bluuebunny/crossref_metadata_2025_split huggingface dataset:
Real time signal detection in communication system with additive non‐gaussian noise
Authors: Amano Masaaki 
Published: July 2007

AbstractGaussian noise usually is considered as additive noise in the communication system. This paper considers the general communication system with non‐Gaussian noise, and discusses the real‐time detection of the random signal in the system from the stand‐point of the theory of Markov process and Bayes decision. A simulation example is presented for the detection of a phase‐modulated signal.
A study of speech scramblers using the transmultiplexer technique
Authors: Azuma Mitsuhiro , Hasebe Takayuki , Torii Naoya , Akiyama Ryota 
Published: July 2007

AbstractThis paper considers a scrambling technique for an analog voice signal, and proposes a band‐split scrambling method applying the transmultiplexer technique, which can effectively divide the voice band using a small amount of signal processing. The principles of signal processing and the filter design are presented. By computer simulation, the quality of the decoded speech, the effect of the channel characteristics and security are evaluated. As a result of quantitative evaluation of the quality of the decoded speech, the relation between the quality of the decoded speech and the number of taps is quantitatively evaluated for fixed and variable scramblings. It is seen that the quality of the voice signal is little affected by the channel characteristics, such as group delay distortion, synchronization error and frequency off‐set. Thus, the proposed method is seen to be economical, without requiring additional circuitry as used by various kinds of cancellers. As for security, an intelligibility test using numbers was executed, and high security, comparable to or better than the conventional method, is obtained. Thus, the proposed secure communication method is shown to be effective, with excellent performance in the quality of the scrambled decoded speech economy and security.
Three‐dimensional (3‐D) facial model‐based description and synthesis of facial expressions
Authors: Choi Chang Seok , Takebe Tsuyosi , Harashima Hiroshi 
Published: July 2007

AbstractThe automatic synthesis of the facial expression by computer is one of the most basic techniques to be applied in various fields. This paper describes a method to synthesize various facial expressions by modification of the 3‐D model of the face. The 3‐D facial model is obtained by adjusting the prepared 3‐D facial shape model to the frontal view of the object neutral face image, and by projecting the gray‐level informations.This paper proposes first a hierarchical model of facial description units for the hierarchical description of the facial expressions. The action unit (AU) of the FACS (facial action coding system), which is known as a systematic method of describing the facial expression, is considered as a layer in the proposed hierarchical model. AU of FACS is realized on a computer, and the automatic synthesis of the expression is executed.This procedure is based on the muscular structure of the face and allows cancellation of individual differences. At present, 34 kinds of AUs are implemented. The graylevel information for wrinkles and teeth, which are important in the synthesis of expressions, is not contained in the 3‐D facial model obtained from the neutral expression. Thus, such information is extracted from other face images, and registered as auxiliary information of the 3‐D facial model. This leads to synthesis of natural expressions. Finally, numerous examples for the facial expression synthesis are presented to show the usefulness of the method.
A method of designing parallel‐type pseudo‐minimum‐phase FIR QMF banks
Authors: Takebe Tsuyoshi , Yoshida Tadashi 
Published: July 2007

AbstractSubband coding (SBC) is a low bit‐rate speed transmission system in which a quadrature mirror filter (QMF) is used to split a speech into frequency bands of the transmitter and recombine these subband signals to form an approximation to the input speech at the receiver. In the conventional parallel‐type FIR multiband pseudo‐QMF bank, the linear‐phase filters are employed in the transmitter and receiver, which gives rise to a problem wherein a large delay is produced.This paper presents a design method for the equal‐bandwidth parallel‐type QMF bank with smaller delay by allowing some delay distortion. The amplitude and phase constraints on each channel filter are shown so that the transmitted signal can be reconstructed approximately at the receiver. A design example is presented for the 4‐band filter bank, where the amplitude constraint is satisfied by a minimum‐phase filter and the phase constraint is satisfied by a allpass circuit cascaded to the filter. The designed low‐pass filter is of 50 percent roll‐off, and the bandpass filter is a 100 percent roll‐off. The attenuation is 43 dB. The delay is almost flat in characteristic and the amount is reduced to approximately one‐half that of the linear‐phase filter bank.
Improvement of one‐dimensional module placement in VLSI layout design
Authors: Ohmura Michiroh , Wakabayashi Shin'Ichi , Miyao Jun'Ichi , Yoshida Noriyoshi 
Published: July 2007

