In this paper we present a case study of a hierarchical statistical analysis. The method which we use here bridges the statistical information between process-level and system-level, and enables us to know the effect of the process variation on the system performance. We use two modeling techniques--intermediate model and response surface model--in order to link the statistical information between adjacent design levels. We show an experiment of the hierarchical statistical analysis applied to a Phase Locked Loop (PLL) circuit, and indicate that the hierarchical statistical analysis is practical with respect to both accuracy and simulation cost. Following three applications are also presented in order to show advantage of this linking method; these are Monte Carlo analysis, worst-case analysis, and sensitive analysis. The results of the Monte Carlo and the worst-case analysis indicate that this method is realistic statistical one. The result of the sensitive analysis enables us to evaluate the effect of process variation at the system level. Also, we can derive constraints on the process variation from a performance requirement.

In the beginning of the new century, many information appliance (IA) products will replace traditional electronic appliances to help people in smart, efficient, and low-cost ways. These successful products must be capable of communicating multimedia information, which is embedded into the electronic appliances with high integration, innovation, and power-throughput tradeoff. In this paper, we develop a codesign procedure to analyze, compare, and emulate the multimedia communication applications to find the candidate implementations under different criteria. The experimental results demonstrate that in general, memory technology dominates the optimal tradeoff and ALU improvements impact greatly on particular applications. The results also show that the proposed procedure is effective and quite efficient.

An inverse scattering problem in three dimensional two layered media is investigated. The shape and the location of the acoustic scatterer buried in one half-space are determined. With some a priori information, it becomes possible to solve this problem in three dimensions. Using the moment method, the scattered field is obtained for the estimated scatterer. An iterative procedure based on the Newton's method for the nonlinear least square problem is able to solve the inverse scattering problem. Some numerical results are presented.

In practice, the DS-CDMA system is equipped with a finite number of Channel Element (CE)s that performs the baseband spread signal processing for a given channel in the base station. In this situation, the call blocking can be caused not only by the insufficient number of channel elements but also by the limit of available traffic channels. In this paper, we focus on analyzing the effect of the limited number of CEs on the Erlang capacity of multimedia DS-CDMA systems in the reverse link when the CDMA cells are sectorized with 3 sectors. For the performance analysis, a multi-dimensional Markov chain model is developed. As a result, the more CE results in the larger Erlang capacity. However, the Erlang capacity is saturated after a certain value of CEs where the call blocking is mainly caused by the insufficient channels per sector.

We study the electromagnetic wave propagation in three-dimensional (3-D) dense random discrete media containing dielectric spheroidal scatterers. We employ a Monte Carlo method in conjunction with the Method of Moments to solve the volume integral equation for the electric field. We calculate the effective permittivity of the random medium through a coherent-field approach and compare our results with a classical mixing formula. A parametric study on the dependence of the effective permittivity on particle elongation and fractional volume is included.

Session length and group size are two most significant factors in achieving efficient scheduling for unicast and multicast traffic in single-hop wavelength division multiplexing (WDM) local area networks (LANs). This paper presents a hybrid protocol to schedule both unicast and multicast traffic in broadcast WDM networks. The protocol makes an important assumption that unicast traffic is the major portion of the overall traffic and is usually scheduled with a pre-allocation-based protocol. On the other hand, multicast traffic is a smaller portion of the overall traffic with multicast sessions and multicast groups, and is scheduled with a reservation-based protocol. The concept of multicast threshold, a function of random variables including the multicast session length and the multicast group size, is also proposed to partition the multicast traffic into two types. If the transmission threshold of a multicast request is larger than the multicast threshold, the request is handled with a reservation-based protocol. Otherwise, the multicast request is handled similar to unicast traffic; that is, each packet in the multicast session is replicated and sent to the unicast queues of destinations. The results show that the hybrid protocol can achieve better channel utilization efficiency and packet delay for unicast traffic under the multicast scenarios with moderate session length and group size. However, separate scheduling or broadcasting will be more suitable for a multicast scenario with very large session length and group size, which is not common on most realistic networks.

