The frequency offset estimation is used to correct any frequency error of the local reference oscillator. In this letter, a frequency offset estimation algorithm utilizing the peak phase error detection and frequency offset smoother is proposed for burst data transmission. The basic idea of frequency offset estimator is to use a curve fitting method. The proposed peak phase error detection avoids a large phase error which yields a bad value for FOE. In order to control the AFC, frequency offset smoother by a simple filter is used. Simulation results show that the proposed algorithm is adequate for frequency offset estimation of burst data transmission.

Effects of the model order estimation error in the TLS-ESPRIT algorithm were investigated. It was found that if the model order is overestimated true signal parameters are preserved even though spurious signals of which power values are negligibly small appear, whereas if the model order is underestimated some signals degenerate to each others, resulting in the erroneous estimates.

This letter introduces a new digital halftoning technique based on error diffusion along a random space-filling curve. The purpose of introducing randomness is to erase regular patterns which tend to arise in an image area of uniform intensity. A simple algorithm for generating a random space-filling curve is proposed based on a random spanning tree and maze traversal. Some experimental results are also given.

This paper proposes a fast synchronization scheme with a short preamble signal for high data rate wireless LAN systems using orthogonal frequency division multiplexing (OFDM). The proposed OFDM burst format for fast synchronization and the demodulator for the proposed OFDM burst format are described. The demodulator, which offers automatic frequency control and symbol timing detection, enables us to shorten the preamble length to one quarter that of a conventional one. Computer simulation results show that the degradation in required Eb/N0 due to the synchronization scheme is less than 1 dB in a selective Rayleigh fading channel.

In learning of feed-forward neural networks, so-called 'training error' is often minimized. This is, however, not related to the generalization capability which is one of the major goals in the learning. It can be interpreted as a substitute for another learning which considers the generalization capability. Admissibility is a concept to discuss whether a learning can be a substitute for another learning. In this paper, we discuss the case where the learning which minimizes a training error is used as a substitute for the projection learning, which considers the generalization capability, in the presence of noise. Moreover, we give a method for choosing a training set which satisfies the admissibility.

We propose a new method of progressive transmission of continuous tone images using multi-level error diffusion method. Assuming that the pixels are ordered and the error is diffused to later pixels, multi-level error-diffused images are resolved into a multiple number of bit planes. In an image with 8 bits per pixel, the number of the bit planes that we construct is 9, and the 2-level, 3-level, 5-level,, error-diffused images are produced by a successive use of the bit planes. The original image is finally achieved precisely.

Three multiplicative algorithms for the floating-point divide operation are compared: the Newton-Raphson method, Goldschmidt's algorithm, and a naive method that simply calculates a form of the Taylor series expansion of a reciprocal. The series also provides a theoretical basis for Goldschmidt's algorithm. It is well known that, of the Newton-Raphson method and Goldschmidt's algorithm, the former is the more accurate while the latter is the faster on a pipelined unit. However, little is reported about the naive method. In this report, we analyze the speed and accuracy of each method and present the results of numerical tests, which we conducted to confirm the validity of the accuracy analysis. Basically, the comparison are made in the context of software implementation (e. g. , a macro library) and compliance with the IEEE Standard 754 rounding is not considered. It is shown that the naive method is useful in a realistic setting where the number of iterations is small and the method is implemented on a pipelined floating-point unit with a multiply-accumulate configuration. In such a situation, the naive method gives a more accurate result with a slightly lower latency, as compared with Goldschmidt's algorithm, and is much faster than but slightly inferior in accuracy to the Newton-Raphson method.

A new method for computing the external Q and unloaded Q of a resonator in the time domain is proposed. The external Q and unloaded Q are derived from the input energy, the field amplitude at the observation point within the resonator and the output power from the port, using the energy relationship during the early stage of the amplitude growth process of the electromagnetic field in the resonator excited by a sinusoidal wave. First, this energy method is applied to the rectangular cavity without a port to carry out a comparison with the analytically derived exact solution. Over the wide range of surface resistances, the unloaded Q can be obtained with an error on the order of 0.01 percent. It is also shown that even if Q is as high as the twelfth power of ten (1012), one is still able to do the calculations with the computation time corresponding to 230 cycles of the resonant frequency. Next, the method is applied to the rectangular cavity with apertures, and the external Q and unloaded Q are computed. Based on these results, the validity of the Q computation is confirmed. This paper also reports the effects of the arrangement types and coarseness of the FDTD grid on the external Q. The rectangular cavity with inductive apertures is computed using the FDTD method. Two types of grid arrangements are used for the coarse, fine and graded meshes. When comparing the external Q's, we found considerable differences between the results obtained when using the type 1 and type 2 grid arrangements, while the difference in resonant frequencies was about 0.1%. It is satisfactory to consider that less power flows out through the aperture in the type 1 arrangement than the actual power flow, while more power flows out in the case of type 2. These facts are important when modeling the conductor corners.

