A performance comparison between TCP and UDP in PHS Internet access is made by experiment from a media synchronization point of view. We consider a situation where PHS mobile terminals access H. 263 video and G. 726 audio stored at a media server by a streaming method. PIAFS is adopted as the data link protocol for the PHS wireless channels. We examined how white noise and Rayleigh fading on the PHS channel as well as the Internet traffic affect the performance. For the comparison, we evaluated several performance measures such as the coefficient of variation of output interval, and found that UDP outperforms TCP in almost all cases.

Visualization of 3-D ultrasound images is a challenging task due to the noisy and fuzzy nature of ultrasound imaging. This paper presents an efficient volume rendering technique for 3-D ultrasound image. A preprocessing technique of 2-D truncated-median filtering is proposed to reduce speckle noise of the ultrasound image. This paper also introduces an adaptive boundary detection method to reduce the computation time for volume rendering of ultrasound image. The proposed technique is compared to the conventional volume rendering methods with respect to the computation time and the subjective image quality. According to the comparison study, the proposed volume rendering method shows good performance for visualization of 3-D ultrasound image.

This paper presents a new data hiding scheme via steganographic image transformation, which is different from conventional data hiding techniques. The transformation is achieved in the frequency domain and the concept of Fourier filtering method is used. An input image is transformed into a fractal image, which can be used in Computer Graphic (CG) applications. One of the main advantages of this scheme is the amount of data to be hidden (embedded) is equal to that of the host signal (generated fractal image) while it is in general limited in the conventional data hiding schemes. Also both the opened fractal image and the hidden original one can be properly used depending on the situation. Unauthorized users will not notice the "secret" original image behind the fractal image, but even if they know that there is a hidden image it will be difficult for them to estimate the original image from the transformed image. Only authorized users who know the proper keys can regenerate the original image. The proposed method is applicable not only as a security tool for multimedia contents on web pages but also as a steganographic secret communication method through fractal images.

In this paper, an algorithm of the median-cut quantization (MCQ) is proposed. MCQ is the technique that reduces multi-valued samples to binary-valued ones by adaptively taking the median value as the threshold. In this work, the search process of the median value is derived from the quick-sort algorithm. The proposed algorithm searches the median value bit by bit, and samples are quantized during the search process. Firstly, the bit-serial procedure is shown, and then it is modified to the bit-parallel procedure. The extension to the multi-level quantization is also discussed. Since the proposed algorithm is based on bit operations, it is suitable for hardware implementation. Thus, its hardware architecture is also proposed. To verify the significance, for the application to the motion estimation, the performance is estimated from the synthesis result of the VHDL model.

This paper describes a new design method for multiply-adders able to process a large quantity of multimedia data. I propose a (signed digits)(unsigned digits) fixed-point multiply-add/subtract unit. The unit eliminates the problems caused by the critical one-bit arithmetic precision drop-off peculiar to the conventional (signed digits)(signed digits) fixed-point multiply scheme. By simultaneously counting in the carry-save form, based on 7-3 counters simultaneously inputting the accumulation terms and the add/sub operation terms of multiplication results, carries are propagated faster than in the conventional method.

In this paper, we propose a dynamic TDMA with priority-based request packet transmission scheme (D-TDMA/PRPTS) which applies priority-based request packet transmission scheme instead of slotted ALOHA (S-ALOHA). D-TDMA/PRPTS can avoid collisions between voice request packets and data request packets and transmit voice request packets preferentially. This makes D-TDMA/PRPTS enlarge the system capacity for voice users with SAD. We analyze voice packet dropping probability and channel utilization for voice traffic by using an appropriate Markov model. We also present simulation results to verify the analysis and to investigate data performances as well, with the voice-data integrated scenario.

Ring head recording on single-layer perpendicular recording media was studied by a simple simulation analysis based on a loop tracing method considering only the perpendicular component. Although the assumed model was primitive, the simulation results qualitatively well explained the experimental results such as a decrease in output at high recording currents and its relaxation upon using a smaller gap-length head. The simulation results revealed that achievable recorded magnetization is, in general, much smaller than the saturation value due to a broad distribution of the ring head field, but a medium with a steeper slope in the perpendicular M-H loop could improve the recording performance. This was confirmed experimentally for the medium with a steeper loop slope, though the medium exhibited a larger medium noise at high densities. It was suggested that the development of perpendicular recording for higher output and lower noise could be performed for both media with a small and steep loop slope. The former should be improved by means of the recording head while the latter by the media. A large improvement is expected for both cases.

In this paper, we will present a study of the time-dependence effect in alumite perpendicular media at different thicknesses. Important parameters of the time-dependence effect such as magnetic viscosity and activation volume are investigated. Viscosity as a function of applied field (viscosity curve) exhibits a short plateau at a low field and then decreases monotonously with increasing field. After correcting for the demagnetizing field, the shape of the intrinsic viscosity curves changes to the well-known shape of the viscosity curve of in-plane media, i. e. , they have a peak near Hc. The intrinsic viscosity curves obtained from the experiments were fitted to an analytical model by Chantrell et al., from which, we found that the effective switching volumes obtained by fitting are much smaller than the column volumes, indicating that the reversal mechanism is incoherent.

