A flat stick-shaped whip antenna was developed for Japanese commercial portable telephones. It provides a high gain even though it is short and retractable. It is an open-sleeve type antenna. i.e., the inductance-loaded dipoleantenna element parallels a twin-lead type feeder. It suppresses the currents on the radio housing even at short antenna lengths. Moreover, it is easy to achieve two resonant characteristies and able to construct retractable type. A relatively high gain is gotten even when the antenna is in a retracted state. This antenna has a suitable configuration for commercial portable telephones. This paper first calculates, the current suppression of the housing on a principal model of this antenna, i.e., without a twin-lead feeder. The second analysis determines the effects of the twin-lead feeder and the dielectric coat on the antenna. Next,the two resonant technique and the configuration for the retractable-type antenna describes. In addition, the return loss and radiation pattern for the trial model were measured. The return loss shows that the two resonant characteristics and the bandwidths of the high and low resonant frequencies are 2.2% and 1.5%(VSWR2), respectively. For when the antenna is extended, radiation patterns are nearly the same as for the case of the 1/2 wavelength dipole antenna, and the antenna efficiencies are -1.6 dB at 820 MHz and -1.1 dB at 950 MHz. Other hand, for the retracted state, they are destroyed by the housing currents, but the efficiency is relatively high of -1.8 dB at 950 MHz. In these experiments, it is clear that the antenna achieves high performances.

Fundamental microwave key devices used in achieving compact mobile/portable telephones (raidio units) are discussed. The historical development flow of the systems and radio units are introduced, with respect to the 800-/900-MHz and 1.5-GHz Japanese cellular radio systems. The design concept of the developed radio units is briefly described. Tehnical requirements for RF circuits are reviewed and the developed key devices are practically applied to the circuits. Key factors in the requirements are also shown. Finally. future trends fro the key devices are surveyed from the stand point of achieving a smaller and more light weight pocket radio unit.

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.

Based on the test method proposed by Sub-Committee G of the International Special Committee on Radio Interference, most telephone receivers in Japan have insufficient immunity to acoustic noise caused by radio-frequency fields. This is because the modulation depth of the RF signal used is too high to accurately simulate the audio-frequency components of TV video signals. Reducing the modulation depth from 80% to 5% produces a more realistic simulation.

Estimating the macroscopic demand for telephones is essential to long-term planning construction of telecommunication networks facilities. Although there are several useful forecast equations, they need some types and/or vast amounts of data that are sometimes unavailable, especially in developing countries. This paper presents a sophisticated telephone demand estimation technique that is based on the demands of residential and business users. It uses several parameters to estimate the increase in telephone demand. A simplified equation is also presented that is a function of only one parameter: normalized gross domestic product (GDP) per capita. This simplified equation is shown to be useful by using data for more than ten countries.

In this paper approaches to language identification based on the sequential information of phonemes are described. These approaches assume that each language can be identified from its own phoneme structure, or phonotactics. To extract this phoneme structure, we use phoneme classifiers and grammars for each language. The phoneme classifier for each language is implemented as a multi-layer perceptron trained on quasi-phonetic hand-labeled transcriptions. After training the phoneme classifiers, the grammars for each language are calculated as a set of transition probabilities for each phoneme pair. Because of the interest in automatic language identification for worldwide voice communication, we decided to use telephone speech for this study. The data for this study were drawn from the OGI (Oregon Graduate Institute)-TS (telephone speech) corpus, a standard corpus for this type of research. To investigate the basic issues of this approach, two languages, Japanese and English, were selected. The language classification algorithms are based on Viterbi search constrained by a bigram grammar and by minimum and maximum durations. Using a phoneme classifier trained only on English phonemes, we achieved 81.1% accuracy. We achieved 79.3% accuracy using a phoneme classifier trained on Japanese phonemes. Using both the English and the Japanese phoneme classifiers together, we obtained our best result: 83.3%. Our results were comparable to those obtained by other methods such as that based on the hidden Markov model.