One of unique features of CDMA slotted ALOHA (CDMA S-ALOHA) is that user must synchronize his transmission to given slot. Thus orthogonal sequence as spreading sequence would achieve ideal throughput if each of packets accomplish perfect synchronization. In the presence of any ambiguity in synchronizations, however, quasi-synchronous (QS) sequences suit well with CDMA S-ALOHA system. In this paper, we introduce new QS-sequences obtained from the orthogonal Gold sequences and discuss their performance when applying to CDMA S-ALOHA systems. As a result, withstanding to access timing error, good performance is ensured with this sequence under the environment of AWGN, MAI (multiple access interference) and frequency non-selective fading, that is, micro or pico cellular systems and indoor wireless LANs.

As the current Internet becomes popular in information access, demands for real-time display and playback of continuous media are ever increasing. The applications include real-time audio/video clips embedded in WWW, electronic commerce, and video-on-demand. In this paper, we present a new control protocol R3CP for real-time applications that transmit stored MPEG video stream over a lossy and best-effort based network environment like the Internet. Several control mechanisms are used: a) packet framing based on the meta data; b) adaptive queue-length based rate control scheme; c) data preloading; and d) look-ahead pre-retransmission for lost packet recovery. Different from many adaptive rate control schemes proposed in the past, the proposed flow control is to ensure continuous, periodic playback of video frames by keeping the receiver buffer queue length at a target value to minimize the probability that player finds an empty buffer. Contrary to the widespread belief that "Retransmission of lost packets is unnecessary for real-time applications," we show the effective use of combining look-ahead pre-retransmission control with proper data preloading and adaptive rate control scheme to improve the real-time playback performance. The performance of the proposed protocol is studied via simulation using actual video traces and actual delay traces collected from the Internet. The simulation results show that R3CP can significantly improve frame playback performance especially for transmission paths with poor packet delivery condition.

This paper investigates the problem of communication errors in distributed shared memory (DSM) systems. Communication errors can introduce two critical problems: damage and loss. The damage problem makes the transmitted data destroyed and then produces incorrect computational results. The loss problem causes the transmitted data lost during transmission and then not received. However, the loss problem can be easily resolved using acknowledgement. Therefore, we focus on how to efficiently handle the damage problem. In DSM systems, the size of data transferred between nodes is larger than the size actually shared between nodes. That is, when a processing node receives data, not all the data items in this received data will be used. Based on this property, we present a new technique to resolve the data damage problem in DSM systems. This technique allows a processing node to continue computation without being blocked to wait for the correct data when it receives damaged data. Therefore, the latency for handling the data damage can be hidden. However, there is an optimistic assumption made in the proposed technique. If this optimistic assumption is not valid, the latency will not be hidden. To show the advantage and the overhead of the proposed technique, we perform extensive trace-driven simulations. The simulation results show that at least 62% of the latency for handling data damage can be hidden.

The fact that bounded interval band orthonormal scaling function shows oversampling property is demonstrated. The truncation error is estimated when scaling function with oversampling property is used to recover signals from their discrete samples.

We derive an upper bound on the size of a block code with prescribed burst-error-correcting capability combining those two ideas underlying the generalized Singleton and sphere-packing bounds. The two ideas are puncturing and sphere-packing. We use the burst metric defined by Gabidulin, which is suitable for burst error correction and detection. It is demonstrated that the proposed bound improves previously known ones for finite code-length, when minimum distance is greater than 3, as well as in the asymptotic forms.

