• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.5A
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• pp.726-742
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• 2000

A novel rte control algorithm consisting of two major components, i.e. a variable encoding frame rate method and a hybrid DCT/wavelet I-frame coding scheme, is proposed in this work for low bit rate video coding. Most existing rate control algorithms for low bit rate video focus on bit allocation at the macroblock level under a constant frame rate assumption. The proposed rate control algorithm is able to adjust the encoding frame rate at the expense of tolerable time-delay. Furthermore, an R-D optimized hybrid DCT/wavelet scheme is used for effective I-frame coding. The new rate-control algorithm attempts to achieve a good balance between spatial quality and temporal quality to enhance the overall human perceptual quality at low bit rates. It is demonstrated that the rate control algorithm achieves higher coding efficiency at low bit rates with a low additional computational cost. The variable frame rate method and hybrid I-frame coding scheme are compatible with the bi stream structure of H.263+.

• Proceedings of the KSPS conference
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• 2007.05a
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• pp.315-318
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• 2007

This paper proposes the methods to enhance the speech quality of source controlled variable bit-rate coder based on the waveform interpolation. The methods are to estimate and generate the parameters that are not transmitted from encoder to decoder by the repetition and extrapolation schemes. For the performance evaluation, the PESQ(Perceptual Evaluation of Speech Quality) scores are measured. The experimental results shows that our proposed method outperforms the conventional source controlled variable bit-rate coder. Especially, the performance of the extrapolation method is better than that of the repetition method.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.142-147
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• 1998

Recently, 8 kb/s CS-ACELP coder of G.729 is atandardized by ITU-T SG15 and it has been reported that the speech quality of G729 is better than or equal to that of 32kb/s ADPCM. However G.729 is the fixed rate speech coder, and it does not consider the property of voice activity in mutual conversation. If we use the voice activity, we can reduce the average bit rate in half without any degradations of the speech quality. In this paper, we propose an efficient variable rate algorithm for G.729. The variable rate algorithm consists of two main subjects, the rate determination algorithm and algorithm, we combine the energy-thresholding method, the phonetic segmentation method by integration of various feature parameters obtained through the analysis procedure, and the variable hangover period method. Through the analysis of noise features, the 1 kb/s sub rate coder is designed for coding the background noise signal. So, we design the 4 kb/s sub rate coder for the unvoiced parts. The performance of the variable rate algorithm is evaluated by the comparison of speed quality and average bit rate with G.729. Subjective quality test is also done by MOS test. Conclusively, it is verified that the proposed variable rate CS-ACELP coder produced the same speech quality as G.729, at the average bit rate of 4.4 kb/s.

• Journal of Korea Society of Industrial Information Systems
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• v.9 no.3
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• pp.1-10
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• 2004

Video-On-Demand servers use compression techniques to reduce the storage and bandwidth requirements. The compression techniques make the bit rates of compressed video data significantly variable from frame to frame. Consequently, Video-On-Demand servers with a constant bit rate retrieval can not maximize the utilization of resources. It is possible that when variable bit rate video data is stored, accurate description of the bit rate changes could be computed a priori. In this paper, I propose a buffer management scheme called MAX for Video-On-Demand server using variable bit rate continuous media. By caching and prefetching the data, MAX buffer management scheme reduces the variation of the compressed data and increases the number of clients simultaneously served and maximizes the utilization of system resources. Results of trace-driven simulations show the effectiveness of the scheme.

• International Journal of Advanced Culture Technology
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• v.6 no.4
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• pp.266-274
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• 2018

Securing field screens through LTE broadcasting is a very valuable technology in news production because of its locality and concurrency. TV has the advantage of being able to deliver news faster than newspapers and make live coverage on the scene of events and events. To maximize this, not only middle-end broadcasters but also existing broadcasting stations are taking an active role in broadcasting using LTE. However LTE broadcasting has yet to secure any conflicting values of broadcasting stability and securing high-definition. The key to solving this difficulty is the transmission rate of Bit Rate. This paper compares the Bit Rate transmission method, which is variable and fixed, to provide more on-site screens through LTE, and presents the Bit Rate adjustment method as a method of screen stabilization through experiment. Consequently, the most stable broadcast screen can be obtained when transmitting and connecting it to broadcasting in a variable 5M method.

