This paper describes a factorization-based algorithm that reconstructs 3D object structure as well as motion from a set of multiple uncalibrated perspective images. The factorization method introduced by Tomasi-Kanade is believed to be applicable under the assumption of linear approximations of imaging system. In this paper we describe that the method can be extended to the case of truly perspective images if projective depths are recovered. We established this fact by interpreting their purely mathematical theory in terms of the projective geometry of the imaging system and thereby, giving physical meanings to the parameters involved. We also provide a method to recover them using the fundamental matrices and epipoles estimated from pairs of images in the image set. Our method is applicable for general cases where the images are not taken by a single moving camera but by different cameras having individual camera parameters. The experimental results clearly demonstrates the feasibility of the proposed method.

When multiple flows including continuous media streams are simultaneously sent from a computer, allocation and management of both processor capacity and network bandwidth need to be considered. We propose a framework of Quality of Service (QoS) management inside a sending host that controls execution of sending threads in consideration of utilization of processor capacity and network bandwidth. To distinguish from flows which require only best-effort service, we call a flow which requires a specific rate of service "reserved flow. " To guarantee QoS of such reserved flow both in processor- and network-intensive cases in a sending host, processor capacity reserve is allocated such that the rate of each reserved flow is attained and non-conforming data are policed before they are transmitted. Processor Capacity Manager and the network device driver exchange information in a cooperative manner to support the rate adaptive allocation of processor capacity reserve. In this paper, we describe design and implementation of our framework on RT-Mach. The results of performance evaluations demonstrate that our scheme performs well for full-duplex Ethernet.

Low bit-rate speech and audio codings are key technologies for multimedia communications. A number of coding scheme have been developed for various applications. In Internet application, good speech and audio quality at very low bit-rate (8-16 kb/s) is valuable. Two recently proposed speech and audio-coding schemes, CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Prediction, standardized by the ITU-T in Recommendation G. 729) and TwinVQ (Transform-domain Weighted INterleave Vector Quantization, one of the candidates for MPEG-4 audio) were compared from the viewpoints of coding schemes and quality. Although there are significant differences in their basic structures and frame lengths, this paper describes that both use the same compression techniques, such as LPC (Linear Predictive Coding)-analysis pitch-period estimation and vector quantization. While CS-ACELP provides toll quality for speech at 8 kb/s, the quality it provides for music signals is insufficient. The TwinVQ transform coder is based on LPC and vector quantization and is also capable of operating at 8 kb/s. Evaluation of these two schemes in terms of their fundamental technologies, quality, delay, and complexity showed that the quality of TwinVQ for music signals is better than that of CS-ACELP, and that the quality of CS-ACELP is better for speech signals. Therefore, TwinVQ may be better suited for one-directional Internet applications, and CS-ACELP may be better for two-directional communication.

Priority control and call admission control are indispensable traffic management methods to guarantee each QoS requirement of connections in ATM networks. The key technique of call admission control under priority control is to estimate required bandwidth of each connection to satisfy all QoSs of calls in progress. In this paper, we propose a novel approximation method to calculate the required bandwidth of ATM connections through priority queues and show a practical call admission control scheme using the proposed method. The essence of the approximation method is to model prioritized parallel queues as a series of queues in tandem with no priority control by focusing on the number of cells in queues. The tandem queue approximation method enables us to model each queue under priority control as a single non-priority FIFO queue in terms of its queue length. This results in that effective bandwidth techniques are applicable to priority queues. The effectiveness of the proposed scheme is evaluated by some numerical examples.

In this paper, Orthogonal Multicode OFDM-DS/CDMA system using Partial Bandwidth Transmission is proposed. By using the flexible carrier allocation of OFDM, Partial Bandwidth Transmission is considered for high quality communication. Furthermore, multicode packet data transmission is presented. Multicode packet data transmission is very effective to handle variable data. Since the proposed system can detect the header information without complex control, it is also suitable for packet data transmission. The computer simulation results show that the BER performance of the proposed system with the ideal channel estimation is improved compared with the case of the conventional Orthogonal Multicode DS/CDMA system with ideal RAKE receivers. Moreover the proposed system with the channel estimation by MLS algorithm also shows the good BER performance. In packet data transmission, the delay and throughput performances are also improved in the proposed system.

A simple near-orthogonal code is used as frequency-hopping patterns for the frequency-hopped multiple access systems. Extended RS code is used as channel coding to deplete the effects of hits from simultaneous users. Packet error probability and channel throughput for the system utilizing the near-orthogonal code are evaluated and compared to the corresponding values obtained from the system utilizing random patterns. Results show that the former can provide substantial improvement over the latter. In our illustrated examples, we also show that under the constraint of packet error probability PE 10-2, the maximum achievable number of users with most (n,k) RS codes of interest is less than the number of distinct codewords in the near-orthogonal code. Thus, the number of codewords of the near-orthogonal code is large enough to support the practical application.

