In this paper, we present a simple codeword length generation algorithm and its hardware implementation. The proposed technique is based on the dividing the Huffman table as two parts; with leading 0'bits and following bits. The method is shown to be efficient in the memory requirement and searching speed since only logic gates are needed in the implementation and searching can be process parallel without looking up the memory table. The total equivalent gates for the implementation are about only 100 gates and critical path delay is 10 ns. The results of experiments show that the proposed algorithm has a very high speed and a good performance. The designed blocks are synthesized by Compass synthesis with 0.5 µm CMOS, 3.3V, technology.

Regular ternary logic functions are one of the most useful special classes of Kleenean functions, and a lot of research has been done on them. However, there has been little work done on incompletely specified regular ternary logic functions. This paper describes the following points: (1) Minimization of incompletely specified regular ternary logic functions. (2) A new definition of incompletely specified fuzzy switching functions and their minimization. (Concretely speaking, minimal disjunctive forms of incompletely specified fuzzy switching functions are represented in formulas of regular ternary logic functions. ) (3) Their application to fuzzy logic circuits such as fuzzy PLAs of AND-OR type.

The formulation of the wall admittance of a circular microstrip antenna by the spectral domain method is presented. The circular microstrip antenna is calculated using the cavity model. The electromagnetic fields within the antenna cavity are determined from the impedance boundary condition at the side aperture. The contribution from the region outside the antenna is taken into account by the wall admittance. The wall admittance is defined by the magnetic field produced by the equivalent magnetic current at the aperture. The magnetic field is calculated by the spectral domain method. The wall admittances obtained by this method are compared with the results calculated by Shen. The calculated input impedances of the microstrip antenna agree fairly well with the experimental data for the substrate thickness of up to 0.048λg. The formulation of wall admittance presented here is easily applicable to arbitrarily shaped microstrip antennas.

This paper proposes a phase-rotating phase-shift keying (PSK) modulation and shows that its narrow-band version is suitable for Viterbi equalization. The proposed PSK has the following features: 1) a spectrum shaping of the transmit/receive filters does not need to be restricted to the Nyquist criterion; 2) the transmitted data sequence is rotated for every symbol in order to reduce noise-correlation at the receiver. First, this paper discusses a performance degradation of bit error rate of Viterbi equalizers in the presence of the sampling timing offset or under time-dispersive frequency selective fading. Next, computer simulation confirms that π/2-shifted binary PSK with narrow-band spectrum shaping filter, which includes offset QPSK for its special case, solves the above mentioned performance degradation, keeping good spectrum efficiency equal to M-ary PSK.

An Acoustic echo canceller has problems adaptating under noisy or double-talk conditions. The adaptation process requires a precise identification of the temporarily changed room impulse response. To do this, both minimizing the step size parameter of the Least Mean Square (LMS) method to be as small as possible and giving up on updating the adaptive filter coefficients have been considered. This paper describes an adaptive cross-spectral technique that is robust to adaptive filtering under noisy or double-talk conditions and for colored signals such a speech signal. The cross-spectral technique was originally developed to measure the impulse response in a linear system. Here we apply in the adaptive cross-spectral technique to solve the acoustic echo cancelling problem. This cross-spectral technique takes the ensemble average of the cross spectrum between input and error signals and the averaged cross spectrum is divided by the averaged power spectrum of the input signal to update the filter coefficients. We have confirmed that the echo signal is suppressed by about 15 dB even under double-talk conditions. We also explain that this method has a systematic error due to using a short time block for estimating the room impulse response. Then we investigate overlapping every last half block by the following first half block in order to reduce the effect of the systematic error. Finally, we compare our method with the Frequency-domain Block LMS (FBLMS) method because both methods are implemented in the frequency domain using a short time block.

This paper proposes an adaptive microphone array using blind deconvolution. The method realizes an signal enhancement based on the combination of blind deconvolution, synchronized summation and DSA (Delay-and-Sum Array) method. The proposed method improves performance of estimation by the iterative operation of blind deconvolution using a cost-function based on the coherency function.

In order that the speech recognition system might have a high performance in the noisy environment, the directional microphone arrays at the input of the system have been broadly investigated. The purpose of this study is to develop a new wide-band directional microphone system in view of advancing to an adaptive one afterwards. In the proposed system, three microphones are arranged on a straight line and the beamforming is accomplished in such a way that the output value of the middle microphone is added to the integrated value of the difference between two microphones at both sides. In this study, the signal processing of microphone outputs is implemented by using active RC circuits. Finally, the objective directivity can be experimentally obtained in wide frequency ranges required for the speech recognition.

