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.

Overlapping multi-pulse pulse position modulation (OMPPM) is a modulation scheme having higher capacity and cutoff rate than other conventional modulation schemes when both off-duration between pulses shorter than a laser pulsewidth and resolution better than a laser pulsewidth are realized [1],[2]. In Refs. [1],[2] erasure events of a few chips that can be decoded correctly is defined as an erasure event. This results in lower bounds on the performance of OMPPM in optical-direct-detection channel in quantum limited case. This paper analyzes more exact performance of OMPPM in optical direct-detection channel in quantum limited case when both off-duration between pulses shorter than a laser pulsewidth and resolution better than a laser pulsewidth are realized. First we derive the error probability of OMPPM with considering what chips are detected or erased. Then we derive the capacity and the cutoff rate of OMPPM using the error probability. It is shown that OMPPM outperforms on-off keying (OOK), pulse position modulation (PPM), multi-pulse PPM (MPPM), and overlapping PPM (OPPM) in terms of both capacity and cutoff rate for the same pulsewidth and the same duty cycle. Moreover, it is shown that OMPPM with fewer slots and more pulses per block has better cutoff rate performance when the average received power per slot is somewhat large.

A multi-channel high temperature superconducting interference device (high Tc SQUID) system with high magnetic field resolution has been developed. Step edge junctions were employed as weakly coupled Josephson junctions for the SQUID. These junctions worked well and their I-V curves fit the resistively shunted junction (RSJ) model. The SQUID design was investigated to improve magnetic field resolution. The size of the SQUID's center hole was investigated, and we found the optimized size of the hole to be about 25 µm. Meissner effect of superconductor was used in order to concentrate magnetic fluxes. A large washer SQUID and a flux concentrating plate was developed to concentrate magnetic flux to the SQUID center hole. The magnetic field resolution became 370 fT/Hz at 10 Hz and 220 fT/Hz at 10 kHz. This field resolution was enough to detect biomagnetic signals such as magnetocardiac signals. The SQUID was mounted on a special chip carrier and was sealed with epoxy resin for protection from humidity. We have designed and developed a 4-channel and a 16-channel high Tc SQUID system. We used them in a magnetically shielded room to measure magnetic signals of the human heart. We obtained clear multi-channel magnetocardiac signals, which showed clear so called QRS and T wave peaks. A clear isofield contour map of magnetocardiac signals was also obtained. These data indicated that high Tc SQUID is feasible for these biomagnetic applications.

Go-Back-N automatic repeat request (GBN ARQ) and Stop-and wait (SW) ARQ schemes are one of fundamental and widely used error control procedures for data communication and computer communication systems. The throughput and delay performances of these ARQ schemes have been analyzed for a random error channel, which could not applicable for a radio channel, for example. In this paper, considering the correlated, noisy channel, we derive the exact formula for the delay of a frame in GBN and SW ARQ schemes. First, the delay formula for the discrete time M[x]/G/1 queueing system with starter. Next, the virtual service time of a frame is found in terms of the decay factor of a two-state Markov chain. As a result, it is shown that the performance of the delay is improved with the larger decay factor.

Recently, a lot of research works have been carried out regarding intelligent communication. If the final information sink is assumed as a human being, a communication channel can be used more effectively when encoders/decoders work "intelligently" or take into account of the semantics of information to be sent. We have been studying error-controlling systems based on different importance of segmental information. The system divides the information input into segments to which individual importance can be assigned. The segments are individually encoded by appropriate error-correcting codes (ECCs) which correspond to their importance among codes with different error-correcting capabilities. For the information that difference of the importance is systematically aligned, conventional UEP (unequal error protection) codes can be applied, but we treat the case that alignment of the importance of the information source is not systematically aligned. Since the system uses multiple ECCs with different (n,k,d) parameters, information regarding what length of the next codeword is required for decoding. We propose error controlling schemes using mulriple ECCs; the first scheme and the second scheme use the obvious codelength identifying information. In the second scheme, information bits are sorted so that segments with the same importance can be encoded by an ECC with the same error-correcting capability. The third scheme is a main proposal in this paper and uses Variable Capability Coding scheme (VCC) which uses some ECCs having different error-correcting capabilities and codelengths. A sequence encoded by the VCC is separable into appropriate segments without obvious codelength identifying information when the channel error probability is low. Subsequently, we evaluate these schemes by coderate when (1) error correcting capability (2) codelength identifying capability are the same. One of the feature of VCC is the capability of resuming from propagative errors because errors beyond the codelength identifying capability occur and the proper beginning of the codeword is lost in the decoder. We also evaluate this capability as (3) resynchronizing capability.

