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.

An unbiased estimation method for symmetric noncausal ARMA model parameters is presented. The proposed algorithm works in two steps: first, a spectrally equivalent causal system is identified by lattice whitening filter and then the equivalent noncausal system is reconstructed. For AR system with noise or ARMA system without noise, the proposed method does not need any iteration method nor any optimization procedure. An estimation method of noise variance when the observation is made in noisy situation is discussed. The potential capabilities of the algorithm are demonstrated by using some numerical examples.

Recently, we proposed a low power consumption FIR switched-capacitor filter constructed with capacitors having capacitances in proportion to square roots of transfer function coefficient values. It is referred to as an FIR semi-parallel cyclic type (SPCT) filter. In this paper, we present IIR SPCT filter. It needs only a single operational amplifier, hence being low power consumption. The IIR SPCT filter has smaller total capacitance than one of the IIR parallel cyclic type (PCT) filter and better high frequency response than one of the IIR transfer function coefficient ratio (TCR) filter. As a whole, the IIR SPCT filter has middle performance of the IIR PCT and TCR filters for the total capacitance, the number of types of clock pulses, and high frequency response.

Optimum filters for an image restoration are formed by a degradation operator, a covariance operator of original images, and one of noise. However, in a practical image restoration problem, the degradation operator and the covariance operators are estimated on the basis of empirical knowledge. Thus, it appears that they differ from the true ones. When we restore a degraded image by an optimum filter belonging to the family of Projection Filters and Parametric Projection Filters, it is shown that small deviations in the degradation operator and the covariance matrix can cause a large deviation in a restored image. In this paper, we propose new optimum filters based on the regularization method called the family of Regularized Projection Filters, and show that they are stable to deviations in operators. Moreover, some numerical examples follow to confirm that our description is valid.

A pipelined architecture is proposed for the normalized least mean square (NLMS) adaptive digital filter (ADF). Pipelined implementation of the NLMS has not yet been proposed. The proposed architecture is the first attempt to implement the NLMS ADF in the pipelined fashion. The architecture is based on an equivalent expression of the NLMS derived in this study. It is shown that the proposed architecture achieves a constant and a short critical path without producing output latency. In addition, it retains the advantage of the NLMS, i. e. , that the step size that assures the convergence is determined automatically. Computer simulation results that confirm that the proposed architecture achieves convergence characteristics identical to those of the NLMS.

Digital Signal Processors with complex arithmetic capability (DSP-C) are useful for various applications. In this paper, we propose a method for the effective implementation of specific circuits with real coefficients on DSP-C. DSP-C has special hardware such as a complex multiplier so that a complex calculation can be performed with only one instruction. First, we show that nodes with two real coefficient input branches can be implemented by complex multiplications. We apply this implementation to 2D circuits and transversal circuits with real coefficients. Next, we introduce a new computational mode (Advanced mode) and a new multiplier into PSI, a kind of DSP-C which has been proposed already, in order to process the circuits effectively. The effectiveness of the proposed method is shown by simulation in the last part.

The system architectures, which allow a high performance fully balanced (FB) system based on ordinary/modified single-ended opamps to be implemented, are investigated and the basic and general requirements are formulated. Two new methods of an FB analog system design, which contribute towards achieving both a high performance IC system implementation and a great reduction of the design time are presented. It is shown that a single-ended system based on any type of opamp (rail-to-rail, constant gm, etc. ), realized in any technology (CMOS, bipolar, BiCMOS, GaAs), can be easily and effectively converted to its FB counterpart in a very practical way. Using the proposed rules, any FB system implementation with opamps (data converter, modulator, filter, etc. ) requires only a single-ended system version design and the drawbacks related to a conventional FB system design are avoided. The principles of the design are pointed out and they are verified by experimental results.

