• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.25-28
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• 2002

This paper presents variable bit rate(VBR) rate control scheme based on MPEG-4 VM8 rate control scheme. An initial Q searching method provides more accurate bit allocation for the first frame. A frame skipping and RD Model update scheme when coded frame quality is too low or high prevents image quality fluctuation.

• Journal of Digital Contents Society
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• v.13 no.1
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• pp.111-118
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• 2012

The burst is to require abruptly high transmission rate in case of transmitting pre-stored variable bit rate video data, and it causes to be inefficient use of network resource, resource reservation. To avoid these problems, smoothing is transmission plan where variable rate video data is converted to a constant bit rate stream. These smoothing algorithms include CBA, MCBA, MVBA and others. To evaluate amount of burst reduction in the existing CBA, MCBA, MVBA algorithm, this paper compares the burst-related-factors of transmission plan in smoothing algorithms with original video sources which were stored Variable Bit Rate. There are maximum frame bytes, maximum GOP bytes, transmission rate variability per frame, transmission rate variability per GOP in burst-related evaluation factors. Experimental result shows burst-related factors of smoothing algorithms which were used for experiment lower than that of pre-stored video data, except special case.

• Journal of Korea Multimedia Society
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• v.21 no.3
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• pp.415-422
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• 2018

Advances of telecommunication technology bring various changes in journalism field. Reporters started to gather, edit, and transmit content to main server in media company using hand-held smart media and notebook computer. This paper tried to testify valid bit-rate of visual news content using LTE network and mobile phone. Field news like natural disasters need real-time transmission of video content. But broadcasting company normally use heavy ENG system and transmission satellite trucks. We prepared and experimented different types of visual content that has different bit-rates. Transmission tool was LU-60HD mobile system of LiveU Corporation. Transmission result shows that bit-rate of 2Mbps news content is not suitable for broadcasting and VBR (Variable Bit Rate) transmission has better definition quality than CBR (Constant Bit Rate) method. Three different bit-rate of VBR transmission result shows that 5Mbps clip has better quality than 1Mbps and 3Mbps. The higher bit-rate, the better video quality. But if the content has much movements, that cause delay and abnormal quality of video. So optimizing the balance between stability of signal and quality of bit-rate is crucial factor of real-time broadcasting news gathering business.

• Journal of Korea Society of Industrial Information Systems
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• v.4 no.4
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• pp.32-39
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• 1999

Video-On-Demand servers have to provide timely processing guarantees and reduce the storage and reduce the storage and bandwidth requirements for continuous media However, compression techniques used in Video-On-Demand servers make the bit rates of compressed video data significantly variable from frame to frame Consequently, most pervious Video-On-Demand servers which use constant bit rate retrieval to guarantee deterministic service under-utilize the system resources and restrict the number of clients. In this paper, I propose a buffer management scheme called CAP(Caching And Prefetching) for Video-On-Demand server using variable bit rate continuous media. By caching and prefetching the data CAP 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.

• Proceedings of the KIEE Conference
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• 2000.07d
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• pp.3166-3168
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• 2000

This paper proposed variable threshold dual rate ADPCM coding method which is modified from the standard ADPCM of ITU G.726 for speech quality improvement. The speech quality of variable threshold dual rate ADPCM is better than single rate ADPCM at noisy environment without increasing the complexity by using ZCR(Zero Crossing Rate). In this case, ZCR is used to divide input signal samples into two categories(noisy & speech). The samples with higher ZCR is categorized as the noisy region and the samples with lower ZCR is categorized as the speech region. Noisy region uses higher threshold value to be compressed by 16Kbps for reduced bit rates and the speech region uses lower threshold value to be compressed by 40Kbps for improved speech quality. Comparing with the conventional ADPCM, which adapts the fixed coding rate. the proposed variable threshold dual rate ADPCM coding method improves noise character without increasing the bit rate. For real time applications, ZCR calculation was considered as a simple method to obtain the background noise information for preprocess of speech analysis such as FFT and the experiment showed that the simple calculation of ZCR can be used without complexity increase. Dual rate ADPCM can decrease the amount of transferred data efficiently without increasing complexity nor reducing speech quality. Therefore result of this paper can be applied for real-time speech application such as the internet phone or VoIP.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1750-1754
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• 2013

A variable bit rate speech coder is presented. The coder is based on the observation that a speech signal can be viewed as a combination of piecewise linear signals in a short time period. The encoder detects the sample points where the slope of the signal changes, which are called the inflection points in this paper. The coder transmits the location and value for the detected inflection sample, but only the location information for the noninflection samples. In the decoder, the noninflection samples are estimated with interpolation of the received information. Several factors affecting the performance of the coder have been tested through simulation. Simulation results show that the linear interpolation produces 1 ~ 5 dB improvement over the cubic spline interpolation. And the -law companding does not provide any benefit when it is applied before the inflection detection. With low threshold values in the inflection point detection, the coder shows better MOS and more than 16 dB improvement in SNR compared to the continuously variable slope delta modulation (CVSDM).

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.8
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• pp.15-21
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• 1996

This paper describes data placement schemes that provide uniform and balanced to multiple disks load for retrievals of VBR (variable bit rate) video at varying retrieval speeds. To support maximum concurent users at arbitrary-speed playbacks in a disk-arry based system, the hot spot disks should be carefully avoided. In this paper, we extend the proposed scheme, prime round-robin(PRR), for VBR video. In addition, we have compared the performance of PRR and PRR (PRR extension).

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.205-209
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• 1994

This paper presents a variable rate speech coding method with phonetic segmentation, called for PSVXC. Multiple access techniques that require efficient encoding of speech to achieve capacity improvements are currently emerging in the cellular telephone system. The variable rate speech coder have the reduced average data rate required to transmit conversational speech. Each frame of active speech is classified into one of four phonetic classes. A distinct coding configuration and bit-rate is applied to each category. And also a split vector quantization is used to accurately quantize the LPC information using LSP parameters.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1487-1504
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• 1997

In this paper, the two-stage variable block-size multiresolution motion algorithm is proposed for an interframe coding scheme in the wavelet decomposition. An optimal bit allocagion between motion vectors and the prediction error in sense of minimizing the total bit rate is obtained by the proposed algorithm. The proposed algorithm consists of two stages for motion estimatation and only the first stage can be separated and run on its own. The first stage of the algorithm introduces a new method to give the lower bit rate of the displaced frame difference as well as a smooth motion field. In the second stage of the algorithm, the technique is introduced to have more accurate motion vectors in detailed areas, and to decrease the number of motion vectors in uniform areas. The algorithm aims at minimizin gthe total bit rate which is sum of the motion vectors and the displaced frame difference. The optimal bit allocation between motion vectors and displaced frame difference is accomplished by reducing the number of motion vectors in uniform areas and it is based on a botom-up construction of a quadtree. An entropy criterion aims at the control of merge operation. Simulation resuls show that the algorithm lends itself to the wavelet based image sequence coding and outperforms the conventional scheme by up to the maximum 0.28 bpp.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.10C
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• pp.945-950
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• 2002

In this paper the performance evaluations on Asynchronous Transfer Mode(ATM) access node are performed in the ATM access network which consists of access node and channel. The performance factors of access node are Cell Loss Ratio(CLR), buffer capacity and delay time. Both the ATM cell-scale queueing model and burst-scale queueing model are considered as the traffic model of access node for various traffic types such as Constant Bit Rate(CBR), Variable Bit Rate(VBR) and random traffic in the ATM access networks. Based on these situations, the relation of CLR, buffer capacity and delay time is analyzed in the ATM access node.