Recently there has been considerable interest in coded modulation schemes that offer multiple levels of error protection. That is, constructions of (block or convolutional) modulation codes in which signal sequences associated with some message symbols are separated by a squared Euclidean distance that is larger than the minimum squared Euclidean distance (MSED) of the code. In this paper, the trellis structure of linear unequal-error-protection (LUEP) codes is analyzed. First, it is shown that LUEP codes have trellises that can be expressed as a direct product of trellises of subcodes or clouds. This particular trellis structure is a result of the cloud structure of LUEP codes in general. A direct consequence of this property of LUEP codes is that searching for trellises with parallel structure for a block modulation code may be useful not only in analyzing its structure and in simplifying its decoding, but also in determining its UEP capabilities. A basic 3-level 8-PSK block modulation code is analyzed under this new perspective, and shown to offer two levels of error protection. To illustrate the trellis structure of an LUEP code, we analyze a trellis diagram for an extended (64,24) BCH code, which is a two-level LUEP code. Furthermore, we introduce a family of LUEP codes based on the |||-construction, using Reed-Muller (RM) codes as component codes. LUEP codes in this family have the advantage of having a well known trellis structure. Their application in constructing LUEP-QPSK modulation codes is presented, and their error performance over an AWGN channel examined.

A new recursive method for obtaining the mean waiting time in a polling system with general service order and gated service discipline is proposed. The analytical approach used to obtain the mean waiting time is via an imbedded Markov chain and a new recursive method is used to obtain the moments of pseudocycle time which are parameters in the formula for the mean waiting time. This method is computationally tractable, so the analytical results can cover a wide range of applications. Simulations are also conducted to verify the validity of the analysis.

A uniquely decodable (UD) code pair (C, S) is considered for the two-user binary adder channel. For a class of linear codes C, the maximum independent set of the graph associated with C, which is the second code S, is evaluated. When the rate R1 of C is less than 0.5, there exist UD codes (C, S)'s such that the rate R2 of S exceeds the Khachatrian's and Guo's results in amount.

A stochastic system represented by an input-output model can be described by mainly two different types of state space representation. Corresponding to state space representations innovation models are examined. The relationship between both representations is made clear systematically. An easy transformation between them is presented. Zeros of innovation models are the same as those of an ARMA model which is stochastically equivalent to innovation models, and related to stable eigenvalues of generalized eigenvalue problem of matrix Riccati equation.

We present an 8-dimensional trellis-coded 8-PSK with a symbol transition constraint that is similar to that of π/4-shift quadrature phase shift keying (QPSK). This scheme can achieve a coding gain of 1.6 to 2.4 dB at the same rate of π/4-shift QPSK on Gaussian channel, and it has also an immunity against the integer multiples of 90 phase ambiguities. In order to label the constellation of the proposed scheme, a constellation partitioning algorithm is presented. This algorithm, on the basis of set partitioning, can be used to label the signal constellation with no coset structure.

This paper proposes and investigates an intelligent error-controlling scheme according to different importance of segmental information. In particular, the scheme is designed for facial images encoded by model-based coding that is a kind of intelligent compression coding. Intelligent communication systems regard the contents of information to be transmitted with extremely high compression and reliability. After highly efficient information compression by model-beaed coding, errors in the compressed information lead to severe semantic errors. The proposed scheme reduces semantic errors of information for the receiver. In this paper, we consider Action Unit (AU) as a segment of model-based coded facial image of human being and define the importance for each AU. According to the importance, an AU is encoded by an appropriated code among codes with different error-correcting capabilities. For encoding with different error controlling codes, we use three kinds of constructions to obtain unequal error protection (UEP) codes in this paper. One of them is the direct sum construction and the others are the proposed constructions which are based on joint and double coding. These UEP codes can have higher coderate than other UEP codes when minimum Hamming distance is small. By using these UEP codes, the proposed intelligent error-controlling scheme can protect information in segment in order to reduce semantic errors over a conventional error-controlling scheme in which information is uniformly protected by an error-correcting code.

