In this work, a new structure of M-channel linear-phase paraunitary filter banks is proposed, where M is even. Our proposed structure can be regarded as a modification of the conventional generalized linear-phase lapped orthogonal transforms (GenLOT) based on the discrete cosine transform (DCT). The main purpose of this work is to overcome the limitation of the conventional DCT-based GenLOT, and improve the performance of the fast implementation. It is shown that our proposed fast GenLOT is superior to that of the conventional technique in terms of the coding gain. This work also provides a recursive initialization design procedure so as to avoid insignificant local-minimum solutions in the non-linear optimization processes. In order to verify the significance of our proposed method, several design examples are given. Furthermore, it is shown that the fast implementation can be used to construct M-band linear-phase orthonormal wavelets with regularity.

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.

A CMOS fully balanced current-mode filter is presented. A fully balanced current-mode integrator which is the basic building block is implemented by adding a very simple common-mode-rejection mechanism to fully differential one. The fully balanced operation can eliminate even order distortion, which is one of the drawbacks in previous continuous current-mode filter. Moreover, the additional circuit can work as not only common-mode-rejection mechanism but also Q-tuning circuit which compensates lossy elements due to finite output impedance of MOS FET. A prototype fifth-order low-pass lad-der filter designed in a standard digital 0.8µm CMOS process achieved a cut-off frequency (fC) of 100MHz; fC was tunable from 75MHz to 120MHz by varying a reference bias current from 50µA to 150µA. Using a single 3V power supply with a nominal reference current of 100µA, power dissipation per one pole is 30mW. The active filter area was 0.011mm2/pole and total harmonic distortion (THD) was 0.73 [%] at 80MHz, 80µA amplitude signal. Furthermore, by adjusting two bias currents, on chip automatic both frequency and Q controls are easily implemented by typical tuning systems, for example master-slave tuning systems [1].

We present a systematic theory for the optimum sub-band interpolation using a given analysis or synthesis filter bank with the prescribed coefficient bit length. Recently, similar treatment is presented by Kida and quantization for decimated sample values is contained partly in this discussion [13]. However, in his previous treatment, measures of error are defined abstractly and no discussion for concrete functional forms of measures of error is provided. Further, in the previous discussion, quantization is neglected in the proof of the reciprocal theorem. In this paper, linear quantization for decimated sample values is included also and, under some conditions, we will present concrete functional forms of worst case measures of error or a pair of upper bound and lower limit of those measures of error in the variable domain. These measures of error are defined in Rn, although the measure of error in the literature [13] is more general but must be defined in each limited block separately. Based on a concrete expression of measure of error, we will present similar reciprocal theorem for a filter bank nevertheless the quantization for the decimated sample values is contained in the discussion. Examples are given for QMF banks and cosine-modulated FIR filter banks. It will be shown that favorable linear phase FIR filter banks are easily realized from cosine-modulated FIR filter banks by using reciprocal relation and new transformation called cosine-sine modulation in the design of filter banks.

This paper discusses the fixed-point smoothing and filtering problems given lumped covariance function of a scalar signal process observed with additive white Gaussian noise. The recursive Wiener smoother and filter are derived by applying an invariant imbedding method to the Volterra-type integral equation of the second kind in linear least-squares estimation problems. The resultant estimators in Theorem 2 require the information of the crossvariance function of the state variable with the observed value, the system matrix, the observation vector, the variance of the observation noise and the observed value. Here, it is assumed that the signal process is generated by the state-space model. The spectral factorization problem is also considered in Sects. 1 and 2.

A filter configuration that allows configuration of any transfer function used the state variable is discribed as an application of the second generation current conveyors (CCIIs) to RC networks. The filter types discussed are low-pass filter (LPF), high-pass filter (HPF), band-pass filter (BPF), all-pass filter (APF), and band-elimination filter (BEF). The filter circuit consists of four CCIIs and allows tandem connections. The device sensitivity and CCII's sensitivity to transfer coefficient are relatively low. The filter circuit that allow simultaneous configuration wewe fabricated. An experimental result at around 10kHz was obtained for the filters. In the case, the LPF, HPF, BPF, APF, and BEF characteristics are obtained at Q value of 5.0.

