In this paper, we present a set of sequence pairs which produce zero correlation windows not only in the middle part of the sum of aperiodic correlation functions, but also in the two terminal parts. We name it "Ear Windows". In asynchronous CDMA communication system, this set of sequences appears to be completely removing the intersymbol interference (ISI) and multiuser interference (MUI) caused by the multipath effect if the maximum delay is shorter than the length of the "Ear Windows". We can also find application in M-ary modulation. The intercode interference will be mitigated drastically.

We propose a motion adaptive sensor for image enhancement and wide dynamic range sensing. The motion adaptive sensor is able to control integration time pixel by pixel. The integration time is determined by saturation and temporal changes of incident light. It is expected to have high temporal resolution in the moving area, high SNR in the static area, and wide dynamic range. We have fabricated a prototype and show some results obtained by our experiments.

We propose a new sampling control system integrated on an image sensor. Contrary to the conventional random access pixels, the proposed sensor is able to read out spatially variant pixels at high speed, without inputting pixel address for each access. The sampling positions can be changed dynamically by rewriting the sampling position memory. Since the proposed sensor has an array memory that keeps the pixel position to be sampled. The sampling position can be dynamically changed by rewriting the memory array. It can achieve any spatially varying sampling patterns. We have made a first prototype and show results obtained by the prototype.

In order to enhance performance of an image sensor, we have been investigating a novel integration of compression and sensing. By this integration, the image signal that has to be readout from the sensor is significantly reduced. Thus, the integration can consequently leads to high pixel rate sensing. The compression scheme we make use of is conditional replenishment that detects and encodes moving areas. We have developed prototypes based on two different architectures, that are pixel parallel and column parallel architectures. We present the two prototypes and their comparisons, and show the results obtained by them.

In this paper, we propose a novel image sensor which compresses image signals on the sensor plane. Since an image signal is compressed on the sensor plane by making use of the parallel nature of image signals, the amount of signal read out from the sensor can be significantly reduced. Thus, the potential applications of the proposed sensor are high pixel rate cameras and processing systems which require very high speed imaging or very high resolution imaging. The very high bandwidth is the fundamental limitation to the feasibility of those high pixel rate sensors and processing systems. Conditional replenishment is employed for the compression algorithm. In each pixel, current pixel value is compared to that in the last replenished frame. The value and the address of the pixel are extracted and coded if the magnitude of the difference is greater than a threshold. Analog circuits have been designed for processing in each pixel. A first prototype of a VLSI chip has been fabricated. Some results of experiments obtained by using the first prototype are shown in this paper.

We propose a novel concept of an integration of compression and sensing in order to enhance performance of an image sensor. By integrating the compression function on the sensor plane, the image signal that has to be read out from the sensor is significantly reduced. Thus, the integration can consequently increase the pixel rate of the sensor. The compression scheme we make use of is conditional replenishment that detects and encodes moving areas. In this paper, we discuss design and implementation of two architectures for on-sensor compression. One is a pixel parallel approach and the other is a column parallel approach. We describe both approaches and design and a prototype of pixel parallel architecture.

In this paper, we propose a novel method of all-focused image acquisition using multiple differently focused images. Based on the assumption that depth of the scene changes stepwise, we derive a formula for reconstruction between the desired all-focused image and multiple acquired images; we can reconstruct the all-focused image by iterative use of the formula. We also introduce coarse-to-fine estimation of point spread functions of the acquired images. We show we can reconstruct an all-focused image for a natural scene.

Wireless ATM networks are motivated by an anticipated demand for wireless extensions of the integrated multimedia services. A TDMA-based wireless ATM is presented for the integration of wireless access with wired B-ISDN/ATM networks. The presented system is characterized by a reservation-based access protocol to handle diverse services and a dynamic duplexing scheme for TDD to support uplink/downlink asymmetric traffic. Resource allocation algorithms which aim at prorating the shared radio resources among ABR message traffic such as files and e-mails are also presented. Simulation results show that the equal sharing algorithm where the resources are equally shared among all mobile terminals can provide a higher average and lower variance of the throughput than the FIFO-based algorithm.

This paper presents a real-time VBR (variable bit rate) transport for next-generation multiservice personal communication networks. After discussing the characteristics of real-time VBR traffic, we propose a statistical multiplexing-based resource allocation scheme for real-time VBR traffic: "guaranteed" resources are assigned to high-priority packets, while "add-on" resources are assigned in a dynamic manner. The proposed scheme have been validated for an integrated voice/video environment, where video is encoded with a H.261-based two-layer video coder. The protocol was designed with the basic premise that there is a relatively large number of unused voice slots on average that can be used to carry "add-on" video packets. Numerical examples show that the prioritized transport under consideration can significantly improve the picture quality, and allow better utilization of wireless resources through statistical multiplexing.

