QoS control for multicast over the Internet is necessary to provide high quality multicast video streaming. With existing DiffServ multicast schemes, multicast traffic in a DiffServ domain increases as the multicast-group membership grows, and consequently packets guaranteed for delivery may be discarded. Since it is hard to guarantee QoS for all flows over the entire network, we propose DERACS a delivery range control scheme which guarantees QoS in a limited delivery range formed by actively discarding packets. In this paper, we show the delivery range of DERACS through simulation and analysis.

To support multicast over mobile IP networks, the IETF proposed two schemes, namely bi-directional tunneling (MIP-BT) and remote subscription (MIP-RS). MIP-BT provides mobility by establishing bi-directional tunnel from home agent to mobile node's care-of address. Although it does not require multicast tree reconstruction, it suffers from redundant routing that causes packet delivery delay and network bandwidth consumption. On the other hand, MIP-RS provides mobility by reconstructing multicast tree when mobile node enters a foreign network. Although MIP-RS provides shortest multicast routing path, it requires the reconstruction of entire multicast tree in case of source mobility and causes service disruption during the process of handover. To cope with those problems, we propose new source mobility scheme that enjoys the advantages of MIP-RS and provides seamless handover with the help of proxy router, which assists handover by swapping network addresses and conducting packet forwarding. We examine the deployment issues of the proposed scheme over the Internet and evaluate the proposed scheme from the viewpoint of routing distance and network bandwidth consumption through simulation experiments with various parameters, and then clarify the effectiveness and efficiency of the proposed scheme.

One of the main issues of mobile IPv6 is handover latency that causes service disruption time. Although plenty of proposals significantly reduce the service disruption time, they suffer from redundant routing that causes packet misordering and bandwidth consumption during the process of inter-domain handover. In this paper, we propose R-MIP, a routing-aware handover scheme for mobile IP, that minimizes the redundant routing during the process of inter-domain handover by utilizing forwarding routers. R-MIP consists of forwarding router discovery and proactive handover. We evaluate R-MIP in the view of packet misordering and bandwidth consumption, and clarify its efficiency. We also evaluate the impact of the forwarding router's capacity since routers have limited resources. By strategically locating forwarding routers, e.g. next to the router that has peering to another domain, the redundant routing caused by inter-domain handover will be efficiently suppressed.

One of the main issues of mobile IPv6 is handover latency that causes service disruption time. Although plenty of proposals significantly reduce the service disruption time, they suffer from redundant routing that causes packet misordering and excess bandwidth consumption. In this paper, we propose a new scheme that minimizes the redundant routing during the process of inter-domain handover by utilizing forwarding routers. Our proposed scheme consists of forwarding router discovery and proactive handover. Furthermore, we evaluate the proposed scheme in the view of packet misordering and bandwidth consumption, and clarify the effectiveness of the proposed scheme.

We have researched on dynamic media access (MAC) protocol for inter-vehicle communication (IVC) for automated vehicle control system (AVCS). In IVC, since there are no nodes that take care of the synchronization of vehicle communication, it is required for each node to sense media information and send its own vehicle information. Our target issues to be solved are the long career sensing time, difficulty of time slot or code allocation in case that vehicles are located so densely. Therefore, we propose a new scheme that solves those issues by changing the media sensing distance and frame transmission distance depending on the density of the nodes. The proposed scheme is basic MAC protocol and hence is adaptive for both collision and reserved MAC protocol. Moreover, we also propose a new scheme that transmits vehicles' own information by utilizing multi-hop communication to fulfil the AVCS system requested transmission distance. Finally, we evaluated our proposed scheme in the case of MCS/CDMA protocol. We clarified that our protocol improved the success ratio of code allocation from 20% to 60%. Moreover we clarified that only two hops is sufficient to fulfil the system requested transmission distance averagely.

