In medical diagnosis, cone beam CT increases the dose absorbed by a patient. However, the radiographic noise (such as quantum noise) in a CT image increases when radiation exposure is reduced. In this paper, we propose a method to improve the CT image degraded by the quantum mottle based on 2-D wavelet transform modulus sum (WTMS). The noise and regular parts of an image can be observed by tracing the evolution of its 2-D WTMS across scales. Our experimental results show that most of the quantum mottle in the 2-D projections is removed by the proposed method and the edges preserved well. We investigate the relation between the number of X-ray photons and the quality of the denoised images. The result shows the possibility that a patient's dose can be reduced about 10% with the same visual quality by our method.

Wireless LAN (WLAN) systems transmit and receive via a common frequency band. In this band, signals of other wireless applications operate on a WLAN beamformer as interferences, and so the problem in adaptive antenna is increasing the canceling performance in the presence of moving interference sources. The performance of conventional adaptive beamformer is severely degraded and the robust adaptive beamformer must be equipped with additional sensors to obtain desired performances. Therefore, in order to avoid having to install additional sensors, an efficient algorithm is necessary. In this paper, we introduce a fast adaptive algorithm with variable forgetting factor, which does not require any further additional modifications. Through computer simulations, we can obtain better performances than those of other techniques under a variety of operating conditions.

This letter addresses the QoS issue of a RSVP flow during handoff events. For the QoS guarantee of real-time applications with RSVP reservation in All-IP wireless networks, mechanisms are required to minimize the resource reservation path changes and the packet loss resulting from handoff events. If the new RSVP reservation is made along the path via a new base station (BS) in advance for soft handoff, on-going RSVP flow session can be kept with the guaranteed QoS. Therefore, we propose a seamless switching scheme of RSVP branch path for soft handoff. We also show that this scheme could provide the QoS guarantee by adaptively adjusting the pilot signal threshold values for soft handoff.

The proxy mechanism widely used in WWW systems offers low-delay data delivery by means of "proxy server." By applying proxy mechanisms to video streaming system, we expect that high-quality and low-delay video distribution can be accomplished without introducing extra load on the system. In addition, it is effective to adapt the quality of cached video data appropriately in the proxy if user requests are diverse due to heterogeneity in terms of the available bandwidth, end-system performance, and user's preferences on the perceived video quality. In this paper, we propose proxy caching mechanisms to accomplish high-quality and low-delay video streaming services. In our proposed system, a video stream is divided into blocks for efficient use of cache buffer. A proxy cache server is assumed to be able to adjust the quality of cached or retrieved video blocks to requests through video filters. We evaluate our proposed mechanisms in terms of the required buffer size, the play-out delay and the video quality through simulation experiments. Furthermore, to verify the practicality of our mechanisms, we implement our proposed mechanisms on a real system and conducted experiments. Through evaluations from several performance aspects, it is shown that our proposed mechanisms can provide users with a low-latency and high-quality video streaming service in a heterogeneous environment.

Takine investigated the continuous time queueing system with single server and Markovian arrival streams (MAS) where service times of customers depend on the states of the underlying Markov chain immediately before and after arrivals and may differ from one another among streams. In this paper, we consider the discrete version of Takine's model. Since a MAS can be used to model the correlated traffics of several classes, we consider the discrete time queueing system with MAS as an input traffic. Moreover packets from different classes of traffic may require different service times. So service times of customers which are different from one another among streams and may depend on the states of the underlying Markov chain immediately before and after arrivals, are also considered. We obtain the distribution of the workload in the system using Neuts' matrix analytic method. Based on this result, we obtain the distributions of the actual waiting time and the sojourn time of a customer of each class and the joint distribution of the numbers of customers of each class. We also provide a recursion formula for computing the joint queue length distribution. We give an application to multimedia conference system and numerical example.

With the advance of high-speed network technologies, availability and popularity of streaming media contents over the Internet has grown rapidly in recent years. Because of their distinct statistical properties and user viewing patterns, traditional delivery and caching schemes for normal web objects such as HTML files or images can not be efficiently applied to streaming media such as audio and video. In this paper, we therefore propose an integrated caching scheme for streaming media with segment-based caching and hierarchically distributed proxies. Firstly, each stream is divided into segments and their caching algorithms are considered to determine how to distribute the segments into different level proxies efficiently. Secondly, by introducing two kinds of segment priorities, segment replacing algorithms are proposed to determine which stream and which segments should be replaced when the cache is full. Finally, a Web-friendly caching scheme is proposed to integrate the streaming caching with the conventional caching of normal web objects. Performance of the proposed algorithms is verified by carrying out simulations.

This paper proposes a new dynamic multicast routing algorithm for layered streams. Since a layered multicast technique accommodates different types of users in the same multicast group, it helps to provide multicast services in a heterogeneous environment. However, this makes it difficult to construct an efficient routing tree when receivers join or leave a multicast session dynamically. In the proposed algorithm, we adopt a pre-determined path approach to handle such dynamic membership of a layered multicast session without the burden of much additional traffic. Simulation results show that the proposed algorithm can minimize the average multicast tree cost, and that it works well on large-scale networks and those with traffic heterogeneity and a small number of routing control messages.

