In this paper,novel techniques for designing Finite Impulse Response (FIR) digital integrators have been given. The design is based on analytical approach for computing the weights required in the structures. Exact mathematical formulas for computing these weights have been derived.

This letter presents an efficient design method of multiplierless 2-D state-space digital filters (SSDFs) based on a genetic algorithm. The resultant multiplierless 2-D SSDFs, whose coefficients are represented as the sum of two powers-of-two terms, are attractive for high-speed operation and simple implementation. The design problem of multiplierless 2-D SSDFs described by Roesser's local state-space model is formulated subject to the constraint that the resultant filters are stable. To ensure the stability for the resultant 2-D SSDFs, a stability test routine is embedded in th design procedure.

This paper describes newly developed miniaturized stepped impedance resonators with a double coaxial structure (DC-SIR's) and their application to bandpass filters. The new DC-SIR's using dielectric material are devised for more compact and lower frequency bandpass filters. Fundamental characteristics such as resonance properties and unloaded-Q make it clear that DC-SIR's have attractive features that miniaturization can be achieved without Q-factor degradation. Trial 400 MHz bandpass filters incorporating DC-SIR's are also made. Experimental results of bandpass filters proved that DC-SIR's are applicable to lower frequency band radio equipment and able to contribute to the expansion of applicable frequency ranges of dielectric coaxial resonators.

In this letter, we introduce a predictor based least square (PLS) algorithm. By involving both order- and time-update recursions, the PLS algorithm is found to have a more stable performance compared with the stable version (Version II) of the RLS algorithm shown in Ref.[1]. Nevertheless, the computational requirement is about 50% of that of the RLS algorithm. As an application, the PLS algorithm can be applied to the fast Newton transversal filters (FNTF). The FNTF algorithms suffer from the numerical instability problem if the quantities used for extending the gain vector are computed by using the fast RLS algorithms. By combing the PLS and the FNTF algorithms, we obtain a much more stable performance and a simple algorithm formulation.

Design techniques are presented for high performance microstrip bandpass filters using GaAs FETs for loss compensation. The filters are based on conventional planar filter topologies with the addition of GaAs FET negative resistance circuits to amplify the signal within the resonators via a reflection-mode of amplification. Three practical filters have been demonstrated using these negative resistance techniques: (1) A filter employing an active loop configuration, (2) a dual-mode microstrip ring resonator filter, and (3) an end-coupled half-wavelength resonator filter. The investigation of this negative resistance method of loss compensation has led to the development of an exciting new type of miniaturised filter which employs MIC microstrip resonators with MMIC negative resistance chips bonded into the filter for loss compensation. This approach has the advantage of combining the proven capabilities of established MIC microstrip filter topologies with the excellent reproducibility of the MMIC loss compensation circuits.

An adaptive joint-process IIR filter with generalized lattice structure is constructed. This filter can borrow both FIR and IIR features and simultaneously holds the well-known merits of lattice structure.

In this paper we present a design method for all-pass networks with consideration of the stability. It is based on the eigen filter method and Remez exchange algorithm is used to obtain the equiripple phase error solution. In the iteration of the proposed algorithm, the eigen values besides maximum eigen value are used in order to obtain a stable all-pass networks.

This paper presents a new structure for noise and echo cancelers based on a combined fast abaptive algorithm. The main purpose of the new structure is to detect both the double-talk and the unknown path change. This goal is accomplished by using two adaptive filters. A main adaptive filter Fn, adjusted only in the non-double-talk period by the normalized LMS algorithm, is used for providing the canceler output. An auxiliary adaptive filter Ff, adjusted by the fast RLS algorithm, is used for detecting the double-talk and obtaining a near optimum tap-weight vector for Fn in the initialization period and whenever the unknown path has a sudden or fast change. The proposed structure is examined through computer simulation on a noise cancellation problem. Good cancellation performance and stable operation are obtained when signal is a speech corrupted by a white noise, a colored noise and another speech signal. Simulation results also show that the proposed structure is capable of distinguishing the near-end signal from the noise path change and quickly tracking this change.

This paper discusses a CMOS differential-difference amplifier circuit suitable for low voltage operation. A new multiple weighted input transconductor circuit structure is suggested to be use in DDA implementation. The proposed DDA can be employed in several analog/digital systems to improve their parameters. Selected examples of the proposed transconductor/DDA applications are also discussed.

We present a method of extracting the digital inphase (I) and quadrature (Q) components from oversampled bandpass signals using narrow-band bandpass Hilbert transformers. Down-conversion of the digitized IF signals to baseband and reduction of the quantization noise are accomplished by the multistage decimator with the complex coefficient bandpass digital filters (BPFs), which construct the bandpass Hilbert transformers. Most of the complex coefficient BPFs in the multistage decimator can be replaced with the lowpass filters (LPFs) under some conditions, which reduces computational burden. We evaluate the signal to quantization noise ratio of the I and Q components for the sinusoidal input by computer simulation. Simulation results show that the equivalent amplitude resolution of the I and Q components can be increased by 3 bits in comparison with non-oversampling case.

A new gradient type adaptive algorithm is proposed in this paper. It is formulated based on the least squares criteria while the conventional gradient algorithms are based on the least mean square criteria. The proposed algorithm has two variable parameters and by changing them we can adjust the characteristic of the algorithm from the RLS to the LMS depending on the environment. This capability of adjustment achieves the possibility of providing better solutions. However, not only it provides better solutions than the conventional algorithms under some conditions but also it provides a very interesting theoretical view point. It provides a unified view point of the adaptive algorithms including the conventional ones, i.e., the LMS or the RLS, as limited cases and it enables us to analyze the bounds for those algorithms.

