MultiWave data compression algorithm is based on the multiresolution wavelet techniqu for decomposing Electrocardiogram (ECG) signals into their coarse and successively more detailed components. At each successive resolution, or scale, the data are convolved with appropriate filters and then the alternate samples are discarded. This procedure results in a data compression rate that increased on a dyadic scale with successive wavelet resolutions. ECG signals recorded from patients with normal sinus rhythm, supraventricular tachycardia, and ventriular tachycardia are analyzed. The data compression rates and the percentage distortion levels at each resolution are obtained. The performance of the MultiWave data compression algorithm is shown to be superior to another algorithm (the Turning Point algorithm) that also carries out data reduction on a dyadic scale.

This paper describes an error-correcting parser (ec-parser) for context-free languages that is an extension of the Leiss's parser. Since the ec-parser uses precomputed informations and a pruning technique by lookahead, the ec-parser is always faster than the Lyon's parser. Several examples are shown.

This paper presents an adaptive rate error control scheme for digital communication over time-varying channels. The cyclic code with majority-logic decoding is used in a cascaded way as an inner code to create a simple and powerful hybrid-ARQ error control scheme. Inner code is used only for error correction and the outer code is used for both error correction and error detection. When an error is detected, retransmission is required. The unsuccessful packets are not discarded as with conventional schemes, but are combined with their retransmitted copies. Approximations for the throughput efficiency and the undetectable error probability are given. A high reliability coupled with a simple high-speed implementation makes it suitable for high data rate error control over both stationary and nonstationary channels. Adaptive error control scheme becomes the best solution for time-varying channels when the optimum code is selected according to the actual channel conditions to enhance the system performance. The main feature of this system is that the basic structure of the encoder and decoder need not be modified while the error-correction capability of the code increases. Results of a comparative analysis show that the proposed scheme outperforms other similar ARQ protocols.

In order to develop silicon ICs operating up to above 450, Integrated Injection Logic (IIL) was chosen. A new structure for IIL was designed through experimental and theoretical studies of pn junctions, transistors, and IIL at high temperatures. A 5-µm design rule was used. The new IIL was fabricated by a specially developed combined process of ion implantation and low temperature epitaxy. The IIL was fully operational from room temperature to 454, and the output amplitude of a nine-stage ring oscillator was about 30 mV at 454. The minimum delay time of the IIL was 22 nsec at 454. The minimum power-delay product was 11 pJ and was one-third of that for IILs fabricated by 10-µm rule at 50.

This paper describes and analyzed several indices in assessing algorithms of data compression of electrocardiograms, such as the cross correlation (CC), the percent root mean square difference (PRD), and a new measure of standardized root mean square difference (SRD). Although these indices are helpful to objectively evaluate the algorithms, the visual examination of the reconstructed waveform is indispensable to decide the optimal compression ratio. This paper presents the clinical significance of selected waveforms which are prone to be distorted or neglected in the restored waveforms but are crucial for cardiologists to diagnose the patient. A database of electrocardiograms is also proposed for the comparative evaluation of compression algorithms.

This paper presents a high-density multiple-valued content-addressable memory (MVCAM) based on a floating-gate MOS device. In the proposed CAM, a basic operation performed in each cell is a threshold function that is a kind of inverter whose threshold value is programmable. Various multiple-valued operations for data retrieval can be easily performed using threshold functions. Moreover, each cell circuit in the MVCAM can be implemented using only a single floating-gate MOS transistor. As a result, the cell area of the four-valued CAM are reduced to 37% in comparison with that of the conventional dynamic CAM cell.

An ECL-CMOS SRAM technology is proposed which features a combination of ECL word drivers, ECL write circuits and low-voltage CMOS cells. This technology assures both ultra-high-speed and high-density. In the ECL-CMOS SRAM,various kinds of noise generated during the write cycle seriously affect the memory performance, because it has much faster access than conventional SRAMs. To overcome this problem, we propose three noise reduction techniques; a noise reduction clamp circuit, an emitter follower with damping capacitor and a twisted bit line structure with "normally on" equalizer. These techniques allow fast accese and cycle times. To evaluate these techniques, a 64-kb SRAM chip was fabricated using 0.5-µm BiCMOS technology. This SRAM has a short cycle time of 2 ns and a very fast access time of 1.5 ns. Evaluation proves the usefulness of these techniques.

This paper proposes a new scheme for concatenative speech synthesis to improve the speech segment selection procedure. The proposed scheme selects a segment sequence for concatenation by minimizing acoustic distortions between the selected segment and the desired spectrum for the target without the use of heuristics. Four types of distortion, a) the spectral prototypicality of a segment, b) the spectral difference between the source and target contexts, c) the degradation resulting from concatenation of phonemes, and d) the acoustic discontinuity between the concatenated segments, are formulated as acoustic quantities, and used as measures for minimization. A search method for selecting segments from a large speech database is also descrided. In this method, a three-step optimization using dynamic programming is used to minimize the four types of distortion. A perceptual test shows that this proposed segment selection method with minimum distortion criteria produces high quality synthesized speech, and that contextual spectral difference and acoustic discontinuity at the segment boundary are important measures for improving the quality.