We have studied the electro-optical properties in Polymer Network Liquid Crystal Displays which are composed of a three-dimensional polymer network formed in continuous liquid crystal layers prepared by photo-polymerization induced phase separation processes. In view of the behavior in the orientation of the liquid crystal molecule at the polymers surface, it can be clarified that the electro-optical properties are strongly governed by the polymer. By a combination of two different characteristics of the polymers with respect to a temperature dependence, a variation in the driving voltage for a temperature change can be improved.

The performance of AC plasma displays has been improved in the area of brightness and contrast, while significant advances in image quality are still required for the HDTV quality. In particular, in full color motion video, motion artifacts and lack of color depth are still visible in some situations. These motional artifacts are mitigated as the number of the subfields increases, usually at the cost of losing brightness or increasing driving circuitry. Therefore, it is still one of our great concerns to find out the optimized subfield configuration through weighting and order of each subfield, and their coding of combination. For evaluation and improvement of motion picture disturbance, we have established a procedure that fully simulates the image quality of displays which utilize the subfield driving scheme. The simulation features virtually located sensor pixels on human retina, eye-tracking sensor windows, and a built-in spatial low pass filter. The model pixelizes the observers retina like a sensor chip in a CCD camera. An eye-tracking sensor window is assigned to every light emission from the display, to calculate the emissions from one to four adjoining pixels along the trajectory of motion. Through this model, a scene from original motion picture without disturbance is transformed into the still image with simulated disturbance. The integration of the light emission from adjoining pixels through the window, also functions as a built-in spatial low pass filter to secure the robust output, considering the MTF of the human eye. Both simulation and actual 42-in-diagonal PDPs showed close results under various conditions, showing that the model is simple, but reasonable. Through the simulation, general properties of the subfield driving scheme for gray scale have been elucidated. For example, a PWM-like coding offers a better performance than an MSB-split coding in many cases. The simulation also exemplifies the motion picture disturbance as a non-linear filter process caused by the dislocation of bit weightings, suggesting that tradeoffs between disturbance and resolution in motion area are mandatory.

A method to improve the reflection efficiency of holographic polymer dispersed liquid crystal (HPDLC) is proposed and its effectiveness is confirmed. Controlling the alignment of liquid crystal (LC) in tiny droplets of HPDLC can increase the refractive-index difference between the LC droplet layer and the polymer layer, causing the peak reflectance and reflective spectral width to expand. We observed experimentally that 96% of the light components excluding the scattering loss can be diffracted in a transmission HPDLC device by ordering the LC. In a reflection HPDLC, we found that reflection could be improved by ordering through an applied shear force. Our findings should lead to an improvement in the quality of reflective display devices.

This paper proposes a novel asynchronous direct sequence/code division multiple access (DS/CDMA) communication system using analog pseudo noise (PN) sequences that have an orthogonal relation for all active users. Analog PN sequences are produced by an adaptive filter called a "code-orthogonalizing filter" (COF). In a base station receiver, the tap coefficients of the COF can be adaptively controlled "to orthogonalize" or "to approach to orthogonalize" various received PN sequences. The elements of the analog PN sequences consist of the tap coefficients of the COF. The analog PN sequence produced is assigned to the transmitter of each user in order. As a result, multiple access interference (MAI) caused by other users can be reduced considerably, and multiple access capacity increased by the proposed system compared with matched filter (MF) reception and COF reception.

In this paper, the bit error rate (BER) performance of the Spread Spectrum communication with Constrained Spreading Code system is studied under the synchronous CDMA system. Particularly, BER considering the tracking error is derived by theoretical analysis. The synchronizing spreading sequence is employed to track the signals in the receiver. As the result, the BER performance is degraded by increasing the number of users. However, the BER performance can be improved by canceling the co-channel interference and by suppressing the cross-correlation value between the information spreading sequence and the synchronizing spreading sequence.

M-ary/SS systems are compared with DS/SS systems applied the multicarrier systems and the RAKE receiver in the presence of AWGN, frequency selective Rayleigh fading and partial-band interference. In particular, the BER performance and SSMA capability are evaluated. Consequently, the M-ary/SSMA system using the multicarrier techniques is subject to the M-ary/SSMA system with the RAKE receiver in the presence of partial-band interference. The BER performance of the M-ary/SSMA system is better than that of the DS/SSMA system when the number of users is smaller than 20. And the spectral efficiency of the multicarrier M-ary/SSMA system is best in the four systems when JSR=20 [dB] and BER=10-3.

