This paper proposes a very small on-chip multimedia real-time OS for embedded system LSIs, and demonstrates its usefulness on MPEG-2 multimedia applications. The real-time OS, which has a conditional cyclic task with suspend and resume for interacting hardware (HW) / software (SW) of embedded system LSIs, implements the minimum set of task, interrupt, and semaphore managements on the basis of an analysis of embedded software requirements. It requires only about 2.5 Kbytes memory on run-time, reduces redundant conventional cyclic task execution steps to about 1/2 for HW/SW interactions, and provides sufficient performance in real-time through implementing two typical embedded softwares for practical multimedia system LSIs: an MPEG-2 system protocol LSI and an MPEG-2 video encoder LSI. This on-chip multimedia real-time OS with 2.5 Kbyte memory will be acceptable for future multimedia embedded system LSIs.

Recently many researches concerning heart sound analysis are being processed with development of digital signal processing and electronic components. But there are few researches about recognition of heart sound, especially full cardiac cycled heart sound. In this paper, three new recognition methods about full cardiac cycled heart sound were proposed. The first method recognizes the characteristics of heart sound by integrating important peaks and analyzing statistical variables in time domain. The second method builds a database by principal components analysis on training heart sound set in time domain. This database is used to recognize new input of heart sound. The third method builds the same sort of the database not in time domain but in time-frequency domain. We classify the heart sounds into seven classes such as normal (NO) class, pre-systolic murmur (PS) class, early systolic murmur (ES) class, late systolic murmur (LS) class, early diastolic murmur (ED) class, late diastolic murmur (LD) class and continuous murmur (CM) class. As a result, we could verify that the third method is better efficient to recognize the characteristics of heart sound than the others and also than any precedent research. The recognition rates of the third method are 100% for NO, 80% for PS and ES, 67% for LS, 93 for ED, 80% for LD and 30% for CM.

In this paper, we describe statistical properties of timing jitter of symbol timing recovery circuit for carrierless amplitude/phase modulation (CAP)-based very high-rate digital subscriber line (VDSL) system. Analytical expressions of the timing jitter for envelope-based timing recovery system, such as squarer-based timing recovery (S-TR) and absolute-value-based timing recovery (A-TR) schemes, are derived in the presence of additive white Gaussian noise (AWGN) or far-end crosstalk (FEXT). In particular, the analytical and simulation results of the timing jitter performance are presented and compared for a 51.84 Mb/s 16-CAP VDSL system. The A-TR system implemented digitally meets the DAVIC's VDSL system requirement, which specifies the maximum peak-to-peak jitter value of 1.5 nsec and the acquisition time of 20 msec.

A new method for active suppression of reflected sound waves is proposed in this paper. The proposed control system is based on the state feedback control. FEM (Finite Element Method) was applied to represent the sound field under the system equations as proposed by Samejima et al. A new performance index was derived so as to minimize the sound intensity leaving a control region, which was set around the control source on a wall. On the basis of the system equations and the new performance index, an optimal feedback law governing suppression of waves reflected from the wall was derived. In order to evaluate the validity of the proposed method, computer simulations in one- and two-dimensional sound fields were executed. In a one-dimensional sound field, the time response was examined, and the distribution of the instantaneous sound intensity was evaluated in a two-dimensional sound field. The results showed that the reflected sound waves can be suppressed quite well in one-dimensional sound fields by using this method and that the proposed method can potentially suppress the reflected sound waves in the two-dimensional sound fields as well.

In this letter, we propose an efficient channel allocation scheme to provide multicast traffic in multitier cellular systems. Our proposed scheme allocates microcell/macrocell channels based on the 'microcell-group size' and probability. Also, we analyze the performance of the scheme in view of the call blocking probability in case of considering overflow traffic or not. Numerical results show that our proposed scheme exhibits a better performance than conventional schemes.

One of the disadvantages of compressed data is their vulnerability, that is, even a single corrupted bit in compressed data may destroy the decompressed data completely. Therefore, Variable-to-Fixed length Arithmetic Coding, or VFAC, with error detecting capability is discussed. However, implementable error recovery method for compressed data has never been proposed. This paper proposes Burst Error Recovery Variable-to-Fixed length Arithmetic Coding, or BERVFAC, as well as Error Detecting Variable-to-Fixed length Arithmetic Coding, or EDVFAC. Both VFAC schemes achieve VF coding by inserting the internal states of the decompressor into compressed data. The internal states consist of width and offset of the sub-interval corresponding to the decompressed symbol and are also used for error detection. Convolutional operations are applied to encoding and decoding in order to propagate errors and improve error control capability. The proposed EDVFAC and BERVFAC are evaluated by theoretical analysis and computer simulations. The simulation results show that more than 99.99% of errors can be detected by EDVFAC. For BERVFAC, over 99.95% of l-burst errors can be corrected for l 32 and greater than 99.99% of other errors can be detected. The simulation results also show that the time-overhead necessary to decode the BERVFAC is about 12% when 10% of the received words are erroneous.