Multilevel RLL (Runlength Limited) sequences are analyzed. Their noiseless capacity and lower bounds on the channel capacity in the presence of additive white Gaussian noise are given. Moreover, the analytical power spectra formulae for those sequences which generalize the previously derived one for binary sequences are newly derived. We conclude from the analysis of the power spectra that multilevel RLL sequences are attractive from the point of view that they increase information rate while keeping low DC-content and self-clocking capability of binary RLL sequences.

We propose a method which can construct LOTOS specifications of communication systems from temporal properties described in Extended branching time temporal logic (EBTL) in this paper. Description of LOTOS, one of Formal Description Techniques, is based on behaviors of systems so that LOTOS permits strict expression. However, it is difficult to express temporal properties explicitly. On the other hand, Temporal logic is based on properties of systems. Thus temporal logic permits direct expression of temporal properties which can be understood intuitively. Accordingly, temporal logic is more useful than FDTs on the first step of systems specification. This method is effective in this point of view. Specifications obtained using this method can have a structured style. When temporal properties are regarded as constraints about temporal order among actions of systems, those specification can have a constraint oriented style. This method is based on sequential composition of Labelled Transition Systems (LTSs) which are semantics of LOTOS. EBTL is also defined on LTSs. In general, many LTSs satisfy the same temporal property. To obtain such the minimal LTS, the standard form of LTSs corresponding to EBTL operators are defined. Those standard LTSs are mainly tailored to the field of communication protocol. We also show conditions that original temporal properties are preserved in case of sequential composition of standard LTSs. In addition, applying this method to the Abracadabra Protocol, we show that this method can construct specifications of concrete protocols effectively.

An annoying problem encountered in automatic seal imprint verification is that for seal imprints may have a lot of variations, even if they are all produced from a single seal. This paper proposes a new automatic seal imprint verification system which adds an imprint quality assessment function to our previous system in order to solve this problem, and also examines the verification performance of this system experimentally. This system consists of an imprint quality assessment process and a verification process. In the imprint quality assessment process, an examined imprint is first divided into partial regions. Each partial region is classified into one of three quality classes (good quality region, poor quality region, and background) on the basis of characteristics of its gray level histogram. In the verification process, only good quality partial regions of an examined imprint are verified with registered one. Finally, the examined imprint is classified as one of two types: a genuine and a forgery. However, as a result of quality assessment, if the partial regions classified as poor quality are too many, the examined imprint is classified as ambiguous" without verification processing. A major advantage of this verification system is that this system can verify seal imprints of various qualities efficiently and accurately. Computer experiments with real seal imprints were performed by using this system, previous system (without image quality assessment function) and document examiners of a bank. The results of these experiments show that this system is superior in the verification performance to our previous system, and has a similar verification performance to that of document examiners (i.e., the experimental results show the effectiveness of adding the image quality assessment function to a seal imprint verification system).

Learning correctly from queries" is a formal learning model proposed by Angluin. In this model, for a class Γ of language representations, a learner asks queries to a teacher of an unknown language Lq which can be represented by some GqΓ, and eventually outputs a language representation GΓ which represents Lq and halts. An algorithm (leaner) A is said to learn a class of languages represented by Γ in the weak definition if the time complexity of A is some polynomial of n and m, where n is the minimum size of the lagunage representations in Γ which represent Lq, and m is the maximum length of the counterexamples returned in an execution. On the other band, A is said to learn represented by Γ in the strong definition if at any point τ of the execution, the time consumed up to τ is some polynomial of n and m, where n is the same as above, and m is the maximum length of the counterexamples returned up to τ. In this paper, adequacy of the model is examined, and it is shown that both in the weak and strong definitions, there exist learners which extract a long counterexample, and identify Lq by using equivalence queries exhaustively. For example, there exists a learner which learns the class CFL of context-free languages represented by the class CFG of context-free grammars in the weak definition using only equivalence queries. Next, two restrictions concerning with learnability criteria are introduced. Proper termination condition is that when a teacher replies with yes" to an equivalence query, then the learner must halt immediately. The other condition, called LBC-condition, is that in the weak/strong definition, the time complexity must be some polynomial of n and log m. In this paper, it is shown that under these conditions, there still exist learners which execute exhaustive search. For instance, there exists a learner which learns CFL represented by CFG in the weak definition using membership queries and equivalence queries under the proper termination condition, and there also exists a learner that learns CFL represented by CFG in the strong definition using subset queries and superset queries under LBC-condition. These results suggest that the weak definition is not an adequate learning model even if the proper termination condition is assumed. Also, the model becomes inadequate in the strong definition if some combination of queries, such as subset queries and superset queries, is used instead of equivalence queries. Many classes of languages become learnable by our extracting long counterexample" technique. However, it is still open whether or not CFL represented by CFG is learnable in the strong definition from membership queries and equivalence queries, although the answer is known to be negative if at least one of (1) quadratic residues modulo a composite, (2) inverting RSA encryption, or (3) factoring Blum integers, is intractable.

