We introduce an iterative decoding algorithm for the decoding of coded interleaved frequency division multiple access (IFDMA) system in this paper. The iterative decoding algorithm incorporates a low complexity a posterior probability (APP) detector based on breadth-first tree search algorithm with a soft-input soft-out (SISO) error-correcting codes (ECC) decoder. The proposed detector combines the principles of the M-algorithm with QR decomposition (QRD-M) to generate soft decision statistics in place of the conventional APP detector which has a high computational cost. We show through simulations that the purposed iterative decoding algorithm is able to achieve significant performance gain using only very small number of iterations. Furthermore, the simulated bit error rate (BER) performance of a system using the proposed algorithm not only exhibits similar diversity order as the hypothetical matched filter bound but also approaches it at high signal-to-noise ratio (SNR).

In this paper, we investigate the use of block spread (BS) and frequency hopping (FH) techniques in coded interleaved frequency division multiple access (IFDMA) systems to achieve maximum channel diversity order. We derive the upper bound on the theoretical bit error rate (BER) that shows the maximum achievable diversity order under maximum likelihood sequence estimation (MLSE). Next, we establish the design criteria for the two different techniques to achieve the maximum diversity order. We also compare the two techniques in terms of their operational complexity and performance. Moreover, we show, through simulation, that the maximum diversity order can be achieved by using a low complexity detector in practical systems instead of the high complexity MLSE.

We address the adaptive allocation of subcarrier, bit and power resources among the base stations (BSs) of a downlink multi-cell OFDMA system using the non-cooperative game theoretic approach. Unlike the information theoretic approach where continuous bit-loading is used, our algorithm allocates integer number of bits to each subcarrier. Using a utility function with linear pricing on both the transmit bit rate and power, we first show that for a simple two-cell system with single user and fixed modulation, a Nash equilibrium (NE) solution always exists. However, for games formulated on more complicated systems such as with more than two cells, with more than one user in a cell, or when adaptive modulation is used, we observed that the existence of NE cannot be guaranteed. If NE points exist, the game can be played repeatedly with each BS (the player) selects its best strategy until the solution converges, or known as the myopic game. Based on the framework of potential games with coupled constraints, we propose an algorithm which can achieve a stable solution among the players despite that NE solution to the originally problem does not necessarily exist. The convergence of the proposed algorithm is verified by simulation.

We address the allocation of subcarriers, bits and power among the base stations (BSs) of downlink multi-cell OFDMA systems using the non-cooperative game theoretic approach. The utility function of a player is defined as the difference between the revenue generated from transmitting the bits and the cost of the power consumed. Unlike the commonly adopted water-filling approach where continuous bit loading is used, we consider assigning integer number of bits to subcarriers. We address the existence of pure Nash equilibrium (NE) of the static games for two-cell modulation adaptive and three-cell single modulation OFDMA systems. Theorems related to the existence of pure NE for the two games are developed. Using these results, we show that the existence of a stable solution for the OFDMA games cannot be guaranteed if minimum rate requirements must be satisfied.

In this paper, a low complexity a posterior probability (APP) detector based on breadth-first tree search algorithm is purposed for coded interleaved frequency division multiple access (IFDMA) system. The proposed detector combines the principles of the M-algorithm with QR decomposition (QRD-M) to generate soft decision statistics in place of the full APP detector which has a high computational cost. We first demonstrate that an IFDMA system can be viewed as a MIMO system at the receiver and then derive the proposed detector. Simulated results show that the bit error rate (BER) performance of a system using the proposed QRD-M detector performs better than a system using frequency domain equalizer, especially for channels with greater diversity. Moreover, the simulated BER performance exhibits similar diversity order as the maximum channel diversity order.

