We present a new space-time successive interference cancellation (ST-SIC) scheme with multiple transceiver antennas for direct-sequence code division multiple access (DS-CDMA) systems. The proposed scheme is computationally very efficient, while maintains the performance close to the previous space-time multiuser detection (ST-MUD) scheme. The bit error rate (BER) performance of the ST-SIC scheme for coherent phase shift keying (PSK) modulation is analytically examined in Rayleigh fading channels, and its validity and usefulness are demonstrated by computer simulations.

Band Division MC-CDM (BD-MC-CDM) has been proposed for high quality wireless communications and has been investigated in terms of link level performance. In this paper, we investigate frequency band and time slot selection technique from the viewpoint of system level performance in order to realize the efficient BD-MC-CDM system under cellular environments. Then a downlink frequency band and time slot selection scheme is proposed for cellular BD-MC-CDM systems. The proposed scheme selects transmission frequency band according to the signal-to-interference ratio (SIR) estimated by using the pilot signal at mobile stations and also selects transmission time slot by using the SIR threshold. Simulation results show that the proposed scheme improves the downlink throughput but degrades delay performance and it has a trade off between throughput and delay performance. By selecting suitable control parameters, the proposed scheme achieves the throughput improvement without sacrificing the delay performance.

A single chip processor suitable for various multimedia communication products has been developed. This chip achieves real-time bi-directional encoding/decoding for CIF resolution video at a frame rate of 30 fr/s, and meets such standards, as H.320 and H.324. The chip is composed of a video-processing unit for MPEG-4 and H.26X standards, a DSP unit for speech codec and multiplex processes, and a RISC unit for managing the whole chip. By heterogeneous multiple processor architecture, careful study of task sharing for each processing unit and bus configuration, a single chip solution can be achieved with reasonable operation speed and low-power consumption suitable for consumer products. Moreover, by applying an original video processing unit architecture, this chip achieves real-time bi-directional encoding/decoding for CIF-resolution video at a frame rate of 30 fr/s. An original video bus was developed to provide high performance and low-power consumption while sharing one external memory which is necessary for various video processes and graphics functions. This shared memory also has the effect of minimizing die size and I/O ports. This chip has been fabricated with 4-metal 0.18 µm CMOS technology to produce a chip area of 10.510.5 mm2 with 1.2 W power dissipation including I/O power, at 1.8 V for internal supply and 3.3 V for I/O power supply.

In this paper we describe a new framework of feature compensation for robust speech recognition, which is suitable especially for small devices. We introduce Delta-cepstrum Normalization (DCN) that normalizes not only cepstral coefficients, but also their time-derivatives. Cepstral Mean Normalization (CMN) and Mean and Variance Normalization (MVN) are fast and efficient algorithms of environmental adaptation, and have been used widely. In those algorithms, normalization was applied to cepstral coefficients to reduce the irrelevant information from them, but such a normalization was not applied to time-derivative parameters because the reduction of the irrelevant information was not enough. However, Histogram Equalization (HEQ) provides better compensation and can be applied even to the delta and delta-delta cepstra. We investigate various implementation of DCN, and show that we can achieve the best performance when the normalization of the cepstra and the delta cepstra can be mutually interdependent. We evaluate the performance of DCN using speech data recorded by a PDA. DCN provides significant improvements compared to HEQ. It is shown that DCN gives 15% relative word error rate reduction from HEQ. We also examine the possibility of combining Vector Taylor Series (VTS) and DCN. Even though some combinations do not improve the performance of VTS, it is shown that the best combination gives the better performance than VTS alone. Finally, the advantage of DCN in terms of the computation speed is also discussed.

A VLSI-specific wavelet processing technique has been developed and implemented as a processor in accordance with the JPEG2000 specification. This proposed procedure of discrete wavelet transforms uses an altered calculation equations and makes use of intermediate results through wavelet calculation. The implementation of the proposed procedure is capable of realizing a highly efficient DWT for large size images in spite of using low hardware costs and a small size buffering memory. In order to obtain fast EBCOT processing, three types of parallel processing are introduced in the EBCOT architecture. The processor performs compression of 720480 pixels images with the speed of 30 frames per second (fps) at a required operating frequency as low as 32 MHz or lower. Furthermore, it need not divide an image into tiles so that the problem of deterioration of image quality due to tile division does not occur. A prototype of this processor has been fabricated in a 0.25-µm 5-layer CMOS process. The chip is 10.210.4 mm2 in size and consumes 2.0 W when supplied with 2.5 V and 32 MHz.