• MALSORI
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• no.65
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• pp.81-91
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• 2008

In this paper, we propose a new variable bit-rate speech coder based on the waveform interpolation concept. After the coder extracted all parameters, the amounts of the distortions between the current and the predicted parameters which are estimated by extrapolation using past two parameters are measured for all parameters. A parameter would not be transmitted unless the distortion exceeds the preset threshold. At the decoder side, the non-transmitted parameter is reconstructed by extrapolation with past two parameters used to synthesize signals. In this way, we can reduce 26% of the total bit rate while retaining the speech quality degradation below 0.1 PESQ score.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1393-1396
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• 2002

무선 ATM(Asynchronous Transfer Mode) 망을 기본으로 하여 ATM망에서 사용하는 서비스 종류인 CBR(Constant Bit Rate), rtVBR(Real-Time Variable Bit Rate), nrtVBR(Non-Real-Time Variable Bit Rate), ABR(Available Bit Rate), UBR(Unspecified Bit Rate)에 따른 플로우를 우선 각 서비스 흐름에 따라서 공평하게 패킷 스케줄링 한다. 그리고 각 서비스 종류에 따라 대역이나 지연에 따라 모든 사용자의 다양한 서비스 트래픽에 공평하게 서비스하는 구조를 갖는다. 아울러 무선 통신망 채널에 에러가 발생할 경우를 고려하여 각 서비스 트래픽에 대한 QoS(Quality of Service)를 보장해주는 무선 ATM 통신망에서 동작하는 스케줄링 알고리즘을 제안하여 시뮬레이션과을 통해 기존의 방법과 성능을 비교하여 기존의 유선 통신망 스케줄링 알고리즘과 달리 무선망의 에러에 대한 각 트래픽의 에러에 대한 성능 향상에 대해 알아본다.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.29A no.8
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• pp.34-40
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• 1992

This paper shows a new relational formula of multiplexing gain versus variable input bit rates useful for designing Nx1 ATM(Asynchronous Transfer Mode) multiplexer which mixes several asynchronous bit streams with different transmission rates. The relation between multiplexing gain and input bit stream speeds is derived from the occupied mean lenght(the width per unit time) of cells and the occupation probability of the number of cells at an arbitrary instant when the rates of the periodic cell strams change randomly. And the relation between multiplexing gain and variable bit rates from different number of input bit streams is analyzed accordingly. Under the condition of unlimited multiplexing speed, the more number of input bit streams increases, the bigger the multiplexing gain becomes. While for the case which restricts the multiplexing speed to a limited value, the multiplexing gain becomes smaller contrarily as the number of input bit streams continues too invrease beyond a boundary value. It is shown that for designing an ATM multiplexer according to the latter case, the combination of input bit streams should be determined such as its total bit rate is lower thean, but most apprpaximate to, the multiplexed output speed. Also the general formula evaluating the most significant parameters which should be needed to design the multiplexer is derived.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2017.05a
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• pp.641-644
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• 2017

Usage of LTE(Long Term Evolution) technology is expanded to industrial an emergency service field beyond commercial communications of mobile internet focused on smartphone. In this paper, transmission performance of LTE technology be supplied to various service area is analyzed on the level of application traffics. Performance is evaluated with compter simulation based NS(Network Simulator)-2, CBR(Constant Bit Rate), VBR(Variable Bit Rate File) traffic is used as simulation target. Results and methods of this paper can be used for research and developmemt of LTE based networks.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.443-451
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• 2003

Since a codebook-based CELP coder models its excitation signal according to one of several bit rates pre-assigned to codebooks and synthesizes speech signal using codebooks, it can not support encoding of speech signal at an arbitrary bit rate in one encoder. The proposed variable bit rate speech coder encodes the excitation signal based on the bit rate assigned to a present frame of speech using one-dimensional SPIHT and wavelet transform. Also it does't need to model excitation signal (or codebook) to some types as CELP coder, and can encode excitation signal at various bit rates without exact pitch information according to user requirement. As a result, since the coder doesn't have a codebook structure, it has relatively low coder complexity and provides equal or better speech quality compared to G.729 and G.723.1 coder.