In this paper, we present a CDN (Content Delivery Network) architecture for mobile streaming service in which content segmentation, request routing, pre-fetch scheduling, and session handoff are controlled by SMIL (Synchronized Multimedia Integration Language) modification. In this architecture, mobile clients simply follow modified SMIL files downloaded from a portal server; these modifications enable multimedia content to be delivered to the mobile clients from the best surrogates in the CDN. The key components of this architecture are 1) content segmentation with SMIL modification, 2) on-demand rewriting of URLs in SMIL, 3) pre-fetch scheduling based on timing information derived from SMIL, and 4) SMIL updates by SOAP (Simple Object Access Protocol) messaging for session handoffs due to client mobility. This architecture enhances streaming media quality for mobile clients while utilizing network resources efficiently and supporting client mobility in an integrated and practical way. The current status of our prototype on a mobile QoS testbed "MOBIQ" is also reported in this paper.

This paper deals with the wave scattering from a periodic surface with finite extent. Modifying a spectral formalism, we find that the spectral amplitude of the scattered wave can be determined by the surface field on only the corrugated part of the surface. The surface field on such a corrugated part is then expanded into Fourier series with unknown Fourier coefficients. A matrix equation for the Fourier coefficients is obtained and is solved numerically for a sinusoidally corrugated surface. Then, the angular distribution of the scattering, the relative power of each diffraction beam and the optical theorem are calculated and illustrated in figures. Also, the relative powers of diffraction are calculated against the angle of incidence for a periodic surface with infinite extent. By comparing a finite periodic case with an infinite periodic case, it is pointed out that relative powers of diffraction beam are much similar in these of diffraction for the infinite periodic case.

A bent-waveguide-based multimode interference (MMI) demultiplexer is designed for the operation at 0.85- and 1.55-µm wavelengths using the three-dimensional semi-vectorial beam-propagation method. First, it is shown that the use of a straight MMI waveguide results in a long coupler length of more than 1000µm for wavelength demulitplexing. To reduce the coupler length, we next introduce a bent MMI waveguide. Bending with a radius of 1500µm leads to a coupler length of less than 200µm. After designing two output waveguides connected to the MMI section, we finally choose a coupler length to be 175µm for efficient demultiplexing properties. Consequently, an output power of more than 90% can be obtained, leading to a low insertion loss of 0.34dB at both 0.85- and 1.55-µm wavelengths. The demultiplexer achieves small polarization dependence, i.e., less than 2dB difference in contrast and 0.02dB difference in insertion loss.

Different quality of service (QoS) requirements must be guaranteed in multimedia code division multiple access (CDMA) mobile communication system to support various applications such as voice, video, file transfer, e-mail, and Internet access. In this paper, we analyze the system in a mixed-traffic environment consisting of voice, stream data, and packet data where preemptive priority is granted to delay-intolerant voice service and a buffer is offered to delay-tolerant stream data services. In the case of best-effort packet data service, the probability of access control by transmission permission is applied to obtain throughput improvement. To analyze the multimedia CDMA mobile communication system, we built a two-dimensional Markov chain model on high-priority voice and stream data services, and then performed numerical analyses in conjunction with packet data services based on a residual capacity equation. We investigate the performance of an integrated voice/stream-data/packet-data CDMA mobile system with the finite buffer size of stream data in terms of voice service blocking probability, average stream data service delay, average packet data service delay, and throughput.

In this paper, we present a novel beam-former capable of tracking a rapidly moving speaker in a very noisy environment. The localization algorithm extracts a set of candidate direction-of-arrival (DOA) for the signal sources using array signal processing methods in the frequency domain. A minimum variance (MV) beam-former identifies the speech signal DOA in the direction where the signal's spectrum entropy is minimized. A fine tuning process detects the MV direction which is closest to the initial estimation using a smaller analysis window. Extended experiments, carried out in the range of 20-0 dB SNR, show significant improvement in the recognition rate of a moving speaker especially in very low SNRs (from 11.11% to 43.79% at 0 dB SNR in anechoic environment and from 9.9% to 30.51% in reverberant environment).

We present an efficient stream authentication scheme using authentication stars. The computation overhead of the proposed scheme on the sender is almost the same as that of the scheme with the smallest overhead. On the receiver's side, the verification probability of the proposed scheme is much higher than that of any other scheme. To show this, we first conducted a mathematical analysis on the verification probability of our scheme and then performed simulation to compare the verification probability of our scheme with those of the previous schemes. Simulation results shows that when the packet loss rate is 50%, the verification probability of our scheme is 73% whereas those of the previous schemes are below 41%.