The effects of carrier frequency offset (CFO), sampling frequency offset (SFO), and channel responses on the received samples are analyzed and explored to develop the turbo joint channel estimation, synchronization and decoding scheme for coded MIMO-OFDM systems. For burst transmission, with initial estimates derived from the pre-amble, the proposed scheme can operate without the need of pilot tones during the data segment. Simulation results also show that the proposed scheme offers fast convergence and low mean-squared-error (MSE) over quasi-static Rayleigh multi-path fading channels, and can be used in a coded MIMO-OFDM transceiver in presence of multipath fading, CFO, and SFO to achieve a bit-error-rate (BER) performance comparable to that in an ideal case of perfect synchronization and channel estimation after three iterations.

This paper investigates inter-carrier interference (ICI) suppression for the uplink of an orthogonal frequency division multiple access (OFDMA) system in a time and frequency selective Rayleigh fading channel. In such a doubly selective channel, channel variation within each OFDMA block disrupts the orthogonality among subcarriers, leading to ICI. We formulate the signal model for OFDMA uplink in a doubly selective Rayleigh fading channel, and then propose a MMSE and a MMSE successive detection (MMSE-SD) scheme to suppress ICI. It is shown that the MMSE scheme is the optimal linear scheme in terms of achievable data rate, while the MMSE-SD scheme is able to further remove ICI and exploit the Doppler diversity embedded in time-varying channels. Simulation results are presented to illustrate the performance improvements that can be obtained from using the proposed suppression schemes.

This paper proposes an accurate UWB multiple-cell localization system which adopts an adaptive TDOA combination algorithm to reduce the localization uncertainty. The time difference of arrival (TDOA) is adaptively combined in such a way that only makes use of the best directional motion increments to reduce both the random channel noise and the geometric errors. Furthermore, maximum signal to noise ratio (SNR) TDOA combining and tracking error controlling are employed to reduce the localization uncertainty. Unlike the conventional TDOA equation solutions, the algorithm avoids using the nonlinear triangular calculation and the TDOA matrix inversions to improve the performance with minimizing dilution of precision (DOP). This method shows a very good competition compared to the existing methods in tracking/filtering process and even achieves a better Cramer-Rao bound (CRB). The hardware including the moving target (tag) and the receiver system had been designed with a very low cost, and long battery life. Finally, after giving the analysis of DOP, the real experimental system reveals that it outperforms the other methods in term of CRB evaluation and error (bias) mitigation.

In this paper, we propose a coverage optimization algorithm based on QoS requirements. The coverage optimization algorithm improves the network QoS performance by the joint-optimization of coverage, routing, energy efficiency and delay. The QoS metrics are created with the priorities of the routing consistence, network lifetime and the coverage model, etc. Also, a new method is developed to take advantages of the dense area to compensate the sparse area with the predictor plus interpolation method. With this method, relaying nodes from dense area are chosen to provide compensating coverage for the sparse area. Furthermore, this method can greatly improve the network lifetime by extending the contingent battery life of the nodes in the spare area. Finally, the simulation results perform the evaluation for our algorithm.

We address the adaptive allocation of subcarrier, bit and power in the downlink of multicell OFDMA systems which are operating at a frequency reuse factor of one. When a centralized scheduler is used, the presence of co-channel cell interference inter-relates the assignment of the subcarriers and powers in all the cells, making the optimal allocation of radio resources a challenging problem. The lack of optimal solution in the literature for such a centralized approach is attributed to the difficulties in solving the mixed integer nonlinear programming problem. We show that through some manipulations, the solution to the original problem can be decoupled into solving two sub- problems sequentially: first solve for power allocation which is a linear programming and then subcarrier-bit allocation which is a binary linear programming. The elimination of the nonlinear constraints in our approach results in the significant reduction in the computation complexity, making optimal solution becomes possible. Although optimal solution not always can be computed in real time when the number of subcarriers is large, it can still serve as a performance benchmark to future developed heuristic algorithms. Finally, results in our paper are more practical since discrete values are used for bit-loading. This is different to most of the reported work in the literature which adopted information theoretic approach.