We carried out a one year field trial of a voice-activated automatic telephone exchange service at KDD Laboratories which has about 200 branch phones. This system has DSP-based continuous speech recognition hardware which can process incoming calls in real time using a vocabulary of 300 words. The recognition accuracy was found to be 92.5% for speech read from a written text under laboratory conditions independent of the speaker. In this paper, we describe the performance of the system obtained as a result of the field trial. Apart from recognition accuracy, there was about 20% error due to out-of-vocabulary input and incorrect detection of speech endpoints which had not been allowed for in the laboratory experiments. Also, we found that the recognition accuracy for actual speech was about 18% lower than for speech read from text even if there were no out-of-vocabulary words. In this paper, we examine error variations for individual data in order to try and pinpoint the cause of incorrect recognition. It was found from experiments on the collected data that the pause model used, filled pause grammar and differences of channel frequency response seriously affected recognition accuracy. With the help of simple techniques to overcome these problems, we finally obtained a recognition accuracy of 88.7% for real data.

A novel low-power dissipation and high-speed converter-control-IC has been developed for the transmitting amplifier in digital portable telephones. The IC consists mainly of CMOS devices to reduce the bias current. To improve circuit speed, bipolar transistors are used in the output stage of the operational amplifier and in the current sources of the oscillator because they have a larger current capability and smaller parasitic capacitance than CMOS devices. The IC has one-fifth the bias current of a conventional control circuit consisting of discrete devices, and it can operate up to a switching frequency of 3MHz. The small bias current increases converter efficiency, and the high switching frequency reduces converter size. Using this IC, converter loss is 17% less than that with a conventional control circuit.

This paper presents the M-LCELP (Multi-mode Learned Code Excited LPC) speech coder, which has been developed for the next generation half-rate digital cellular telephone systems. M-LCELP develops the following techniques to achieve high-quality synthetic speech at 4kb/s with practically reasonable computation and memory requirements: (1) Multi-mode and multi-codebook coding to improve coding efficiency, (2) Pitch lag differential coding with pitch tracking to reduce lag transmission rate, (3) A two-stage joint design regular-pulse codebook with common phase structure in voiced frames, to drastically reduce computation and memory requirements, (4) An efficient vector quantization for LSP parameters, (5) An adaptive MA type comb filter to suppress excitation signal inter-harmonic noise. The MOS subjective test results demonstrate that 4.075kb/s M-LCELP synthetic speech quality is mostly equivalent to that for a North American full-rate standard VSELP coder. M-LCELP codec requires 18 MOPS computation amount. The codec has been implemented using 2 floating-point dsp chips.

The relation between the radiation pattern and the dimension of the conducting box for a portable telephone is illustrated both theoretically and experimentally. The Galerkin-moment method using the Fourier series expansion for the surface current of the conducting box, which has a great advantage of having a high accuracy, is employed to obtain the radiation pattern. As an example of antennas, a quarter-wavelength monopole antenna having a sinusoidal current distribution is used. As a result, it is pointed out that the radiation pattern of a monopole antenna mounted on the box tends to tilt in a lower direction both in theory and in experiment as well. The relation between the radiation pattern and the location of the monopole antenna is also described. An asymmetrical, or distorted pattern is observed when the monopole antenna moves away from the center of the top plane.

This paper discusses the fully three-dimensional finite difference time domain (FDTD) method to analyze a monopole antenna mounted on a rectangular conducting box covered with a layer of dielectric. The effects of the conductivity and the permittivity of the dielectric layer are investigated. It is shown that all calculation results agree very well with the measured data.

This paper analyses the amplification characteristics of a two-loop controlled switching power amplifier for a digital portable telephone and presents the amplifier which has a flat gain and small phase delay from dc to 100kHz. This amplifier is a modification of a switching regulator and it uses two-loop control to achieve a wideband amplification characteristic. Optimum amplification characteristics, however, can't be designed by using the conventional method for designing a switching regulator because a flat gain and small phase delay in an amplification characteristic has not been considered for most switching regulators. This paper analyses in detail the small-signal transfer functions of the switching power amplifier and shows the behaviour of zero and poles. It also shows the boundary condition of large-signal operation. A new design procedure of a switching power amplifier is presented, and the analytical results are verified by experiments.

This letter proposes an LSP quantizing method which uses interframe correlation of the parameters. The quantized parameters are represented as a moving average of code vectors. Using this method, LSP parameters are quantized efficiently and the degradation of decoded parameters caused by bit errors affects only a few following frames.

The states-of-the-art in visual communication in Japan are described. First the status of networks, which is a basis for offering visual communication service, is outlined. Visual communication service being developed on the basis of ISDN is described. The future service can be represented by NTT's service vision VI&P. Visual communication technologies and services being studied are surveyed.