We propose Optimal Temporal Decomposition (OTD) of speech for voice morphing preserving Δ cepstrum. OTD is an optimal modification of the original Temporal Decomposition (TD) by B. Atal. It is theoretically shown that OTD can achieve minimal spectral distortion for the TD-based approximation of time-varying LPC parameters. Moreover, by applying OTD to preserving Δ cepstrum, it is also theoretically shown that Δ cepstrum of a target speaker can be reflected to that of a source speaker. In frequency domain interpolation, the Laplacian Spectral Distortion (LSD) measure is introduced to improve the Inverse Function of Integrated Spectrum (IFIS) based non-uniform frequency warping. Experimental results indicate that Δ cepstrum of the OTD-based morphing spectra of a source speaker is mostly equal to that of a target speaker except for a piecewise constant factor and subjective listening tests show that the speech intelligibility of the proposed morphing method is superior to the conventional method.

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.

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.

This letter handles symbolic simulation for high-level hardware design descriptions including uninterpreted functions. Two new heuristics are introduced, which are named "symbolic function table" and "synchronization". In the experiment, the equivalence of a hardware/software codesign was checked up to a given finite number of cycles, which is composed of a behavioral design, that is, a small DSP program written in C, and its register-transfer-level implementation, a VLIW architecture with an assembly program. Our symbolic simulator succeeded in checking the equivalence of the two descriptions which were not tractable without the heuristics.

This letter presents a significant property of the mapping parameters that play a central role to represent a given signal in Fractal Interpolation Functions (FIF). Thanks to our theoretical analysis, it is derived that the mapping parameters required to represent a given signal are also applicable to represent the upsampled signal of a given one. Furthermore, the upsampled signal obtained by using the property represents the self-affine property more distinctly than the given signal. Experiments show the validity and usefulness of the significant property.

In this letter, we present the new type parallel-coupled band-pass filter (BPF) which uses the dielectric guide in coupled sections with finite metallization thickness. A mode-matching method has been used to analyze this new structure and the simulation results are shown and validated through comparison with other available data. The results in this letter show that the dielectric guide of coupled lines with finite metal strips can be newly added to the design parameters of the parallel-coupled BPF structure and other microwave applications.

Channel estimation is one of the key technologies in mobile communications. Channel estimation is critical in providing high data rate services and to overcome fast fading in very high-speed mobile communications. This paper presents a novel channel estimation based on hybrid spreading of I and Q signals (CEHS). Simulation results show that it can effectively mitigate the influence of fast fading and enable to provide high data rates for very high speed mobile systems.

Lasers have been established as a unique nanoprocessing tool due to its intrinsic three-dimensional (3D) fabrication capability and the excellent compatibility to various functional materials. Here we report two methods that have been proved particularly promising for tailoring 3D photonic crystals (PhCs): pinpoint writing via two-photon photopolymerization and multibeam interferential patterning. In the two-photon fabrication, a finely quantified pixel writing scheme and a method of pre-compensation to the shrinkage induced by polymerization enable high-reproducibility and high-fidelity prototyping; well-defined diamond-lattice PhCs prove the arbitrary 3D processing capability of the two-photon technology. In the interference patterning method, we proposed and utilized a two-step exposure approach, which not only increases the number of achievable lattice types, but also expands the freedom in tuning lattice constant.

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.

We propose a new protocol to achieve fault recovery of multicast applications in IP internetwork with mobile participators. Our protocol uses the basic unicast routing capability of IETF Mobile IP as the foundation, and leverages existing IP multicast models to provide reliable multicast services for mobile hosts as well. We believe that the resulting scheme is simple, scalable, transparent, and independent of the underlying multicast routing facility. A key feature of our protocol is the use of multicast forwarding agent (MFA) to address the scalability and reliability issues in the reliable mobile multicast applications. Our simulation results show the distinct performance advantages of our protocol using MFAs over two other approaches proposed for the mobile multicast service, namely Mobile Multicast Protocol (MoM) and bi-directional tunneling, particularly as the number of mobile group members and home agents (HAs) increases.

Interval arithmetic is able to be applied when we include the ranges of various functions. When we include them applying the interval arithmetic, the serious problem that the widths of the range inclusions increase extremely exists. In range inclusion of polynomials particularly, Horner's method and Alefeld's method are well known as the conventional methods which mitigates this problem. The purpose of this paper is to propose the new methods which are able to mitigate this problem more efficiently than the conventional methods. And in this paper, we show and compare the efficiencies of the new methods by some numerical examples.

We propose two new adaptive minimum symbol error rate algorithms based on biased and unbiased decision rule respectively for M-ary PAM equalizer systems. The proposed algorithms can be processed either on-line or off-line depending on the availability of the information on channel impulse response. Comparisons are made between our algorithms with other existing algorithms. Computer simulations are performed to present performance results and some important algorithm properties including the effect of varying equalizer length and SNR values.

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.

