband and direct sequence spread spectrum method in order to improve performance and covertness simultaneously. Turbo coding method with 1/3 coding rates is used for channel coding algorithm, and turbo equalization method which iterately exchange probabilistic information between equalizer and decoder is used for receiver side. After optimal threshold value was set in Rake processing, this paper analyzed the performance by varying the number of chips were 8, 16, 32 and the number of bands were from 1 to 4. Through the simulation results, we confirmed that the performance improvement was obtained by increasing the number of bands and chips. 2~3 dB of performance gain was obtained when the number of chips were increased in same number of bands.

• The Journal of the Acoustical Society of Korea
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• v.40 no.4
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• pp.370-381
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• 2021

This paper presented an efficient transceiver structure of multiband Frequency Shift Keying (FSK) signals with direct sequence spread spectrum for maintaining covertness and performance. In aspect to covertness, direct sequence spread spectrum method, which multiplying by Pseudo Noise (PN) codes whose rate is much higher than that of data sequence, is employed. In aspect to performance, in order to overcome performance degradation caused by multipath and Doppler spreading, we applied multiband, turbo equalization, and weighting algorithm are applied. Based on the simulation results, by applying 4 number of multiband and number of chips are 8 and 32, experiments were conducted in a lake with a distance of moving from 300 m to 500 m between the transceivers. we confirmed that the performance was improved as the number of bands and chips are increased. Furthermore, the performance of multiband was improved when the proposed weighting algorithm was applied.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.11 no.1
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• pp.35-42
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• 2001

The primitive polynomial on GF(q) is used in the area of the scrambler, the error correcting code and decode, the random generator and the cipher, etc. The algorithm that generates efficiently the primitive polynomial on GF(q) was proposed by A.D. Porto. The algorithm is a method that generates the sequence of the primitive polynomial by repeating to find another primitive polynomial with a known primitive polynomial. In this paper, we propose the algorithm that is improved in the A.D. Porto algorithm. The running rime of the A.D. Porto a1gorithm is O($\textrm{km}^2$), the running time of the improved algorithm is 0(m(m+k)). Here, k is gcd(k, $q^m$-1). When we find the primitive polynomial with m odor, it is efficient that we use the improved algorithm in the condition k, m>>1.

• The Journal of the Acoustical Society of Korea
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• v.42 no.3
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• pp.169-180
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• 2023

The performance of underwater acoustic communication is greatly affected by multipath propagation and Doppler spread. This paper proposes a new communication technique, the Triangle Spread Carrier (TSC) technique, by modifying the existing Sweep Spread Carrier (SSC) technique that is strong in a multipath propagation environment. The proposed TSC technique is a form in which the up-chirp and down-chirp signals have repeated carriers, and each correlation function characteristic is used to estimate and correct the Doppler shift frequency of the receiving signal. To demonstrate the performance of the proposed TSC technique, we present the results of simulations using underwater channel simulators and sea trial conducted in the East Sea. When demodulating using only the estimated Doppler shift frequency as a result of the sea trial, the uncoded bit error rate was up to 0.194, but when the proposed method was applied, the uncoded bit error rate was reduced to 0.001.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.51-62
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• 2003

This paper presents a novel survivor memeory management and decoding techniques with sequential backward state transition control in the trace back Viterbi decoder. The Viterbi algorithm is an maximum likelihood decoding scheme to estimate the likelihood of encoder state for channel error detection and correction. This scheme is applied to a broad range of digital communication such as intersymbol interference removing and channel equalization. In order to achieve the area-efficiency VLSI chip design with high throughput in the Viterbi decoder in which recursive operation is implied, more research is required to obtain a simple systematic parallel ACS architecture and surviver memory management. As a method of solution to the problem, this paper addresses a progressive decoding algorithm with sequential backward state transition control in the trace back Viterbi decoder. Compared to the conventional trace back decoding techniques, the required total memory can be greatly reduced in the proposed method. Furthermore, the proposed method can be implemented with a simple pipelined structure with systolic array type architecture. The implementation of the peripheral logic circuit for the control of memory access is not required, and memory access bandwidth can be reduced Therefore, the proposed method has characteristics of high area-efficiency and low power consumption with high throughput. Finally, the examples of decoding results for the received data with channel noise and application result are provided to evaluate the efficiency of the proposed method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7C
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• pp.691-699
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• 2006

Despite of the in-depth effort to cantrol the variability in IP networks, quality of service (QoS) is still not guaranteed in the IP networks. Thus, it is necessary to deal with the audible artifacts caused by packet lasses. To overcame the packet loss problem, most speech coding standard have their own embedded packet loss concealment (PLC) algorithms which adapt extrapolation methods utilizing the dependency on adjacent frames. Since many low bit rate CELP coders use predictive schemes for increasing coding efficiency, however, error propagation occurs even if single packet is lost. In this paper, we propose an efficient PLC algorithm with consideration about the speech characteristics of lost frames. To design an efficient PLC algorithm, we perform several experiments on investigating the error propagation effect of lost frames of a predictive coder. And then, we summarize the impact of packet loss to the speech characteristics and analyze the importance of the encoded parameters depending on each speech classes. From the result of the experiments, we propose a new PLC algorithm that mainly focuses on reducing the error propagation time. Experimental results show that the performance is much higher than conventional extrapolation methods over various frame erasure rate (FER) conditions. Especially the difference is remarkable in high FER condition.

• Journal of radiological science and technology
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• v.38 no.4
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• pp.437-442
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• 2015

Accurate volume measurement of the prostate is a significant role in determining the result of diagnosis and treatment of benign prostate hyperplasia. The purpose of this study was to determine, when measuring prostate volume by TRUS, whether location is more accurately determined by transaxial or longitudinal scanning. With reference to the patient's image, it was produced six prostate model. It compares the actual volume and the measurement volume, and find the optimal measurement position of each specific model. Prostate volume measured by TRUS closely correlates with prostate phantom volume. There was no significant difference(p = .156). To measure the accurate volume of prostate with focal protrusion, its length should be measured exclude the protrusions.