• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.9
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• pp.63-69
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• 1998

In this paper, we propose a new Bit-Rate control algorithm based on bit usage matching to substitute encoded GOP(s) for new GOP(s) in MPEG-2 bitstream. It iteratively encodes current picture according to quantization value of previous picture and records bit-usage of each slice until nearly target bits are used. With target bits falling in two output bits, quantization value of slice should be changed to alleviate output bit error. We use recorded bit-usage information to decide which slices should be encoded with one quantization value and others with another. As every macroblock has different activity, we change macroblock quantization value using slice quantization value and activity value. The simulation results demonstrate that the fluctuation of the output bits can be kept within few-several tens of bits while maintaining the quality of the reconstructed pictures at a relatively stable level.

• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.9
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• pp.1325-1332
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• 2013

The first frame of a GOP, an I frame, is encoded in intra mode which generates a larger number of bits. In addition, the I frame is used for the inter mode encoding of the following frames. Thus the intial QP for the I frame affects the first frame as well as the following frames. In our previous work, we analyzed the number of bits for an I frame and showed that the ratio of the number of bits which maximizes the PSNR of a GOP maintains similar value regardless of GOP's. In this paper, we propose a R-Q model which can be used for the calculation of the initial QP given the amount of bits for an I frame. The proposed model is simple and adaptively modifies model parameters, so it can be applicable to the real-time application. It is shown by experimental results that the proposed model captures initial QP characteristics effectively and the proposed method for model parameters accurately estimates the real values.

• The Journal of the Korea Contents Association
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• v.12 no.12
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• pp.14-24
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• 2012

DVC (Distributed Video Coding) techniques need feedback channel for parity bit control to achieve the good RD performances, however, this causes the DVC system to have high decoding latency. In order to implement in real time environments and to accelerate commercializations, many research works have been focusing on the development of fast video decoding algorithm. As one of the real time implementations, this paper deals with a novel DVC scheme suitable for some application areas where source statistics such as motion information can be provided to the encoder side from the decoder side. That is, this paper propose a fast distributed video decoding scheme to improve the decoding speed by using the feedback of motion information derived in SI generation. Through computer simulations, it is shown that the proposed method outperforms the conventional fast DVC decoding schemes.

• The Transactions of the Korea Information Processing Society
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• v.4 no.10
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• pp.2629-2639
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• 1997

The video coding for very low bit-rate has recently received considerable attention, but the conventional coding schemes with block based transform suffer from the blocky effect for the constraints of limited bit-rate. In this paper, we present a video coding system based on wavelet transform and multiresolution motion estimation/compensation for very low bit-rate video. The proposed scheme uses the wavelet transform which is flexible to represent non-stationary image signals and adaptable to the human visual characteristics. The wavelet transformed coefficients are coded by various coding modes in accordance with the sum of absolute error after motion estimation/compensation in wavelet decomposed domain. And simple buffer control technique is applied to handle constant image quality. It is shown that the presented scheme has more acceptable image quality without blocky effects than conventional block based transform video coding.

• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.188-198
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• 2000

As MPEG-2 is introduced, a new editing system that works at the compressed domain is required. It is efficient to paste the two different video sequences in the compressed domain, but after repeated pasting, the result sequence can violate MPEG-2 rate control algorithm. In other words, it causes the underflow of the buffer during the decoding time. The earlier study inserted an additional fade-in sequence but it is not natural when we make a program. In this paper, a nobel method using requantization of the underflow sequence is presented.

• 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.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.7 no.2
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• pp.206-212
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• 2006

Many image and video compression algorithms work by splitting the input image into blocks and producing variable-length coded bits for each block data. If variable-length coded data are transmitted consecutively, then the resulting coder is highly sensitive to channel errors. Therefore, most image and video techniques for providing some protection to the stream against channel errors usually involve adding a controlled amount of redundancy back into the stream. Such redundancy might take the form of resynchronization markers, which enable the decoder to restart the decoding process from the known state, in the event of transmission errors. The Error Resilient Entropy Code (EREC) is a well known method which can regain synchronization without any redundant information to convert from variable-length code to fixed-length code. This paper proposes an enhancement to EREC, which greatly improves its transmission ability for the compressed image quality without any redundant bits in the event of errors. The simulation result shows that the both objective and subjective quality of transmitted image is enhanced compared with the existing EREC at the same BER(Bit Error Rate).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.7C
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• pp.654-662
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• 2009

Video streams for mobile video streaming can be encoded to fit the available network bandwidth by controlling three factors: temporal resolution, spatial resolution, and picture quality. The controlling of picture quality by modifying the quantization parameter (QP) is most widely used. In this paper, we demonstrate that reducing the spatial resolution adaptively can be more efficient in terms of picture quality and energy consumption in low bit-rate environment, and present a model to find the optimal spatial resolution for the available bandwidth. Adaptive spatial resolution control scheme is especially effective when the bandwidth between the video server and the mobile device varies considerably with time, and when the mobile device is sensitive to energy consumption. Our scheme can improve the picture quality by approximately O.5dB and reduce energy consumption by more than 50% compared to the conventional video coding in low bit-rate environment.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.