This paper examines the effect of segmentation mismatch on audio-video transmission by Bluetooth. We focus on the segmentation mismatch caused by the difference between the RFCOMM Maximum Frame Size and the baseband packet payload size. By experiment, we assessed the maximum throughput and media synchronization quality for various types of ACL packets. In the experiment, a media server transferred stored video and audio streams to a single terminal with point-to-point communication; we supposed no fading environment and added white noise by which interference from DSSS systems is modeled. The experiment showed that the effect of segmentation mismatch is large especially when the total bit rate of the two streams is near the channel transmission rate. We also observed that the media synchronization control is effective in compensating for the disturbance by the segmentation mismatch in noisy environments.

In this paper, two efficient VLSI architectures for biorthogonal wavelet transform are proposed. One is constructed by the filter bank implementation and another is constructed by the lifting scheme. In the filter bank implementation, due to the symmetric property of biorthogonal wavelet transform, the proposed architecture uses fewer multipliers than the orthogonal wavelet transform. Besides, the polyphase decomposition is adopted to speed up the processing by a factor of 2. In the lifting scheme implementation, the pipeline-scheduling technique is employed to optimize the architecture. Both two architectures are with advantages of lower implementation complexity and higher throughput rate. Moreover, they can also be applied to realize the inverse DWT efficiently. Based on the above properties, the two architectures can be applied to time-critical image compressions, such as JPEG2000. Finally, the architecture constructed by the lifting scheme is implemented into a single chip on 0.35 µm 1P4M CMOS technology, and its area and working performance are 5.005 5.005 mm2 and 50 MHz, respectively.

This paper proposes an efficient query evaluation scheme for a mediator system intended to integrate heterogeneous computing environment in terms of operating systems, database management systems, and other software. Most of mediator systems transform a global query into a set of sub-queries based on their target remote servers. Each sub-query is evaluated by the query modification method to evaluate a global query. However, it is possible to reduce the evaluation cost of a global query when the results of frequently requested sub-queries are materialized in a mediator. In a mediator, its integrating schema can be incrementally modified and the evaluation frequency of a global query can also be continuously varied. In order to select the optimized set of materialized sub-queries with respect to their current evaluation frequencies, the proposed method applies a decay factor for modeling the recent access behavior of each sub-query. In other words, the latest access of a sub-query gets the highest attention in the selection process of materialized sub-queries. As a result, it is possible to adjust the optimized set of materialized sub-queries adaptively according to the recent changes in the evaluation frequencies of sub-queries. Since finding the optimum solution of this problem is NP-hard, it takes too long to be used in practice when the number of sub-queries is large. Consequently, given the size of mediator storage, the rank-based selection algorithm proposed in this paper finds the set of materialized sub-queries which minimizes the total evaluation cost of global queries in linear search complexity.

This paper considers the problem of providing relative service differentiation in IEEE 802.11 Wireless LAN by using different Medium Access Control (MAC) parameters for different service classes. We present an analytical model which predicts the saturation throughput of IEEE 802.11 Distributed Coordination Function with multiple classes of service. This model allows us to show that relative service differentiation can be achieved by varying the initial contention window alone. In this case, the saturation throughput of a station can be shown to be approximately inversely proportional to the initial contention window size being used by that station. The simulation results validate our analytical model.

A new inter-client synchronization framework employing a server-client coordinated adaptive playout and error control toward one-to-many (i.e., multicast) media streaming is discussed in this paper. The proposed adaptive playout mechanism controls the playout speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. By coordinating the playout speed of each client, the inter-client synchronization with respect to the target presentation time is smoothly achieved. Furthermore, RTCP-compatible signaling between the server and group-clients is performed to achieve the inter-client synchronization and error recovery, where the exchange of controlling message is restricted. Simulation results show the performance of the proposed multicast media streaming framework.

In this article we analyze the performance of a space division output buffered multichannel switch operating in an ATM multimedia environment as follows. Fixed size packets belonging to two classes arrive onto the switch inputs in each time slot. Class-1 packets, representing real time traffic such as live audio and live video communications, are sensitive to delay but insensitive to loss and have their own service time needs. Class-2 packets, representing nonreal time communications such as file transfers, are insensitive to delay but sensitive to loss and have their different service time needs. To respond to the class-1 delay sensitivity, the switch gives class-1 packets higher service priority over class-2. And to respond to the difference in service time needs, the switch operates at two service rates, one for each class. This latter assumption is the major feature of the article, as previous studies have usually assumed that the two classes have the same service needs and thus the same service rate. For the purpose of the analysis, the switch is modelled as a priority, discrete time, batch arrival, multiserver queueing system, with infinite buffer and two geometric service times with two parameters. Performance measures analyzed are system occupancy and packet waiting time.