This paper proposes a method to transform a CORDIC ARMA lattice filter, which is originally realized for signal analysis, into a signal synthesis lattice filter (CORDIC ARMA lattice synthesis filter). In order to perform such a transformation and then obtain the CORDIC ARMA lattice synthesis filter, we must implement the followings with CORDIC: (1) the structure of the altered lattice filter; and (2) an angle calculation module. However, we cannot achieve such an implementation as an extension of the CORDIC ARMA lattice filter algorithm. Therefore, this paper proposes CORDIC implementation schemes for both the structure and module, and then we realize the CORDIC ARMA lattice synthesis filter. By using CORDIC processors, the elementary sections of the CORDIC ARMA lattice synthesis filter are efficiently implemented without any multipliers. Since the obtained signal synthesis lattice filter consists of dedicated CORDIC processors, it keeps the advantage of the CORDIC ARMA lattice filter, that is a simple structure.

A novel method is proposed to calculate the distributed coupling of dual-modes in a circular resonator. New theoretical expressions are devised to accumulate the infinitesimal coupling between orthogonal modes and their validity is justified by the FD-TD analysis and experiments. The distributed coupling concept of a circular disk resonator is applied to a square disk resonator to calculate its resonant frequency. We have fabricated two types of low-profile dual-mode square dielectric disk resonator BPF, using high dielectric constant material (εr = 93) having a dimension of 5 mm 5 mm 1 mm. The filter characteristics are explained by the transmission line circuit model.

Adaptive infinite impulse response (IIR) digital filter implemented using a cascade of second order direct form allpass filters and a finite impulse response (FIR) filter, has the property of its poles converging to those of the unknown system. In this paper we implement the adaptive allpass-FIR digital filter using a lattice allpass filter with minimum number of multipliers. We then derive a simple adaptive algorithm, which does not increase the overall number of multipliers of the proposed adaptive digital filter (ADF) in comparison to the ADF that uses the direct form allpass filter. The proposed structure and algorithm exhibit a kind of orthogonality, which ensures convergence of the poles of the ADF to those of the unknown system. Simulation results confirm this convergence.

We present a receiver structure with joint blind equalization, carrier recovery, and timing recovery. The blind equalizer employs a decomposition transversal filtering technique which can reduce the complexity of convolution to about a half. We analyze the performance surface of the equalizer cost function and show that the global minima correspond to perfect equalization. We also derive proper initial tap settings of the equalizer for convergence to the global minima. We describe the timing recovery and the carrier recovery methods employed. And we describe a startup sequence to bring the receiver into full operation. The adaptation algorithms for equalization, carrier recovery, and timing recovery are relatively independent, resulting in good operational stability of the overall receiver. Some simulation results for cable-modem type of transmission are presented.

Basic characteristics of a short-ended coplanar waveguide (CPW) resonator of good spurious suppression property is studied. The resonator is loaded with open tubs at its middle position and makes a fully planar structure. The length of the resonator is shortened almost by half and the first spurious resonance goes up to more than 3 times of the fundamental resonant frequency without degradation of unloaded Q(Q0). The origin and property of spurious response is thoroughly investigated to show the advantage and the limit of this configuration. The external Q(Qe) and fundamental resonant frequency of the resonator are also clarified theoretically and experimentally. Using those result, a bandpass filter (BPF) is designed on the basis of the narrow band approximation is realized and its transmission characteristics are examined theoretically and experimentally. The spurious suppression characteristics have been realized by the present filter in accordance with the expectation.

Call tracking data contains a calling address, called address, service type, and other useful attributes to predict a customer's calling activity. Call tracking data is becoming a target of data mining for telecommunication carriers. Conventional data-mining programs control the number of association rules found with two types of thresholds (minimum confidence and minimum support), however, often they generate too many association rules because of the wide variety of patterns found in call tracking data. This paper proposes a new method to reduce the number of generated rules. The method proposed tests each generated rule based on Akaike Information Criteria (AIC) without using conventional thresholds. Experiments with artificial call tracking data show the high performance of the proposed method.

We propose and demonstrate a simple polarization-independent construction of a local node for optical WDM ring networks using a centralized multiwavelength light source (MWLS). The node is simply composed of a 4-port optical circulator, an add/drop multiplexing (ADM) filter, a reflective modulator, and a drop fiber Bragg grating (FBG). A Faraday rotator mirror (FRM) is used to enable an LiNbO3 intensity modulator to operate in the polarization-independent mode. We examine three ADM filters, an interference filter, a fiber Fabry-Perot (FP) filter, and a set of FBG's. An optical WDM system experiment is performed to demonstrate the feasibility of the proposed node construction.

