Irreducible components of canonical graphs for second order spectral null constraints at a rational submultiple of the symbol frequency fsk/n are studied where fs is the symbol frequency. We show that if n is prime then a canonical graph consists of disjoint irreducible components. We also show that the number of irreducible components of a canonical graphs is finite if n is prime. For the case n = 2 and p O mod n, all aperiodic irreducible components are identified explicitly where p is a parameter of a canonical graph.

Recently, the low earth orbit satellite communications has been attracting much attention. These communications have many strong features, however, the communication performances are influenced by carrier frequency offset (CFO) and, particularly, it is hard to acquire the synchronization. A large number of publications have so far been made on the synchronization acquisition of DS/SS systems under CFO and most of them make use of the maximum likelihood decision in finding the maximum values of Fourier transform outputs. However, the implementations of Fourier transforms usually require high cost and large space. In this paper, we propose a new simple acquisition scheme using half-symbol differential decoding technique for DS/SS systems under CFO. This scheme makes use of the addition and subtraction of baseband signals and their delayed versions, (omitting Fourier transforms), together with integrations by recursive integrators, and thus resulting in much simpler implementation. In general, it is shown that the proposed scheme can acquire the code synchronization under carrier frequency offset with much smaller computational complexities and the sacrifice of longer acquisition time.

In this paper we discuss the binary spreading sequences whose spectral distributions are DC free and spectral distribution's shapes can be easily controlled by a certain parameter denoted by δ. The newly developed sequences, referred to as modified antisymmetric M-sequences, are modified-versions of the conventional antisymmetric (AS)M-sequences. The proposed sequences are designed to increase the varieties of spectral distribution's shapes and improve the correlation properties when compared to those of the FM coded M-sequences which have already proposed by Tsuzuki et al. Some typical line coded M-sequences, i.e. the (differential) Manchester coded M-sequences and the FM coded M-sequences, and the conventional AS M-sequences are included in the set of proposed sequences. The improvement of the average BER (bit error rate) performance for asynchronous DS/SSMA (direct sequence/spread spectrum multiple access) systems using the proposed sequences in comparison to the system using the conventional AO/LSE (auto-optimal phase with least sidelobe energy) M-sequences is also shown.

This paper proposes a new speech codec based on CELP for PHS multimedia communication. PHS portable terminals should consume as little power as possible, and the codec used in them has to be robust against channel errors. Therefore, the proposed codec operates with low computational complexity while reducing the deterioration in speech quality due to channel errors. This codec uses two new schemes to reduce computational complexity. One is moving average scalar quantization for the filter coefficients of the synthesis filter. This scheme requires 90% less complexity to quantize synthesis filter coefficients compared to the widely used vector quantization. The other is pre-selection for selecting an algebraic codebook used as random excitation source. An orthogonalization scheme is used for stable pre-selection. Deterioration of speech quality is suppressed by using CRC and parameter estimation for error protection. Two types of codec are proposed: a 10-ms frame type that transmits 160 bits every 10-ms and a 15-ms frame type that transmits 160 bits every 15 ms. The computational complexity of these codecs is less than 5 MOPS. In a nochannel error environment, the speech quality is equal to that of ITU-TG.726 at 32.0 kbit/s. With 0.3% channel error, both codecs offer more comfortable conversation than G.726. Moreover, at 1.0% channel error, the 10-ms frame type still provides comfortable conversation.

In this paper, we consider conventional signaling link fault tolerance and error correction mechanisms to provide reliable services of mobile multimedia telecommunication network based on ATM (Asynchronous Transfer Mode) technology. Also, we propose an efficient signaling protocol interworking architecture and a reliable distributed interworking network architecture between SS7 based FPLMTS and ATM networks. Besides, we evaluate the performance of proposed method through computer simulation. According to the results, proposed signaling architecture shows efficient and fast fault restoration characteristics than conventional MTP-3/3b based network. Functional signaling protocol stack and network architecture of proposed fast rerouting mechanism provide reliable and efficient restoration performance in view of interworking between SS7 based FPLMTS and ATM networks.

Distributed multimedia applications are often sensitive to the Quality of Service (QoS) provided by the communication network. They usually require guaranteed QoS service, so that real-time communication is possible. However, searching a route with multiple QoS constraints is known to be a NP-complete problem. In this paper, we propose a new simple and efficient distributed QoS routing algorithm, called "DQoSR," for supporting real-time communication in high-speed networks. It searches a route that could guarantee bandwidth, delay, and delay jitter requirements. Routing decision is based only on the modified cost, hop and delay vectors stored in the routing table at each node and its directly connected neighbors. Moreover, DQoSR is proved to construct loop-free routes. Its worst case message complexity is O(|V|2), where |V| is the number of nodes in the network. Thus DQoSR is fast and scales well to large networks. Finally, extensive simulations show that average rate of establishing successful connection of DQoSR is very near to optimum (the difference is less than 0.4%).