This paper discusses a new design method for 2-D variable FIR digital filters, which is an extension of our previous work for 1-D case. The method uses a 3-D prototype FIR filter whose cross-sections correspond to the desired characteristics of 2-D variable FIR filters. A 2-D variable-angle FIR fan filter is given as a design example.

In this paper we propose a method for increasing the reliability in multiprocessor realization of lowpass and highpass FIR digital filters possessing a maximally flat magnitude response. This method is based on the use of array realization of the filter which has been proposed earlier by the authors. It is shown that if a processing module of the array functions erroneously, it is possible to exclude the module and still obtain a lowpass FIR filter. However, as a price we should tolerate a slight degradation in the magnitude response of the filter that is equivalent to a wider transition band. We also analyze the behavior of the filter when our proposed schemes are implemented on more than one module. The justification of our approach is based on that a slight degradation of the spectral characteristics of a filter may be well tolerated in most filtering applications and thus a graceful degradation in the frequency domain can sufficiently reduce the vulnerability to errors.

The factored state space approach (FSS) can be a powerful mathematical tool for the synthesis and analysis of non state space digital filters. In the following letter, this technique is used for the rederivation of some classes of low sensitivity filters described by a II-cascade two-pair structure. This method leads to a simplified synthesis algorithm (with applications to automated synthesis procedures for many classes of non state space digital filters) as well as a straightforward analysis of roundoff noise and norm scaling problems.

A bounded complex (BC) digital transfer function realized with a II-cascade structure of Lossless Bounded Complex (LBC) two-pairs is known to have low magnitude sensitivity. In this letter, it is shown that the two-pairs parameters depend directly on some invariants of the transfer function corresponding to the transmission zeros of the structure. An analysis of the existence and the numbering of these invariants leads to a simplified automated LBC filter structure design avoiding the need for polynomial manipulations. These results are also easily applied for the real filtering case.

The scope of this paper is the realization of FIR digital filters with an emphasis on linear phase and maximally flat cases. The transfer functions of FIR digital filters are polynomials and polynomial evaluation algorithms can be utilized as realization schemes of these filters. In this paper we investigate the application of a class of polynomial evaluation algorithms called "recursive triangles" to the realization of FIR digital filters. The realization of an arbitrary transfer function using De Casteljau algorithm, a member of the recursive triangles used for evaluating Bernstein polynomials, is studied and it is shown that in some special and important cases it yields efficient modular structures. Realization of two dimensional filters based on Bernstein approximation is also considered. We also introduce recursive triangles for evaluating the power basis representation of polynomials and give a new multiplier-less maximally flat structure based on them. Finally, we generalize the structure further and show that Chebyshev polynomials can also be evaluated by the triangles. This is the triangular counterpart of the well-known Chebyshev structure. In general,the triangular structures yield highly modular digital filters that can be mapped to an array of concurrent processors resulting in high speed and effcient filtering specially for maximally flat transfer functions.

This paper proposes a novel method to realize highly linear MOS circuits using MOSFETs in the nonsaturation region. The proposed method is based on the cancellation of nonlinearity of two MOSFETs by using a current inversiontype negative impedance converter. First, grounded and floating resistor realizations are discussed. Next, by exploiting the MOS resistor circuits, gyrators and inductors are realized. As an application example, a third-order doubly-terminated LC filter is simulated. SPICE analysis shows low total harmonic distortions, excellent controllability and small gain error in the passband.

The requirement of structural boundedness or passivity leads to important classes of digital filters among which are the wave digital (WD) filters and the LBR cascade structures having low coefficient sensitivity. Contrary to the WD filters, the LBR filters are directly synthesized in z-domain and several authors presented different approaches for a better understanding of the synthesis procedure especially for complex transfer functions. Some tentatives were also made to give parallels between passive analog and digital filters (i.e. WD or LBR filters). A general approach to LBR synthesis with transmission zeros not necessarily on the unit circle is presented along with some explicit expressions for the LBR (and the generalized complex counterpart LBC) filter parameters for the realization of an input transfer function. The results can be of interest in automated procedures for low sensitivity digital filter design.

This paper presents a design of low-power CMOS OTA-C filters suitable for on-chip integration of advanced monolithic system LSIs that have analog I/O and digital signal processing capability. First, we discuss the distortion of MOS cross-coupled circuits which have a quite low distortion when the MOS FETs have the square law characteristics. Considering the nonidealities of MOS FET, however, we find that the third harmonic component of signal dominates the total harmonic distortion (THD) of the cross-coupled pair circuit. We propose a new architecture to reduce the 3rd harmonic component. Low distortion operational transconductance amplifiers (OTA) which consist of the proposed low distortion cross-coupled pair are applied to the realization of OTA-Capacitor filters. The SPICE simulation shows that the THD of the filter is 0.0047% and the power dissipation is 22.6 mW.

It is often desired to change the cutoff frequencies of digital filters in some applications like digital electronic instruments. This paper proposes a design of variable lowpass digital filters with wider ranges of cutoff frequencies than conventional designs. Wave digital filters are used for the prototypes of variable filters. The proposed design is based on the frequency scaling in the s-domain, while the conventional ones are based on the z-domain lowpass-to-lowpass transformations. The first-order approximation by the Taylor series expansion is used to make multiplier coefficients in a wave digital filters obtained from a frequency-scaled LC filter become linear functions of the scaling parameter, which is similar to the conventional design. Furthermore this paper discusses the reduction of the approximation error. The curvature is introduced as the figure of the quality of the first-order approximation. The use of the second-order approximation to large-curvature multiplier coefficients instead of the first-order one is proposed.