Low-density parity-check (LDPC) codes are one of the most promising next-generation error-correcting codes. For practical use, efficient methods for encoding of LDPC codes are needed and have to be studied. However, it seems that no general encoding methods suitable for hardware implementation have been proposed so far and for randomly constructed LDPC codes there have been no other methods than the simple one using generator matrices. In this paper we show that some classes of quasi-cyclic LDPC codes based on circulant permutation matrices, specifically LDPC codes based on array codes and a special class of Sridhara-Fuja-Tanner codes and Fossorier codes can be encoded by division circuits as cyclic codes, which are very easy to implement. We also show some properties of these codes.

We had recently measured a chromatic dispersion of optical fiber and a time delay of chirped fiber grating based on a bidirectional modulation of an optical intensity modulator. In this paper, we analyze characteristics of the chromatic dispersion measurement method using a bidirectional modulation of an optical intensity modulator, and give a detailed explanation about the selection of measurement setup parameters to achieve an accurate measurement. We also propose a modified measurement system to decrease relative intensity noise caused by the bidirectional transmission through a device under test.

Time-Division-Multiplexing (TDM) is usually aimed at digital signals, while Continuous Wave Time-Division- Multiplexing (i.e. CWTDM) presented in this paper mainly addresses the problem of multiple continuous signals to share a channel. According to the idea in Ref.[1], this paper proposes a novel method for implementing CWTDM, which can make multiple band-limited continuous signals time-division-multiplexed into one continuous signal without significantly expanding the frequency band. The new method has several important applications. In particular, it can be used to implement an on-board FDMA-CWTDM conversion to develop a new system of satellite communications with more efficient performance.

This paper proposes a novel speech synthesis method to generate human-like natural speech. The conventional unit-selection-based synthesis method selects speech units from a large database, and concatenates them with or without modifying the prosody to generate synthetic speech. This method features highly human-like voice quality. The method, however, has a problem that a suitable speech unit is not necessarily selected. Since the unsuitable speech unit selection causes discontinuity between the consecutive speech units, the synthesized speech quality deteriorates. It might be considered that the conventional method can attain higher speech quality if the database size increases. However, preparation of a larger database requires a longer recording time. The narrator's voice quality does not remain constant throughout the recording period. This fact deteriorates the database quality, and still leaves the problem of unsuitable selection. We propose the plural unit selection and fusion method which avoids this problem. This method integrates the unit fusion used in the unit-training-based method with the conventional unit-selection-based method. The proposed method selects plural speech units for each segment, fuses the selected speech units for each segment, modifies the prosody of the fused speech units, and concatenates them to generate synthetic speech. This unit fusion creates speech units which are connected to one another with much less voice discontinuity, and realizes high quality speech. A subjective evaluation test showed that the proposed method greatly improves the speech quality compared with the conventional method. Also, it showed that the speech quality of the proposed method is kept high regardless of the database size, from small (10 minutes) to large (40 minutes). The proposed method is a new framework in the sense that it is a hybrid method between the unit-selection-based method and the unit-training-based method. In the framework, the algorithms of the unit selection and the unit fusion are exchangeable for more efficient techniques. Thus, the framework is expected to lead to new synthesis methods.

This paper describes an approach to generating speech with emotional expressivity and speaking style variability. The approach is based on a speaking style and emotional expression modeling technique for HMM-based speech synthesis. We first model several representative styles, each of which is a speaking style and/or an emotional expression, in an HMM-based speech synthesis framework. Then, to generate synthetic speech with an intermediate style from representative ones, we synthesize speech from a model obtained by interpolating representative style models using a model interpolation technique. We assess the style interpolation technique with subjective evaluation tests using four representative styles, i.e., neutral, joyful, sad, and rough in read speech and synthesized speech from models obtained by interpolating models for all combinations of two styles. The results show that speech synthesized from the interpolated model has a style in between the two representative ones. Moreover, we can control the degree of expressivity for speaking styles or emotions in synthesized speech by changing the interpolation ratio in interpolation between neutral and other representative styles. We also show that we can achieve style morphing in speech synthesis, namely, changing style smoothly from one representative style to another by gradually changing the interpolation ratio.