We present a new method for estimation of spectrum transition of nonstationary signals in cases of low signal-to-noise ratio (SNR). Instead of the basic functions employed in the previously proposed time-varying autoregressive (AR) modeling, we introduce a spectrum transition constraint into the cost function described by the partial correlation (PARCORR) coefficients so that the method is applicable to noisy nonstationary signals of which spectrum transition patterns are complex. By applying this method to the analysis of vibration signals on the interventricular septum (IVS) of the heart, noninvasively measured by the novel method developed in our laboratory using ultrasonics, the spectrum transition pattern is clearly obtained during one cardiac cycle for normal subjects and a patient with cardiomyopathy.

In this paper, the correlation properties are used to develop two efficient encoding schemes for speech line spectrum frequency (LSF) parameters. The first scheme (1D KL), which exploits the intraframe correlation, is based on one-dimensional Karhunen-Loeve (KL) transformation; the second scheme, which requires some coding delays to further utilize the interframe correlation, uses two-dimensional (2D KL) transform in the frequency domain or one-dimensional KL transform co-operating with DPCM in the time domain. Moreover, since the KL transform is globally optimal, which is sensitive to the change of input data statistics, further two adaptive transform coding systems are also investigated in this paper. The performance of all systems for different bit rates is investigated and adequate comparisons are made. It is shown that the gain of using KL transformation to exploit the intraframe and interframe correlation is 3 and 4 bits/speech frame, respectively.

This paper proposes a method of improving the accuracy of the attenuation constant estimate obtained by using the cross-spectral technique. In the cross-spectral technique, the envelope of the estimated impulse response is deformed due to the use of a time window. As a result, the estimated impulse response decays more rapidly than the real impulse response does, and the attenuation constant obtained by the estimated impulse response becomes larger than the real value. This paper first describes how the attenuation constant changes in the process of impulse response estimation. Next, we propose a method of improving the accuracy of the estimation. The effect of the proposed method is confirmed by computer simulation.

The conventional linear prediction can be viewed as a constrained blind equalization problem that has gained a lot of interests along with development of telecommunication networks. Because the blind equalization or deconvolution is a general framework of the inverse problem, the reliable and faster algorithm is requested in many applications. This paper proposes an orthogonal wavelet transform domain realization of a blind equalization technique termed as EVA, and presents an application to speech analysis. An orthogonal transformation has no influence to the equalization result in general, but we show that a particular wavelet makes the matrix in EVA nearly lower triangular that promotes the faster convergence in the estimation of maximum eigenvalue and its associate vector in EVA iteration. The experiments with the Japanese vowels show that the the proposed method effectively separates the glottis and vocal tract information, hence is promising for speech analysis.

It is confirmed by simulation that SOI-DRAMs can be operated at high speed by using the floating body structures. Several floating body effects are analyzed. It is described that the dynamic retention characteristics are not dominated by capacitive coupling and hole redistribution. And it is described that those characteristics are determined by the leakage current in the small pn-junction region of the floating body. Reducing pn junction leakage current is important for realizing a long retention time. If the pn junction leakage is suppressed to 10-18 A/µm, a dynamic retention time of 520 sec at a VBSG of 0.5 V can be achieved at 27. On those conditions, the refresh current of SOI-DRAMs is reduced by 54% compared with bulk-Si DRAMs.

This paper describes a newly developed FET Coupled Logic (FCL) circuit that operates at very high frequencies with very low supply voltages below 3.3 V. An FCL circuit consists of NMOS source-coupled transistor pairs for current switches, load resistors, emitter followers and current sources that are controlled by a band-gap reference bias generator. The characteristics and performance are discussed by comparing this circuit with other high-speed circuits. The optimal circuit parameters for FCL circuits are also discussed, and the fact is noted that a larger swing voltage enhances the circuit's performance. The simulated delay of a 0.25 µm FCL circuit is less than 15 ps for a 2.5 V power supply, and the simulated maximum toggle frequencies are over 5 GHz and 10 GHz at 2.5 V and 3.3 V power supply, respectively. The simulation results show that FCL circuits achieve the best performance among the current mode circuits, which include ECL circuits, NMOS source-coupled logic circuits. The delay of the FCL circuit is less than half that of an ECL circuit. The maximum toggle frequency of the FCL circuit is about triple that of NMOS source-coupled logic circuit. Because the FCL circuit uses low-cost CMOS-based BiCMOS technologies, its cost performance is superior to ECL circuits that require expensive base-emitter self-aligned processes and trench isolation processes. Using depletion-mode NMOS transistors for current switches can lower the minimum supply voltage for FCL circuits and it is below 1.5 V. The FCL circuit is a promising logic gate circuit for multi-Gbit/s tele/data communication LSIs.