A new class of nonlinear filters called Vector Median Rational Hybrid Filters (VMRHF) for multispectral image processing is introduced and applied to color image filtering problems. These filters are based on Rational Functions (RF). The VMRHF filter is a two-stage filter, which exploits the features of the vector median filter and those of the rational operator. The filter output is a result of vector rational function operating on the output of three sub-functions. Two vector median (VMF) sub-filters and one center weighted vector median filter (CWVMF) are proposed to be used here due to their desirable properties, such as, edge and details preservation and accurate chromaticity estimation. Experimental results show that the new VMRHF outperforms a number of widely known nonlinear filters for multispectral image processing such as the Vector Median ilter (VMF) and Distance Directional Filters (DDf) with respect to all criteria used.

In this paper, system capacity of the reverse link of a wireless multimedia CDMA system with transmission power control is analysed for receivers with and without CCI cancellers. For N classes of users, system capacity is represented by a point in an N-dimensional space. It is shown that system capacity is improved considerably with CCI cancellers, that system capacity region is non-convex in general, and that its boundary is well approximated with a unique hyper plane when CCI cancellers are fully employed.

Both frequency- and time-domain analyses of glide signals from a PZT glide-slider flying over a laser zone-textured (LZT) thin film disk medium were used to determine the slider vibration at a small disk-slider clearance. Slider vibration was found to be particularly dependent on the uniformly placed laser bump and the effects due to the air-bearing stiffness over the LZT medium. We found that a high density of small, pointed laser bumps (10X) has a more distinct impact on airflow than large, jagged-rim craterlike laser bumps (1X) on the slider. We therefore investigated the effect of laser bump density on the slider vibration, and found that marginally higher laser bump density (3X versus 2X) results in higher slider vibration. While resonant vibration has been a major glide problem, the effects of laser bump density have also recently become important in the face of ultralow glide height, 0.5 µ" (12 nm). Its influence can be clearly observed when the disk-slider clearance becomes very small. At such an ultrasmall disk-slider clearance, even minimal slider vibration can be detrimental to the head-disk interface. Taking into account the various contributions of slider vibration and considering possible damage to the head-disk interface, it is clear that the optimization of laser bump design should go beyond just the glide height and coefficient of stiction. It should take into account the effects of laser bump height, density and spatial distribution on vibration-induced flying height variation while maintaining a low glide height and coefficient of stiction. An ideal LZT medium should therefore have low bump height to enable low glide height, i. e. , 0.5 µ" (12 nm), but specific bump shapes and sufficient density to achieve low stiction. Laser bump density should, however, be controlled to moderate its effect on slider vibration and possibly disk-slider collision (297 words).

Traffic management schemes such as Connection Admission Control (CAC), policing, and traffic shaping are important to provide multimedia communications with better Quality of Service (QoS). In the conventional model, admission control and policing are done at intermediate nodes, and traffic shaping is done at the edge of a network. However, QoS of communications should be defined between tasks or threads rather than between hosts. Therefore traffic management inside a host is as important as that in networks. We propose software-based traffic management architecture over a real-time microkernel. The architecture focuses on the interface between a network driver and user threads calling the driver. We categorized services of communication threads into three classes: Real-Time at Guaranteed Rate (RT-GR), Real-Time at Available Rate (RT-AR), and Best-Effort (BE). Our architecture is designed for an environment containing a mixture of these services. In the architecture, a sender periodic thread of RT-GR or RT-AR is executed such that the sending rate matches a user-specified rate. The network driver monitors the per-flow rate of injected data and discards the data if the injected rate exceeds the user-specified rate. To avoid the continuous discarding of data, the sending thread can adjust its sending rate by periodically looking at logged data concerning the rate. RT-AR service can achieve more than the specified rate when bandwidth is available. The scheme of software traffic management is effective in attaining higher throughput not only for full-duplex Ethernet but also for ATM because the difference of rate between hardware and software is reduced. In this paper, we describe the design and implementation of the software-based traffic management architecture on Real-Time Mach. The results of performance evaluations demonstrate that our traffic management scheme performs well for full-duplex Ethernet.

In future multimedia mobile communication systems, heterogeneous users will require very different and time-varying transmission rates. Thus, they should be supported with a novel approach that can accommodate the fluctuating data rates in wireless channels. In this paper, a simple method that is used to admit the rates successfully is proposed. The method is also distributed in a sense that it can be implemented at each mobile without any interaction with other mobiles. The method consists of power control with a higher SIR (signal to interference ratio) target and periodic adjustment of the data rate. We prove a good rate admission property of the proposed method and verify its performance with numerical investigation.