A text-to-speech conversion system for Japanese has been developed for the purpose of producing high-quality speech output. This system consists of four processing stages: 1) linguistic processing, 2) phonological processing, 3) control parameter generation, and 4) speech waveform generation. Although the processing at the first stage is restricted to the texts on general weather conditions, the other three stages can also cope with texts of news and narrations on other topics. Since the prosodic features of speech are largely related to the linguistic information, such as word accent, syntactic structure and discourse structure, linguistic processing of a wider range than ever, at least a sentence, is indispensable to obtain good quality speech with respect to the prosody. From this point of view, input text was restricted to the weather forecast sentences and a method for linguistic processing was developed to conduct morpheme, syntactic and semantic analyses simultaneously. A quantitative model for generating fundamental frequency contours was adopted to make a good reflection of the linguistic information on the prosody of synthetic speech. A set of prosodic rules was constructed to generate prosodic symbols representing prosodic structures of the text from the linguistic information obtained at the first stage. A new speech synthesizer based on the terminal analog method was also developed to improve the segmental quality of synthetic speech. It consists of four paths of cascade connection of pole/zero filters and three waveform generators. The four paths are respectively used for the synthesis of vowels and vowel-like sounds, nasal murmur and buzz bar, friction, and plosion, while the three generators produce voicing source waveform approximated by polynomials, white Gaussian noise source for fricatives and impulse source for plosives. The validity of the approach above has been confirmed by the listening tests using speech synthesized by the developed system. Improvements both in the quality of prosodic features and in the quality of segmental features were realized for the synthetic speech.

The concept of functional memory was proposed over nearly four decades ago. However, the actually usable products have not appeared until the 1980s instead of the long history of development. Functional memory is classified into three categories; there are a general functional memory, a processing element array with small size memory and a special purpose memory. Today a majority of functional memory is an associative memory or a content addressable memory (CAM) and a special purpose memory based on CAM. Due to advances in fablication capability,the capacity of CAM LSI has increased over 100 K bits. A general purpose CAM was developed based on SRAM cell and DRAM cell, respectively. The typical CAM LSI of both types, 20 K bits SRAM based CAM and 288 K bits DRAM based CAM, are introduced. DRAM based CAM is attractive for the large capacity. A parallel processor architecture based on CAM cell is proposed which is called a Functional Memory Type Parallel Processor (FMPP). The basic feature is a dual character of a higher performance CAM and a tiny processor array. It can perform a highly parallel operation to the stored data.

An approximate but sufficiently accurate high frequency solution is developed in this paper for analyzing the problem of electromagnetic plane wave scattering by an open-ended, perfectly-conducting, semi-infinite parallel-plate waveguide with a thin layer of lossy or absorbing material on its inner wall, and with a planar termination inside. The high frequency solution combines uniform geometrical theory of diffraction (UTD) and aperture integration (AI) methods. The present method has several advantages in comparison with other methods.

This paper describes a novel method to calculate the fields scattered by a polyhedron structure for an incident plane wave. In this method, the fields diffracted by an edge are calculated using the equivalent edge currents which are separated into components dependent on each of the two surfaces which form the edge. The separated equivalent edge currents are based on the Geometrical Theory of Diffraction (GTD). Using this Separated Equivalent Edge Current Method (SEECM) , fields scattered by a polyhedron structure can be calculated without special treatment of the singularity in the diffraction coefficient. This method can be also applied successfully to structures with convex surfaces by modeling them as polyhedron structures.

A soft-decision decoding algorithm for binary linear block codes is proposed. This algorithm seeks to minimize the block error probability. With careful examinations of the first hard-decision decoded results, the candidate codewords are efficiently searched for. Thus, we can reduce the decoding complexity (the number of hard-decision decodings) and lower the block error probability. Computer simulation results are presented for the (23, 12) Golay code. They show that the decoding complexity is considerably reduced and the block error probability is close to that of the maximum likelihood decoder.

A new system for speech synthesis by concatenating waveforms selected from a dictionary is described. The dictionary is constructed from a two-hour speech that includes isolated words and sentences uttered by one male speaker, and contains over 45,000 entries which are identified by their average pitch, dynamic pitch parameter which represents micro pitch structure in a segment, duration and average amplitude. Phoneme duration is set according to phoneme environment, and phoneme power is controlled, by both pitch frequency and phoneme environment. Tests show the average errors in vowel duration and consonant duration are 28.8 ms and 16.8 ms respectively, and the vowel power average error is 2.9 dB. The pitch frequency patterns are calculated according to a conventional model in which the accent component is abbed to a gross phrase component. Set a phoneme string and prosody information, the optimum waveforms are selected from the dictionary by matching their attributes with the given phonetic and prosodic information. A waveform selection function, which has two terms corresponding to prosody and phonological coincidence between rule-set values and waveform values from the dictionary, is proposed. The weight coefficients used in the selection function are determined through subjective hearing tests. The selected waveform segments are then modified in waveform domain to further adjust for the desired prosody. A pitch frequency modification method based on pitch synchronous overlap-add technique is introduced into the system. Lastly, the waveforms are interpolated between voiced waveforms to avoid abrupt changes in voice spectrum and waveform shape. An absolute evaluation test of five grades is performed to the synthesized voice and the mean of the score is 3.1, which is over "good," and while the original speaker quality is retained.