The jamming robustness of frequency hopping (FH) systems with M-ary frequency shift keying (MFSK) modulation may be potentially neutralized by a follower partial-band jammer. In this paper, a maximum likelihood (ML)-based algorithm that uses a two-element array is proposed for joint follower jamming rejection and symbol detection in slow FH/MFSK systems over quasi-static flat Rayleigh fading channels. The algorithm is derived by treating both the received jamming components and the unknown data symbols as deterministic quantities to be jointly estimated in an integrated ML operation. In addition, an approximate expression for the symbol error rate (SER) of the proposed scheme is derived when BFSK signaling is employed in a jamming dominant scenario. Analytical and simulated results show that the proposed approach is able to remove jamming and outperform the conventional and sample matrix inversion (SMI)-based beam-formers in the presence of a follower partial-band jammer.

This paper proposes a pilot-aided algorithm for the joint estimation of the channel impulse response (CIR), carrier frequency offset (CFO) and sampling frequency offset (SFO) in burst-mode multiple-input multiple-output (MIMO) - orthogonal frequency division multiplexing (OFDM) systems. We employ the vector recursive least-squares (RLS) algorithm that can function as an adaptive filter with multiple outputs for the joint estimation problem in MIMO scenarios. Specifically, based on the received signal samples that contain pilot tones in the frequency domain, we formulate a cost function that includes the multi-antenna channel impulse responses and frequency offsets for the vector RLS-based joint estimation of CIR CFO and SFO. Analytical and simulated results show that, over large ranges of CFO and SFO values, the proposed estimation and tracking approach offers fast convergence, high stability, and provides a near optimum receiver performance that is remarkably close to the ideal one in the case of perfect channel estimation and synchronization over Rayleigh multi-path fading channels.

This paper proposes a pilot-aided joint channel estimation and synchronization scheme for burst-mode orthogonal frequency division multiplexing (OFDM) systems. The scheme eliminates the need of an IFFT block while keeping the low number of parameters to be estimated for low complexity without sacrificing the performance and convergence speed. For fast convergence and high performance, we develop a linearized cost function of the carrier frequency offset (CFO), sampling clock frequency offset (SFO) and channel impulse response (CIR) coefficients based on received signal samples and pilot tones in frequency domain and the corresponding recursive least square (RLS) estimation and tracking algorithm. For channel responses, CIR coefficients are estimated to benefit their low number and then transformed to the channel transfer function in order to keep low complexity. The ICI introduced by rotation due to CFO and SFO is analyzed and modeled, and a simple maximum-likelihood (ML) scheme based on the preamble is developed for coarse estimation of initial CFO and SFO values to be used in suppression of dominant ICI effects and in fine RLS estimation and tracking. Simulation results demonstrate that, in large practical ranges of CFO and SFO values, the proposed pilot-aided joint channel estimation and synchronization scheme provides a receiver performance remarkably close to the ideal case of perfect channel estimation and synchronization.

10.1109/PIMRC.2007.4394856

It is well known that motion estimation is the most computationally intensive processing unit of the H.264 video encoder. Various fast motion estimation algorithms have been proposed to reduce its complexity. Generally, these approaches achieve speedup by reducing the number of candidate search points within the search window. In this paper, we propose a new method, which uses the Sum-of-Absolute-Differences mapping (SAD map) to dynamically cache the SAD values and then reuse them for different block sizes. Experimental results on standard video sequences verified that the proposed method is capable of increasing the encoder speed by up to 15% without any loss in PSNR value or increase in bit rate. Due to its generic nature, this method can be applied in any fast motion estimation methods although it is especially effective in the full search motion estimation method.

As one important component in H.264 video encoder, interpolation of half and quarter pixels is also computational intensive. Compared to integer pixel motion estimation, "finer" interpolation provides better block match. However, this good motion compensation performance is obtained at the expense of increased complexity. Based on our previous work, this paper presents an improved fast and adaptive interpolation method that further reduces the complexity of video encoding process. Experimental results on typical video sequences demonstrate that the proposed method is able to increase encoder speed ranging from 10% to 22% compared with our previous work without any PSNR loss or bit rate increase.