P2P content distribution networks are collecting interest to efficiently deliver rich contents such as audio and video. Stream interruption caused by relay peers dropping out of the P2P network is a problem in P2P live streaming systems. In this paper, we propose a P2P live streaming system with low stream interruption. In the proposed system, peers are connected to the peers with low drop-out probability and a wide and short P2P network is constructed. Also, multiple paths to the source is considered. We show the improvement in the interruption time by our algorithm through simulations.

The demand to access multimedia data such as live video is increasing due to the spread of broadband access network. However, the Internet only provides the so called best-effort service. The network condition of the Internet shows fluctuation. Delivering video without any control causes degradation of the final video quality. To avoid such degradation, we must control video traffic to adapt to the fluctuating available bandwidth. In this paper, we propose a technique to find the available bandwidth for end-to-end stream connections over the Internet. We claim that the correlation coefficient of the network delay of an end-to-end stream is closely related to the bottleneck-link load for the end-to-end stream connection. We show and prove the relationship through analysis and simulations.

Vector representation of graphics has the advantage that the image can be displayed at any size. When the resolution of a bitmap image is changed, the lack of a line segment arises. In addition, moire occurs when the resolution of an image with halftone dots is changed. We propose a new technique to convert a binary image with halftone dots into its vector representation. Resolution conversion of a binary image can easily be performed without moire by using a continuous tone approximation of the halftone dots. First, we separate the area of halftone dots and line drawings in the image. Next, a continuous tone approximation is applied to the area of halftone dots. Then, conventional vectorization is applied to both continuous tone areas and line drawings. Finally, these components are mixed and reconstructed. Our approach provides an efficient way of displaying cartoon-like images at any size with a limited amount of data.

The demand to access multimedia data such as live video is increasing due to the spread of broadband access network. However, the Internet only provides so called best-effort service. The network condition of the Internet shows fluctuation. Delivering video without any control causes degradation of the final video quality. In this paper we propose a stream-rate control algorithm RC/sup 3/ that uses correlation of network delay to evaluate available bandwidth. We show performance of RC/sup 3/ derived from simulations.

In this paper, we propose a content based video categorizing method focusing broadcasted sports videos using camera motion parameters. We define two new features in the proposed method; "camera motion extraction ratio" and "camera motion transition". Camera motion parameters in the video sequence contain very significant information for categorization of broadcasted sports video, because in most of sports video, camera motions are closely related to the actions taken in the sports, which are mostly based on a certain rule depending on types of sports. Based on the characteristics, we design a sports video categorization algorithm for identifying 6 major different sports types. In our algorithm, the features automatically extracted from videos are analyzed in a statistical manner. The experimental results show a clear tendency the applicability of the proposed method for sports genre identification.

A new system architecture called decoder-downloadable system is described. The purpose of this system is to provide a uniform platform for the multimedia world based on image compressions using characteristics of the contents. This system enables dynamic decoder downloading by negotiation between servers and clients for seamless and minimum delay playback. This negotiation scheme is implemented by Java and CORBA (Common Object Request Broker). Thus, servers and clients in this system are independent from OSes and hardware specifications. This system is suitable for TV broadcasting as well as Internet streaming. We improve a decoder architecture in order to make the system more flexible. A decoder has some functions such as bitstream parsing, image compression algorithms (DCT, wavelet, motion estimation and so on). However, conventional decoders are monolithic software. Thus, it is impossible to share parts of decoders. The proposed method enables a module-based decoder and sharing some modules among some decoders. Consequently, the flexible decoder can make downloading time shorter by avoidance to download redundant parts of decoders. It also achieves scalability enabling this system to be used in some multimedia applications, such as a multimedia content search system, as well as a multimedia player.