Taking a 1.5-GHz SAW antenna duplexer for PDC, we have developed a new configuration for the transmitter final stage filter and a new weighting technique for the receiver top filter. These transmitter and receiver filters provide insertion losses as low as 0.8 and 1.6 dB, respectively. Combining the filters, we have developed a miniature antenna duplexer of which size is 1.40.60.2 cm3 , several-time smaller than that of a conventional dielectric-filter duplexer. It also ensures sufficient power-handing capabilities.

A TE010 mode dielectric resonator is proposed to be used in a millimeter-wave filter. The resonator was fabricated using the photolithographic technique, and high unloaded Q of 1610 was obtained at 60 GHz. A planar circuit type millimeter-wave filter, using TE010 mode dielectric resonators, was fabricated using NRD guides as input and output circuits. The measured filter characteristics agreed with calculated values well. The filter can be applicable to future millimeter-wave mobile communications systems.

An extended configuration of a stepped impedance comb-line filter is presented. The parallel stripline sections of stepped impedance resonators are coupled electromagnetically and a coupling capacitor is introduced. The creation of an attenuation pole near the passband was detailed. A design procedure for the two-pole extended filter is derived from an analysis using even-and odd-mode impedances. Experimental filters were constructed by ceramic lamination technique. They exhibited excellent performances suitable for portable telephones.

A new superresolution technique is proposed for high-resolution estimation of the scattering analysis. For complicated multipath propagation environment, it is not enough to estimate only the delay-times of the signals. Some other information should be required to identify the signal path. The proposed method can estimate the frequency characteristic of each signal in addition to its delay-time. One method called modified (Root) MUSIC algorithm is known as a technique that can treat both of the parameters (frequency characteristic and delay-time). However, the method is based on some approximations in the signal decorrelation, that sometimes make problems. Therefore, further modification should be needed to apply the method to the complicated scattering analysis. In this paper, we propose to apply a time-domain null filtering scheme to reduce some of the dominant signal components. It can be shown by a simple experiment that the new technique can enhance estimation accuracy of the frequency characteristic in the Root-MUSIC algorithm.

A novel method of multichannel surface EMG processing has been developed to compensate for the distortion in bipolar surface EMG signals due to the movement of innervation zones. The distortion of bipolar surface EMG signals was mathematically described as a filtering function. A compensating technique for such distorted bipolar surface EMG signals was developed for the brachial biceps during dynamic contractions in which the muscle length and tension change. The technique is based on multichannel surface EMG measurement, a method for estimating the movement of an innervation zone, and the inverse filtering technique. As a result, the distorted EMG signals were compensated and transformed into nearly identical waveforms, independent of the movement of the innervation zone.

It is known that an anticausal IIR filter can be realized in real time by using the time reversed section technique. When combined with a casual IIR filter, the overall transfer function can yield exact linear phase characteristic in theory. This paper presents a new method for designing complex IIR digital filters with exact linear phase. The design problem of IIR filters with exact linear phase can be reduced to magnitude-only filter design. The proposed procedure is based on the formulation of an eigenvalue problem by using Remez exchange algorithm. By solving the eigenvalue problem to compute the real maximum eigenvalue, the solution of the rational interpolation problem can be achieved. Therefore, the optimal filter coefficients are easily obtained through a few iterations. The proposed design algorithm not only retains the speed inherent in Remez exchange algorithm, but also Simplifies the interpolation step because is has been reduced to the computation of the real maximum eigenvalue. Several examples are presented to demonstrate the effectiveness of the proposed method.