For mobile communication systems with code division multiple access (CDMA), a new modulation scheme, π/2-shift BPSK, is proposed. The performance has been evaluated in terms of relative out-of-band power, bit-error rate (BER), and spectral efficiency. As the result, it is shown that the proposed scheme has an advantage over conventional BPSK, conventional QPSK, and π/4-shift QPSK under nonlinear amplification.

The pattern spectrum has been proposed to represent morphological size distribution of an image. However, the conventional pattern spectrum cannot extract approximate shape information from image objects spotted by noisy pixels since this is based only on opening. In this paper, a novel definition of the pattern spectrum, morphological multiresolution pattern spectrum (MPS), involving both opening and closing is proposed. MPS is capable of distinguishing details from approximate information of the image.

Linear programming techniques are useful in many diverse applications such as: production planning, energy distribution etc. To find an optimal solution of the linear programming problem, we have to repeat computations and it takes a lot of processing time. For high speed computation of linear programming, special purpose hardware has been sought. This paper proposes a systolic array for solving linear programming problems using the revised simplex method which is a typical algorithm of linear programming. This paper also proposes a modified systolic array that can solve linear programming problems whose sizes are very large.

The parameter estimation problem of point spread function is one of the most challenging and important task for image restoration. A new method for the parameter estimation in the case of motion blur is presented here. It is based on the principle that the power spectrum of the motion blurred image contains periodical minima relevant directly to the motion derection and length. Though the principle is very simple and effective in certain cases, the direct use of it may lead to poor performance an the signal-to-noise ratio (SNR) gets lower. To improve the estimation accuracy, by analyzing image noise effect on the detection of the minima, we propose a method to greatly reduce spectral noise, and give the lowest allowed SNR at which the minima may still be identified reliably. We also estimate the power spectrum of the original image, which is a must for the Wiener restoration filter, from the noisy blurred image based on a noncasual autoregressive model. Once above parameters are decided, the Wiener filter is used to restore the noisy blurred image. Our method is very practical; no parameter needs to be known a priori, or to be adjusted manually to fit into various application problems. The proposed method is finally applied to systhesized and real motion blurred images to demonstrate its effectiveness.

High-speed display of 3-D objects in virtual reality environments is one of the currently important subjects. Shape simplification is considered an efficient method. This paper presents a method of hierarchical cube-based segmentation for shape simplification and multiresolution model construction. The relations among shape simplification, resolution and visual distance are derived firstly. The first level model is generated from scattered range data by cube-base segmentation with the first level cube size. Multiresolution models are then generated by re-sampling polygonal patch vertices of each former level model with hierarchical cube-based segmentation structure. The results show that the algorithm is efficient for constructing multiresolution models of free-form shape 3-D objects from scattered range data and high compression ratio can be obtained with little noticeable difference during the visualization.

In the field of speech recognition, many researchers have proposed speech recognition methods using auditory information like acoustic signal or visual information like shape and motion of lips. Auditory information has valid features for speech recognition, but it is difficult to accomplish speech recognition in noisy environment. On the other side, visual information has advantage to accomplish speech recognition in noisy environment, but it is difficult to extract effective features for speech recognition. Thus, in case of using either auditory information or visual information, it is difficult to accomplish speech recognition perfectly. In this paper, we propose a method to fuse auditory information and visual information in order to realize more accurate speech recognition. The proposed method consists of two processes: (1) two probabilities for auditory information and visual information are calculated by HMM, (2) these probabilities are fused by using linear combination. We have performed speech recognition experiments of isolated words, whose auditory information (22.05kHz sampling, 8-bit quantization) and visual information (30-frame/s sampling, 24-bit quantization) are captured with multi-media personal computer, and have confirmed the validity of the proposed method.

This paper proposes a nonlinear signal processing by using a three layered network which is trained with self-organized clustering and supervised learning. The network consists of three layers, i.e., self-organized layer, an evaluation layer and an output layer. Since the evaluation layer is designed as a simple perceptron network and the output layer is designed as a fixed weight linear node, the training complexity is the same as a conventional one consisting of self-organized clustering and a simple perceptron network. In other words, quite high speed training can be realized. Generally speaking, since the data range is arbitrary large in signal procession, the network shoulk cover this range and output a value as accurately as possible. However, it may be hard for only a node in the network to output these data. Instead of this mechanism, if this dynamic range is covered by using several nodes, the complexity of each node is reduced and the associated range is also limited. This results on the higher performance of the network than conventional RBFs. This paper introduces a new non-linear spectrum estimation which consists of LPC analysis and RBF network. It is shown that accuracy spectrum envelopes can be obtained since a new RBF network can estimate some nonlinearities in a speech production.