Mass measurement system for the measurement of mass of substances placed in randomly vibrating circumstances has been developed. Mass measurement range was defined from 0 g to 400 g for the primary model with the measurement error of approximately 3% when randomly directional vibration of 6 m/sec2 acceleration was applied to the system.

In order that speech recognition system may have a high recognition rate in a noisy environment, a wide-band sharp directional microphone system is required at the input for securing a high S/N ratio. The authors have already reported the realization of a wide-band uni-directional microphone system by three-microphone integration method. In this paper, we intend to describe the derivation of a sharp directivity function and the realization of its microphone system. First, setting the shape of the characteristic function to bring a sharp directional pattern and then expanding it into the Fourier series, we derive a new directivity function. Next, on the basis of this directivity function, we will present a sharp directional microphone system with only three non-directional microphones and the subsequent analog signal processing. And also, the directional pattern acquired by the proposed method and the effect of the dispersion in the sensitivity of the constituent microphones on the directivity are discussed in detail.

A K-band MMIC subharmonically pumped mixer integrating local oscillator (LO) amplifier has been developed. For up-converter application, it is necessary to reduce the leakage of second harmonic component of LO frequency to RF port, which is generated by nonlinear operation of LO amplifier. A quasi-lumped short-circuited stub using microstrip structure has been successfully applied to the MMIC mixer to enhance 2fLO-suppression. We propose a new configuration of a quasi-lumped short-circuited stub, which reduces the influence of parasitic elements of via-holes. The developed MMIC has a one-stage LO amplifier and it has shown about 10 dB-improvement of 2fLO-suppression compared to conventional configuration using a quarter-wavelength short-circuited stub.

In this paper, we propose a Call Admission Control (CAC) scheme for the Direct Sequence-Code Division Multiple Access (DS-CDMA) systems supporting voice and data services and analyze the Erlang capacity under the proposed CAC scheme. Service groups are classified by Quality of Service (QoS) requirements such as the required Bit Error Rate (BER) and information bit rate, and Grade of Service (GoS) requirement such as required call blocking probability. Different traffics require different system resources based on their QoS requirements. In the proposed CAC scheme, some system resources are reserved exclusively for handoff calls to have high priority over new calls. Additionally, the queueing is allowed for both new and handoff data traffics that are not sensitive to delay. As a performance measure of the suggested CAC scheme, Erlang capacity is introduced. For the performance analysis, a four-dimensional Markov chain model is developed. As a numerical example, Erlang capacity of an IS-95B type system is depicted, and optimum values of system parameters such as the number of reservation channels and queue lengths are found. Finally, it is observed that Erlang capacity is improved more than 2 times by properly selecting the system parameters with the proposed CAC scheme. Also, the effect of handoff parameters on the Erlang capacity is observed.

This paper proposes an optical power limiter composed of serially connected two tapered velocity couplers consisting partly of nonlinear material. The method of device designing is explained and it is exemplified that the optical output can be regulated stably to a prescribed value over a wide range of optical input. The device performance is simulated by means of FD-BPM algorithm.

Off-line state identification methods for a sequential machine using a homing sequence or an adaptive homing sequence (AHS) are well-known in the automata theory. There are, however, so far few studies on the subject of the on-line state estimator such as a state observer (SO) which is used in the linear system theory. In this paper, we shall construct such an SO for a Moore machine based on the state identification process by means of AHSs, and discuss the convergence property of the SO.

To satisfy huge service demand and multi-traffic requirements with limited bandwidth, this paper proposes two different procedures of multi-channel multiple access schemes with the slotted ALOHA operation for both data and voice traffic and presents an exact analysis to numerically evaluate the performance of the systems. In scheme I, there is no limitation on access between data transmissions and voice transmissions, i.e., all channels can be accessed by all transmissions. In scheme II, a channel reservation policy is applied, where a number of channels are used exclusively for voice packets while the remaining channels are used for both data packets and voice packets. We call the system using scheme I "Non-reservation system" and call the system using scheme II "Reservation system. " Performance characteristics we obtained include loss probability for voice traffic, average packet delay for data traffic and channel utilization for both traffic. The performance of the schemes and the effects of the design parameters are numerically evaluated and compared to a wide-bandwidth conventional single-channel slotted ALOHA system with single data traffic. The analysis presented in this paper will be not only useful for the performance evaluation and the optimum design of multi-channel multi-traffic systems in wireless environments, but also applicable to evaluate other performance measures in priority networks, cellular mobile radio networks or multi-hop wireless networks.