A new scheme is proposed in this paper for OFDMA (orthogonal frequency division multiple access) downlink resource allocation. It seeks to maximize the total system capacity subject to constraints on total available power, BER and rate proportions. Future communication systems should take both capacity and fairness into account together. Therefore, the new scheme has a novel step to rearrange the subcarriers among users with least capacity loss and most fairness gain to achieve a certain tradeoff between enhanced rate (Hui and Zhou, 2006) and proportional fairness (Shen et al., 2003). This tradeoff can be easily controlled by the introduced tradeoff-factor (TF). Simulation results show that this new capacity and fairness controllable scheme may be suitable for future quality of service (QoS) adjustable OFDMA systems.

Adaptive subcarrier-and-bit allocation in orthogonal frequency division multiplexing (OFDM) systems can significantly improve spectral efficiency and reduce power consumption. The combinatorial optimization on the power, bit and subcarrier allocation in multiclass multiuser OFDM systems normally involved nonlinear objective functions and constraints, resulting in highly complex and computation intensive optimization problems. In this paper, we propose a method to linearize and simplify the optimization problem formulated on subcarrier-and-bit allocation in multiclass multiuser OFDM systems. With a drastically reduced complexity, the converted linear optimization problem can be solved in close to real time and the solutions found are optimal, as no relaxation is made during the linearization. Different approximate relationships between symbol error rate (SER) and bit error rate (BER) are also used to investigate the calculation on the required transmit power.

In this paper, we proposed a new multicarrier, direct sequence code division multiple access, the so-called premulticoded MC-DS-CDMA system. In this scheme, the basic MC-DS-CDMA system is augmented with a precoder and a multicode encoder at the transmitter. The multicode encoder is an inverse Walsh-Hadamard transform (IWHT) unit, while the precoder is a recently suggested ldquoconstant amplituderdquo encoder that guarantees that the IWHT output stream has binary level. While IWHT, multicarrier (MC) transmission, and direct sequence spreading provide us with frequency diversity, the constant amplitude precoder ensures that the transmitted waveform is spread BPSK signal, minimizing peak to average-power ratio (PAPR) problem. The multicode and (MC) combination also addresses the need for flexible high data rate services in present day wireless communications. In the receiver, after matched filtering or correlation demodulation for each subcarrier and WHT operation, a detection strategy based on minimum-mean-square-error (MMSE) criterion is used. We have derived an upper bound for the bit error rate (BER) performance in Nakagami fading channel. Numerical results are compared with those obtained from a single carrier DS-CDMA system (SC-DS-CDMA) and a MC-DS-CDMA scheme without the precoding and IWHT units. We show that our proposed MC-DS-CDMA scheme has a much better BER performance compared with both of these schemes.

The performance of frequency hopping (FH) systems with M-ary frequency shift keying (MFSK) modulation may be significantly degraded in the presence of a follower jammer. In this paper, we introduce a signal model that takes into consideration the effect of a follower jammer explicitly, and then propose a maximum likelihood (ML) based interference cancellation and detection scheme for slow FH/MFSK systems. Based on using a simple 2-element antenna array, the scheme attempts to eliminate the unknown interfering follower jamming signal, and performs ML detection in a single integrated operation. Furthermore, an approximate expression of symbol error rate (SER) of the ML-based detection scheme is also derived in BFSK signaling. Simulation results show that the proposed scheme is able to remove jamming and outperform beamformers in the presence of a follower partial-band jammer

In this paper, multiple access interference (MAI) suppression in synchronous interleaved orthogonal frequency division multiple access (OFDMA) system is investigated. Based on the signature vector formulated for each user, a new user detector is proposed. It is shown that when matched filtering is employed to suppress MAI in the proposed detector, under imperfect power control, strong users can severely degrade the performance of its neighboring users. To improve the system performance, we derive a subspace zero-forcing and a subspace MMSE MAI suppression scheme for the proposed detector. We show that only the knowledge of the desired user's CFO is required in MAI suppression and the system can be rendered almost MAI-free provided that the CFO estimation is accurate enough.