This paper presents a new approach to computing symbol error probability of fast frequency-hopped M-ary frequency shift keying (FFH/MFSK) systems with majority vote under multitone jamming. For illustrating the applications, we first consider the case in which the source data rate is fixed and the hopping rate is allowed to vary. In this case, the optimum orders of diversity for several values of M are examined. Results show that M=4 outperforms other values. Then, we treat another case in which the hopping rate is fixed and the data rate is adjusted so as to obtain maximum throughput under a given constraint of error probability. In addition to the case of diversity alone, we also evaluate the performances of the fixed hopping rate case with channel coding using convolutional code and BCH code.

Unequal error protection (UEP) is a very promising coding technique for satellite broadcasting, as it gradually reduces the transmission rate. From the viewpoint of bandwidth efficiency, UEP should be achieved in the context of multilevel coded modulation. However, the conventional mapping between encoded bits and modulation signals, usually realized for multilevel block modulation codes and multistage decoding, is not very compatible with UEP coding because of the large number of resulting nearest neighbor codewords. In this paper, new coded modulation schemes for UEP based on unconventional partitioning are proposed. A linear operation referred to as interlevel combination is introduced. This operation generalizes previous partitioning proposed for UEP applications and provides additional flexibility with respect to UEP capabilities. The error performance of the proposed codes are evaluated both by computer simulations and a theoretical analysis. The obtained results show that the proposed codes achieve good tradeoff between the proportion and the error performance of each error protection level.

In this paper, we study unequal error protection (UEP) capabilities of punctured convolutional codes. For constructing the good UEP convolutional codes, the conditional weight distributions of UEP convolutional codes are defined and evaluated. The conditional weight distributions are computed by using the transfer functions of time-varying trellis structures of punctured convolutional codes. The best UEP convolutional codes from the viewpoint of the weight distributions are listed.

This paper proposes and investigates a coding and decoding scheme to achieve adaptive unequal error protection (UEP) using several convolutional codes which have different error-correcting capabilities. An appropriate encoder is selected to unequally protect each frame of information sequence according to the importance of the frame. Since the supplemental information of selected encoder is not sent for the sake of reducing redundancy, we assume that the decoder does not know which encoder was used, and the decoder has to estimate the used encoder. In order to estimate which encoder was used, the method using biased metric in Viterbi decoding is proposed. In decoding, however, there is a problem of Decoder-Selection-Error (DSE), which is an error that the decoder selected in a receiver does not correspond to the encoder used in a transmitter. An upper bound of DSE rate in decoding is derived. The proposed decoding scheme using the biased metric in a trellis can improve DSE rate and BER performance, because transition probability of encoders is taken into account in calculating likelihood by means of making branch or path metric biased. Computer simulation is employed to evaluate the BER performance and DSE rate of the proposed scheme. The performance is compared with a conventional equal error protection scheme and a UEP with the supplemental information on the used encoder. It is found that the proposed scheme can achieve better performance than them in case N=2.

We discuss the concept of coding over the ring of integers modulo m. This method of coding finds its origin in the early work by Varshamov and Tenengolz. We first give a definition of the codes followed by some general properties. We derive specific code constructions and show computer-search results. We conclude with applications in 8-phase modulation and peak-shift correction in magnetic recording systems.

The DC component suppressing method, called Guided Scrambling (GS), has been proposed, where a source bit stream within a data block is subjected to several kinds of scrambling and a RLL (Run Length Limited) coding to make the selection set of channel bit streams, then the one having the least DC component is selected. Typically, this technique uses a convolutional operation or GF (Galois field) conversion. A review of their respective symbol error properties has revealed important findings. In the former case, the RS (Reed-Solomon) decoding capability is reduced because error propagation occurs in descrambling. In the latter case, error propagation of a data block length occurs when erroneous conversion data occurs after RS decoding. This paper introduces expressions for determining the decoded symbol error probabilities of the two schemes based on these properties. The paper also discusses the difference in code rates between the two schemes on the basis of the result of calculation using such expressions.

In this paper, we discuss the error correcting capability of the constant amplitude coding. In orthogonal multi-code CDMA systems, the transmitted signal has large amplitude fluctuation and the amplitude fluctuation causes large nonlinear distortion. In order to avoid the amplitude fluctuation, we have proposed "constant amplitude coding" in our previous study. The constant amplitude coding can achieve constant amplitude transmission by utilizing the redundancy of the coding, i. e. , the coding can have an error correcting effect. In this paper, we examine the effectiveness of the error correcting capability of the constant amplitude coding and show that the constant amplitude coding is very effective against nonlinear distortion under the consideration of the error correcting capability.