We developed a compact, 16-channel integrated optical subscriber module for one-fiber bi-directional optical access systems. They can support more subscribers in a limited mounting space. For ultimate compactness, we created 8-channel integrated super-compact optical modules, 4-channel integrated limiting amplifiers, and 4-channel integrated LD drivers for Fast Ethernet. We introduce a new simulation method to analyze the electrical crosstalk that degrades sensitivity of the optical module. A new IC architecture is applied to reduce electrical crosstalk. We manufactured the optical subscriber module with these optical modules and ICs. Experiments confirm that the module offers a sensitivity of -27.3 dBm under 16-channel 125 Mbit/s simultaneous operation.

In this paper, the space time transmit diversity (STTD) decoding combined with minimum mean square error (MMSE) equalization is presented for MC-CDMA downlink and uplink in the presence of multiple receive antennas. The equalization weights that minimize the MSE for each subcarrier are derived. From computer simulation, it was found that the BER performance of STTD decoding combined with MMSE equalization and Mr-antenna diversity reception using the weights derived in this paper provides the same diversity order as 2Mr-antenna receive diversity with MMSE equalization but with 3 dB performance penalty and is always better than that with no diversity. The uplink BER performance can also be improved with STTD, but the error floor still exists. However, with 2-receive antennas in addition to 2-antenna STTD, the BER floor can be reduced to around 10-5 even for the uplink.

Video transmission over Terrestrial Trunked Radio (TETRA) mobile channel employing MPEG-4 visual coding standard is proposed in this paper. Detail parameters of the proposed systems are discussed in this paper. Performance of the proposed systems was evaluated in Average Peak Signal to Noise Ratio (APSNR) versus Signal to Noise Ratio (SNR) and Bit Error Rate (BER). In particular, the video quality that can be achieved at different channel conditions and employing different combinations of MPEG-4 visual error resilient tools is presented in this paper. Results obtained show that higher video bitrate does not necessarily lead to higher video quality at the receiver as the received video quality depends on the bit error pattern or the number of error free video packets.

Classical studies of Asynchronous Transfer Mode (ATM) switching elements and in particular the buffer behavior of the Shared Buffer Memory (SBM), assume that all read and write operations of cells to, respectively from, the SBM are executed simultaneously. However, in a real switching element, the inlets (outlets) are scanned sequentially for arriving (departing) cells during the so-called input (output) cycle. Furthermore, the input and output cycles are intermingled, each read operation being followed by a write operation. This is referred to as the Timeslot Interchange Mechanism (TIM). In this paper, we present the analysis of a queueing model that includes the TIM. We model the cell arrival processes on the inlets of the switching element as independent Bernoulli arrival processes. Moreover, we assume that cells are routed from the inlets to the outlets of the switching element according to an independent and uniform process, i.e., the destinations of consecutive cell arrivals on any given inlet are independent and for a given cell all destinations are equiprobable. Under these assumptions, we will derive expressions for the probability generating functions of the queue length in an individual routing group (a logical queue that contains all cells scheduled for the same destination), the (total) queue length in the SBM, and the cell waiting time. From these results, expressions for the mean values and the tail distributions of these quantities are calculated, and the influence of the TIM on the buffer behavior is studied through comparison with a model where all read and write operations occur simultaneously.

Software applications for the transfer of money or personal information are increasingly common on the Internet. These applications require user authentication for confirming legitimate users. One-time password authentication methods risk a stolen-verifier problem or other steal attacks because the authentication on the Internet server stores the user's verifiers and secret keys. The SAS-2 (Simple And Secure password authentication protocol, ver.2) and the ROSI (RObust and SImple password authentication protocol) are secure password authentication protocols. However, we have found attacks on SAS-2 and ROSI. Here, we propose a new method which eliminates such problems without increasing the processing load and can perform high security level same as S/Key systems without resetting the verifier.

