A narrowband interference in direct sequence spread spectrum communication systems also affects the characteristics of a delay lock loop. In this paper, the delay errors of a baseband delay lock loop (DLL) in the presence of the interference which consists of a narrowband Gaussian noise and several tones are examined, and when a filter is used to reject the interference, the characteristics of the DLL are analyzed using the Fourier method. Furthermore, from the calculation results of the delay error in case where a prediction error filter with two-sided taps is used as the rejection filter, it is shown that the filter is necessary to keep the DLL in the lock-on state.

In this paper, we propose a spread spectrum pulse position modulation (SS-PPM) system, and describe its basic performances. In direct sequence spread spectrum (DS/SS) systems, pseudo-noise (PN) matched filters are often used as information demodulation devices. In the PN matched filter demodulation systems, for simple structure and low cost of each receiver, it is desired that each demodulator uses only one PN matched filter, and that signals transmitted from each transmitter are binary. In such systems, on-off keying (SS-OOK), binary-phase-shift keying (SS-BPSK) and differential phase-shift keying (SS-DPSK) have been conventionally used. As one of such systems, we propose the SS-PPM system; the SS-PPM system is divided into the following two systems: 1) the SS-PPM system without sequence inversion keying (SIK) of the spreading code (Without SIK for short); 2) the SS-PPM system with SIK of the spreading code (With SIK for short). As a result, we show that under the same bandwidth and the same code length, the data transmission rate of the SS-PPM system is superior to that of the other conventional SS systems, and that under the same band-width, the same code length and the same data transmission rate, the SS-PPM system is superior to the other conventional SS systems on the following points: 1) Single channel bit error rate (BER) (BER characteristics of the SS-PPM system improve with increasing the number of chip slots of the SS-PPM system, and as the number of chip slots increases, it approaches Shannon's limit); 2) Asynchronous CDMA BER; 3) Frequency utilization efficiency. In addition, we also show that With SIK is superior to Without SIK on these points.

The spread spectrum system (abbreviated as SS system) is known to be an excellent communication system which resists jamming. Recently, its application to a simplified wireless communication system has been considered to be suited for consumer communication. In Japan, SS wireless LAN system has got the approval on 2.4GHz ISM band already. A compact SS transceiver for the SS wireless LAN is realized, whose data ratio is 230kbps. The SS transceiver is based on a direct sequence for the modulation, and the demodulation is carried out by a specially developed SAW device. In the first part of this paper, the technical conditions of the SS wireless LAN are mentioned. Then the SAW device and the principle of the demodulation are discussed. Finally, the configuration of the SS transceiver and the protocol of the SS wireless LAN are presented.

In this paper, we attempt the comparison of the image/signal restoration between Projection Filter, which is regarded as one of the linear optimal filters, and the non-linear filter based on MEM. From the simulation, we show the advantage of MEM restoration filter in restoring noisy degraded images.

This review presents research topics and results on digital signal processing in the last twenty years in Japan. The main parts of the review consist of design and analysis of multidimensional digital filters, multiple-valued logic circuits and number systems for signal processing, and general purpose signal processors.

An automatic tap assignment method in sub-band adaptive filter is proposed in this letter. The number of taps of the adaptive filter in each band is controlled by the mean-squared error. The numbers of taps increase in the bands which have large errors, while they decrease in the bands having small errors, until residual errors in all the bands become the same. In this way, the number of taps in a band is roughly proportional to the length of the impulse response of the unknown system in this band. The convergence rate and the residual error are improved, in comparison with existing uniform tap assignment. Effectiveness of the proposed method has been confirmed through computer simulation.

It is known that the resolution conversion based on orthogonal transform has a problem that is difference of luminance between the converted image and the original. In this paper, the scale factor of the system employing various orthogonal transforms is generally formulated by considering the DC gain, and the condition of alias free for DC component is indicated. If the condition is satisfied, then the scale factor is determined by only the basis functions.