An adaptive transmission scheme, in which the work in [3] is extended to multiuser environment, is proposed for LDPC-coded MIMO-OFDMA cellular systems that employ FDD by considering active user selection and sub-channel power allocation. The performance of the proposed scheme is obtained from simulation and compared with that of the conventional scheme using mean SNR only. It is shown that the proposed scheme can provide up to 5.5 dB gain over the conventional scheme at the expense of only 6 more bits in feedback information.

In order to simultaneously combat both of the inter-carrier interferences (ICIs) and multiple access interferences (MAIs) to achieve reliable performance in multi-carrier code division multiple access (MC-CDMA) systems, this letter proposes a maximum likelihood based scheme for joint frequency offset estimation and multiuser symbol detection. To reduce the computational complexity called for by the joint decision statistic without extra mechanisms, the genetic algorithm (GA) is employed to solve the nonlinear optimization involved. Due to the robustness of the GA, the joint decision statistic can be efficiently solved, and, as shown by furnished simulation results, the proposed approach can offer satisfactory performance in various scenarios.

To take intercarrier interference (ICI) attributed to time variations of the channel into consideration, the time- and frequency-selective (doubly-selective) channel is parameterized by a finite parameter model. By capitalizing on the finite parameter model to approximate the doubly-selective channel, a Kalman filter is developed for channel estimation. The ICI suppressing, reduced-complexity Viterbi-type Maximum Likelihood (RML) equalizer is incorporated into the Kalman filter for recursive channel tracking and equalization to improve the system performance. An enhancement in the channel tracking ability is validated by theoretical analysis, and a significant improvement in BER performance using the channel estimates obtained by the recursive channel estimation method is verified by Monte-Carlo simulations.

A polynomial expansion linear detector (PELD) based turbo receiver is proposed for single-carrier zero-padding block transmission (ZP-SCBT) systems over the single-input single-output (SISO) and multiple-input multiple-output (MIMO) frequency selective channels. The performance is compared with the minimum mean square error (MMSE) and match filter (MF) based turbo receivers. It is shown that the PELD-based turbo receiver provides a good trade-off between performance and complexity compared with the other two alternatives.

This letter deals with our investigations into improving the performance of a wireless uplink system when an orthogonal frequency-division multiple access (OFDMA) is used as an access scheme. To do this, the OFDMA-based uplink system adopts a frequency diversity coupled with a cyclic time shift (CTS) at the transmitter, which is named as the FD-OFDMA system with CTS. It is found that the multi-user FD-OFDMA system equipping with CTS can decrease the probability of destroying the orthogonality among the users and provide the MAI-robust reception without decreasing the bandwidth efficiency of the system.

We studied 10 Gbit/s-based time-spreading and wave-length-hopping (TS-WH) optical code division multiplexing (OCDM) using fiber Bragg gratings (FBGs). To apply it to such the high bit rate system more than ten gigabit, two techniques are adopted. One is encoding with the maximum spreading time of 400 ps, which is four times as data bit duration, to encode without shortening chip duration. Another is encoder design. The apodized refractive index profile to the unit-gratings composing the encoder is designed to encode the pulses with 10-20 ps width at 10 Gbit/s rate. Using these techniques, 210 Gbit/s OCDM is demonstrated successfully. In this scheme, transmission distance is limited due to dispersion effect because the signal has wide bandwidth to assign a wavelength-hopping pattern. We use no additional devices to compensate the dispersion, in order to construct simple and cost-effective system. Novel FBG encoder is designed to incorporate both encoding and compensating of group delay among chip pulses within one device. We confirm the extension of transmission distance in the TS-WH OCDM from the demonstration over 40 km-long single mode fiber.