Streaming services and visual communication services delivered over the Internet have become popular in recent years. In the future, broadband services using MPEG2/4 will become the dominant type. These services will require transport protocols that provide high quality and high throughput from end to end of the system. We propose a new transfer method that allows the network load to be adaptively balanced according to the network's state. We built a prototype of an actual MPEG2 streaming system and used it to estimate the effectiveness of this method.

The handoff in Mobile IP networks causes packet sequence disruption during a packet forwarding procedure and may result in performance degradation in higher layer protocols. We investigate the impact of handoff in the Mobile IPv6 networks, where an optimized routing with the smooth handoff is adopted. The impact on the packet sequence is measured by an 'unstable time period (UTP)' and a 'silence time period (STP).' The UTP explains the time duration of out-of-sequence packets while the STP reflects the blackout duration of a mobile node after the initiation of handoff procedure. With the analysis on the UTP and STP, the total transient time period (denoted as handoff time period or HTP) after the handoff initiation can be estimated. In our previous work, focusing on the UTP, the packet flow sequence under the smooth handoff is analyzed for the Mobile IPv4 networks. The proposed queuing-based analysis is extended in this work for the Mobile IPv6 networks. That is, several modifications are made to conform to Mobile IPv6 and at the same time the queuing analysis itself is improved to better model the handoff procedure. The numerical results show that the queuing delay for the handoff packets (affected by background traffic) and the involved link (or route) capacities affect the estimated UTP, STP, and HTP. In addition, two schemes such as priority queuing and buffered packet forwarding are introduced to reduce the transient period and the improvements are analyzed for comparison.

A time-domain equalization (TEQ) algorithm is presented to shorten the effective channel impulse response to increase the transmission efficiency of the 54 Mbps IEEE 802.11a orthogonal frequency division multiplexing (OFDM) system. In solving the linear equation Aw = B for the optimum TEQ coefficients, A is shown to be Hermitian and positive definite. The LDLT and LU decompositions are used to factorize A to reduce the computational complexity. Simulation results show high performance gains at a data rate of 54 Mbps with moderate orders of TEQ finite impulse response (FIR) filter. The design and implementation of the algorithm in field programmable gate array (FPGA) are also presented. The regularities among the elements of A are exploited to reduce hardware complexity. The LDLT and LU decompositions are combined in hardware design to find the TEQ coefficients in less than 4 µs. To compensate the effective channel impulse response, a radix-4 pipeline fast Fourier transform (FFT) is implemented in performing zero forcing equalization. The hardware implementation information is provided and simulation results are compared to mathematical values to verify the functionalities of the chips running at 54 Mbps.

In this letter, we propose an enhanced time-based registration method and analyze the performance numerically. In the analysis, we assume Poisson call arrival distribution and exponential cell resident time. The performance of the enhanced time-based registration method is compared with the performance of the original time-based registration method. In the comparisons, we see that in a certain range of parameters, the enhanced time-based registration method has better performance.

By means of the three-dimensional (3D) finite-difference time domain (FDTD) method, we have investigated in detail the optical properties of a two-dimensional photonic crystal (PC) surface-emitting laser having a square-lattice structure. The 3D-FDTD calculation is carried out for the finite size PC slab structure. The device is based on band-edge resonance, and plural band edges are present at the corresponding band edge point. For these band edges, we calculate the mode profile in the PC slab, far field pattern (FFP) and polarization mode of the surface-emitted component, and photon lifetime. FFPs are shown to be influenced by the finiteness of the structure. Quality (Q) factor, which is a dimensionless quantity representing photon lifetime, is introduced. The out-plane radiation loss in the direction normal to the PC plane greatly influences the total Q factor of resonant mode and is closely related with the band structure. As a result, Q factors clearly differ among these band edges. These results suggest that these band edges include resonant modes that are easy to lase and resonant modes that are difficult to lase.

