• The KIPS Transactions:PartC
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• v.9C no.5
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• pp.679-686
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• 2002

The transmission of prerecorded and compressed video data without degradation of picture quality requires video servers to cope with large fluctuations in bandwidth requirement. Bandwidth smoothing techniques can reduce the burst of a variable-bit rate stream by prefetching data at a series of fixed rates and simplifying the allocation of resources in the video servers and the network. In this paper, the proposed smoothing algorithm results in the optimal transmission plans for (1) the smallest bandwidth requirements, (2) the minimum number of changes in transmission rate, and (3) the minimum amount of the server process overhead. The advantages of the proposed smoothing algorithm have been verified through the comparison with the existing smoothing algorithms in diverse environments.

• Journal of Digital Contents Society
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• v.8 no.1
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• pp.9-15
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• 2007

Smoothing is a transmission plan where variable rate video data is converted to a constant bit rate stream. Among them are CBA, MCBA, MVBA, e-PCRTT and others. e-PCRTT algorithm, which was improved from PCRTT, restricts the number of rate changes with fixed-size run. This causes unnecessary rate changes when run size is small and buffer size is large. In this paper, to overcome a shortcoming of e-PCRTT algorithm, a smoothing algorithm is proposed, which is improved from e-PCRTT, where a transmission rate transmits more intervalsl as possible. Experiments demonstrated that the proposed algorithm outperformed e-PCRTT algorithm. In order to show the performance, various evaluation factors were used such as the number of transmission rate changes, peak rate, transmission rate variability and so on.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10c
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• pp.331-333
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• 2000

최근 VOD(Video On Demand)나 AOD(Audio On Demand) 서비스가 급증하면서 멀티미디어 스트리밍 기술에 대한 관심이 커지고 있다. 기존의 스트리밍 서비스는 정적으로 멀티미디어를 전송하므로 네트워크 환경의 대역폭(bandwidth) 변화에 대처하는데 한계가 있다. 이러한 문제점을 해결하기 위하여 본 논문에서는 적응형 스트리밍 서비스(ASS; Adaptive Streaming Service) 시스템을 제안한다. ASS 시스템은 가변 비트율을 통한 미디어 스케일링방식을 이용하여 멀티미디어 데이터를 동적으로 전송한다. 또한 프레임 조정방식과 화질 조정방식을 동시에 채택하여 적용함으로써 고화질이 보장되는 전송환경과 느린 회선에서도 끊김없이 안정적인 스트리밍 서비스를 제공받을 수 있도록 한다.

• Journal of Internet of Things and Convergence
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• v.8 no.1
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• pp.53-58
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• 2022

Smoothing is a transmission plan that converts video data stored at a variable bit rate into a constant bit rate. In the study of [6-7], when a data rate increase is required, the frame with the smallest increase is set as the start frame of the next transmission rate section, when a data tate decrease is required. the frame with the largest decrease is set as the start frame of the next transmission rate section, And the smoothing algorithm was proposed and performance was evaluated in an environment where network traffic is not considered. In this paper, the smoothing algorithm of [6-7] evaluates the adaptive CBA algorithm and performance with minimum frame rate, average frame rate, and frame rate variation from 512KB to 32MB with E.T 90 video data in an environment that considers network traffic. As a result of comparison, the smoothing algorithm of [6-7] showed superiority in the comparison of the minimum refresh rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.4B
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• pp.278-287
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• 2012

In this paper, we propose an allocation scheme for variable message packings to increase efficiency of military operation using Link-16 which is well-known for tactical data link by delivering imagery information rapidly. We propose a variable message packing scheme using COC waveform to support variable data rate under some coverage limitation. Variety of message packing makes Link-16 vary transmission rate appropriately for tactical environment. We also propose a allocation scheme to assign message packing to time slot properly. Finally we verify the performance and superiority of proposed ideas by simulations.

• Journal of IKEEE
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• v.2 no.2 s.3
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• pp.233-238
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• 1998

In this paper, the implementation of dual rate ADPCM using G.726 16Kbps and 40Kbps speech codec algorithm is handled. For small signals, the low rate 16Kbps coding algorithm shows almost the same SNR as the high rate 40Kbps coding algorithm , while the high rate 40Kbps coding algorithm shows the higher SNR than the low rate 16Kbps coding algorithm fur large signal. To obtain the good trade-off between the data rate and synthesized speech quality, we applied low rate 16Kbps for the small signal and high rate 40Kbps for the large signal. Various threshold values determining the rate are applied for good trade-off between data rate and speech quality. The simulation result shows the good speech quality at a low rate comparing with 16Kbps & 40Kbps.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.18 no.1
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• pp.129-136
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• 2004

In this paper, we proposed quad variable rates ADPCM coding method for efficient speech transmission and real time porcessing is implemented on TMS320C6711-DSP. The modified ADPCM with four variable coding rates, 16[kbps], 24[kbps], 32[kbps] and 40[kbps] are used for speech window samples for good quality speech transmission at a small data bits and real time encoding and decoding is implemented using DSP. ZCR is used to identify the influence of the noise on the speech signal and to decide the rate change threshold. For noise superior signals, low coding rates are applied to minimize data bit and for noise inferior signals, high coding rates are applied to enhance the speech quality. In most speech telecommunications, silent period takes more than half of the signals, speech quality close to 40[kbps] can be obtained at comparabley low data bits and this is shown by simulation and experiments. TMS320C6711-DSK board has 128K flash memory and performance of 1333MIPS and has meets the requirements for real time implementation of proposed coding algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.6
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• pp.621-628
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• 1992

This paper proposes a progressive Image transmission method with a variable magnification hierarchical structure for image processing system. As Introduced in the literature, the progressive image transmission method, uses a fixed magnification rates of either 4 or 1/4. Thus, a sudden in-crease In resolution Is obtained due to a sudden Increase in information. By adapting a variable magnification hlerarchical structure In this research, a gradual increase in resolution Is possible by slowly inrireasing information between hierarchical levels. The simulation results show that a 5.7dB SNR improvemr'nt Is obtained with an Improved compression rate by 0.7 Ult /pel compare to the LP method. It also gives about 1 dB SNR improvement compare to the PCS method at intermediate hierarchical levels.

• Journal of Korea Multimedia Society
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• v.5 no.3
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• pp.316-322
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• 2002

In this paper we consider continuous block allocation scheme of UNIX file system to support real time retrieval service. The proposed block allocation scheme is designed to place real time data at appropriate disk block location in considering the consume-rate that is given with real time data. To effectively determine the disk block location we analyze the relationship between consume-rate and the two variable factors that are the number of continuous blocks and the cylinder distance of logically consecutive data. In traditional UNIX block allocation scheme it is in fact impossible to find continuous free disk blocks in a specific cylinder location. Thus we propose new bitmap-based free block allocation scheme that enables to determine whether a block in specific cylinder location is free state, or not.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.2
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• pp.159-165
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• 2010

The Multi-code (Mc) modulation has been developed for high-speed data transmission over the wireless environments, but it suffers two critical problems due to the limited resource of Orthogonal Codes (OC) and high Peak-Average Power Ratio (PAPR). In this paper, we propose a novel modulation technique called AOCG [1] (Advanced Orthogonal Code Group)-OFDM (Orthogonal Frequency Division Multiplexing) to solve the above problems and obtain the variable high bit rates which can be controlled by the four parameters depending on the quality of services (QoS) required by users.