As an appliance that achieves rate reduction of MPEG-2 multi-program transport streams, MPEG-2 multi-program transport stream trans coder has been proposed. This transcoder needs to multiplex packets without any STD buffer failure. However, no efficient packet-multiplexing scheme has been reported until now. In this paper, we propose packet-multiplexing scheme that no STD buffer failure occurs in MPEG-2 multi-program transport stream transcoder. This proposed scheme is implemented by the following ways: 1) PSI packets are output in constant interval, 2) audio packets are output in the same PCR as that of input transport stream, 3) video packets are output so that the intervals of each output packet for every video elementary stream should be constant. From simulation experiments, we show the status of STD buffer and show the effectiveness for our proposed scheme.

A coding scheme specifically designed for animation images is proposed. Taking the characteristics of animation images into account, lines and homogeneous color regions are extracted from animation images. Lines and contours of the homogeneous color regions are approximated by straightline and spline functions. We found that smoothing operations are effective to extract homogeneous regions from background images. A connected filter is used for the smoothing operation while keeping edges of the original image correctly. Precise approximation of the contours is also required. For this purpose, we propose to use dynamic programming with a feedback algorithm. As a result, an animation image can be represented by two layers; significant points as a base layer, and discrete cosine transform (DCT) blocks for an additional layer. For the compensation of the loss caused by the smoothing operation, the DCT is applied to the difference between the original images and lines and homogeneous color regions images. High-quality images can be obtained by adding this differential data to the approximated images.

With the increasing popularity of high-performance DV cameras and video editing applications on PC, desktop videos have become more attractive. Video editing and summary, however, still poses problems as it lacks useful support tools, which can efficiently show the video contents to editors. Against this background, our goal is to build a useful video-editing system grounded on techniques of content analysis. On the basis of our video document architecture concept, we regard camera work parameters as a means of reference to construct video. Camera work parameters can be defined follows: (1) a variety of camera work patterns (such as panning and zooming, etc.); and (2) the quality of camera work of speed and camera shake rate. By calculating these parameters, we aim to build a system, enabling one to detect significant scene candidates from the original shot sequence as well as edit videos automatically.

In this paper, we propose a scalable video transmission system based on transcoding. First, we focus on the transcoder to reduce the bitrate of MPEG-2 bitstreams. We show that it is realized the layered coding by using the differential data between input and output of a transcoder. Next, we show the coding algorithm of this differential data. This algorithm can improve the coding efficiency by removing unnecessary code for differential data. By simulation experiments, we show that the proposed coding algorithm prevents the loss of coding efficiency in traditional hierachical coding, and show the effectiveness of proposed scheme.

MPEG-2 Multi-program Transport stream (TS) achieves improvement of transmission efficiency by multiplexing several MPEG-2 streams. In this paper, we propose a transcoder which achieves rate reduction of MPEG-2 multi-program TS. For the purpose of realizing MPEG-2 multi-program TS transcoder, this transcoder requires a rate control method and re-multiplexing method: The former improves average SNR values in total of streams, and the latter achieves the evasion from failure of STD buffer. Next, from simulation experiments, we compare the conventional rate control methods to the proposed one. On the other hand, we show the state of STD buffer. Finally, we show the effectiveness for our proposed scheme.

In recent years, it has become possible to carry out on-line lectures using network technology. A student who is distant from a teacher can attend class by watching video of a lecture which is distributed through the network. However, many of this type of online lecture systems only use videos, and do not use the network to distribute teaching materials. In this paper, we prepare a method of distributing teaching materials automatically for the online lecture system, and state the technical improvements which are thought to be needed for this system.

In this paper, we focus on the video transcoder as a bit rate reducer in a network node and propose a rate control scheme for a low-delay MPEG-2 video transcoder. First, we summarize the requirements of a rate control algorithm for low-delay transcoding. Next, based on these requirements, we describe the proposed rate control scheme in detail. Then, we analyze the input and output buffer delay and calculate the total delay time of the proposed transcoder. Finally, we evaluate the proposed scheme from the simulation results of experiments on picture quality, transcoding delay time and GOP (group of pictures) structure information (N/M value) of the input bit stream. Consequently, we showed that the proposed scheme can provide the same picture quality as a traditional scheme and is independent of the GOP structure.