The optical network is a promising approach for realizing a scalable backbone network. In backbone networks, survivability is very important because great volumes of traffic incur damage from faulty equipment. To address this issue, various recovery schemes have been proposed for optical backbone networks. Among those schemes, shared mesh restoration utilizes link bandwidth efficiently because the backup lightpaths share link bandwidth if they protect against different failures and are never utilized simultaneously. However, a route computation method for the backup lightpaths that promotes such bandwidth sharing is necessary to achieve efficient bandwidth utilization. This paper proposes a distributed route computation method for the backup lightpaths in shared mesh restoration. In this method, the link weight is estimated to be smaller if a backup lightpath newly established can share the link bandwidth with the backup lightpaths already accommodated in that link. The link weight can be calculated using the Markov Decision Theory. The bandwidth sharing between the backup lightpaths can be promoted by selecting the shortest route based on such modified link weights. The proposed method effectively realizes efficient utilization of the link bandwidth and achieves low loss rate of reliable lightpath establishment requests under the same traffic load. The proposed method restricts the amount of link state information advertised by the routing protocol and achieves a sufficiently small amount of route calculation.

Phonetic transcription of text is an indispensable component of text-to-speech (TTS) systems and is used in acoustic modeling for speech recognition and other natural language processing applications. One approach to the transcription of written text into phonetic entities or sounds is to use a set of well-defined context and language-dependent rules. The process of transcribing text into sounds starts by preprocessing the text and representing it by lexical items to which the rules are applicable. The rules can be segregated into phonemic and phonetic rules. Phonemic rules operate on graphemes to convert them into phonemes. Phonetic rules operate on phonemes and convert them into context-dependent phonetic entities with actual sounds. Converting from written text into actual sounds, developing a comprehensive set of rules, and transforming the rules into implementable algorithms for any language cause several problems that have their origins in the relative lack of correspondence between the spelling of the lexical items and their sound contents. For Standard Malay (SM) these problems are not as severe as those for languages of complex spelling systems, such as English and French, but they do exist. In this paper, developing a comprehensive computerized system for processing SM text and transcribing it into phonetic entities and evaluating the performance of this system, irrespective of the application, is discussed. In particular, the following issues are dealt with in this paper: (1) the spelling and other problems of SM writing and their impact on converting graphemes into phonemes, (2) the development of a comprehensive set of grapheme-to-phoneme rules for SM, (3) a description of the phonetic variations of SM or how the phonemes of SM vary in context and the development of a set of phoneme-to-phonetic transcription rules, (4) the formulation of the phonemic and phonetic rules into algorithms that are applicable to the computer-based processing of input SM text, and (5) the evaluation of the performance of the process of converting SM text into actual sounds by the above mentioned methods.

An efficient hybrid image vector quantization (VQ) technique based on a classification in the DCT domain is presented in this letter. This algorithm combines two kinds of VQ, predictive VQ (PVQ) and discrete cosine transform domain VQ (DCTVQ), and adopts a simple classifier which employs only three DCT coefficients in the 88 block. For each image block, the classifier switches to the PVQ coder if the block is relatively complex, and otherwise switches to the DCTVQ coder. Experimental results show that the proposed algorithm can achieve higher PSNR values than ordinary VQ, PVQ, JPEG, and JPEG2000 at the same bit-rate.

This investigation proposes a virtual-FIFO (VFIFO) back-off algorithm for wireless networks. The proposed scheme takes advantage of the central unit (CU) in a wireless network to broadcast a common back-off window size to all the users, significantly alleviating the unfairness of bandwidth utilization in conventional binary exponential back-off (BEB) algorithms. The proposed scheme exploits the CU's capability for collision detection to estimate the number of simultaneously competing users. Additionally, packets generated in a given cycle are split into groups according to their times of arrivals and are guaranteed to be serviced one after another within the next cycle. Although the proposed algorithm is not strictly first come fist served, the FIFO principle is virtually accomplished. Simulation results demonstrate that the standard deviation of delay can be improved by more than two orders and the throughput can be maintained at 0.42 when the number of users approaches infinity. The capture effect even further improves system performance.

