Most digital signal processing (DSP) algorithms for multimedia and communication applications require multiplication and addition operations. Especially matrix-matrix or matrix-vector the multiplications frequently used in DSP implementations needs inner product arithmetic which takes the most processing time. Also multiplications for the DSP algorithms for software defined radio (SDR) applications require different input bitwidths. Therefore, the multiplications for inner product need to be sufficiently flexible in terms of bitwidths to utilize hardware resources efficiently. This paper proposes a novel reconfigurable inner product architecture based on a pipelined adder array, which offers increased flexibility in bitwidths of input arrays. The proposed architecture consists of sixteen 44 multipliers and a pipelined adder array and can compute the inner product of input arrays with any combination of multiples of 4 bitwidths such as 44, 48, 412, ... 1616. Experimental results show that the proposed architecture has latency of maximum 9 clock cycles and throughput of 1 clock cycle for inner product of various bitwidths of input arrays. When TSMC 0.18 µm libraries are used, the chip area and critical path of the proposed architecture are 186,411 gates and 2.79 ns, respectively. The proposed architecture can be applied to a reconfigurable arithmetic engine for real-time SDR system designs.

We study a modification method for constructing low-density parity-check (LDPC) codes for solid burst erasures. Our proposed modification method is based on a column permutation technique for a parity-check matrix of the original LDPC codes. It can change the burst erasure correction capabilities without degradation in the performance over random erasure channels. We show by simulation results that the performance of codes permuted by our method are better than that of the original codes, especially with two or more solid burst erasures.

Based on the Khatri-Rao matrix product, we propose a novel unitary space-time modulation design called KR-USTM in this paper. Different from existing USTM schemes, such as the systematic approach and space-time frequency-shift keying (ST-FSK), KR-USTM does not require any computer search and can be applied to any number of transmit antennas. Moreover, the special structure of KR-USTM also makes it a high-rate scheme and achieve full antenna diversity as well as lower decoding complexity. Simulation results show that the proposed KR-USTM constellation achieves error performance comparable to existing USTM designs at low rates, while it outperforms them at high rates.

The hand posture estimation system by searching a similar image from a vast database, such as our previous research, may cause the increase of processing time, and prevent realtime controlling of a robot. In this study, the authors proposed a new estimation method of human hand posture by rearranging a large-scale database with the Self-Organizing Map including self-reproduction and self-annihilation, which enables two-step searches of similar image with short period of processing time, within small errors, and without deviation of search time. The experimental results showed that our system exhibited good performance with high accuracy within processing time above 50 fps for each image input with a 2.8 GHz CPU PC.

In this paper, a chromatic adaptation model (CAM) based on the dependence on LMS cone responses of the human visual system (HVS) is proposed for TV and PC monitors under a variety of viewing conditions. We derived the proposed CAM based on Breneman's corresponding color data. The results of the experiments were carried out to assess the proposed model performance in terms of color fidelity by comparing complex images on a LCD monitor. We confirmed that the proposed model performed better to predict corresponding colors under various viewing conditions. Therefore, the reproduced colors, which are viewed in real surround viewing conditions, are perceived the same as original object colors, when the proposed CAM was applied to color display devices such as CRT, LCD, and PDP.

We propose transformation of a parity-check matrix of any low-density parity-check code. A code with transformed parity-check matrix is an equivalent of a code with the original parity-check matrix. For the binary erasure channel, performance of a message-passing algorithm with a transformed parity-check matrix is better than that with the original matrix.

This is commonly thought that CASE tools reduce programming efforts and increase development productivity. However, no paper has provide quantitative data supporting the matter. This paper discusses productivity improvement through the use of an integrated CASE tool system named EAGLE (Effective Approach to Achieving High Level Software Productivity), as shown by various data collected in Hitachi from the 1980s to the 2000s. We have evaluated productivity by using three metrics, l) program generation rate using reusable program skeletons and components, 2) fault density at two test phase, and 3) learning curve for the education of inexperienced programmers. We will show that productivity has been improved by the various facilities of EAGLE.