The maximum amounts of embedded information that is important in practical system design of two watermarking methods based on the parallel combinatorial spread spectrum (PC/SS) scheme are discussed in this paper. One is a private watermarking method proposed in this paper and has a practical strong point to make the quality of the watermarked image to be constant in any images. The other is a public watermarking method that was previously proposed by the authors. Through this study, the minimum number of orthogonal sequences for embedding the required amount of information under the condition that quantization noise is only assumed is found in each watermarking method.

The normalized least mean square (NLMS) algorithm has the drawback that the convergence speed of adaptive filter coefficients decreases when the reference signal has high auto-correlation. A technique to improve the convergence speed is to apply the decorrelated reference signal to the calculation of the gradient defined in the NLMS algorithm. So far, only the effect of the improvement is experimentally examined. The convergence property of the adaptive algorithm to which the technique is applied is not analized yet enough. This paper first defines a cost function properly representing the criterion to estimate the coefficients of adaptive filter. The name given in this paper to the adaptive algorithm exploiting the decorrelated reference signal, 'normalized least mean EE' algorithm, exactly expresses the criterion. This adaptive algorithm estimates the coefficients so as to minimize the product of E and E' that are the differences between the responses of the unknown system and the adaptive filter to the original and the decorrelated reference signals, respectively. By using the cost function, this paper second specifies the convergence condition of the normalized least mean EE' algorithm and finally presents computer simulations, which are calculated using real speech signal, to demonstrate the validity of the convergence condition.

The performance of a twisted-pair channel under ADSL environment is assumed to be dominated by far end crosstalk (FEXT) and additive white Gaussian noise (AWGN). In this paper, we study the channel capacity of the copper twisted pair and the optimum input power spectral density distribution at this channel capacity in the presence of ADSL environment. The channel capacity under different loop length and different input power will also be given. The simulation results show that the distribution of the optimum input power spectral density in the presence of AWGN and FEXT is not uniform. This is different from the situation where AWGN is the only interference, where the input power distribution is approximately uniform.

In this letter, the bandwidth adaptation of an adaptive IIR multiline enhancer is explored. In addition to the problem of bias cancellation of the main structure, different aspects of the proposed filter such as noise equivalent bandwidth and optimal bandwidth are considered and compared with the half power bandwidth of the adaptive multiline enhancer. Since the center frequency of the multiple sinusoids of the input signal is estimated with no bias, with the assumption that the center frequency of the incoming signal is accurately adapted, the error surface of the algorithm is calculated analytically as a function of the filter bandwidth. Computer simulations are used to compare optimum and adapted bandwidths.

The hardware implementation of a proposed high dimensional discrete torus knot code was successfully realized on an ASIC chip. The code has been worked on for more than a decade since then at Aichi Prefectural University and Nagoya Institutes of Technology, both in Nagoya, Japan. The hardware operation showed the ability to correct the errors about five to ten times the burst length, compared to the conventional codes, as expected from the code configuration and theory. The result in random error correction was also excellent, especially at a severely degraded error rate range of one hundredth to one tenth, and also for high grade characteristic exceeding 10-6. The operation was quite stable at the worst bit error rate and realized a high speed up to 50 Mbps, since the coder-decoder configuration consisted merely of an assemblage of parity check code and hardware circuitry with no critical loop path. The hardware architecture has a unique configuration and is suitable for large scale ASIC design. The developed code can be utilized for wider applications such as mobile computing and qualified digital communications, since the code will be expected to work well in both degraded and high grade channel situations.

In this paper, important methods for cryptanalysis of the stream cipher based on a class of keystream generators are discussed. These methods employ an approach called the fast correlation attack. This cryptographic problem is treated by considering its equivalent channel coding approach, namely decoding of certain very low rate codes in presence of very high noise. A novel family of algorithms for the fast correlation attack is presented. The algorithms are based on the iterative decoding principle in conjunction with a novel method for constructing the parity-checks. A goal of this paper is to summarize reported results and to compare some of the recent ones. Accordingly, the family is compared with recently proposed improved fast correlation attacks based on iterative decoding methods. An analysis of the algorithms performances and complexities is presented. The corresponding trade-offs between performance, complexity and required inputs are pointed out.

The block cipher CAST-256 based on CAST-128 was a candidate algorithm for the AES round 1. In this paper we present a first result of a differential attack on CAST-256 reduced to 9 quad-rounds. One of the three round functions of CAST-256 has differential characteristics, for which a non-zero inputxor results in a zero outputxor, with high probability. This type of characteristic is the most useful for differential attack. We also show that CAST-256 has weak keys with respect to differential attack. Thus CAST-256 reduced to 9 quad-rounds can be attacked using 2123 chosen plaintexts in the case of differentially weak keys. The time complexity is about 2100 encryptions. Immunity to differential cryptanalysis of CAST-256 is not necessarily improved compared with CAST-128. Only 5 rounds of CAST-128 can be attacked using a similar differential characteristic.