GSRI Pragmatic TCM, which is a Pragmatic Trellis Coded Modulation allowing bandwidth expansion, has been proposed. In [1], it is shown that this scheme can achieve higher performance than conventional Pragmatic TCM scheme. On the other hand, a real-time video multimedia communication is one of the possible applications for the third generation mobile communication systems. This video multimedia communication system needs a multiplexer which mixes various types of media such as video, voice and data into a single bitstream. ITU-T has standardized H.223 Annex A, B, C and D multimedia multiplexing protocols for low bit-rate mobile communications. This paper evaluates the performance of the GSRI Pragmatic TCM with an application of a mobile multimedia system using H.223 Annex D multiplexing scheme and MPEG-4 video coding.

This paper focuses on a single BSA (Basic Service Area) in an infrastructure network and studies the performance of the IEEE 802.11 standard MAC protocol by means of simulation. The MAC protocol supports DCF (Distributed Coordination Function) and PCF (Point Coordination Function). The simulation model includes both data transmission with the DCF and H.263 video transmission with the PCF. In the simulation we assume that the channel transmission rate is 2 Mbps and use the system parameters specified in the standard for the DSSS (Direct Sequence Spread Spectrum) physical layer. We evaluate the performance of this protocol in terms of throughput and MPDU (MAC Protocol Data Unit) delay for various values of the CFP (Contention Free Period) repetition interval and the CFP maximum duration. Numerical results show that if the CFP repetition interval is set too long, video MPDU delay becomes very large periodically; therefore, average video MPDU delay deteriorates. We also find that as the CFP maximum duration decreases, the number of video terminals that can be accommodated in the system decreases. Furthermore, how channel transmission errors affect the performance of the protocol is examined. A two-state continuous-time Markov model is used as a burst error model. As a result, we see that for a small number of video terminals, the average video-MPDU-delay performance does not deteriorate drastically for larger values of bit error rate.

This paper proposes a new parallel high speed mobile radio transmission scheme using cyclic-shifted-codes generated from a modified M-sequence. The modified M-sequence is biased with constant direct current (dc) on an M-sequence and is inserted the guard chips before and after this biased M-sequence. The proposed system has the following features: i) Orthogonality of the codes is kept not only between direct waves of each parallel channels but also between direct and delayed waves within the guard chips; ii) It is possible to reduce the number of kinds of codes allocated to one user; and iii) It is easy to recover both code and chip timings. In this paper, moreover, the performance of the proposed system was evaluated in terms of bit error rate (BER) under additive white Gaussian noise (AWGN), non-selective one path Rayleigh fading and double-spike Rayleigh fading channels. As a result, the proposed transmission scheme can transmit several Mbps in a high-speed double-spike Rayleigh fading channel with better quality in comparison with a conventional multicode CDM transmission scheme based on M-sequences.

Medium noise is the dominant noise in ultrahigh density disk recording systems. The peak, width and jitter noise are analyzed by micromagnetic simulations. Four different media, with a fixed grain size of 135 and a coercivity of 2900 Oe, are chosen for medium noise analysis. The linear recording density is increased from 340 KFCI (Kilo flux-changes per inch) to 750 KFCI, while the area density goes up from 14.3 Gb/in2 to 31.5 Gb/in2. The peak-amplitude noise is studied by the distribution of the peak magnetization Mp in each bit. The distribution of Mp develops from a delta-function around the remanence Mr at low densities to a flat distribution at extremely high densities. It is found that the transition a-parameter is no longer proportional to the square root of Mrδ, as given in the William-Comstock approximation. The peak-jitter noise in the read back voltage is analyzed by the percentage of the transition jitter in a bit length.