Acoustic modeling in speech recognition uses very little knowledge of the speech production process. At many levels our models continue to model speech as a surface phenomenon. Typically, hidden Markov model (HMM) parameters operate primarily in the acoustic space or in a linear transformation thereof; state-to-state evolution is modeled only crudely, with no explicit relationship between states, such as would be afforded by the use of phonetic features commonly used by linguists to describe speech phenomena, or by the continuity and smoothness of the production parameters governing speech. This survey article attempts to provide an overview of proposals by several researchers for improving acoustic modeling in these regards. Such topics as the controversial Motor Theory of Speech Perception, work by Hogden explicitly using a continuity constraint in a pseudo-articulatory domain, the Kalman filter based Hidden Dynamic Model, and work by many groups showing the benefits of using articulatory features instead of phones as the underlying units of speech, will be covered.

This paper describes a multiband vector quantization (VQ) technique based on inner product for wideband speech coding at 16 kb/s. Our approach consists of splitting the input speech into two separate bands and then applying an independent coding scheme for each band. A code excited linear prediction (CELP) coder is used in the lower band while a transform based coding strategy is applied in the higher band. The spectral components in the higher frequency band are represented by a set of modulated lapped transform (MLT) coefficients. The higher frequency band is divided into three subbands, and the MLT coefficients construct a vector for each subband. Specifically, for the VQ of these vectors, an inner product-based distance measure is proposed as a new strategy. The proposed 16 kb/s coder with the inner-product based distortion measure achieves better performance than the 48 kb/s ITU-T G.722 in subjective quality tests.

In order to judge the goodness of zero correlation zone sequence sets, a new concept, called ZCZ characteristic, is proposed. Then by defining a sequence operation, i.e. correlation product, and establishing its basic properties, a new approach to construct sets of sequences with a large zero correlation zone is presented.

Trellis diagrams of lattices and the Viterbi algorithm can be used for decoding. It has been known that the numbers of states and labels at every level of any finite trellis diagrams of a lattice L and its dual L* under the same coordinate system are the same. In the paper, we present concrete expressions of the numbers of distinct paths in the trellis diagrams of L and L* under the same coordinate system, which are more concrete than Theorem 2 of [1]. We also give a relation between the numbers of edges in the trellis diagrams of L and L*. Furthermore, we provide the upper bounds on the state numbers of a trellis diagram of the lattice L1L2 by the state numbers of trellis diagrams of lattices L1 and L2.

In this paper, a cell-driven multiplier generator is developed that can produce high-performance gate-level netlists for multiplier-related arithmetic functional units, including multipliers, multiplier and accumulators (MAC) and dot product calculator. The generator optimizes the speed/area performance both in the partial product compression and in the final addition stage for the specified process technology. In addition to the conventional CMOS full adder cells, we have also designed fast compression elements based on pass-transistor logic for further performance improvement of the generated multipliers. Simulation results show that our proposed generator could produce better multiplier-related functional units compared to those generated using Synopsys Designware library or other previously proposed approaches.

In order to control a sound field using multiple sources and microphones, we must choose the optimum values of parameters such as the numbers of sources and microphones, the location of the sources and the microphones and the filter tap length. Because there is a huge number of possible combinations of these conditions, the boundary surface control principle can be useful as a basis of a design method of such a system. In this paper, a design method of sound field reproduction and active noise control based on the BSC principle are described and several example of its application are presented.

In this paper, we propose a new on-line adaptive relaxation algorithm for an inverse filter in a multichannel sound reproduction system. The fluctuation of room transfer functions degrades reproduced sound in conventional sound reproduction systems in which the coefficients of the inverse filter are fixed. In order to resolve this problem, an iterative relaxation algorithm for an inverse filter performed by truncated singular value decomposition (adaptive TSVD) has been proposed. However, it is difficult to apply this method within the time duration of the sound of speech or music in the original signals. Therefore, we extend adaptive TSVD to an on-line-type algorithm based on the observed signal at only one control point, normalizing the observed signal with the original sound. The result of the simulation using real environmental data reveals that the proposed method can always carry out the relaxation process against acoustic fluctuation, for any time duration. Also, subjective evaluation in the real acoustic environment indicates that the sound quality improves without degrading the localization.