AbstractThis paper considers the building block method, which is one of the methods in VLSI layout design, and discusses the improvement of module placement assuming that a one‐dimensional placement of the modules is given as the initial placement. Discussions are made of the following:
The case where net list given as the input is composed only of two‐terminal nets, which connect the external terminals and the modules. The placement improvement problem PMO is formulated, where the virtual connection length defined only by × coordinate is to be minimized while retaining the placement order of the modules. The algorithm to solve this problem, as well as the proof for the optimality, are presented. Then a problem PM1 with a constraint on the range of displacement of the module is considered, and it is shown that the problem can be solved by extending the algorithm for PMO.
A problem PM2 is considered, which is an extension of PM1 by introducing in general the multiterminal net connecting modules and external terminals. It is shown that PM2 can be reduced to PM1.
A problem PM3 is considered, where river routing is adopted as the connection model, and the connection length is to be minimized considering also y coordinates (separation). It is shown that the optimal solution can be obtained in a polynomial time of the number of modules and the number of nets.
Theoretical study on reconstruction of waveforms by a natural observation filter
Authors: Iijima Taizo 
Published: July 2007

AbstractA few years ago a fundamental theory on natural observation of waveforms was published as the foundation of a new effective method instead of Fourier analysis. In this paper a natural observation filter is proposed based on the theory. The filter has a typical property with regard to perfect reconstruction for a certain characteristic waveform group. The algebraic relation between the structure of the filter and the corresponding waveform group is theoretically clarified. The result of the present paper is expected to be applicable for recognition problems of various waveforms.
Analytical least‐square design of three‐dimensional linear‐phase FIR digital filters
Authors: Muneyasu Mitsuji , Hinamoto Takao 
Published: July 2007

AbstractThis paper discusses the optimal design of linear‐phase three‐dimensional (3‐D) FIR digital filters in the least‐square sense. The design procedure is as follows. The linear‐phase 3‐D polynomial which is symmetric with respect to the origin is decomposed into four polynomials from the viewpoint of the symmetry (cubic symmetry, ω1 and ω2 axes skew symmetry, ω2 and ω3, axes skew symmetry and ω1 and ω3 axes skew symmetry). Each of those resulting polynomials is then used as the filter model. Next, the filter parameters are determined analytically so as to minimize the square errors between the actual magnitude response of the filter and the ideal magnitude response.Finally, two numerical examples are given to demonstrate the validity of the theory. The proposed method allows us to reduce the computational complexity in the design since it is not necessary to use various optimization procedures or to calculate the inverse matrix. The filter with the desired response can be designed in a short time.
A method for solving reliability optimization problem by fuzzy multiobjective 0–1 linear programming
Authors: Sasaki Masato , Gen Mitsuo , Ida Kenichi 
Published: July 2007

AbstractOptimization problems related to system reliability such as unit selection and redundant allocation are generally considered to be 0–1 linear programming (0–1 LP) problems. Recently an efficient method which considers the particular structure called generalized upper bounding (GUB) was proposed. On the other hand, it is increasingly necessary to deal with such reliability optimization problem as multiple objective decision making (MODM) problems with multiple conflicting functions. But the above method (GUB) is limited to solving single‐objective or bicriteria 0–1 LP problems. In this paper we propose a new fuzzy multiple objective 0–1 LP (0–1 FMLP) method which can adapt to changes in the utilization of system resources and the economic environment by making use of tolerances in the system control parameters. At the same time, the method considers the variability which arises from the subjective judgment of the system decisionmaker (DM) by introducing fuzzy logic into the goal specification. This method is an extension to the more general MODM problem of the efficient algorithm which takes into account the GUB structure that arises in the solution of a system reliability optimization problem. Further, in order to show the effectiveness of the method proposed here, we consider the problem of optimally planning unit selection and redundant distribution in order to maximize system availability.
Simulation of neural network digital adder and multiplier
Authors: Sakai Ken‐Ichiro , Morisue Mititada , Koinuma Hideyuki 
Published: July 2007

AbstractA novel adder and multiplier using Hopfield‐type neural networks for which a technique is introduced to avoid the false operations is proposed, and the simulation results for these operations are illustrated. The main feature of these circuits is high‐speed operation which is achieved at the almost constant computation time due to their parallel calculation ability even if the number of bits in digital circuits increases.The previously proposed adder has a problem in that some false operations occur in the special cases of addition. The error between the correct and false values in the operations is always only a binary value of ±1 once the false operation occurs.This paper describes in detail the technique to avoid the false operations. It is important to determine the ratio of weight of the object function to constraint function, taking into consideration the energy function that should be chosen. Computer simulations for the adder verify that the correct operation can be achieved by choosing the special ratio of the weight for these functions.On the other hand, a novel multiplier is described using a Hopfield‐type network which is composed of the proposed adder and a neural network to perform a logical product. The simulation results for the multiplier show that the correct operations of a multiplication can be achieved with high‐speed performance. It is clear from these results that a high‐performance digital adder and multiplier can be constructed by using the Hopfield‐type neural network model.
A proposal on cooperative development environment for next‐generation circuit simulator
Authors: Ootani Jun'Ichi , Konno Daisuke , Adachi Takehiko 
Published: July 2007