The medium noise of single-layer perpendicular recording media is known to be suppressed by reducing the magnetic domain size and achieving a higher squareness ratio (Mr/Ms = SQ) in the perpendicular M-H loop. The media with smaller domain sizes exhibit a small slope at Hc in the M-H loop due to exchange de-coupling between adjacent grains, which requires a sharp head field to acquire high recording performances. Reduction of the medium thickness would be effective for recording as only a sharp head field near the head surface could be used. Thus, the effects of reduced recording layer thickness in single-layer perpendicular recording media on read/write performances were investigated using Co/Pd multilayer media with a small loop slope having thickness, δ, of 46, 22 and 10 nm, and with a steeper loop slope having δ of 40 and 10 nm. It was found that the recording performance on small loop slope media could be improved in terms of signal level by reducing the recording layer thickness, which indicated that the recording on the media was sensitive to the recording head field. The results in the simulation analysis were similar to those obtained experimentally, indicating that the change in recording layer thickness could be mainly regarded as that in the head-medium spacing. Thinner media with steeper loop slopes could acquire a narrower dipulse width. The recording resolution of the present media, however, was determined under the influence of the domain structure and the size. Finally, for media with small loop slopes, the same SNR of 38 dB at 100 kFRPI was obtained for thicknesses of 22 and 10 nm, which was larger than that for a thick medium of 46 nm thickness by 8 dB. For both the steep loop slope media, the obtained SNR was 35 dB at 100 kFRPI.

As technology moves at an annual area density increase rate of 80-120% and channel density moves beyond 3, micro-fluctuation of media recording performance and the homogeneity of media's recording capability become serious reliability concerns in future high density magnetic recording systems. Two concepts are proposed in this work for the characterization of the micro-recording performance fluctuation at high bit and channel densities: recording performance roughness analysis and dynamic magnetic roughness analysis. The recording performance roughness analysis is based on an in-situ measurement technique of the non-linear transition shift (NLTS). Relationship between the performance roughness and the roughness of dynamic magnetic parameters are studied. Results of experimental investigations indicate that the NLTS based performance roughness analysis can reveal more details on media's recording capability and the capability fluctuation--the macro and micro fluctuation of recording performance. The dynamic magnetic roughness analysis is read/write operation based and can be used to characterize the macro and micro fluctuation of media's dynamic magnetic properties. The parameters used for the analysis include media's dynamic coercivity and the dynamic coercive squareness. Here, "dynamic" refers to the dynamic performance measured at MHz frequency. The authors also noticed in their technology development process that further methodology development and confirmation are necessary for media's dynamic performance analysis. Therefore, the work also extends to the accuracy analysis of the playback amplitude based methods for the analysis of the dynamic coercive squareness and dynamic hysteresis loop. A method which is of smaller testing error is identified and reported in this work.

It is known that amorphous continuous media such as TbFeCo have extremely low noise characteristics because of the structure of the continuous grainless medium. There is great interest in the use of amorphous media in magnetic recording. This study investigated the recording characteristics of the amorphous continuous medium by computer simulation using the Landau-Lifshitz-Gilbert equation. It was shown that the transition of the continuous medium is very sharp and the noise level very low. It was also shown that the recorded magnetization patterns of the continuous medium are distinct at the high recording density of 380 Gbit/inch2. We concluded that the continuous medium has great potential for use in ultra-high density recording.

We have developed an IP-over-Ethernet-based ultra high-speed multimedia wireless LAN prototype operating in the 60-GHz band. It employs a media-access-control (MAC) protocol based on reservation-based slotted idle signal multiple access (RS-ISMA), which was implemented in the former prototype, for supporting various IP traffic such as real-time AV traffic and best-effort web traffic. The protocol also has a new function called NACK sensing for the efficient retransmission of wireless multicast packets. It was demonstrated that the prototype can provide the world's fastest radio transmission speed of 128 Mbps for two-way communications. We have measured the throughput and latency of the prototype LAN for Ethernet-frame transmission in a point-to-point baseband-connected environment. The measurement showed that the prototype LAN provides a maximum throughput of 30 Mbps, and that the measured throughput agrees with the theoretically predicted throughput. It also showed that the maximum latency, which includes switching and routing latency in the wired part, is below 1 msec.

Recent development of compact and powerful portable computers and mobile phones and proliferation of the Internet will enable mobile multimedia communications. From the viewpoint of implementing multimedia services into mobile communications, it allows us to predict that traffic characteristics of mobile networks change. For planning, designing, and operating mobile multimedia networks, it is important to investigate traffic models which take the effect of multimedia services into consideration. This paper investigates population of active users in a micro-cell and proposes a traffic model for mobile multimedia networks. This model describes a population process of active users in a micro-cell in diffusion model, and its characteristics include self-similarity and activity of mobility. We also made an evaluation of network performance by using simulation, in order to show that characteristics of the proposed traffic model have impact on planning and designing networks.