This paper proposes a novel FEC (forward error correction) scheme for high-speed wireless systems aiming at mobile computing applications. The proposed scheme combines inner nonredundant error correction with outer parallel encoding random FEC for differentially detected QPSK (quadrature phase shift keying) signals. This paper, first, examines error patterns after the differential detection with nonredundant error correction and reveals that particular double symbol errors occur with relatively high probability. To improve the outer FEC performance degradation due to the double symbol errors, the proposed scheme uses I and Q channel serial to parallel conversion in the transmission side and parallel to serial conversion in the receiving side. As a result, it enables to use simple FEC for the outer parallel encoding random FEC without interleaving. Computer simulation results show the proposed scheme employing one bit correction BCH coding obtains a required Eb/No improvement of 1.2 dB at a Pe of 10-5 compared to that with the same memory size interleaving in an AWGN environment. Moreover, in a Rician fading environment where directional beam antennas are assumed to be used to improve the degradation due to severe multipath signals, an overall Eb/No improvement at Pe of 10-5 of 3.0 dB is achieved compared to simple differential detection when the condition of delay spread of 5 nsec, carrier to multipath signal power ratio of 20 dB and Doppler frequency at 20 GHz band of 150 Hz.

This paper presents a generating method of LSP parameter sequences for speech synthesis by rule. In our method, neural networks are schemed to generate LSP parameter sequences of Vowel-Consonant-Vowel (VCV) units. The quality of synthesized speech by concatenation way of VCV units through table-look-up technique can not be improved so much owing to the distortion appearing on VCV units junction. In our method, the neural networks concatenate VCV units step by step with less distortion on VCV units junction, which synthesizes good quality speech.

This paper presents the evaluation for personal communication systems (PCS). Two types of PCS are supposed for low and high degree of mobility. The service area with 30km radius is covered by a multiple hexagonal cells, which are micro cells and macro cells for the low mobility and high mobility planes respectively. As for a traffic distribution, uniform and exponentially tapered traffic distributions are assumed. After defining the system model, cost evaluation form along with capacity has been derived. The evaluation and discussions are made in terms of cost economy, capacity and spectrum usage in various conditions. It is shown that there exist the optimum cell radius for the prescribed subscriber numbers and the integration of two systems is desirable for the support of full mobility with cost-effectiveness and spectrum efficiency.

This paper discusses the features of SOI/CMOS circuits in comparison with bulk/CMOS circuits. We have to design circuits with small fan outs and short wires to take advantage of high-speed and low-power SOI/CMOS devices to their fullest. We can take advantage of the SOI/CMOS structure if the ratio of the source/drain capacitances to the gate capacitances is much greater in the load capacitance. Thus, we propose a new flip-flop circuit with a smaller gate capacitance. The flip-flop circuit operates 30% faster than the previous circuit at 2.0 V. We also propose a buffer circuit having less delay disparity between the complementary output signals. The buffer circuit has the delay disparity of 18 ps at 0.2 pF and 2.0 V. We fabricated an 8-bit frequency divider and a 4-bit demultiplexer using the proposed circuits and 0.35 µm SOI/CMOS process. The 8-bit frequency divider and the 4-bit demultiplexer operate at 2.8 GHz and 1.6 GHz, respectively, at 2.0 V.

Spectral transform methods have been widely studied for classification and analysis of logic functions. Spectral methods have also been used for logic synthesis, and by use of BDDs, practical-sized synthesis problems have been solved. Wavelet theory has recently attracted the attention of researchers in the signal processing field. The Haar function is used in both spectral methods and in signal processing to obtain spectral coefficients of logic functions of signals. In this paper spectral transform-based analysis of neural nets verifying signal processing and discrete function is presented. A neural net element is defined as a discrete function with multi-valued input signals and multi-valued or binary outputs. The multi-valued variable is realized as a variable (V, W) formed by a pair of a binary value and a multi-value pulse width. The multi-valued encoding is used with the multi-valued Haar function to give meanings to the wavelet coefficients from the view of Boolean algebra. A design example shows that these conceptually different concepts are closely related.

Genetic algorithms were introduced by Holland in 1975 as a method of solving difficult optimization problems by means of simulated evolution. A major drawback of genetic algorithms is their slowness when emulated by software on conventional computers. Described is an adaptation of the original genetic algorithm that is advantageous to hardware implementation along with the architecture of a hardware framework that performs the functions of population storage, selection, crossover, mutation, fitness evaluation, and survival determination. Programming of the framework is illustrated with the set coverage problem that exhibits a 6,000 speed-up over software emulation on a 100 MHz workstation.

The Resource Auction Multiple Access (RAMA) protocol was proposed by N. Amitay for fast resource allocations to mobile terminals. We have proposed an Enhanced RAMA (ERAMA) protocol yielding lower system delay and delay variations than does the RAMA protocol. In this paper, we model a two-stage queueing network to evaluate the performance of the proposed protocol in terms of mean access delay, mean buffering delay, talk spurt loss ratio, and channel utilization, under homogeneous voice connections. The analytical results yield upper estimates for the various performance indices, compared with those of the simulations.