This paper investigates modified random timers based on uniform and exponentially distributed timers for feedback scalability for large groups. We observe the widely-used probability distribution functions and propose new ones that are aware of network delays. The awareness of network delays of our proposed modified p.d.fs proves to be able to achieve lower expected number of messages compared to the original ones given that the parameters are optimized for the network variables: the number of receivers, and the network delay. In our analysis we derive an equation to estimate the optimized parameter based on these network variables. We also simulate the p.d.fs for heterogenous network delays and find that each receiver only needs to be aware of its network delay.

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.

The multimedia applications have recently generated much interest in wireless network infrastructure with supporting the quality-of-service (QoS) communications. In this paper, we propose a lantern-tree-based QoS on-demand multicast protocol for wireless ad hoc networks. Our proposed scheme offers a bandwidth routing protocol for QoS support in a multihop mobile network, where the MAC sub-layer adopts the CDMA-over-TDMA channel model. The QoS on-demand multicast protocol determines the end-to-end bandwidth calculation and bandwidth allocation from a source to a group of destinations. In this paper, we identify a lantern-tree for developing the QoS multicast protocol to satisfy certain bandwidth requirement, while the lantern-tree is served as the multicast-tree. Our lantern-tree-based scheme offers a higher success rate to construct the QoS multicast tree due to using the lantern-tree. The lantern-tree is a tree whose sub-path is constituted by the lantern-path, where the lantern-path is a kind of multi-path structure. This obviously improves the success rate by means of multi-path routing. In particular, our proposed scheme can be easily applied to most existing on-demand multicast protocols. Performance analysis results demonstrate the achievements of our proposed protocol.

The main issue in real time voice applications over Internet is to investigate a lossless playout without jitter while maintaining playout delay as small as possible. Existing playout algorithms estimate network delay by using timestamps and determine the playout schedule only at the beginning of each talkspurt. Also their scheduled playout time is determined based on a fixed upper playout delay bound over a talkspurt. The sliding adaptive playout algorithm we propose can estimate jitter without using timestamps and its playout time is allowed to slide to a certain extent. The aim of sliding playout schedule is to determine the scheduled playout time at the beginning of each talkspurt based on the playout delay expected under the normal condition where the degree of actual jitter is below the magnitude which is not quite large in relation to variations in the "baseline" delays. Then the proposed algorithm can be effectively applied to minimize the scheduled playout delay while improving the voice quality against a spike which may occur at the beginning of a talkspurt as well as a spike which occurs in the middle of a talkspurt. We develop an analytical model of the general adaptive playout algorithms and analyze the playout buffer performance for different degrees of jitter, different values of the scheduled playout delay, different maximum lengths of delay spikes, and arbitrary tolerable ranges of sliding. Based on the analytical results, we suggest the specific values of parameters used in the sliding algorithm.

One of the key issues in the next generation Internet is end-to-end Quality of Service (QoS) provisioning for real-time applications. The Differentiated Services (DiffServ) architecture offers a scalable alternative to provide QoS in the Internet. However, within this architecture, an efficient scheduling mechanism is still needed to ensure such QoS guarantees. In this paper, scheduling mechanism for supporting QoS differentiation among multiple traffic classes in IP differentiated services networks is studied. A scheduling algorithm called Multiclass Efficient Packet Fair Queueing (MEPFQ) is proposed that enables fair bandwidth sharing while supporting better bounds on end-to-end network delay for QoS-sensitive applications such as voice over IP (VoIP) within the DiffServ framework. The mechanism allows to create service classes and assign proportional weights to such classes efficiently according to their resource requirements. Besides, MEPFQ tries to ensure that packets from low priority class will not be starved even under extreme congestion cases. The results from the simulation studies show that the mechanism is able to ensure both the required end-to-end network delay bounds and bandwidth fairness for QoS-sensitive applications based on the specified service weights under various traffic and network conditions. Another important aspect of the MEPFQ algorithm is that the scheme has lower implementation complexity, along with scalability to accommodate the growing traffic flows at the core routers of high-speed Internet backbone.

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.

The IP routing lookup problem is equivalent to finding the longest prefix of a packet's destination address in a routing table. It is a challenging problem to design a high performance IP routing lookup architecture, because of increasing traffic, higher link speed, frequent updates and increasing routing table size. At first, increasing traffic and higher link speed require that the IP routing can be executed at wire speed. Secondly, frequent routing table updates require that the insertion and deletion operations should be simple and low delay. At last, increasing routing table size hopes that less memory is used in order to reduce cost. Although many schemes to achieve fast lookup exist, less attention is paid on the latter two factors. This paper proposed a novel pipelined IP routing lookup architecture using selective binary search on hash table organized by prefix lengths. The evaluation results show that it can perform IP lookup operations at a maximum rate of one lookup per cycle. The hash operation ratio for one lookup can be reduced to about 1%, less than two hash operations are needed for one table update and only 512 kbytes SRAM is needed for a routing table with about 43000 prefixes. It proves to have higher performance than the existing schemes.