We propose a smart design methodology for advanced ULSI memories to reduce the turn around time(TAT) for circuit revisions with no area penalty. This methodology was executed by distributing extra gate-arrays, which were composed of the n-channel and p-channel transistors, under the power line and the signal line. This method was applied to the development of a 16 Mb DRAM with double metal wiring. The design TAT can be reduced to 1/8 using 1500 gates. This design methodology has been confirmed to be very effective.

A new automated main distributing frame (AMDF) system is developed that reduces operating costs in metallic-cable main distributing frames (MDFs) used for communication networks. In this AMDF system, a robot inserts connecting-pins into the crosspoint holes of matrix-boards. This process allows jumpering to be completed within three minutes and the route-setting for line testing within one minute. The AMDF system provides approximately 2,100 office equipment cable-terminals. Parallel installation of several AMDF systems allows larger MDF systems to be constructed. This system reduces costs and achieves high reliability through three new technologies: high-density matrix-board, precision pin-handling, and a highly reliable system control. Test results for a prototype AMDF system confirm their effectiveness.

Prosodic features of the spoken Japanese play an important role in the transmission of linguistic information concerning the lexical word accent, the sentence structure and the discourse structure. In order to construct prosodic rules for synthesizing high-quality speech, therefore, prosodic features of speech should be quantitatively analyzed with respect to the linguistic information. With a special focus on the fundamental frequency contour, we first define four prosodic units for the spoken Japanese, viz., prosodic word, prosodic phrase, prosodic clause and prosodic sentence, based on a decomposition of the fundamental frequency contour using a functional model for the generation process. Syntactic units are also introduced which have rough correspondence to these prosodic units. The relationships between the linguistic information and the characteristics of the components of the fundamental frequency contour are then described on the basis of results obtained by the analysis of two sets of speech material. Analysis of weathercast and newscast sentences showed that prosodic boundaries given by the manner of continuation/termination of phrase components fall into three categories, and are primarily related to the syntactic boundaries. On the other hand, analysis of noun phrases with various combinations of word accent types, syntactic structures, and focal conditions, indicated that the magnitude and the shape of the accent components, which of course reflect the information concerning the lexical accent types of constituent words, are largely influenced by the focal structure. The results also indicated that there are cases where prosody fails to meet all the requirements presented by word accent, syntax and discourse.

When orignal signals are contaminated by both additive and signal-dependent noise components, the test statistics of locally optimum detector are obtained for detection of weak composite signals based on the generalized Neyman-Pearson lemma. In order to consider the non-additive noise as well as purely-additive noise, a generalized observation model is used in this paper. The locally optimum detector test statisics are derived for all different cases according to the relative strengths of the known signal, random signal, and signal-dependent noise components. Schematic diagrams of the structures of the locally optimum detector are also included. The finite sample-size performance characteristics of the locally optimum detector are compared with those of other common detectors.

The performance of ASIPs (Application Specific Integrated Processors) is heavily affected by the design of their instruction set architecture. In order to maximize the performance of ASIP, it is essential to design an architecture that has an optimum instruction set. This paper descibes a new method that automates the design of optimum instruction set of ASIP. This method solves the Instruction set implementation Method Selection Problem(IMSP). IMSP is to be solved in the instruction set architecture design. Frse, the IMSP is formalized as an integer programming problem, which is to maximize the perfomance of the CPU under the constraints of chip area and power consumption. Then, a branch-and-bound algorithm to solve IMSP is described. According to the experimental results, the proposed algorithm is quite effective and efficient in solving the IMSP. The presented method automates a complex part of the ASIP chip design and is also a good design tool that enables designer to predict the performance of their design before completion.

The discrete time analysis of logic circuits is usually more efficient than the continuous time analysis, but the preciseness of the discrete time analysis is not guaranteed. The paper shows a method to decide a unit time for a logic circuit under which the analysis result is the same as the result based on the continuous time. The delay time of an element is specified with an interval between the minimum and maximum delay times, and we assume an analysis method which enumerates all possible delay cases under the deisrete time. Our main theorem is as follows: refine the unit time by a factor of 1/2, and if the analysis result with a unit time u and that with a unit time u/2 are the same, then u is the expected unit time.