This paper presents Advanced Audio Zip (AAZ), a fine grained scalable to lossless (SLS) audio coder that has recently been adopted as the reference model for MPEG-4 audio SLS work. AAZ integrates the functionalities of high-compression perceptual audio coding, fine granular scalable audio coding, and lossless audio coding in a single framework, and simultaneously provides backward compatibility to MPEG-4 Advanced Audio Coding (AAC). AAZ provides the fine granular bit-rate scalability from lossy to lossless coding, and such a scalability is achieved in a perceptually meaningful way, i.e., better perceptual quality at higher bit-rates. Despite its abundant functionalities, AAZ only introduces negligible overhead in terms of lossless compression performance compared with a nonscalable, lossless only audio coder. As a result, AAZ provides a universal yet efficient solution for digital audio applications such as audio archiving, network audio streaming, portable audio playing, and music downloading which were previously catered for by several different audio coding technologies, and eliminates the need for any transcoding system to facilitate sharing of digital audio contents across these application domains.

This paper derives a unified representation of the Cramer-Rao lower bounds (CRLBs) for semi-blind maximum-likelihood (ML) channel estimation in time-hopping (TH) ultra-wideband systems with pulse amplitude and position modulation, pulse amplitude modulation and pulse position modulation (PPM). These bounds subsume the existing results for both pilot-based and blind ML channel estimation in TH-PPM as special cases. A numerical example is used to illustrate the effects of different data modulation formats and system parameters on the CRLBs derived

In this paper, a joint symbol timing and frequency-offset estimator based on the maximum likelihood (ML) criterion, is derived for OFDM systems over frequency selective fading channels. .We use the linear minimum mean square estimation technique (LMMSE) to estimate the nuisance parameters such as transmitted signals and fading gains, and arrive at an algorithm to jointly estimate symbol timing and frequency offset. Simulation results show that our proposed symbol timing estimator performs better than the estimator reported by van de Beek et al (1997), while our proposed frequency offset estimator achieves a lower MSE at higher SNR (/spl ges/ 15 dB). Simulation results also show that our proposed symbol timing and frequency offset estimator has better MSE performance than the cyclostationarity-based estimator studied in T. Pollet and M. Moeneclaey (1995) at higher SNR (for example 15 dB). However, in terms of probability of correct timing synchronization and MSE, our symbol timing estimator achieves roughly the same performance as the estimator by Lee et al (2002), although a different approach is used in our estimator. Our proposed estimator is next modified to two practical ones independent of SNR, furthermore, one of which needs only the knowledge of the maximum delay spread to perform estimation. We also extend our proposed estimator to applications where multiple OFDM symbols can be observed to perform synchronization.

The recently introduced MPEG standard for lossless audio coding, MPEG-4 Audio Scalable to Lossless (SLS) coding technology, provides a universal audio format that integrates the functionalities of lossy audio coding, lossless audio coding and fine granular scalable audio coding in a single framework. We propose two coding methods that improve the coding efficiency of SLS, namely, a context-based arithmetic code (CBAC) method and a low energy mode code method. These two coding methods work harmonically with the current SLS framework and preserve all its desirable features, such as fine granular scalability, while successfully improving its lossless compression ratio performance.