This paper reports on the properties of thin-film thermal switches that are monolithically fabricated on high-Tc superconductive filter. Operating at a wide temperature range of 50-77 K, it was found that the switch could control the center frequency by -10 MHz with an increased insertion loss of less than 0.7 dB. In an on-off switching operation of filter characteristics using thin-film switches, power consumption was approximately 20 mW at 77 K, and the signal decay time as a switching speed was 30 ms at 76 K with a switch current of 70 mA. The decay time decreased exponentially as the switch current or the temperature setting increased.

In this paper, the Pisarenko and the Constrained Yule-Walker (CYW) estimators of a tone frequency are first newly derived from the viewpoint of using directly the autocorrelation coefficients. Then, simulation of these two estimators is carried out in some detail. The simulation results show that compared with the Pisarenko estimator the CYW estimator, which has not been adequately studied, works poorly for low and moderate Signal-to-Noise Ratio (SNR) values. However, in case of high SNR value, it yields very small bias and comparable estimation variance, and thus produces more accurate tone frequency estimates.

The magnitude of low frequency noise is studied in a Nb-(nanoconstrictions)-NbN system with adjustable current-voltage characteristics. We find that the magnitude of low frequency noise decreases sharply with increasing the subgap conductivity of the device. We suggest a qualitative explanation of this observation in terms of gradual build up of the nanoconstriction region by field assisted growth. The decrease of low frequency noise is related to the "cleanliness" of the system as measured by the amount of Andreev reflection-related conductivity. The scaling of the magnitude of low frequency noise with device resistance is also discussed.

This paper presents new go-back-N ARQ protocols for point-to-multipoint communications over broadcast channels such as satellite or broadcast radio channels. In the conventional go-back-N ARQ protocols for multidestination communications, usually only error detection codes are used for error detection and m copies of a frame are transmitted at a time. In one of our protocols, a bit-by-bit majority-voting decoder based on all of the m copies of a frame is used to recover the transmitted frame. In another protocol, a hybrid-ARQ protocol, which is an error detection code concatenated with a rate repetition convolutional code with the Viterbi decoding, is used. In these protocols, a dynamic programming technique is used to select the optimal number of copies of a frame to be transmitted at a time. The optimal number is determined by round trip propagation delay of the channel, the error probability, and the number of receivers that have not yet received the message. Analytic expressions are derived for the throughput efficiency of the proposed protocols. The proposed point-to-multipoint protocols provide satisfactory throughput efficiency and perform considerably better than the conventional protocols under high error rate conditions, especially in environments with a large number of receivers and large link round trips. In this paper we analyze the performances of the proposed protocols upon the random error channel conditions.

This paper gives field experimental results on 16-ary quadrature amplitude modulation/time division multiple access (16QAM/TDMA) and trellis coded 16QAM/TDMA systems for land mobile communications in order to evaluate its capability of achieving large capacity and high quality data transmission. Pilot symbol aided space diversity and symbol timing synchronization based on maximum likelihood (ML) estimation are applied to both 16QAM/TDMA and trellis coded 16QAM/TDMA to improve transmission quality. For the trellis coded 16QAM/TDMA, trellis coding with Viterbi decoding and 2-frame symbol interleaving are further employed. The field experiments were conducted in the Tokyo metropolitan area of Japan. The results show that 16QAM/TDMA and trellis coded 16QAM/TDMA are practical modulation/access schemes for land mobile communication systems.