We have been developing new cycling environments by using knowledge sharing and speech communication. We have offered multimodal knowledge contents to share knowledge on safe and exciting cycling. We accumulated 140 contents, focused on issues such as riding techniques, trouble shootings, and preparations on cycling. We have also offered a new way of speech communication using an ad-hoc wireless LAN technology for safe cycling. Group cycling requires frequent communication to lead the group safely. Speech communication achieves spontaneous communication between group members without looking around or speaking loudly. Experimental result through actual cycling has shown the effectiveness of sharing multimodal knowledge contents and speech communication. Our new developed environment has an advantage of increasing multimodal knowledge through the accumulation of personal experiences of actual cycling.

Connectedness oriented communication denotes a mode of communication in which the activities of communication are more important than the contents of communication. It is targeted at maintaining and enhancing human social relationships. As our lifestyles and societies are shifting along with the progress of Information Technology, communication media that are connectedness oriented will play an important role. In this paper we propose a media called FaintPop, which is an example of such new media that are suitable for connectedness oriented communication. It is a communication media designed for a community, with which the sense of connectedness can be shared among members. Furthermore, it provides a general overview of the communication activities occurring in the community. We discuss several principles and points in designing the media, especially about the interaction of the users. Results and findings from the experiment using the media are reported.

In story-based communication, where a message is conveyed in story form, it is important to embody the story with expressive materials. However, it is quite difficult for users to create rich multimedia contents using multimedia editing tools. This paper proposes a web-based multimedia environment, SPOC (Stream-oriented Public Opinion Channel), aiming at helping non-skillful people to convert their stories into TV-like programs very easily. The system can produce a digital camera work for graphics and video clips as well as generate an agent animation automatically according to a narration text. Findings in evaluation experiments showed that SPOC is easy-to-use and easy-to-learn for novice users. Given a short instruction, the subjects not only mastered the operations of the software, but also succeeded in creating highly original programs. In subjective evaluation, the subjects answered that they enjoyed using the software without feeling difficulty. These results suggest that this system reduces user's cost in making a program, and encourages communication in a network community.

IP multicast is seen as an efficient way of encouraging multimedia services such as Internet TV and Videoconferencing because it can deliver packets to multiple users while efficiently using network resources. Source Specific Multicast (SSM) is suggested as the IP multicast routing protocol and it can construct multicast trees efficiently. However it increases multicast forwarding table entries and fails to handle source mobility. This paper proposes the Unicast Extension Multicast Protocol (UMP) to solve these problems. In the protocol, only the routers that act as branching points keep multicast forwarding table entries, and packets are delivered between these routers using IP unicast. This prevents the multicast forwarding table entries from burdening other non-branch routers. Additionally, UMP supports source mobility by using the recursive join messages to prevent the creation of redundant paths while supporting source mobility.

In this letter, we propose a packet scheduling algorithm to support both real-time voice and video traffic for wireless multimedia data service in orthogonal frequency division multiple access (OFDMA) system. Our design objective is to maximize the number of real-time service users that can be supported in the system subject to QoS requirement of packet loss rate (PLR). Both time slots and subcarriers are taken into account as the basic resource allocation unit in OFDMA/FDD system. The simulation results show that our proposed algorithm can dramatically increase the number of users satisfying the underlying QoS requirement for the real-time service, as compared to the existing algorithm.

In communication systems such as mobile telecommunication systems and the Internet, resource sharing among coexisting real-time and non-real-time services is extremely important to provide multimedia services. This paper analytically investigates the performance of the packet data control algorithm proposed in. This algorithm efficiently uses radio resources by utilizing the remaining capacity that is not used by real-time services. The state probability vectors and transition probability matrices of both the real-time and non-real-time services are first derived and then the delay characteristics, the outage probability of voice users, and the outage probability of data users are evaluated. A performance analysis with high bit-rate non-real-time services is also presented.

Future mobile communication systems are expected to support multimedia applications (audio phone, video on demand, video conference, file transfer, etc.). Multimedia applications make a great demand for bandwidth and impose stringent quality of service requirements on the mobile wireless networks. In order to provide mobile hosts with high quality of service in the next generation mobile multimedia wireless networks, efficient and better bandwidth reservation schemes must be developed. A novel traffic-based bandwidth reservation scheme is proposed in this paper as a solution to support quality of service guarantees in the mobile multimedia wireless networks. Based on the existing network conditions, the proposed scheme makes an adaptive decision for bandwidth reservation and call admission by employing fuzzy inference mechanism, timing based reservation strategy, and round-borrowing strategy in each base station. The amount of reserved bandwidth for each base station is dynamically adjusted, according to the on-line traffic information of each base station. We use the dynamically adaptive approach to reduce the connection-blocking probability and connection-dropping probability, while increasing the bandwidth utilization for quality of service sensitive mobile multimedia wireless networks. Simulation results show that our traffic-based bandwidth reservation scheme outperforms the previously known schemes in terms of connection-blocking probability, connection-dropping probability, and bandwidth utilization.