Measuring and modeling network traffic is of key importance for the traffic engineering of IP networks, due to the growing diversity of multimedia applications and the need to efficiently support QoS differentiation in the network. Several recent measurements have shown that Internet traffic may incorporate long-range dependence and self-similar characteristics, which can have significant impact on network performance. Self-similar traffic shows variability over many time scales, and this behavior must be taken into account for accurate prediction of network performance. In this paper, we propose a new parameter fitting procedure for a superposition of Markov Modulated Poisson Processes (MMPPs), which is able to capture self-similarity over a range of time scales. The fitting procedure matches the complete distribution of the arrival process at each time scale of interest. We evaluate the procedure by comparing the Hurst parameter, the probability mass function at each time scale, and the queuing behavior (as assessed by the loss probability and average waiting time), corresponding to measured traffic traces and to traces synthesized according to the proposed model. We consider three measured traffic traces, all exhibiting self-similar behavior: the well-known pOct Bellcore trace, a trace of aggregated IP WAN traffic, and a trace corresponding to the popular file sharing application Kazaa. Our results show that the proposed fitting procedure is able to match closely the distribution over the time scales present in data, leading to an accurate prediction of the queuing behavior.

In future mobile networks, new technologies will be needed to enable a mobile host to move across heterogeneous wireless access networks without disruption of the connection. In the past, many researchers have studied handover in such IP networks. In almost all cases, special network devices are needed to maintain the host's mobility. Moreover, a host cannot move across heterogeneous wireless access networks without degradation of the goodput for real-time communication, although a mobile host with multiple network interfaces can connect to multiple wireless access networks. For these reasons, we consider that a mobile host needs to manage seamless handover on an end-to-end basis. In this paper, we propose a multi-path transmission algorithm for end-to-end seamless handover. The main purpose of this algorithm is to improve the goodput during handover by sending the same packets along multiple paths, minimizing unnecessary consumption of network resources. We evaluate our algorithm through simulations and show that a mobile host gains a better goodput.

Recently, space-time multiple trellis coded modulation (ST-MTCM) has been introduced in order to achieve maximum transmit diversity gain and larger coding gain with the existance of parallel paths, which can not be achieved with STTCM system. In order to achieve good performance, it is crucial to maximize the intra-distance, which is defined by parallel paths and determine the performance. Conventional ST-MTCM uses a generator matrix G for coded modulation; however, we find that no matrix can be designed which can maximize the intra-distance by computer search. In this paper, we focus on maximizing the intra-distance and the diversity gain, and hence design a new coded modulation scheme. We use trellis codes in this paper which cannot be described by a matrix G. The proposed codes can achieve the maximum intra-distance and thus good coding gain, which may not be achieved by conventional codes. We also show that the proposed code can achieve good performance both in quasi-static and fast flat fading channels without the need for changing the codes as is necessary in the conventional ST-MTCM scheme.

This letter presents a new concatenated code and a new criterion for the new concatenated code in fast Rayleigh fading channel. The new concatenated code consists of the cascade of a new space-time trellis code (STTC) as an inner code and a new convolutional code as an outer code. The new criterion maximizes the minimum free distance for the new convolutional code and both the minimum trace and the average trace of distance matrix for the new STTC. The new concatenated code improves the frame error rate (FER) performance significantly with low complexity. The new STTC and convolutional code are designed so as to satisfy the new criterion for 4-state 4 phase shift keying (PSK). The results of the suggested concatenated code are obtained using two transmit antennas, and shown to be significantly superior to the new and existing STTCs. As the number of receive antennas increases, the performance of the new concatenated code significantly improves, for instance, reaches FER = 10-3 at signal-to-noise ratio (SNR) = 5.2 dB for four receive antennas. Note that the proposed concatenated code also improves significant FER performance by using only one receive antenna for high SNR.

This paper presents a transmit diversity scheme that allocates space-time block codes (STBC) to beamspace and spreading codes for two-dimensional spreading orthogonal frequency-division multiplexing code-division multiplexing (OFDM-CDM) downlink transmission. In this scheme, the STBC output symbols are beam-steered using a pair of neighboring beams selected via closed-loop beam selection. The beam-steered symbols in two adjacent time slots are spread by two distinct spreading codes and multiplexed in the same spreading segment. User signals transmitted from different pairs of beams, but that share the same beam, interfere with each other when decoding STBC. Spreading codes are thus allocated to users according to beam pairs used. This is to suppress the interference in time-direction despreading that precedes decoding of STBC. Simulation results demonstrated that the proposed scheme provides beam gains or beam diversity gains or both and that it alleviates inter-code interference by spatially separating user signals by using transmit beam. The proposed scheme also provides high tolerance to large Doppler spread.