A novel method of providing eye-contact for multiple participants in a teleconferencing system is presented. This method proposes a simple idea to fulfil the requirement of eye contact, for two or more local and remote participants simultaneously, by using half mirrors and cameras placed at the common points of the extended lines of gaze of each participant. In contrast to the commonly available one to one eye contact video phone systems, which also use half mirrors, the proposed system has the advantage of serving more than one person at a given location, hence named the multiple person eye contact (MPEC) teleconferencing system. We observed the resultant images by using a preliminary experiment and also building a prototype system. The results from both of these experiments proved the merits of the proposed idea.

In this paper a novel method of providing eye contact for multiple participants in a teleconferencing system is presented. This method proposes a simple idea to fulfill the requirement for eye contact, for two or more local and remote participants simultaneously, by using half mirrors and cameras placed at the common points of the extended lines of gaze of each participant. In contrast to the commonly available one to one eye contact of video phone systems, which also use half mirrors, the proposed system has the advantage of serving more than one person at a given location, hence named multiple person eye contact (MPEC) teleconferencing system. We observed the resultant images by using a preliminary experiment and also by building a prototype system. The results from both of these experiments proved the merits of the proposed idea. One of the main features of this system is that, it gives any participant, a feeling of "being looked at" if any remote participant is actually gazing at him. On the other hand if none of the remote participants are looking at him, then he gets the feeling of "not being looked at" which is also extremely important for effective inter personal communication. Furthermore, this system preserves the spatial continuity of neighboring participants, since all the participants at a given location are captured by each video camera.>

A scheme of motion estimation using multiple image sensors is proposed, in which the multiple image sensors work at different integration intervals or different sampling instances. Because combined use of multiple image sensors virtually increases the temporal resolution while keeping signal to noise ratio high enough, the accuracy of the estimation can be improved. Both gradient-based and block-matching-based methods are formulated for the images obtained by the multiple image sensors. Tile experimental results show that the proposed scheme significantly improves the accuracy and reduces computational complexity.>

A new structure of fractionally spaced equalizer (FSE) is described. A conventional FSE has taps spaced equally T/M, where T is the symbol spacing and M is an integer. Although there are many advantages of FSE in digital data transmission, the computational requirements increase with factor M. We describe a non-equally spaced FSE (NFSE) with less computational requirements. A novel tap control algorithm for NFSE under some restrictions is also presented.>

We propose a method of using face direction information to make the local participant's video display of a video conferencing environment change according to his direction of view, in order to give him/her (the local participant) an improved visual sensation. We have named this method "steerable viewing window" (SVW). We also discuss an image processing method used to detect a person's face direction, without using special head mounted equipment. Further, we propose a method of reducing the data rate for transmission by coding the area around the gaze center with high resolution and the remaining with a lesser resolution. This method of saving bytes in video conferencing is named line of gaze centered coding (LOGCC).>

A novel image sensor on which video signals can be compressed is proposed. Since the video signal is compressed on an imager plane by using fast analog processing, the amount of image data read out from the imager can be significantly reduced with very small latency. The proposed system can be potentially applied to high pixel rate cameras such as those for high speed imaging and high resolution imaging. Conditional replenishment is employed for the video compression algorithm. Analog circuits are designed both for processing in each pixel and for controlling the entire data rate. The behavior of the circuit is investigated on the basis of both an analog circuit simulator and a scale-up-circuit. A VLSI chip is designed and is under fabrication.>

The authors describe a method for estimating camera parameters from image sequences for application to emerging model-based video coding systems. The parameters to be estimated include focal length, zoom, and 3-D rotation parameters. The method consists of first establishing a correspondence and then, estimating the parameters by fitting the correspondence data to a transformation model based on a perspective mapping model and a 3-D rotation and zoom operation model. They show by simulations and experiments that the proposed method successfully estimates the focal length from the image sequences, it explains very well the induced motion field of images undergoing camera operation (3-D rotation and zoom), and that it significantly outperforms conventional estimation methods, especially for wide-angled images. It is anticipated that the proposed method will be successfully applied to compensating for the motion field induced by camera operation in extracting a 3-D object model and a 3-D object motion, and to synchronizing the viewing direction and scale of an image, in model-based video coding technology.>

VIWOB is an integrated environment for programming and executing simulations interactively on distributed computational resources. A visual dataflow oriented language allows a user to assemble application programs by simply connecting processing nodes. Single nodes can be described by subgraphs (modules) permitting the implementation of complex programs in a hierarchical structured manner, as well as taking advantage of previously developed modules, while hierarchical ordered libraries provide convenient access to the available modules. The run-time environment will break up the application and execute the components transparently on a distributed computing facility. An abstract computational model simplifies adding new primitive modules.>

In the realization of code-division multiple access based on a spread-spectrum communication system, i.e. spread-spectrum multiple access (SSMA), reduction of cochannel interference is an important problem. An adaptive array antenna system is proposed that includes a cancellor of cochannel interference, which can improve performance by a combination of temporal and spatial filtering. While the adaptive array suppresses interference sources with arrival angles different from those of the desired user, the adaptive digital filter-canceller rejects those whose arrival angles are the same as those of the desired user. The proposed system can achieve stable acquisition and low error rate of demodulated data even in a heavy-interference channel where a conventional array antenna system cannot achieve satisfactory acquisition.>