We study the user reneging behavior in a hybrid Multimedia-on-Demand (MoD) system, which provides both interactive and batch services. Reneging happens when users become reluctant to wait for a batch service and quit the service. They will either join the interactive service or quit the system. In this letter, the analytical model for the MoD system with user reneging behavior is provided. Numerical results show that the reneging behavior has a significant impact on system performance. Therefore, careful consideration on reneging behavior should be made for designing a MoD system.

Band Division MC-CDM (BD-MC-CDM) has been proposed for high quality wireless communications and has been investigated in terms of link level performance. In this paper, we investigate frequency band and time slot selection technique from the viewpoint of system level performance in order to realize the efficient BD-MC-CDM system under cellular environments. Then a downlink frequency band and time slot selection scheme is proposed for cellular BD-MC-CDM systems. The proposed scheme selects transmission frequency band according to the signal-to-interference ratio (SIR) estimated by using the pilot signal at mobile stations and also selects transmission time slot by using the SIR threshold. Simulation results show that the proposed scheme improves the downlink throughput but degrades delay performance and it has a trade off between throughput and delay performance. By selecting suitable control parameters, the proposed scheme achieves the throughput improvement without sacrificing the delay performance.

A VLSI-specific wavelet processing technique has been developed and implemented as a processor in accordance with the JPEG2000 specification. This proposed procedure of discrete wavelet transforms uses an altered calculation equations and makes use of intermediate results through wavelet calculation. The implementation of the proposed procedure is capable of realizing a highly efficient DWT for large size images in spite of using low hardware costs and a small size buffering memory. In order to obtain fast EBCOT processing, three types of parallel processing are introduced in the EBCOT architecture. The processor performs compression of 720480 pixels images with the speed of 30 frames per second (fps) at a required operating frequency as low as 32 MHz or lower. Furthermore, it need not divide an image into tiles so that the problem of deterioration of image quality due to tile division does not occur. A prototype of this processor has been fabricated in a 0.25-µm 5-layer CMOS process. The chip is 10.210.4 mm2 in size and consumes 2.0 W when supplied with 2.5 V and 32 MHz.

Video transmission over Terrestrial Trunked Radio (TETRA) mobile channel employing MPEG-4 visual coding standard is proposed in this paper. Detail parameters of the proposed systems are discussed in this paper. Performance of the proposed systems was evaluated in Average Peak Signal to Noise Ratio (APSNR) versus Signal to Noise Ratio (SNR) and Bit Error Rate (BER). In particular, the video quality that can be achieved at different channel conditions and employing different combinations of MPEG-4 visual error resilient tools is presented in this paper. Results obtained show that higher video bitrate does not necessarily lead to higher video quality at the receiver as the received video quality depends on the bit error pattern or the number of error free video packets.

A single chip processor suitable for various multimedia communication products has been developed. This chip achieves real-time bi-directional encoding/decoding for CIF resolution video at a frame rate of 30 fr/s, and meets such standards, as H.320 and H.324. The chip is composed of a video-processing unit for MPEG-4 and H.26X standards, a DSP unit for speech codec and multiplex processes, and a RISC unit for managing the whole chip. By heterogeneous multiple processor architecture, careful study of task sharing for each processing unit and bus configuration, a single chip solution can be achieved with reasonable operation speed and low-power consumption suitable for consumer products. Moreover, by applying an original video processing unit architecture, this chip achieves real-time bi-directional encoding/decoding for CIF-resolution video at a frame rate of 30 fr/s. An original video bus was developed to provide high performance and low-power consumption while sharing one external memory which is necessary for various video processes and graphics functions. This shared memory also has the effect of minimizing die size and I/O ports. This chip has been fabricated with 4-metal 0.18 µm CMOS technology to produce a chip area of 10.510.5 mm2 with 1.2 W power dissipation including I/O power, at 1.8 V for internal supply and 3.3 V for I/O power supply.

Artificial electromagnetic structures have significantly broadened the range of wave propagation phenomena available. In particular, it has been shown that metamaterials can be constructed for which the index-of-refraction is negative over a finite band of frequencies. In this paper, we present the design, fabrication and characterization of a metamaterial that exhibits negative refraction. The metamaterial design we explore is anisotropic in the plane of propagation. Based on our analysis and supporting simulations and measurements, we demonstrate that for the geometry considered, the anisotropic metamaterial has the identical negative refraction properties as would an isotropic negative index metamaterial.

A new sorting algorithm and architecture for fast median filter are proposed. This algorithm results in low area VLSI architecture producing low switching activity and without using feedback. The main idea is to employ the extra matrix for fast search operation of rank of oldest window element. We simulated and synthesized this algorithm using SYNOPSYSTM and showed the sufficiency in real time operation.