AbstractWith the recent development of large‐scale, high‐level functions and high‐speed VLSI circuits, intensive research and development efforts have been made to obtain new circuit simulation techniques. This paper proposes a cooperative development environment, which is indispensable in developing the next‐generation circuit simulation technique. Assuming UNIX and C language, a proposal for the following scheme is presented.Various circuit simulation functions are realized as independent programs (simulation commands) as far as possible. By giving specifications for those commands in a unified way, data are shared among different simulation techniques. The input file for the simulation command is constructed by filing three kinds of data: circuit connection data; the device model data; and the data for the control. The core file is used as the output file. Then the function and realization of the simulation shell, which is to execute the simulation command interactively or to efficiently share the data among the commands, are discussed.Finally, the method of sharing the circuit simulation program using the shared library function is proposed. Experimental examples are presented for the simulation commands such as shared library function and transient analysis.
Nonlinear equation and dynamics of tracking loop in spread spectrum communication
Authors: Kimura Naoki , Miyashita Teruo , Sato Chikara 
Published: July 2007

AbstractThis paper aims at the representation and analysis of the mathematical model for the tracking of a receiver in the spread spectrum communication using the deterministic differential equation. The tracking system is based on the 2Δ‐type baseband delay‐lock loop using the pseudorandom waveform by M‐sequence. Since the M‐sequence has the periodicity, the correlator output is approximated by the Fourier series expansion with the power series coefficients and is treated as a function of the time and the phase difference between the pseudorandom waveforms.By those approaches, the basic equation for the tracking system is derived as a second‐order nonautonomous nonlinear differential equation for the phase difference, including the nonlinear restoring force, the forced input, and the parameter excitation. A computer simulation is applied to the derived nonlinear differential equation, and it is verified that the synchronization process is represented clearly as a trajectory on the phase plane.The behavior in the neighborhood of the phase difference zero is analyzed, and it is shown that the delay‐lock loop may become unstable, resulting in the one‐half subharmonic oscillation. The Poincaré map is constructed by a computer simulation to verify the fore‐mentioned property.
Design of IIR fan filters using iir hilbert transformer
Authors: Nishikawa Kiyoshi , Morita Eiko 
Published: July 2007

AbstractThis paper describes a method of synthesis for the IIR fan filter which applies the IIR Hilbert transformer to the one‐dimensional prototype filter. The method is more direct than the traditional method, which derives the fan filter from the lowpass filter. The design of the quadrant fan filter also is discussed. The bilinear z transform is applied to the analog Hilbert transformer, which is already known as the 90‐deg constant‐phase function, and the result is used as the IIR Hilbert transformer. Only the odd‐order function can be used for the IIR Hilbert transformer for the IIR fan filter, but even‐ as well as odd‐order functions can be used for the quadrant fan filter. The final IIR fan filter transfer function is obtained as the difference of half‐plane type two‐dimensional all‐pass functions. A Chebyshev‐type function is used for the IIR Hilbert transformer. Consequently, the fan filter has the amplitude response with a ripple in the stopband which is one‐half the phase ripple of the Hilbert transformer, while the response is almost flat without a ripple in the passband.
Adaptive equalization and trellis‐coded modulation for selective fading channel
Authors: Ogiwara Haruo , Aniya Mitsuru 
Published: July 2007

AbstractIn high bit‐rate mobile communication, selective fading due to multipath phenomena is one of the causes of communication quality degradation. In selective fading, a time‐varying intersymbol interference occurs.This paper discusses the suitable trellis‐coded modulation code for the channel. Two design rules are discussed: one is to maximize the minimum value of the average squared Euclidean distance between sequences over fading; and the other is to maximize the averaged value of the minimum distance between received sequences over fading and information sequences.Both rules conclude that the optimum code for the Gaussian channel without fading also is optimum for the selective fading channel, that is, the code with the maximum free Euclidean distance is optimum for the fading channel. Simulation results show that the codes designed by the rules give the effective coding gain and the design method is useful.The decoding complexity increases exponentially depending on the sum of the code constraint length and the length of intersymbol interference. To lighten the decoding burden, a simplified decoding method of a decreased survival path is proposed.
Acoustic parameter estimation for design of piezoelectric telephones by the nonlinear optimization method
Authors: Hoshijima Keizo , Simada Osamu , Kitao Masachika , Nomura Yasuo , Ohta Yuka , Ohga Juro 
Published: July 2007

