A dual transformation incorporating the frequency-dependent scaling factor with the impedance dimension is proposed to synthesize the current-mode counterpart of a voltage-mode original. A general class of current-mode active-RC biquadratic filters and a switched-capacitor low-pass biquad are derived to demonstrate the synthesis procedure. Their simulation and test results show that the current transfer functions are the same as the voltage transfer functions of the originals, and thus confirm the validity of the procedure. The dual trasformation described herein is general in that with the scaling factor chosen appropriately it can meet a wide variety of circuit transformation, and thus useful also for circuit classification and identification.

This paper examines the key technologies and applications of optical frequency division multiplexing (OFDM) systems. It is clarified that a 100-channel OFDM system is feasible as a result of multichannel frequency stabilization, common optical amplification and channel selection utilizing a tunable optical filter. Transmission limitation due to fiber four-wave mixing is also described. Major functions and applications of the OFDM are summarized and the applicability of OFDM add/drop multiplexing is examined.

This paper proposes a designing algorithm for quadrilateral recursive filters which consist of four quarter-plane filters in the four quadrants. This can realize a perfect zero-phase filtering which is essential for image processing. Furthermore, several parallel processing algorithms capable of performing under very high parallel efficiency are developed on line-connected and mesh-connected processor arrays. By these proposals, the advantage of two-dimensional non-causal zero-phase recursive digital filters is made clear.

It is sometimes required to change the frequency characteristics of a digital filter during its operation. In this paper a new synthesis of variable even-order IIR digital filters is proposed. The cut-off frequency of the filter can be changed by a single parameter. The fundamental filter structure is a cascade of second-order sections. The multiplier coefficients of each section are determined by using the Taylor series expansion of the lowpass to lowpass frequency transformation. For this method any second-order section can be used as a prototype, but here in this paper only the direct form and the lattice form are described. Unlike the conventional method, any transfer functions can be used for the proposed method. Finally a designed example shows that the proposed filter has wider tuning range than the conventional filter, and the advantage of the proposed filters is confirmed.

This paper proposes a technique for designing two-dimensional (2-D) digital filters approximating an arbitrary magnitude function. The technique is based on 2-D spectral factorization and rational approximation of the complex exponential function. A 2-D spectral factorization technique is used to obtain a recursively computable and stable system with nonsymmetric half-plane support from the desired 2-D magnitude function. Since the obtained system has an exponential function type transfer function and cannot be realized directly in a rational form, a class of realizable 2-D digital filters is introduced to approximate the exponential type transfer function. This class of filters referred to as two-dimensional log magnitude approximation (2-D LMA) filters can be viewed as an extension of the class of 1-D LMA filters to the 2-D case. Filter coefficients are given by the 2-D complex cepstrum coefficients, i.e., the inverse Fourier transform of the logarithm of the given magnitude function, which can be efficiently computed using 2-D FFT algorithm. Consequently, computation of the filter coefficients is straightforward and efficient. A simple stability condition for the 2-D LMA filters is given. Under this condition, the stability of the designed filter is guaranteed. Parallel implementation of the 2-D LMA filters is also discussed. Several examples are presented to demonstrate the design capability.

Unlike a noise removal recursive or averaging filter, this letter presents a temporal filter which attenuates temporal high frequency components and improves visual effects. Although temporal aliasing occurs, the proposed filter proceeds temporal bandlimitation not affected by them. To reduce effects caused by aliasing components, a spatial filtering which is applied along the trajectory of motion is investigated. The proposed filter presents a de-aliasing and effective bandlimiting characteristics as well as reducing of noises.

Binaural effects in two measures are studied. One measure is the detectable limen of click sounds under lateralization of diotic or dichotic noise signals, and the other is phoneme articulation score under localization or lateralization of speech and noise signals. The experiments use a headphones system with listener's own head related transfer function (HRTF) filters. The HRTF filter coefficients are calculated individually from the impulse responses due to the listener's HRTF measured in a slightly sound reflective booth. The frequency response of the headphone is compensated for using an inverse filter calculated from the response at the subject's own ear canal entrance point. Considering the speech frequency band in tele-communication systems is not sufficiently wide, the bandwidth of the HRTF filter is limited below 6.2 kHz. However, the experiments of the localization simulation in the horizontal plane show that the sound image is mostly perceived outside the head in the simulated direction. Under simulation of localization or lateralization of speech and noise signals, the phoneme articulation score increases when the simulation spatially separates the phonemes from the noise signals while the total signal to noise ratio for both ears is maintained constant. This result shows the binaural effect in speech intelligibility under the noise disturbance condition, which is regarded as a part of the cocktail party effect.

There have been several studies related to a reduction of the amount of computational resources used by Turing machines. As consequences, Linear speed-up theorem", tape compression theorem" and reversal reduction theorem" have been obtained. In this paper, we discuss a leaf reduction theorem on alternating Turing machines. Recently, the result that one can reduce the number of leaves by a constant factor without increasing the space complexity was shown for space- and leaf-bounded alternating Turing machines. We show that for time- and leaf-bounded alternating Turing machines, the number of leaves can be reduced by a constant factor without increasing time used by the machine. Therefore, our result says that a constant factor on the leaf complexity does not affect the power of time- and leaf-bounded alternating Turing machines.

A new MOS linear operational transconductance amplifier (OTA) for the up-to-4 MHz range OTA-C filters is proposed. The proposed OTA is designed using a new linearizing technique based on bias-current modulation, to compensate nonlinearities in the transfer characteristic of the conventional current-source-biased source-coupled pair. The design and SPICE simulation are presented in detail, assuming the implementation by the typical p-well CMOS process. The simulation of a 3.58 MHz OTA-C band-pass filter built with the proposed OTAs showed close agreement with the desired performance.