A high-speed dynamic reference single-ended ECL input-interface circuit has been fabricated for advanced ATM switching MCMs. To raise the limit on the number of I/O pins, this circuit operates with a reference signal directly generated from the input signal itself. The reference level is changed dynamically to achieve a larger noise margin for operation. Experimental results show that operation up to 3.4 Gbps with a large level margin can be attained. We deploy this circuit to the input interface LSIs of an 80-Gbps ATM switching MCM.

Fluctuation characteristics of character centroid intervals in laterally written Japanese sentences are investigated by means of their spatial frequency characteristics. Power spectra of character centroid intervals in their longitudinal and transverse directions are obtained for handwritten and word processor printed sample sentences. It is shown that for fluctuation characteristics in the longitudinal direction, power spectra are inversely proportional to their spatial frequencies for handwritten sentences and proportional to them for word processor printed sentences, and there exists a remarkable difference between handwritten and word processor printed sentences. It is also shown that for fluctuation characteristics in the transverse direction, power spectra are proportional to their spatial frequencies for both handwritten and word processor printed sentences, and there is no remarkable difference between handwritten and word processor printed sentences.

A novel method for the radiated immunity test is proposed. The method is to generate controlled electromagnetic fields applying in arbitrary directions to an under test. The fields rotate at a low speed controlled electrically so that the immunity characteristics may be known in more detail. The primal characteristics of the fields generated by a trial benchtop setup are investigated.

A high-speed static logic circuit, the 1:8 demultiplexer (DEMUX), fabricated using single-gate CMOS technology (single-gate means the structure consisting of n+ poly-Si gate for both NMOS and PMOS transistors) has been demonstrated. To suppress short-channel effects in PMOS transistors, we only used the low-energy ion implantation (I/I) of BF2 at 10 keV for counterdoping of the channel and that at 5 keV for source/drain (S/D) extension. To control the threshold voltage Vth of PMOS transistors precisely, the channel dopants were implanted after the growth of the gate oxide because of the suppression of the transient-enhanced diffusion (TED) of boron, and the suppression of boron out-diffusion. A tree-type 1:8 DEMUX circuit composed of 0. 134 µm gate CMOS transistors operates at a high speed of 3.1 GHz and consumes a low power of 35.5 mW/GHz at VDD = 2.0 V. In this single-gate CMOS circuit, down to this small gate length, the maximum operating frequency of the DEMUX circuit increases proportionally with an increase of the inverse of the gate length without an increase of power consumption per GHz. At a practical 2.48832 Gb/s operation, the power consumption was 88 mW, and the phase margin between the input clock signal and the input data signal was 260 ps. It is suggested that a circuit composed of a single-gate CMOS transistor with 0.15 µm gate generation can be applicable to high speed ICs.

The lock-up time of a PLL frequency synthesizer mainly depends on the total loop gain. Since the gain of the conventional phase detector is constant, it is difficult to improve the lock-up time by the phase detector. In this paper, we reconsider the operation of the phase detector and propose the PLL frequency synthesizer with multi-phase detector in which the gain of phase detector is increased by using four stage phase detectors and charge pumps. Then, a higher speed lock-up time and good spurious characteristics can be achieved.

Wavelength Division Multiplex (WDM) photonic networks are expected as key for global communication infrastructure. The accurate measurement methods for AWG-MUX/DMUX are desirable for WDM network design. We measured a transfer function matrix of an AWG-MUX to find that polarization mode dispersion (PMD) and polarization dependent loss (PDL) shows the bandpass characteristics, which may limit the maximum size and the bit rate of the system. These bandpass characteristics of PMD and PDL are reproduced by a simple AWG-MUX model: The phase constant difference of 0.5% between orthogonal modes in arrayed waveguides is sufficient to obtain the measured passband characteristics of PMD and PDL. We find phase distribution difference between two orthogonal modes in the arrayed waveguide grating gives arise to complex PMD.

In this paper a new snake model for image morphing with semiautomated delineation which depends on Hermite's interpolation theory, is presented. The snake model will be used to specify the correspondence between features in two given images. It allows a user to extract a contour that defines a facial feature such as the lips, mouth, and profile, by only specifying the endpoints of the contour around the feature which we wish to define. We assume that the user can specify the endpoints of a curve around the features that serve as the extremities of a contour. The proposed method automatically computes the image information around these endpoints which provides the boundary conditions. Then the contour is optimized by taking this information into account near its extremities. During the iterative optimization process, the image forces are turned on progressively from the contour extremities toward the center to define the exact position of the feature. The proposed algorithm helps the user to easily define the exact position of a feature. It may also reduce the time required to establish the features of an image.