The ATM Wireless Access (AWA) System allows portable terminals such as notebook PCs to provide up to 10Mbits/s to each user. AWA will be one of the last hops of the fiber system; it seamlessly provides wireless terminals with most of the services available in the fiber system. A prototype is developed to confirm system realization and the technical feasibility of the radio transmission rate of 80 Mbit/s, the highest yet reported in wireless access systems, by employing ATM technology to support multimedia communication with different communication quality requirements. The prototype uses TDMA as the multiple access method. This paper proposes the system concept and technical issues of the AWA system. The design and performance of the AWA prototype are clarified. It is confirmed that the target performance of the prototype can be achieved and technical issues are feasible.

In this paper, we propose a fast algorithm to realize parallel median filter for processing 1-D and 2-D signal. In the proposed pipelined architecture, m-passes are employed for filtering signal while word resolution is m bits. One pass employs one processing element (PE), and the number of PEs is independent of the number of samples. Therefore, we only need m PEs for real-time operation. With 8-bits resolution, the system gate-count is less than 5 k. Moreover, this median architecture could be easily modified to consist of the programmable feature that may choose the better sampling number to filter signal. It should be also noted that our proposed processing flow has a progressive property, which is very suitable for bandwidth-limited channel application.

File system buffers provide memory space for data being transferred to and from disk and act as caches for the recently used blocks, and the buffer manager usually reads ahead data blocks to minimize the number of disk accesses. However, if several multimedia files with different consumption rates are accessed simultaneously from the file system in which LRU buffer replacement strategy is used, the read-ahead blocks of the low rate file are unloaded from memory to be used for loading a data block of a high data rate file, therefore they should be reloaded again into memory from disk when these blocks are actually referenced. This paper proposes and implements a new buffer cache management scheme for a multimedia file system and analyzes the performance of the proposed scheme by modifying the file system kernel of FreeBSD. In this proposed scheme, initially, some buffers are allocated to each opened multimedia file, privately, then these buffers are reused for other data blocks of that file when they are loaded from the disk. Moreover, the number of private buffers allocated for the file is dynamically adjusted according to its data rate. An admission control scheme is also proposed to prevent opening of a new file which may cause overloads in the file system. Experimental results comparing proposed scheme with the original FreeBSD and a simple CTL-based model show that the proposed buffer management scheme could support the realtime play back of several multimedia files with various data rates concurrently without helps of a realtime CPU and disk scheduling.

In this paper, we propose the high-level service architecture supporting multimedia multicast. The proposed architecture specifies network-oriented and lightweight communication management, which includes group management, multiparty call control and multicast connection/multiple connections control. Many of the existing approaches handling multimedia multicast applications are dependent on specific transport technologies, such as the Internet and ATM, and lack effective communication management. On the other hand, our approach defines flexible and extensible communication management that can be applied to a variety of multimedia multicast applications, independently of transport technologies. Our architecture supports the separation of control from a terminal, which enables remote control and control mobility, so that a user can use a multicast service in a more various way. The architecture is overlaid above legacy transport networks so that the existing network protocols are used for connection control. This minimizes the modification to a legacy transport network and enhances the practicality of the architecture. In addition, terminal manager and virtual device concepts are introduced that hide the details of physical devices from an application designer. The architecture consists of several service components that effectively interact with each other on a distributed platform. To verify and evaluate our architecture, we have prototyped the high-level service architecture on a CORBA platform and analyzed the architecture using a simulation.

This paper presents an object-oriented model to handle the temporal relationship for all of the multimedia objects at the presentation platform. Synchronization of the composite media objects is achieved by ensuring that all objects presented in the upcoming "manageable" period must be ready for execution. To this end, the nature of overlays is first investigated for various types of objects. Critical overlaps which are crucial in synchronization are also defined. The objective of synchronization is to ensure that the media objects can be initiated precisely at the critical point of the corresponding critical overlap. The concept of manageable presentation interval is introduced and the irreducible media group is defined. The resource scheduling of each presentation group for media object pre-fetch time versus buffer occupancy is also examined. Accordingly, a new model called group cascade object composition Petri-net (GCOCPN) is proposed and an algorithm to implement this temporal synchronization scheme is presented.