This letter presents a new algorithm for echo cancellers, which prevents the reduction of echo return loss due to a double-talk. The essence of the algorithm is to introduce signal delays to avoid the reduction. A convergence condition in the algorithm was examined by using the IIR filter expression of the NLMS algorithm, and it was concluded that the IIR filter should be a low pass filter with unity gain. The condition is accomplished by selecting a small step gain.

In this letter, a new structure of adaptive IIR notch filter is presented. The structure is based on direct form realization and uses the similar adaptation algorithm given in Ref. (4). A quantitative analysis for convergence properties is developed. It is shown that the proposed structure shows superior performance comparing with previously proposed designs. The results of computer simulations are presented to substantiate the analysis.

The approximation of the gain characteristics of linear phase FIR digital filters is reduced to the approximation by cosine polynomials. Therefore we can easily obtain an optimum solution under the LMS of Chebyshev error criterion. However the optimum solution does not always meet practical specifications, especially in the case where the gain is specified strictly at some angular frequencies. On the other hand in such a case, it is known that interpolation technique can be suitably applied for the approximation mentioned above. However, in this case, we encounter another difficulty in the approximation caused by interpolation. In order to overcome the above difficulty, this paper proposes a new method utilizing both of the interpolation and LMS techniques. Some parameters included in approximating functions are used to satisfy prescribed interpolating conditions and the other parameters are used to minimize the approximation error under the LMS criterion. In addition, interpolation technique is extended to include the case in which also higher derivatives are taken into interpolation conditions to make smooth interpolation. An example is shown to illustrate the effectiveness of the proposed method.

A least squares approximation method of recursive digital filters for finite interval response with zero value outside the interval is presented. According to the characteristic of the method, the modified Gauss Method is utilized in iteratively determining design parameters. Convergence, together with the stability of the resulting filter, are guaranteed.

This paper presents newly developed very small MMIC bandpass filters along with novel positive and negative feedback techniques. In order to maintain the expected Q factor without unwanted oscillations in the positive feedback loop, the unity-coupler principle is proposed to stabilize the constituent amplifier. A prototype bandpass filter is monolithically integrated in a very small area of only 0.1 mm2 on a GaAs substrate. A sharp factor as high as 5.6/1-30 dB is achieved near the frequency range of 1 GHz. The other technique presented in this paper is to achieve the bandpass function without using any positive feedback. This is negative feedback consisting of feedback elements with the unique variable transfer function of b/(1as). A variable bandpass filter based on this design concept is also fabricated in a 1.21.3 mm2 area on a GaAs substrate. It has both a varactor and varistor integrated in the circuit, resulting in an independently controllable center frequency and Q factor. It is shown experimentally that the Q factor is controllable over a remarkable range of 20 to 400 and the center frequency is broader than 100 MHz at the 1 GHz band. By cascading two of the fabricated MMIC chips, a forth-order frequency response is successfully obtained along with a 35-40 dB forward gain and an in-band gain flatness of 0.35 dB.

This paper describes bandpass filters using linear tapered transmission line resonators (LTLR's). Bandpass filters are designed on the basis of the approximate description of LTLR's with cascaded multi-sections of uniform transmission lines whose widths are slightly different. By this design method, the fundamental characteristics of LTLR's and filter design parameters can be easily obtained using a general-purpose microwave circuit simulator. Trial LTLR bandpass filters showed excellent performance such as low insertion losses and the ability to control spurious responses, then their measured responses indicated close correspondence with the design results.

We applied a filip-chip-bonding technique to GHz-band SAW filters. The SAW filters mounted by the stud-bump-bonding (SBB) technique which is a kind of flip-chip-bonding technique showed almost the same frequency characteristics as those mounted by the conventional wire-bonding technique at 1.5 GHz. The SAW filter configuration, fabrication process using the SBB, and its electrical characteristics are described and discussed. The SBB technique has a lot of potential to reduce the size and weight even above GHz frequencies.