AbstractUntil now, no formula has been derived for computing the acoustic parameters of the transmitting and receiving systems in the actual piezoelectric telephone handset of small size and light weight. The lack of such a formula was a significant hindrance to the design of the handset.In this paper, estimation of these parameters is made possible by the application of the nonlinear optimization technique. In this optimization problem, convergence is extremely slow since there is a magnitude difference of about 1010 among the parameters. Hence, for each parameter, a hypothetical maximum and minimum are assumed so that the problem is replaced with a hypothetical conditional optimization problem.By means of the change of variables, the variable ranges of all the parameters are aligned. At the same time, the problem is transformed back to the optimization problem without constraints so that the convergence is accelerated.Since the performance function is constructed in which all the measured characteristics of the transmitting and receiving sensitivities under different measurement conditions are used as esimating information, the resultant optimization problem is of a significant size. The simulation characteristics computed based on the obtained estimation values agreed well with the measured characteristics. This implies that the estimated value derived by the present method is useful in actual design of the handset.
Generalization of the non‐gaussian, nonlinear AR time series model type and its application to acoustic environments
Authors: Ohta Mitsuo , Ikuta Akira 
Published: July 2007

AbstractThe traditional standard time series model, such as the AR model, usually assumes the Gaussian property for the fluctuation distribution. It assumes also the linear regression function for the time series, and the well‐known least‐square method is applied based only on the linear correlation data. In the actual sound environment, however, the stochastic process exhibits various non‐Gaussian distributions, and there potentially exist various nonlinear correlations between two or more time points in addition to the linear correlation between time series. Consequently, often the time series in the actual phenomenon cannot be represented by a simple model such as the AR model.This paper derives the new time series model without any preset artificial simplification in a form matched to the complexity and multiplicity of the fluctuation distribution. More precisely, a time series regression model is proposed based on the Bayes theorem, where no regression function is assumed between time series, while reflecting hierarchically all such various correlation information. The proposed method is applied to the actual data in the sound environment, and the practical usefulness is verified.
A theory of initial synchronization and error‐control coding
Authors: Imai Hideki , Shishino Shinichi 
Published: July 2007

AbstractThe error‐control code is used widely at present in digital communication and storage systems. When the error‐control code is applied to the actual system, however, several problems arise that cannot be solved by the traditional code theory.One such problem is synchronization. In the usual application of the error‐control code, it is not tacitly assumed that the synchronization is already established. Such an assumption is not always valid in the actual situation. If the synchronization is not established, it is difficult for the error‐control code to exhibit its full error correction ability.From such a viewpoint, this paper does not use the assumption that the synchronization is established. Both the initial synchronization and error control are considered, i.e., the establishment of synchronization from the unsynchronized state and the execution of the correct decoding are discussed.This paper proposes the initial synchronization decoding model. Two code structures are considered, and the decoding method as well as the asymptotic properties of the error exponent are discussed. As a result, a practical theory for the synchronization and error control is shown, considering simultaneously the initial synchronization and error control. Its features are the high efficiency and the decoding which is not very complex.
Even‐length real‐valued cyclically orthogonal pseudonoise sequences with small absolute value
Authors: Tanada Yoshihiro , Yoshida Hiroshi , Nagasawa Yoji 
Published: July 2007

AbstractThis paper discusses the even‐length real‐valued cyclically orthogonal pseudonoise (PN) sequences with small absolute values as the PN sequence for spread spectrum communication.First, the sequence of maximum absolute value 2 with cubic or inverse phase parameters is considered. By replacing the sequence variable by the power of the primitive root and by regarding the power as the sequence variable, an orthogonal PN sequence is obtained. There exists the same number of such sequences as the number of primitive roots, and the cross‐correlation function is relatively small if the period is large.By convolving a sequence with itself, a set of sequences is obtained which has relatively small absolute value of the sequence and relatively small absolute value of cross correlation. It is proposed also to convolve a sequence with other sets of sequences. The sequence considered in this paper is suited to the digital correlation processing using the fast‐Fourier transform when the period is 4, 16, 256, 65, 536, ….
Multiterminal zero error data transmission with linked encoders
Authors: Oohama Yasutada 
Published: July 2007

AbstractIn the coding of information source and the channel, a problem arises where the transmission rate is discussed under the zero error probability, called “error‐free problem.” In the multiterminal information transmission system considered by Slepian and Wolf (called the SW system), the error‐free problem has not been solved.This paper extends the SW system and defines anew a system where encoders are linked (called the SWL system). The error‐free region (admissible rate region under the zero error probability) is determined for the defined system, and the error‐free problem is given a complete answer. It is shown that the forementioned region is wider than the error‐free region of the SW system. It is interesting that the foregoing results are in a sense matched to the result obtained in the coding of the one‐input/one‐output channel. Furthermore, it is shown that the error‐free region in the SWL system can be achieved by the universal code.