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 describes a CDMA cellular system based on adaptive interference cancellation (CDMA-AIC) with a large capacity. In the CDMA-AIC, each base station employs a single-user type adaptive interference canceller (AIC), which consists of a fractionally chip-spaced code-orthogonalizing filter (COF) and a coherent detector. The AIC adaptively removes power-dominant multiple-access interferences (MAIs) in the cellular system, regardless of whether they are intra-cell interferences or inter-cell interferences, without any information about them, such as spreading codes, signal received timings and channel parameters. Evaluation under the multiple-cell environment demonstrates that the reverse link capacity of the CDMA-AIC with QPSK modulation is 3.6 times as large as the capacity of the CDMA without MAI cancellation. Further, the capacity is less sensitive to transmission power control errors than that of the conventional CDMA systems.

According as the fine LSI process technique develops, the technique to reduce power dissipation of high-frequency integrated analog circuits is getting more important. This paper describes a design of high-frequency integrator with low power dissipation for monolithic leapfrog filters. In the design of the conventional monolithic integrators, there has been a great dfficulty that a high-frequency integrator which can operate at low supply voltage cannot be realized without additional circuits, such as unbalanced-to-balanced conversion circuits and common-mode feedback circuits. The proposed integrator is based on the Miller integrator. By a PNP current mirror circuit, high CMRR is realized. However, the high-frequency characteristic of the integrator is independent of PNP transistors. In addition, it can operate at low supply voltage. The excess phase shift of the integrator is compensated by insertion of the compensation capacitance. The effectiveness of the proposed technique is confirmed by PSPICE simulation. The simulation results of the integrator shows that the common-mode gain is efficiently low and the virtual ground is realized, and that moderate phase compensation can be achieved. The simulation results of the 3rd-order leapfrog filter using the integrator shows that the 50 MHz-cutoff frequency filter is obtained. Its power dissipation in operating 2 V-supply voltage is 5.22 mW.

This paper proposes a topology-independent predistortion for filters using integrators. This employs integrators having the same structure, the same-value elements and an electrically controllable unity-gain frequency and compensates for the deviation of frequency characteristics due to excess phase shifts of integrators without knowledge of a filter topology. The effectiveness of the proposed method is demonstrated through SPICE simulations.

In general, a time-limited signal such as a single sinusoidal waveform framed by a frame period T can be utilized for conveying a multi-level symbol in data transmission. If such a signal is analyzed by the conventional DFT (Discrete Fourier Transform) analysis, the infinite number of frequency components with frequency spacing fD = T1 is needed. This limits the accuracy with which the original frequency of the unframed sinusoidal waverform can be identified. It is especially difficult to identify two similar framed sinusoids whose frequency spacing is narrower than fD. An analytical principle for time-limited signals is therefore proposed by introducing the concept of an Extended Frame into DFT. Waveform analysis more accurate than DFT is achieved by taking into account multiple correlations between extended frames made of an input frame signal and the element frequency components corresponding to the length of each extended frame. In this approach, it is possible to use arbitrary element frequency spacing less than fD. It also allows an element frequency to be selected as a real number times of fD, rather than as an integer times of fD that is used for DFT. With this analyzing mechanism, it is verified that an input frame signal with only the frequency components which coincide with any of the element frequencies can be exactly analyzed. The disturbance caused by the input white noise is examined. As a result, it is found that the superior noise suppression function is achieved by this method over a conventional matched filter. In addition, the error caused by using a finite number of element frequencies and the A/D conversion accuracy required for sampling an input signal are examined, and it is shown that these factors need not impede practical implementation. For this reason, this principle is useful for multi-ary transmission systems, noise tolerant receivers, or systems requiring precise filtering of time limited waveforms.

This paper proposes an integrated interference suppression scheme which realizes interference-resistant satellite digital signal transmission systems. It employs a notch filter in the receiving side to suppress the co-channel interference (CCI) signal. Moreover, the proposed scheme employs an adaptive equalizer combined with a forward error correction (FEC) scheme to improve the Pe (probability of error) performance degradation due to the inter-symbol interference caused by notch filtering of the desired signal. In the typical frequency modulation (FM) CCI environment with a BWi/FN of 2.3 (BWi: interference signal required bandwidth, fN: one half the Nyquist bandwidth of the desired signal), a Δf / fN of 1.05 (Δf: interference frequency offset) and a D/U of 3 dB (desired to undesired (interference) signal power ratio), the proposed scheme improves the required Eb/NO by 1.5 dB at a Pe of 10-4 compared to that without an adaptive equalizer.