A wideband high power amplifier design using a novel band-pass filter with FET's parasitic reactances has been developed. The feature of this design is in that it can provide wide bandwidth and high gain of high power amplifiers. Furthermore, the lower cutoff frequency and bandwidth can be varied independently. With the use of this design, a Ku-band two-stage high power amplifier having a bandwidth of 18% has achieved a linear gain of 9.751.75 dB, a saturated output power of greater than 37 dBm, and a power-added efficiency of greater than 10.4%.

This letter presents a technique to cancel the parasitic effects of operational amplifier (op amp) in active filter design. To minimize the effects, an op amp model considering the parasitics (i.e. both parasitic poles and zeros) is utilized. It is shown that undesirable factors in the transfer function due to the parasitics can be canceled well by predistorting the passive element values of the circuit. As an example, an active-R highpass filter is evaluated both theoretically and numerically. In this way, the proposed technique can be effectively incorporated into the design of active filters.

This paper addresses onboard processing architecture employing direct regeneration. The advantage of direct regeneration is its hardware simplicity, even though the bit error rate performance is slightly inferior to that of demodulation-remodulation scheme with coherent detection. The channel filtering schemes as well as achievable capacities are examined by computer simulation. It is found that the system with direct regeneration has advantage in channel capacity and transmit earth station e.i.r.p. for small earth stations. A possible configuration of direct regeneration onboard in future satellite systems is proposed.

We propose a design method for a two-channel perfect reconstruction FIR filter banks employing linear-phase filters. This type of filter bank is especially important in splitting image signals into frequency bands for subband image cording. Because in such an application, it is necessary to use the combination of linear-phase filters and symmetric image signal, namely linear phase signal to avoid the increase in image size caused by filtering. In this paper, first we summarize the design conditions for two-channel filter banks. Next, we show that the design problem is reduced to a very simple linear equation, by using a half-band filter as a lowpass filter. Also the proposed method is available to lead filters with fewer complexity, which enable us to use simple arithmetic operations. For subband coding, the property is important because it reduces hardware complexity.

This paper studies a method for a parallel implementation of digital half toning technique, which converts continuous tone images into monotone one without losing fidelity of images. A new modified algorithm for half toning is proposed, which is able to be implemented on a rectangular or one dimensional parallel multi-processor array as a part of extensions of space partitioning image processings. The purpose of this paper is primarily to apply space partitioning local image processing technique to nonlinear recursive algorithms. The target is to achieve a fast half toning with high quality. For that propose, local directional error diffusion techniques will be introduced, which enable original recursive error diffusion half toning to be converted into a local processing algorithm without losing its original advantages of producing high quality images. The characteristics of proposed methods will be analyzed and the advantages of our algorithm of high speed processing and high quality will be demonstrated by showing the results of simulations for typical examples.

The evolution of Multiple-Valued Logic (MVL) circuits has been inexorably tied to the rapid technological changes induced by evolving needs and emerging developments in computing methodologies. Unfortunately for MVL, the numbers of designers of technologies and circuits whose lives are dedicated to the improvement of binary techniques, are large and overwhelming. Correspondingly, technological developments in MVL typically await the appearance of a problem or technique in the larger binary world to motivate and/or make possible some new advance. Such opportunities are inevitably quite transient since each such problem is simultaneously attacked by many others of a more conventional bent, and, as well, each technological change begets yet another, quickly. It is in the sensing of this reality that the present paper is written. Correspondingly, its thrust is two-fold: One target is the possibility of encouraging a leap ahead through modest technological projection. The other is the possibility of identifying application areas that already exist in this unbalanced competition, but which are specially suited to multiple-valued solutions. For example, it has been clear for decades that one such area is that of arithmetic. Correspondingly, we in MVL must strive quickly to concentrate our efforts on applications that exploit such demonstrable strengths. Some such applications are includes here; others are visible historically, many probably remain to be found: Search on!