The new video coding standard, H.264/MPEG-4 AVC, uses variable block sizes ranging from 4/spl times/4 to 16/spl times/16 in interframe coding. This new feature has achieved significant coding gain compared to coding a macroblock (MB) using fixed block size. However, this feature results in extremely high computational complexity when brute force rate distortion optimization (RDO) algorithm is used. This paper proposes a fast intermode decision algorithm to decide the best mode in intercoding. It makes use of the spatial homogeneity and the temporal stationarity characteristics of video objects. Specifically, spatial homogeneity of a MB is decided based on the MB's edge intensity, and temporal stationarity is decided by the difference of the current MB and it colocated counterpart in the reference frame. Based on the homogeneity and stationarity of the video objects, only a small number of intermodes are selected in the RDO process. The experimental results show that the fast intermode decision algorithm is able to reduce on the average 30% encoding time, with a negligible peak signal-to-noise ratio loss of 0.03 dB or, equivalently, a bit rate increment of 0.6%.

We propose a new method for DOA estimation in the presence of multipath propagation and mutual coupling (MC) for a frequency hopping (FH) system. We take MC and multipath propagation into account, and derive a maximum likelihood (ML) estimator for both MC matrix and DOA estimation. We then formulate an iterative alternating minimization (AM) algorithm for finding the MC and DOA parameters in an alternate manner. Simulation results illustrating the performance of the algorithm and comparison with the Cramer-Rao bound (CRB) are also presented.

In this paper, we deal with the problem of document image rectification from images captured by digital cameras. The improvement on the resolution of digital camera sensors has brought more and more applications for non-contact text capture. Unfortunately, perspective distortion coupled with resulting images makes it harder to properly identify the contents of captured texts using the traditional optical character recognition (OCR) system. We propose in this work a new technique, which is capable of removing distortion and recovering the fronto-parallel view of text with a single image. Different from reported approaches in the literature, the image rectification is carried out using character boundary and tip point, which are extracted from character strokes based on multiple fuzzy sets and morphological operators. The algorithm needs neither camera calibration nor high-contrast document boundary. Experimental results show our rectification process is fast and robust.

The modified discrete cosine transform (MDCT) has been widely used in many transform audio coding algorithms such as MPEG-1/2 layer III (mp3), MPEG-2/4 AAC, Dolby AC2/AC3, and numerous experimental audio coding algorithms. In this paper, we study the probabilistic distribution properties of the MDCT coefficient for audio signals. It is shown that the generalized Gaussian function with distribution parameter r=0.5 or r=1 (Laplacian) provides a good approximation to the distributions of MDCT coefficients for a variety of audio signals. Results from our study also show that although the distribution of these coefficients is not strictly Laplacian, the divergence between them is in fact very small. Therefore, it leads to only marginal redundancy if these coefficients are simply coded with some low-complexity codes designed for Laplacian sources.

Audio-visual speech recognition employing both acoustic and visual speech information is a novel extension of acoustic speech recognition and it significantly improves the recognition accuracy in noisy environments. Although various audio-visual speech-recognition systems have been developed, a rigorous and detailed comparison of the potential geometric visual features from speakers' faces is essential. Thus, in this paper the geometric visual features are compared and analyzed rigorously for their importance in audio-visual speech recognition. Experimental results show that among the geometric visual features analyzed, lip vertical aperture is the most relevant; and the visual feature vector formed by vertical and horizontal lip apertures and the first-order derivative of the lip corner angle leads to the best recognition results. Speech signals are modeled by hidden Markov models (HMMs) and using the optimized HMMs and geometric visual features the accuracy of acoustic-only, visual-only, and audio-visual speech recognition methods are compared. The audio-visual speech recognition scheme has a much improved recognition accuracy compared to acoustic-only and visual-only speech recognition especially at high noise levels. The experimental results showed that a set of as few as three labial geometric features are sufficient to improve the recognition rate by as much as 20% (from 62%, with acoustic-only information, to 82%, with audio-visual information at a signal-to-noise ratio of 0 dB).