Induced noises from breaking contact arc discharge and sliding contact discharge of dc motor are measured by pick up coil and current probe. Statistical properties, amplitude distribution probability (APD), of induced noise waveform are analyzed by simple method using intermediate frequency of spectrum analyzer. It is shown that APD characteristics can be used to estimate statistical characteristics and peak value of induced noise. Simulation model of the noise made by the combination of Gaussian noise is mentioned. The model called the composite noise generator (CNG) can be good fit to the real characteristics of both noises from breaking arc and dc motor. Applications of the CNG for noise filter using toroidal coil shows that the CNG is useful to realize the test of noise suppression characteristics. What parameters of the CNG should be considered is described for further applications.

The purpose of this paper is to provide a practical tool for performing a shift operation in orthonormal compactly supported wavelet bases. This translation τ of a discrete sequence, where τ is a real number, is suitable for filter bank implementations. The shift operation in this realization is neither related to the analysis filters nor to the synthesis filters of the filter bank. Simulations were done on the Daubechis wavelets with 12 coefficients and on complex valued wavelets. For the latter ones a real input sequence was used and split up into two subsequences in order to gain computational efficiency.

We present a new method to cancel interfering sinusoidal signals. In this method, the Interpolated FFT (IpFFT) algorithm is used to estimate the parameters of the interference signal: frequency, amplitude and phase. The cancellation is then performed in the time domain. In order for the IpFFT to perform reliably, accurate spectral information about the interference signal is needed. Since, the information signal masks the interference signal, it becomes difficult to estimate the parameters of the interference signal. To alleviate this masking effect, two techniques are discussed here. These techniques involve frame update of interference spectral information of the interference signal, and adaptive averaging. Significant improvement over conventional frequency domain filterings is achieved. The price paid is only little, beyond the computation of the FFT. Comparison with the conventional frequency domain filter shows that our system has approximately 5dB better cancellation capability for a single interfering signal.

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 letter proposes evolutionary digital filters (EDFs) as new adaptive digital filters. The EDF is an adaptive filter which is controlled by adaptive algorithm based on the evolutionary strategies of living things. It consists of many linear/time-variant inner digital filters which correspond to individuals. The adaptive algorithm of the EDF controls and changes the coefficients of inner filters using the cloning method (the asexual reproduction method) or the mating method (the sexual reproduction method). Thus, the search algorithm of the EDF is a non-gradient and multi-point search algorithm. Numerical examples are given to show the effectiveness and features of the EDF such that they are not susceptible to local minimum in the multiple-peak performance surface.

An adaptive algorithm for a single sinusoid detection using IIR bandpass filter with parallel block structure has been proposed by Nishimura et al. However, the algorithm has three problems: First, it has several input frequencies being impossible to converge. Secondly, the convergence rate can not be higher than that of the scalar structure. Finally, it has a large amount of computation. In this paper, a new algorithm is proposed to solve these problems. In addition, a new structure is proposed to reduce the amount of computation, in which the adaptive control signal generator is realized by the paralel block structure. Simulation results are given to illustrate the performance of the proposed algorithm.

This paper proposes an algorithm for the design of FIR digital filters whose coefficients have CSD representations. The total number of nonzero digits is specified. A set of filters whose frequency responses have less than or equal to a given Chebyshev error have their coefficients in a convex polyhedron in the Euclid space. The proposed algorithm searches points where a coefficient is maximum or minimum in the convex polyhedron by using linear programing. These points are connected whih the origin to make a convex cone. Then the algorithm evaluates CSD points near these edges of the cone. Moving along these edges means the scaling of frequency responses. The point where the frequency response is the best among all the candidates under the condition of specified total number of nonzero digits is selected as the solution. Several techniques are used to reduce the calculation time. Design examples show that the proposed method can design better frequency responses than the conventional methods.

A cumulant-based lattice algorithm for multichannel adaptive filtering is proposed in this paper. Proposed algorithm takes into account the advantages of higer-order statistics, that is, improvement of estimation accuracy, blindness to colored Gaussian noise and the possibility to estimate the nonminimum-phase system etc. Without invoking the Instrumental Variable () method as used in other papers [1], [2], the algorithm is derived directly from the recursive pseudo-inverse matrix. The behavior of the algorithm is illustrated by numerical examples.