We introduce a method, so called FC method, for maintaining software resources, such as source codes and design documents, in consumer electronics products. Because a consumer electronics product is frequently and rapidly revised, software components in such product are also revised in the same way. However, it is not so easy for software engineers to follow the revision of the product because requirements changes for the product, including the changes of its functionalities and its hardware components, are largely independent of the structure of current software resources. FC method lets software engineers to restructure software resources, especially design documents, stepwise so as to follow the requirements changes for the product easily. We report an application of this method in our company to validate it. From the application, we can confirm that the quality of software was improved about in twice, and that efficiency of development process was also improved over four times.

The history of forward error correction in optical communications is reviewed. The various types of FEC are classified as belonging to three generations. The first generation FEC represents the first to be successful in submarine systems, when the use of RS(255, 239) became widespread as ITU-T G.975, and also as G.709 for terrestrial systems. As WDM systems matured, a quest began for a stronger second generation FEC. Several types of concatenated code were proposed for this, and were installed in commercial systems. The advent of third-generation FEC opened up new vistas for the next generation of optical communication systems. Thanks to soft decision decoding and block turbo codes, a net coding gain of 10.1 dB has been demonstrated experimentally. That brought us a number of positive impacts on existing systems. Each new generation of FEC was compared in terms of the ultimate coding gain. The Shannon limit was discussed for hard or soft decision decoding. Several functionalities employing the FEC framing were introduced, such as overall wrapping by the FEC frame enabling the asynchronous multiplexing of different clients' data. Fast polarization scrambling with FEC was effective in mitigating polarization mode dispersion, and the error monitor function proved useful for the adaptive equalization of both chromatic dispersion and PMD.

This paper studies a cellular system with mobile customers. The network system consists of cells, the tagged cell and the adjacent cells which surround the tagged one. Each cell has a finite number of channels that give calls to the mobile customers. The service (holding) time distribution of the calls is general. Customers in the adjacent cells inflow into the tagged cell according to a Poisson process. The sojourn time distribution of each customer in the tagged cell is general. Each customer without call in progress generates his call according to a Poisson process. It is proved that the steady state distribution in the tagged cell is the generalized Erlang loss formula which is the joint distribution of the number of customers with calls and the number of customers without calls. The distribution depends on the service time distribution and the sojourn time distribution only through their means.

In this paper, we study trellis properties of the tensor product (product code) of two linear codes, and prove that the tensor product of the lexicographically first bases for two linear codes in minimal span form is exactly the lexicographically first basis for their product code in minimal span form, also the tensor products of characteristic generators of two linear codes are the characteristic generators of their product code.

A smell reproduction technique is useful in the field of virtual reality. We have developed the system called an odor recorder for reproducing the smell recorded using the odor sensing technique. We proposed the new type of the odor recorder using the inkjet devices together with a mesh heater. Droplets with tiny volume were forcibly evaporated to generate smell rapidly and reproducibly. Moreover, the mesh heater was directly connected to the sensors without plumbing tubes and the sensors were placed away from the wall of sensor cell. The recording time of the odor with high odor intensity became much shorter than that of the previous system. Then, the recipe of jasmine scent composed of benzyl acetate and Ylang Ylang was successfully determined using the proposed system.

The methodologies of product-line engineering emphasize proactive reuse to construct high-quality products more quickly that are less costly. Requirement engineering for software product families differs significantly from requirement engineering for single software products. The requirements for a product line are written for the group of systems as a whole, with requirements for individual systems specified by a delta or an increment to the generic set. Therefore, it is necessary to identify and explicitly denote the regions of commonality and points of variation at the requirement level. In this paper, we suggest a method of producing requirements that will be a core asset in the product line. Briefly, requirements for families of similar systems (i.e. domain) are collected and generalized which are then analyzed and modeled. The domain requirement as a core asset explicitly manages the commonality and variability. Through this method, the reuse of domain requirements can be enhanced. As a result, the cost and time of software development can be reduced and the productivity increased while significantly reducing error in the requirements.