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.

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.

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%.

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).

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.

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.

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.

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.

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.

Adaptive antenna is a promising to increase the spectral efficiency of mobile radio systems. We developed a compact, cost effective planar Butler Matrix as a beam forming network of a multi beam antenna. This circuit consists of a thin substrate that the conductor attaches to both sides, and two thick substrates that the ground conductor attaches to one side. In this circuit, coupling by crossover causes amplitude and phase error of the Butler Matrix. By narrowing the strip width of the crossover, crossover coupling can be suppressed 10 dB. The measurement results of the experimental 88 Butler Matrix were 0.75 dB amplitude deviation, 9.5 degree phase deviation and VSWR of less than 1.15 within the relative bandwidth of 10% at 900 MHz band.

To support video streaming over the current Internet, a media server usually needs to perform adaptive streaming in combination with congestion control. While existing streaming schemes are mainly designed for particular congestion algorithms and rate shaping techniques, this paper proposes a general buffer-driven adaptive streaming scheme based on control theory. Our scheme can be applied in combination with different congestion control algorithms and different rate shaping (and source coding) techniques. It is also amenable to analysis and easy to implement. Analytical and simulation results show that the proposed scheme gives satisfactory video quality under various Internet conditions while fully utilizing the available network bandwidth.

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.

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.

This paper describes an analytical method to estimate the amount of interference in multibeam non-geostationary satellite systems. The performance of CDMA is compared with that of FDMA (or TDMA) by employing the maximum acceptable number of users per cell in uplink as a measure. Numerical examples shows that the maximum acceptable number of users in FDMA (or TDMA) systems varies according to the altitude of the satellites, while the performance is insensitive to the altitude in CDMA systems. Then, it is found that the superior multiple access scheme depends on the altitude of the satellite.

A slot pair array using a post-wall waveguide is a promising candidate for a Fixed Wireless Access (FWA) sector antenna to be used in a base station. This array is formed by a traveling wave antenna, and therefore its main beam direction varies with frequency. To overcome this difficulty, we propose a new structure that comprises of a cosecant array and an additional Talor array. This structure can fix the main beam in a constant direction whilst maintaining a cosecant radiation pattern. We conducted an investigation based on an array factor, and the validity of the method was confirmed by experiment.

This paper presents a beamspace-time transmit diversity scheme that uses a space-time block code (STBC) and a fixed multi-beam transmit array with low sidelobes for time-domain spreading orthogonal frequency-division multiplexing code-division multiplexing (OFDM-CDM) downlink transmission. The scheme assigns space-time-coded signals to a pair of neighboring beams via closed-loop beam selection. Time-domain spreading provides non-frequency selectivity in each spreading region, which makes it possible for multiple STBCs to share any beam and to be decoded after despreading. Simulation results demonstrated that multiple transmit beams and multiple receive antennas provide large beam gains and/or a high order of diversity gains. In addition, the proposed scheme spatially separates users by beam and thus alleviates multi-user interference.

This paper presents architecture design, FPGA implementation, and measurement results of a real-time signal processing circuit for WCDMA uplink baseband receiver. To enhance uplink signal-to-interference-plus-noise ratio (SINR) performance, a four-element antenna array and a four-finger Rake combiner are integrated in the proposed receiver. Moreover, a low-complexity beamforming architecture using a correlator-based beam searcher, a decision-directed carrier synchronization loop, and a matched-filter based channel estimator is also designed. Simulations are based on the standard Doppler-fading scalar channel models provided by 3GPP and an extension to vector channel models that specify angle of arrival for each path is also made for beamformer simulation. Simulation and hardware emulation results show that the proposed architecture meets the specified requirements. In addition, this architecture, with its correlator-based beamformer weights, achieves such performance improvement with relatively low hardware complexity.

Adaptive beamforming, using the Conjugate Gradient Normal Equation Residual problem, is applied to a base station array, in the UTRA-TDD up-link. A Wideband Directional Channel Model is used, characterising specific micro-cell, street-type scenarios. These differ in the number of mobile terminals, grouped and placed along the street axis, and on their distances to the base station. Time- and angle-of-arrival spreads, and on-the-air interference content are the main parameters inherently varied and analysed. The average beamforming gain and signal-to-noise ratio are evaluated, also varying the number of array elements. The high number of arriving correlated and closely correlated signals, together with the composed nature of the correlation matrix in the algorithm's cost function, result in that other than the MMSE solutions may lead to the best interference suppression, for the tested scenarios. Among the several weighted interfering power components, the most relevant is due to the delayed signals from all the other links. The combination of the number of arriving orthogonal codes, time-of-arrival and angle-of-arrival spreads condition beamforming performance: the number of array elements affect performance, depending on the mobile terminal distance to the base station, and on the number of active links; for short distances and large number of users, larger time-of-arrival spread degrades beamformer performance, over the opposing effect of angle-of-arrival spread; the number of active users affects beamforming gain especially in the case that delay spreads are larger, i.e., for shorter mobile terminal distances to the base station.