In this paper we present a Quality of Service (QoS) management architecture for distributed multimedia applications in heterogeneous communication environments of wired and wireless networks. Gaps in network performance such as bandwidths and error rates between wired and wireless networks, as well as gaps in terminal performance in media handling between desktop computers and handheld computers, bring about heterogeneities. Furthermore, even performance gaps among various desktop computers cause heterogeneities. As a result of these heterogeneities in network and terminal performances and various user preferences, the QoS requirement from each receiver is different. Therefore, mechanisms that adjust and satisfy each QoS requirement are needed. We propose a proxy server called Communication Coordination Server (CCS), which intermediates a video server and a receiver and manages the QoS coordination. The CCS performs QoS admission, adjustment, and allocation mechanisms to satisfy the user's QoS requirement. Then transcoding is used to realize the allocated QoS, and it decodes the input video stream from the video server and encodes it within the CCS. A QoS mapping mechanism that translates application-level QoS into resource-level QoS is needed for the QoS admission. We also propose a new QoS mapping mechanism using spline functions that enables a continuous QoS translation. We have built a CCS prototype in our laboratory testbed, and have verified that the CCS can resolve the heterogeneities between the server and receiver by the QoS adjustment mechanism of the transcoding and the QoS admission.

Synchronization and continuity are essential for multimedia presentation, but because network resources and available bandwidth are both limited, synchronization quality and continuity quality have to be traded off in response to the fluctuating network conditions. This paper therefore introduces an algorithm for intramedia synchronization with adaptive quality of service (QoS) control handled at different layers of multimedia streams. The work described here is an extension of our earlier proposal of a synchronization algorithm by delay compensation protocol with two resynchronization mechanisms: retrieval offset adjustment and data unit skipping. That algorithm has been extended by the introduction of QoS control mechanisms in the QoS plane of a distributed control platform. The extended approach results not only in better synchronization and continuity, but also integrates the QoS adjustment into the existing architecture. Unexpected QoS variations are coped with by an adaptive QoS control designed to maintain the desired application qualities within the fluctuating environment. Simulations implemented on a UDP/IP network have verified the effectiveness of the proposed scheme.

Multimedia communications over mobile networks suffer from fluctuating channel degradation. Conventional error handling schemes consist of the first stage error correction decoding in wireless interface and the second stage error correction decoding in multimedia demultiplexer, where the second stage decoding result is not used to improve the first stage decoding performance. To meet the requirements of more powerful error protection, we propose iterative soft-input/soft-output error correction decoding in multimedia communications, where the likelihood output generated by the error correction decoding in multimedia demultiplexer is fed back to the decoding in wireless interface and the decoding procedure is iterated. The performances were evaluated by MPEG-4 video transmission simulation over mobile channels.

Experiment is reported of enhanced backscattering of light in binary and ternary suspensions of rutile and/or alumina particles. With a conventional CCD camera system for observing the phenomena, the angular line shape and the enhancement factor were agreed with the theoretically predicted curve and value. Observation of the angular distribution scattered at the backscattered direction supported the hypothesis proposed by Pine et al. , in which the transport mean free path of the polydisperse mixture can be expressed in terms of summing its reciprocal values weighted over the particle sizes.

This paper proposes a novel MAC scheme over APON based on the cell arrival timing information to provide residential and small business customers with multimedia services. The proposed scheme supports the frame format of ITU-T recommendation G.983 and also provides diverse ATM service classes such as CBR, rtVBR, nrtVBR, ABR, and UBR traffics. Each service is allocated on the basis of priority and cell arrival timing information. Especially, the CBR and rtVBR services, which are sensitive to delay and CDV, are allocated with higher priority and more exact arrival timing resolution which is achieved with specific coding and ranging procedure. For the proposed MAC scheme, we present grant field format, minislot format, and bandwidth allocation algorithm. Computer simulation results show that the performance of the proposed scheme is significantly improved in terms of CDV and delay time in case of CBR and rtVBR services, comparing with the normal FIFO scheme.

In multi-media systems, the type of interactive communication channels is found almost everywhere and plays an important role, as well as the type of unilateral communication channels. In this report, we shall construct a fluctuation theory based on the concept of set-valued mappings, suitable for evaluation, control and operation of interactive communication channels in multi-media systems, complicated and diversified on large scales. Fundamental conditions for availability of such channels are clarified in a form of fixed point theorem for system of set-valued mappings.

We propose a new bit timing recovery (BTR) scheme, called perturbed sampling BTR (PSBTR), that can operate near the symbol rate in high-bit-rate wireless systems. A peculiar sample clock, the duty factor of which is not 50%, is used in the PSBTR scheme. We call this type of clock a perturbed sample clock and use it for clock recovery. In PSBTR, there is no cycle slip of the sample clock, and the PSBTR circuit is mostly digital. We examine the performance of the PSBTR scheme under additive white Gaussian noise (AWGN) by computer simulation and experiment, and from these results, clarify the relationship between the performance and circuit parameters of the PSBTR circuit. The overall results indicate that the PSBTR scheme performs well and can be employed as a BTR scheme for high-bit-rate wireless systems.