As the transform method of video bit stream to the required bit stream format, the video transcoder has been adopted in various applications. In this paper, we focus on a MPEG video transcoder that can reduce the required bit rate by using re-quantization in the DCT domain. Furthermore, we propose a rate control method for this MPEG video transcoder that takes into consideration the estimation of drift-error propagation. Next, from simulation experiments, we compare the proposed rate control method to the traditional rate control method in terms of complexity, required buffer size and picture quality.

We propose a layered image mosaicing method from a video sequence to acquire an adaptive resolution image. In research of art, for example, researchers often desires the full view of a painting and the details of certain regions, where fine brush-strokes are in it. The proposed method is to acquire the region of special interest at a fine resolution and the remaining image space at a coarser resolution by using a hand-held video camera which repeats moving, panning, tilting, zooming in and zooming out. To integrate video frames, image mosaicing methods can be used. However, existing mosaicing methods, in general, have a defect, where they perform inefficiently because wide angle image frames are registered after magnification. To solve this problem, the layered image mosaicing method is proposed. The proposed method can reduce processing time while it makes a mosaic image accurately. In this paper, the concept, algorithm and experiments are described. The experiments showed that the method was efficient and accurate for acquisition of an adaptive resolution image.

When the size of an image such as a picture or a painting is beyond the scope of a camera, it is difficult to take the whole picture which is a high resolution image all at once. In this case, each part of the image is taken by a camera one by one. A part of the image is an image patch. Using image patches, the whole picture is reproduced. This method is called image mosaicing. However the method has disadvantages for the reproduction of a high resolution image as follows. 1. When there are regions that do not have features or intensity gradients, it is hard to reproduce the original image because they are used to register each part of the image. 2. The higher the number of image patches, the more the image distortion is cumulated because errors on the borders between image patches are added. To solve these problems, the layered image mosaicing method is proposed. In this paper, the concept, implementation and an experiment are described. The experiment showed that the method was effective for image mosaicing.

An important step in the communication of text including unknown characters is the representation of characters. To realize bilateral exchange of texts, representation of unknown characters, such as foreign characters, created characters, design characters, and historical characters which are not being widely used nowadays is an essential subject. In particular, although many Asian countries are using Chinese characters commonly, the fonts and glyphs are different. A method to recognize the configuration by composition of common components called strokes is proposed. In this method, a character is first analyzed into small elements called parts. Then, parts are merged to form a stroke. Experimental results on Chinese characters are given.>

A very low bit-rate video coding algorithm is proposed. The algorithm is a hybrid motion compensation and DCT (discrete cosine transform) coding. However a progressive transmission scheme with a frequency-oriented coding of DCT coefficients is proposed to cope with partial freeze due to the buffer overflow. In addition, some methods to improve motion estimation are proposed. The simulation results show the proposed algorithm shows perceptually better decoded images than the traditional RM8 (reference model 8) algorithm.

A stored program-controlled electronic switching system (ESS) requires a large-capacity memory to store various kinds of programs and data. This paper describes a hierarchy arrangement including a high-speed memory and a low-cost largecapacity magnetic-drum memory for reduction of the cost of the ESS. The paper also describes the design philosophy, reliability, structure, and evaluation of the system. A magnetic drum with hydrodynamic floating heads (memory capacity:3.7 \times 10^{7}bits, clock rate: 2.2 MHz) was employed. The magnetic-drum memory system can provide transfer of individual words (each consisting of 41 bits), transfer of variable-length data by flag bit control, and content-addressable access. Furthermore, the system is designed to be most suitable for transfer of the large amount of data required by the switching system. The data throughput of the magnetic-drum memory system is 210 kbytes/s in a burst mode. The mean time between failures (MTBF) of the system is approximately3.5 \times 10^{3}h. This system assures reliability required for the ESS and high performance for application of the ESS to stored data switching as well as telephone switching.