This paper reviews the historical development of adaptive antennas in Japan. First of all, we watch basic adaptive algorithms. In 1980s, particularly, the following issues were a matter of considerable concern to us; (a) behavior to the coherent interference like multipath waves or radar clutters, (b) signal degradation in case that the direction of arrival (DOA) of desired signal is different from the DOA specified beforehand in the adaptive antennas with the DOA of the desired signal as a prior knowledge, and (c) performance of adaptive antennas when the desired signal and interference are broadband. Although there are a lot of development and modification of adaptive algorithms in Japan, we refer in this paper only to the above-mentioned topics. Secondly, our attention is paid to implementation of adaptive antennas and advanced technologies. A large number of researches on the subjects have been carried out in Japan. Particularly, we focus on the initiative studies in Japan toward mobile communication application. They include researches of mobile radio propagation for adaptive antennas, calibration methods, and adaptive antenna for mobile terminals. As a matter of course, we also refer to adaptive antenna technologies for advanced communication schemes such as CDMA, SDMA, OFDM and so on. Finally, we take notice of some pilot products which were developed to verify the effect of the adaptive antenna in the practical environments. As the initiative ones, a couple of equipments are introduced in this paper.

In this paper, we describe an adaptive integration method for an audio-visual speech recognition system that uses not only the speaker's audio speech signal but visual speech signals like lip images. Human beings communicate with each other by integrating multiple types of sensory information such as hearing and vision. Such integration can be applied to automatic speech recognition, too. In the integration of audio and visual speech features for speech recognition, there are two important issues, i.e., (1) a model that represents the synchronous and asynchronous characteristics between audio and visual features, and makes the best use of a whole database that includes uni-modal, audio only, or visual only data as well as audio-visual data, and (2) the adaptive estimation of reliability weights for the audio and visual information. This paper mainly investigates two issues and proposes a novel method to effectively integrate audio and visual information in an audio-visual Automatic Speech Recognition (ASR) system. First, as the model that integrates audio-visual speech information, we apply a product of hidden Markov models (product HMM), the product of an audio HMM and a visual HMM. We newly propose a method that re-estimates the product HMM using audio-visual synchronous speech data so as to train the synchronicity of the audio-visual information, while the original product HMM assumes independence from audio-visual features. Second, for the optimal audio-visual information reliability weight estimation, we propose a Gaussian mixture model (GMM) based-MCE-GPD (minimum classification error and generalized probabilistic descent) algorithm, which enables reductions in the amount of adaptation data and amount of computations required for the GMM estimation. Evaluation experiments show that the proposed audio-visual speech recognition system improves the recognition accuracy over conventional ones even if the audio signals are clean.

It is believed that distant-talking speech recognition in a noisy environment requires a large-scale microphone array. However, this cannot fit into small consumer devices. Our objective is to improve the performance with a limited number of microphones (preferably only left and right). In this paper, we focused on a profile that is the shape of the power distribution according to the beamforming direction. An observed profile can be decomposed into known profiles for directional sound sources and a non-directional background sound source. Evaluations confirmed this method reduced the CER (Character Error Ratio) for the dictation task by more than 20% compared to a conventional 2-channel Adaptive Spectral Subtraction beamformer in a non-reverberant environment.

We propose a new algorithm for blind source separation (BSS), in which independent component analysis (ICA) and beamforming are combined to resolve the low-convergence problem through optimization in ICA. The proposed method consists of the following two parts: frequency-domain ICA with direction-of-arrival (DOA) estimation, and null beamforming based on the estimated DOA. The alternation of learning between ICA and beamforming can realize fast- and high-convergence optimization. The results of the signal separation experiments reveal that the signal separation performance of the proposed algorithm is superior to that of the conventional ICA-based BSS method.

This letter presents the implementation framework of a video streaming server which uses an optical disk library as a source of media archiving. In order to handle the optical storage subsystem in the framework of disk-based stream service model, we have devised an effective stream scheduling, disk caching, and admission control mechanism. The proposed system has been implemented and its key principles are validated with real experiments.

Unicasting video streams over TCP connections is a challenging problem, because video sources cannot normally adapt to delay and throughput variations of TCP connections. This paper describes a method of extending TCP so that TCP connections can effectively carry hierarchically-encoded layered video streams, while being friendly to other competing connections. We call the method Receiver-based Delay Control (RDC). Under RDC, a TCP connection can slow down its transmission rate to avoid congestion by delaying ACK packet generation at the TCP receiver based on congestion notifications from routers. We present the principle behind RDC, argue that it is TCP-friendly, describe an implementation that uses 1-bit congestion notification from routers, and demonstrate by simulations its effectiveness in streaming hierarchically-encoded layered video.

Transmission characteristics of a high permittivity NRD guide were investigated. A preferable operating mode of the high permittivity NRD guide was newly identified and the wide bandwidth and low loss nature of the millimeter-wave region were observed. Moreover, a technique for construction of a millimeter-wave antenna was developed based on the high permittivity NRD guide. The novelty of the present technique lies in the use of a simple radiator, which consists of a tapered dielectric strip of simple structure which has good compatibility with millimeter wave integrated circuits. Since this radiator has a broad radiation pattern, a new type of antenna compatible with millimeter-wave integrated circuits for marine radar use was fabricated by locating the radiator at the focal point of a cylindrical parabolic reflector. Suitable beam patterns with half-power beam widths of 4in the azimuth plane and 38in the elevation plane can be obtained at 35 GHz.