A design of current-mode continuous-time filters for low voltage and high frequency applications using complementary bipolar current mirror pairs is presented. The proposed current-mode filters consist of simple bipolar current mirrors and capacitors and are quite suitable for monolithic integration. Since the filters are based on the integrator type of realization, the proposed method can be used for a wide range of applications. The frequency of the filters can easily be changed by the DC controlling current. A fifth-order Butterworth and a thirdorder leapfrog filter with tunable cutoff frequencies from 20 MHz to 100 MHz are designed as examples and simulated by SPICE using standard bipolar parameters.

This paper presents a 15-GHz MMIC direct optical injection-locked oscillator (MMIC OILO) with very-wide locking range that uses photosensitive HBTs. The MMIC OILO consists of an HBT and a positive feedback circuit including a Q-damping variable resistor. By utilizing the high-fT/fmax photosensitive HBT, we realize both high-frequency oscillation of 15 GHz and increased equivalent electrical injection power. In addition to increasing the RF injection power, the Q-damping variable resistor effectively reduces the quality-factor of the oscillator, thus realizing the very wide locking range (f) of 567 MHz (f/fosc3.8%). The locking bandwidth of 3.8% is over 10 times wider than that of any yet reported microwave direct OILO. Furthermore, it is shown that the MMIC OILO can also work as a high-gain Q-variable filter photoreceiver by increasing a Q-damping variable resistance over the self-oscillation suppression range.

A voltage-mode resonant forward converter with capacitor-input filter is proposed, and its static and dynamic characteristics for both half-wave type and full-wave type are revealed by analysis and experiment. As a result, this converter has prominent features of simplicity of circuit configuration, isolation between input and output and high stability.

A design method of 2-D lattice digital filter using the Genetic Algorithm (GA) is proposed. By using the GA. 2-D all-pole lattice filter with the cascade connection of transversal (all-zoro) filter is designed directly from a given desired frequency responce.

We have developed a method to automatically generate a multi-input-adder circuit for an irregular array of partial products. "FASTOOL," an FIR Filter Automatic Synthesis TOOL for an HDL design environment, is proposed for use with this method and with conventional filter coefficient design programs. Filter design from specifications to the structure of Verilog-HDL has been automated. It is possible for a system designer to quickly perform filter LSI optimization by balancing cost and performance.

In this paper we propose a new method for removing the characteristic of the piezoelectric transducer from the received signal in the pulse-echo method so that the time resolution in the determination of transit time of ultrasound in a thin layer is increased. The total characteristic of the pulse-echo system is described by cascade of distributed-constant systems for the ultrasonic transducer, matching layer, and acoustic medium. The input impedance is estimated by the inverse matrix of the cascade system and the voltage signal at the electrical port. From the inverse Fourier transform of input impedance, the transit time in a thin layer object is accurately determined with high time resolution. The principle of the method is confirmed by simulation experiments.

The technique for estimating the parameters of multiple waves provides a convenient tool for analysis of multiple wave-fields and eventually for actual applications to mobile communications. Several algorithms have been proposed for those purposes. However, the best tactics to resolve multiple wave-fields are still imperfectly understood at present. This paper proposes a new method for estimating the angles and power levels of arrival waves based on the extended Kalman filter. A space-variable model which we call a spatial state equation is derived using array element locations and incident angles. It has been shown that by means of the model, the estimation of incident waves can be transformed into the problem of parameter identification in linear system which can be carried out by the extended Kalman filter conveniently. The algorithm is initiated directly by the signal received at each array element. The detailed procedure of an extended Kalman filter approach is given in the paper. The performance of the proposed approach is examined by a simulation study with two signals model. The simulation results show a good estimate performance, even in the case that two waves arrive from close directions.