This paper proposes a scalable multiecho cancellation system based on multiple autonomic and distributed echo canceler units. The proposed system does not have any common control section. Distributed control sections are equipped with in multiple echo cancelers operating autonomically. Necessary information is transferred from one unit to the next one. When the number of echoes to be canceled is changed, the necessary number of echo canceler units, each of which may be realized on a single chip, are simply plugged in or unplugged. The proposed system also provides fast convergence thanks to the novel coefficient location algorithm which consists of flat-delay estimation and constrained tap-position control. The input signal is evaluated at each tap to determine when to terminate flat-delay estimation. The number of exchanged taps is selected larger in flat-delay estimation than in constrained tap-position control. The convergence time with a colored-signal input is reduced by approximately 50% over STWQ, and 80% over full-tap NLMS algorithm. With a real speech input, the proposed system cancels the echo by about 20 dB. Tap-positions have been shown to be controlled correctly.

This paper proposes a novel asynchronous direct sequence/code division multiple access (DS/CDMA) communication system using analog pseudo noise (PN) sequences that have an orthogonal relation for all active users. Analog PN sequences are produced by an adaptive filter called a "code-orthogonalizing filter" (COF). In a base station receiver, the tap coefficients of the COF can be adaptively controlled "to orthogonalize" or "to approach to orthogonalize" various received PN sequences. The elements of the analog PN sequences consist of the tap coefficients of the COF. The analog PN sequence produced is assigned to the transmitter of each user in order. As a result, multiple access interference (MAI) caused by other users can be reduced considerably, and multiple access capacity increased by the proposed system compared with matched filter (MF) reception and COF reception.

In this paper, we present a numerical design method for two-dimensional (2-D) FIR linear-phase (LP) quincunx filter banks (QFB) with equiripple magnitude response and perfect reconstruction (PR). The necessary conditions for the filter length of analysis filters are derived. A dual affine scaling variant (DASV) of Karmarkar's algorithm is employed to minimize the peak ripples of analysis filters and an approximation scheme is introduced to satisfy the PR constraint for the 2-D filter banks (FB). The simulation examples are included to show the effectiveness of this proposed design technique.

Since IIR filters have lower computational complexity than FIR filters, some design methods for IIR filter banks have been presented in the recent literatures. Smith et al. have proposed a class of linear phase IIR filter banks. However this method restricts the order of the numerator to be odd and has some drawbacks. In this paper, we present two design methods for linear phase IIR filter banks. One is based on Lagrange-Multiplier method, and optimal IIR filter banks in least squares sense are obtained. In an other approach, IIR filter banks with the maximum number of zeros are derived analytically.

Large-area high-temperature superconducting films and damage-free processing techniques have been developed to fabricate low insertion loss and sharp skirt filters for mobile telecommunication. An off-axis-type dc sputtering method was employed to deposit Y-Ba-Cu-O films on both sides of the substrate. The surface resistance of the films was about 0. 35 mΩ(at 70 K and 10 GHz). An 11-pole bandpass receiving filter for the IS-95 telecommunication system was designed and fabricated using 60 mm 50 mm YBCO films on a 0. 5-mm-thick MgO substrate. The passband insertion loss at 70 K was about 0. 1 dB with 0. 1 dB ripple. The third-order intercept point of the filter was 49. 5 dBm. We have assembled the filter and a low-noise amplifier in a dewar with a cryocooler. Ultralow-noise performance (noise figure: 0. 5 dB at 70 K) was presented by the cryogenic filter subsystem.

A new or future system of mobile telecommunication is built by new digital technologies to provide an improved and more consistent quality of service for the customers. These digital systems can provide greater number of transmission channel allocation for their subscribers and security. On the digital communication system, distortion of transmitted signal should be eliminated as much as possible for high communication quality. However, the need to both minimize distortion of signal amplifiers and continue to provide good filtering protection can become difficult to achieve with conventional high power amplifiers and filters. In this paper, the application of high-Tc superconducting (HTS) power filters on such digital communication systems and recent progress of filter device developments for those are discussed.