In the future cellular mobile systems, the steadily growing mobile subscriber community and their demand for diversity of service place great challenge on the bandwidth utilization, especially in the wireless network part, as radio spectrum is a limited resource. Carefully planned radio usage is critical for both system capacity and service quality. Current research work in literature cares about two aspects of the service provision capability of cellular networks. One is capacity related that emphasizes the user admission capability and the other is service quality related that targets the connection continuity. However, actually achievable user accommodation capability is a cooperative result of both aspects. This paper mathematically reveals the impact of handoff protection, which is introduced to enhance connection robustness, on the capacity of cellular mobile systems. We first extract the basic mobility characteristics from the real world cellular environment to establish an ideal traffic model. Then a Markov approach is proposed to analyze the correlation between the user admission capability and the channel reservation, which is one strategy for handoff protection, and furthermore to answer the question how the user accommodation capability is affected by channel reservation in this ideal model. Simulation outcomes are provided for the creditability verification on the theoretical results. We find that system capacity and service quality are two conflicting objectives and tradeoff is inevitable. Finally, a dynamic channel reservation scheme is described to provide a mechanism for the tradeoff coordination between system revenue and service quality.

We introduce a PDA-based live video streaming system on GPRS network based on MPEG-4 video compression standard. Due to the limited computational resources of PDA, all the key modules of MPEG-4 codec are efficiently implemented and optimized such as multithreading, buffer design, wireless communication, encoder and decoder. Several novel techniques are developed in the coding, streaming as well as the post- processing stages of the system.

This paper presents advanced audio zip (AAZ), an audio codec that provides the fine granular bit-rate scalability from lossy to lossless coding. Perceptually embedded coding principle is employed in AAZ to provide lossy reconstruction with optimal perceptual quality at intermediate bit-rates. AAZ also provides the backward compatibility where the lossless bit-stream embeds a compliant MPEG-4 AAC bit-stream.

Personal area networks (PANs) are expected to play an important role in future mobile ad-hoc wireless communications and information systems. This paper proposes an UWB-based alternate PHV layer for 802.15.4 low rate wireless personal area network (LR-WPAN) system with the intention to enhance such WPAN system with better location positioning capability. The proposed UWB transceiver operates in the first 500 MHz band, 3.1 - 3.6 GHz, within the FCC allocated UWB band. Original 802.15.4 code position modulation scheme is slightly modified to 2-PPM to cater UWB transmission. A simple peak detector based receiver is employed to sample and decode the transmitted symbol. Intensive simulation over IEEE 801.15.3a UWB channel models showed that the proposed UWB based alternate PHY layer can support 802.15.4 250 kbps 16-byte packet transmission over WPAN range of 5 m with 10% PER. It is also illustrated that, in general, PER increases over longer communication range and over longer channel delay spread.

This paper proposes a cascaded RLS-LMS predictor for lossless audio coding. In this proposed predictor, a high-order LMS predictor is employed to model the ample tonal and harmonic components of the audio signal for optimal prediction gain performance. To solve the slow convergence problem of the LMS algorithm with colored inputs, a low-order RLS predictor is cascaded prior to the LMS predictor to remove the spectral tilt of the audio signal. This cascaded RLS-LMS structure effectively mitigates the slow convergence problem of the LMS algorithm and provides superior prediction gain performance compared with the conventional LMS predictor, resulting in a better overall compression performance.

A theoretical analysis has been made to compare the capacity performances of transmit diversity and beamforming in the downlink of direct-sequence code-division multiple-access (DS-CDMA) system. The performances of selection transmit diversity (STD), beamforming (BF), and beam selection transmit diversity (BSTD), which is a hybrid scheme of STD and BF, have been studied at system level with capacity as the performance index. Both fading and interference are jointly considered in multipath propagation environment. It was found that the comparison result strongly relates to the availability of path diversity. The system capacity is maximized by the BF in the presence of path diversity, otherwise it is the BSTD that achieves the best overall capacity performance.