In this paper, we present a numerical design method for two-dimensional (2-D) FIR linear-phase (LP) quincunx filter banks (QFB) with equiripple magnitude response and perfect reconstruction (PR). The necessary conditions for the filter length of analysis filters are derived. A dual affine scaling variant (DASV) of Karmarkar's algorithm is employed to minimize the peak ripples of analysis filters and an approximation scheme is introduced to satisfy the PR constraint for the 2-D filter banks (FB). The simulation examples are included to show the effectiveness of this proposed design technique.

This paper explores a relationship between parameters for the context tree weighting and weights for a general model weighting technique. In particular, an algorithm is proposed that approximately computes the parameters from the weights, and a condition under which no error for the approximation occurs is derived.

This paper formulates problems of fitting two corresponding sets of points by translation, rotation and scaling, and proposes efficient algorithms for the fitting. The algorithms are based on the theory of lower envelopes, or Davenport-Schinzel sequences, and linearization techniques in computational geometry, and are related to dynamic furthest Voronoi diagrams.

This paper presents a new approach to computing symbol error probability of fast frequency-hopped M-ary frequency shift keying (FFH/MFSK) systems with majority vote under multitone jamming. For illustrating the applications, we first consider the case in which the source data rate is fixed and the hopping rate is allowed to vary. In this case, the optimum orders of diversity for several values of M are examined. Results show that M=4 outperforms other values. Then, we treat another case in which the hopping rate is fixed and the data rate is adjusted so as to obtain maximum throughput under a given constraint of error probability. In addition to the case of diversity alone, we also evaluate the performances of the fixed hopping rate case with channel coding using convolutional code and BCH code.

Unequal error protection (UEP) is a very promising coding technique for satellite broadcasting, as it gradually reduces the transmission rate. From the viewpoint of bandwidth efficiency, UEP should be achieved in the context of multilevel coded modulation. However, the conventional mapping between encoded bits and modulation signals, usually realized for multilevel block modulation codes and multistage decoding, is not very compatible with UEP coding because of the large number of resulting nearest neighbor codewords. In this paper, new coded modulation schemes for UEP based on unconventional partitioning are proposed. A linear operation referred to as interlevel combination is introduced. This operation generalizes previous partitioning proposed for UEP applications and provides additional flexibility with respect to UEP capabilities. The error performance of the proposed codes are evaluated both by computer simulations and a theoretical analysis. The obtained results show that the proposed codes achieve good tradeoff between the proportion and the error performance of each error protection level.

In this paper, we study unequal error protection (UEP) capabilities of punctured convolutional codes. For constructing the good UEP convolutional codes, the conditional weight distributions of UEP convolutional codes are defined and evaluated. The conditional weight distributions are computed by using the transfer functions of time-varying trellis structures of punctured convolutional codes. The best UEP convolutional codes from the viewpoint of the weight distributions are listed.

This paper proposes and investigates a coding and decoding scheme to achieve adaptive unequal error protection (UEP) using several convolutional codes which have different error-correcting capabilities. An appropriate encoder is selected to unequally protect each frame of information sequence according to the importance of the frame. Since the supplemental information of selected encoder is not sent for the sake of reducing redundancy, we assume that the decoder does not know which encoder was used, and the decoder has to estimate the used encoder. In order to estimate which encoder was used, the method using biased metric in Viterbi decoding is proposed. In decoding, however, there is a problem of Decoder-Selection-Error (DSE), which is an error that the decoder selected in a receiver does not correspond to the encoder used in a transmitter. An upper bound of DSE rate in decoding is derived. The proposed decoding scheme using the biased metric in a trellis can improve DSE rate and BER performance, because transition probability of encoders is taken into account in calculating likelihood by means of making branch or path metric biased. Computer simulation is employed to evaluate the BER performance and DSE rate of the proposed scheme. The performance is compared with a conventional equal error protection scheme and a UEP with the supplemental information on the used encoder. It is found that the proposed scheme can achieve better performance than them in case N=2.

This paper proposes a high-speed connection admission control scheme, named PERB CAC (CAC based on Prior Estimation for Residual Bandwidth). This scheme estimates the residual bandwidth in advance by generating a series of virtual requests for connection. When an actual connection request occurs, PERB CAC can instantaneously judge if the required bandwidth is larger than the estimated residual bandwidth, so the connection set-up time can be greatly reduced. Therefore, PERB CAC can realize high-speed connection set-up.