The authors propose and investigate an adaptive canceller of intersymbol and cochannel interference due to channel distortion and cross-correlation among pseudonoise sequences assigned to individual users of a DS-SSMA (direct-sequence spread-spectrum multiple-access) system. In order to implement a local area network (LAN) by using a power line installed in a building wall as a transmission channel, the authors have investigated utilization of DS-SSMA which has advantages such as robustness against narrow-band interference and noise and realization of asynchronous code division multiple access. In a power line, however, restriction of transmission bandwidth for communications makes it difficult to suppress cochannel interference and the channel is also time-varying due to fluctuation of loads. Since the proposed canceller adaptively eliminates cochannel interference as well as intersymbol interference, it can facilitate synchronization and increase the number of the simultaneously accessing users on a power line with restricted processing gain. The error probability in the output of the canceller is theoretically calculated for the steady-state case by using a Markov model. Computer simulations illustrate stable convergence properties of the canceller.>

A set of N-1 orthogonal sequences of period N/sup 2/ is proposed, where N is a natural number. Each orthogonal sequence proposed can be modulated by N complex numbers of absolute value 1, so the modulated sequence is also orthogonal. When N is an odd prime number, the absolute value of the cross-correlation function between any two of the N-1 orthogonal sequences is constant and satisfies the mathematical lower bound. This property of the cross-correlation function is not changed when each of the two orthogonal sequences is modulated by N complex numbers of absolute value 1. Two spread-spectrum multiple-access (SSMA) systems using these sequences are proposed. One system is an asynchronous SSMA system, using the proposed sequences unmodulated. The cochannel interference peak between any two channels in this system realizes the mathematical lower bound for an asynchronous SSMA system using a set of orthogonal sequences. The other system is a synchronous SSMA system without cochannel interference which uses the modulated form of the proposed sequences.>

N-shift cross-orthogonal sequences and a complete complementary code derived from them are defined and discussed. A general method for generating this code is also discussed. A synchronous multiple-user spread-spectrum multiple system is proposed that uses N-shift cross-orthogonal sequences.>

The tap selecting digital adaptive FM multipath distortion canceller is described. In order to implement hardware of FM multipath distortion canceller, the convergence property of it is examined by computer simulation with finite bit length variables. As the result, a good DU ratio and fast convergence speed is confirmed. Moreover, simplification of hardware and structure of it is examined in practice.

This paper proposes a new digital filter adaptation algorithm, suitable for those applications where the required numberMof active taps is considerably smaller than the total numberNof possible tap positions. The tap positions are selected according to the "scrub taps waiting in a queue (STWQ)" tap selection algorithm, which is proposed in this paper. With this algorithm, the amount of multiplications and additions can be reduced by a factor ofM/N, because the number of operations is proportional toM. The conditionM\llNmay be satisfied in applications, such as echo canceller, FM multipath distortion canceller (FM-MPDC), etc. The above mentioned applications have been investigated by computer simulation.

Effective utilization of an error-correcting code (ECC) is proposed in order to make a low-hit-rate adaptive differential PCM (ADPCM) system robust against bit errors in noisy channels, such as mobile radio. In 16 kb/s ADPCM systems, the inverse-quantizers with a small number (four) of levels lead to large offsets in stepsizes due to channel errors, and high-order all-pole predictors are very sensitive to them. For the purpose of solving these problems, we present a robust ADPCM system using an ECC, in which the reliability of words decoded by the ECC decoder is utilized to control the updating of the inverse-quantizer and also to stabilize the predictor in the receiver. Moreover, on condition that a sampling rate of the system is reduced from 8 to 6.4 kHz, the proposed system is compared with a forward adaptation system [8]. A computer simulation shows a maximum of 1.8 dB improvement over an usual system using the same ECC in segmental SNR forBER\geq10^{-2}.

A new matrix factorization is proposed for DCT-IV, which is the basis of fast algorithms for many sinusoidal transforms. A new fast algorithm for complex-data DFT based on the new factorization requires the same number of multiplications and far fewer additions than the Preuss algorithm. A new fast algorithm for real-data DFT based on a new algorithm for the discrete Hartley transform is also proposed.

A new proposed automatic equalizer including a decoder of error-correcting code (AEDEC) is obtained by the joint adaptive operation of an automatic equalizer and a decoder. This AEDEC can reduce misadjustment of tap values and error propagation owing to decision errors, and stabilize convergence even without a known training signal in noisy channels.

A class of binary sequences whose elements are either1or-1and whose autocorrelation function is0for all even shifts except the zero shift is discussed. These sequences will be calledEsequences. It is proved that everyEsequence is paired with anotherEsequence, its mate, so that the cross-correlation function between them is0for all even shifts including the zero shift. Using these sequences, new complete orthogonal function sets are derived. Several other important properties of these sequences are also obtained and the relations among these sequences, Golay's complementary sequences and Welti's quaternary codes are discussed.