The performance of decision-feedback equalization combined with maximum-likelihood detection (DFE/ML) using the fixed-delay-tree-search/decision feedback (FDTS/DF) algorithm was estimated analytically in terms of the length of the feedback-filter and the depth of the ML-detector. Performance degradation due to error propagation in the feedback-loop and in the ML-detector was taken into account by using a Markov process analysis. It was quantitatively shown that signal-to-noise-ratio (SNR) performance in high-density magnetic recording channels can be improved by combining an ML-detector with a feedback-filter and that the error propagation in the DFE channel can be reduced by using an ML-detector. Finally, it was found that near-optimum performance with regard to channel SNR and error propagation can be achieved, over the channel density range from 2 to 3, by increasing the sum of the feedback-filter length and the ML-detector depth to six bits.

Conventional subband ADF's (adaptive digital filters) using filter banks have shown a degradation in performance because of the non-ideal nature of filters. To solve this problem, we propose a new type of subband ADF incorporating two types of analysis filter bank. In this paper, we show that we can design the optimum filter bank which minimizes the LMSE (least mean squared error). In other words, we can design a subband ADF with less MSE than that of conventional subband ADF's.

This paper proposes a new algorithm called the fast Projection algorithm, which reduces the computational complexity of the Projection algorithm from (p+1)L+O(p3) to 2L+20p (where L is the length of the estimation filter and p is the projection order.) This algorithm has properties that lie between those of NLMS and RLS, i.e. less computational complexity than RLS but much faster convergence than NLMS for input signals like speech. The reduction of computation consists of two parts. One concerns calculating the pre-filtering vector which originally took O(p3) operations. Our new algorithm computes the pre-filtering vector recursively with about 15p operations. The other reduction is accomplished by introducing an approximation vector of the estimation filter. Experimental results for speech input show that the convergence speed of the Projection algorithm approaches that of RLS as the projection order increases with only a slight extra calculation complexity beyond that of NLMS, which indicates the efficiency of the proposed fast Projection algorithm.

In this paper, a new switched-current auto-tuning filter is proposed. Switched-current (SI) is a current-mode analog sampled-data circuit technique. An SI circuit can be realized using only standard digital CMOS technologies, and is capable of realizing high frequency circuits. The proposed filter is composed of SI-OTA (operational transconductance amplifier) integrators. The gain of an SI-OTA integrator can be electronically controlled by the bias current. The proposed filter is a current controlled filter (CCF) and a PLL technique was used as its tuning method. A 2nd-order SI auto-tuning low-pass filter with 100kHz cutoff frequency was designed assuming a 2µm CMOS process. The characteristics of this SI filter and its tuning characteristics were confirmed by SPICE simulations.

This paper describes modifications to a previously proposed 8-kb/s 4-ms-delay CELP speech coding algorithm with a view to improving the speech quality while maintaining low delay and only moderately increasing complexity. The modifications are intended to improve the effectiveness of interframe pitch lag prediction and the sub-optimality level of the excitation coding to the backward adapted synthesis filter by using delayed decision and joint optimization techniques. Results of subjective listening tests using Japanese speech indicate that the coded speech quality is significantly superior to that of the 8-kb/s VSELP coder which has a 20-ms delay. A method that reduces the computational complexity of closed-loop 3-tap pitch prediction with no perceptible degradation in speech quality is proposed, based on representing the pitch-tap vector as the product of a scalar pitch gain and a normalized shape codevector.

A Ku-band transversal filter with a center frequency of 12 GHz and a bandwidth of 6 GHz using directional couplers made of a multilayer ceramic has been proposed and developed. The directional coupler can realize a wide range of frequency characteristics, e.g. coupling of 3.50.5 dB in the frequency range of 10 to 17 GHz and a wide range of coupling values, i.e. 3 to 35 dB. Calculations have confirmed that the increase in insertion loss due to a decreasing Q-factor can be much less than that for a resonator filter. The transversal filter was fabricated without additional tuning, and measured results agreed well with calculated values.

An adaptive digital filter with adaptive sampling phase is proposed. The structure of the filter makes use of an adaptive delay device at the input of the filter. The algorithm is derived to determine the value of the delay and the filter coefficients by minimizing MSE (mean square error) between the desired signal and the filter output. The computer simulation of the convergence of the proposed adaptive filter with the input of sinusoidal wave and BPSK modulated wave are shown. According to the simulation, the MSE of the proposed adaptive delay algorithm is lower than that of the conventional LMS algorithm.