In this paper, we briefly describe situations that may cause HFN de-synchronization for ciphering applications in UMTS. Detection methods of HFN de-synchronization are discussed and the lower bound of the HFN de-synchronization perceptibility is derived. A supporting simulation result of the perceptibility is given. Then, an Automatic Recovery of HFN Synchronization (ARHS) algorithm is presented. The average lost PDU number of the ARHS algorithm is derived and supported by simulation results. The average lost SDU number is used as the figure of merit for HFN synchronization recovery procedures. Simulation results of the average lost SDU number show that the ARHS algorithm is quite effective to recover HFN synchronization after HFN de-synchronization situations happen.

In this paper, a semi-dynamic protection path configuration method is proposed for WDM optical networks. In the method, the protection path is established by connecting several sub-lightpaths from the source node to the destination node of the original working lightpath, as opposed to conventional path restoration method where a single protection lightpath between the source-destination pair performs restoration. The proposed method provides enhanced flexibility in protection path configuration and relieves the cost of spare capacity reservation. This paper also studies the effects of wavelength conversion capability of intermediate optical cross-connect nodes on protection path routing and spare capacity utilization. In terms of spare capacity utilization, the proposed method shows substantial reduction of spare capacity overhead compared with dedicated path restoration in all optical networks without wavelength conversion, and shows similar capacity efficiency compared with shared path restoration in opaque networks with full wavelength conversion capability. In terms of robustness, the proposed method shows nearly the same restoration ratio for double-link failure as that of dynamic restoration method.

Recently, multiple-input multiple-output (MIMO) systems that realize high bit rate data transmission with multiple antennas at both transmitter and receiver have drawn much attention for their high spectral efficiency. In MIMO systems, space division multiplexing (SDM) has been researched widely. In SDM, the input data symbols are transmitted from multiple transmit antennas at the transmitter, and the output data symbols are extracted by the signal processing at the receiver. In recent wireless communications, the environments that the number of transmit antennas is larger than that of receive antennas often exist. Under such environments, the MIMO system that transmits independent data streams from each transmit antenna simultaneously cannot separate the received signals, and the signal quality deteriorates largely. Therefore, we need the scheme that attains high quality and high throughput data transmission under such environments. In this paper, we propose a throughput maximization transmission control scheme for MIMO systems. The proposed transmission control scheme selects a transmission scheme (a set of transmit antennas, modulation schemes, and coding rates) with maximum throughput based on output signal to interference and noise ratio (SINR) and output signal to noise ratio (SNR). We show that the proposed transmission control scheme attains high throughput by our computer simulation.

There is increasing evidence from in vivo recordings in monkeys trained to respond to stimuli by making left- or rightward eye movements, that firing rates in certain groups of neurons in oculo-motor areas mimic drift-diffusion processes, rising to a (fixed) threshold prior to movement initiation. This supplements earlier observations of psychologists, that human reaction-time and error-rate data can be fitted by random walk and diffusion models, and has renewed interest in optimal decision-making ideas from information theory and statistical decision theory as a clue to neural mechanisms. We review results from decision theory and stochastic ordinary differential equations, and show how they may be extended and applied to derive explicit parameter dependencies in optimal performance that may be tested on human and animal subjects. We then briefly describe a biophysically-based model of a pool of neurons in locus coeruleus, a brainstem nucleus implicated in widespread norepinephrine release. This neurotransmitter can effect transient gain changes in cortical circuits of the type that the abstract drift-diffusion analysis requires. We also describe how optimal gain schedules can be computed in the presence of time-varying noisy signals. We argue that a rational account of how neural spikes give rise to simple behaviors is beginning to emerge.

In this paper, we propose a novel architecture for optical transport networks and its operation scheme guaranteeing the QoS requirements based on real-time traffic measurement. The key concept of the proposed architecture, which we call hybrid optical transport network (HOTNET), is to adopt both optical circuit switching and optical message switching in an optical network. To implement two different switching technologies in a single network, we modify the optical burst switching scheme and merge it into a TDM wavelength routed network. Then, we propose a control framework and an architecture of a switching node for this hybrid switching paradigm. We also discuss a real-time bandwidth provisioning scheme which utilizes the advantages of two respective switching schemes for traffic engineering. Finally, we evaluate the performance of the proposed scheme via computer simulation and the results show that it can guarantee the traffic QoS requirements while maintaining high channel utilization.