A new multi-carrier, direct sequence code division multiple access (MC-DS-CDMA) system with multi-code is proposed. Transmitted data are first serial-to-parallel converted to D parallel data streams and a pre-coding with a rate of D/M is followed. Next, these pre-coded parallel streams are multiplexed by an inverse Walsh-Hadamard transform (IWHT) (M/spl times/M) and M super-bit streams are formed. Due to using the pre-coding, the super-bit streams will be two-level symbol streams. Then, the super-bits are once more serial-to-parallel converted to form M parallel super-bit sub-streams. Finally, all the M parallel super-bit streams are direct sequence spread spectrum (DS-SS) modulated and transmitted using M orthogonal carriers. In the receiver, after demodulation and WHT operation, the strategy of minimum mean square error (MMSE) is used for detection. Based on a correlator receiver for each carrier, its upper bound of bit error rate (BER) performance over Nakagami fading channel is analyzed. Numerical results are compared with those of a single carrier DS-CDMA system (SC-DS-CDMA). It is shown that the proposed MC-DS-CDMA scheme has much better BER performance compared to the SC-DS-CDMA system.

A novel audio coding algorithm is proposed where the warped-linear prediction (WLP) technique is employed to construct a perceptual pre- and post-filter for subband audio coding. A modified signal-to-mask ratio (SMR) calculation is given for subband coding of the WLP residuals of audio signals. The concept of perceptual entropy (PE) is extended to subband coding, resulting in the subband perceptual entropy (SPE), which gives a short time estimate of the lowest possible bit rate for transparent subband audio coding. Two WLP models with frequency responses approximating the spectral shape of the masking threshold are investigated and it is found that the residual signals of both models contain less SPE compared with that of the original audio signals. Subjective tests show that the proposed audio codec operating at 56 kbps has a perceptual quality comparable to MPEG-1 audio Layer II operating at 64 kbps.

In this paper we address the problem of scheduling a large size divisible load on a hypercube cluster of processors. Unlike in earlier studies in the divisible load theory (DLT) literature, here, we assume that the processors have finite-size buffers. Further we impose constraints on the extent to which the load can be divided, referred to as granularity constraint. We first present the closed-form solutions for the case with infinite-size buffers. For the case with load granularity constraint, we propose a simple algorithm to find the sub-optimal solution and then we analyze the case when these buffer are of finite size. For this case, we present an elegant strategy, referred to as incremental balancing strategy (IBS), to obtain an optimal load distribution. Based on the rigorous mathematical analysis, a number of interesting and useful properties exhibited by the algorithm are proven. Numerical examples are presented for the ease of understanding.

We discuss a type of hidden Markov model (HMM) based on fuzzy sets and fuzzy integral theory which generalizes the classical stochastic HMM. The Choquet integral is used as a fuzzy integral which relaxes one of the two independence assumptions that we had with the classical HMM. We apply this new model to speech recognition and compare the performance with the classical HMM. In this research, the main innovation is that this new generalized fuzzy HMM is applied for the first time to speech recognition. Due to the fuzziness of the model, an interesting gain can be observed in terms of a lower computation time.

The use of soft computing for virtual object implementation is a novel technique that has received a lot of attention in the last few years. This field is very useful especially for applications related to computer games and virtual environments. One of the application areas is the use of real time lip tracking information for the implementation of the virtual lips in animated characters. The increased computing power of computers has made it possible to do real time lip tracking. This real time lip tracking information can be used to implement and control a virtual lip. On the other hand, the use of soft computing to represent the real time lip parameters enables us to have a more robust and flexible system which can compensate for the potential errors of lip tracking. In this paper we present a system which generates real time lip information for soft computing based virtual animated lips.

In a spectrally overlaid multi-band CDMA (SOM-CDMA) system, wide-band, medium-band and narrow-band CDMA subsystems with different spreading bandwidths are spectrally overlaid with each other. This system architecture has an advantage that different traffic types can exist together within a given band of spectrum, without having to allocate a separate spectrum band for each type of traffic. However, spectrally overlaid subsystems may also induce more interference on particular reference mobiles. Under this scenario, to study the effect of interference on the system capacity becomes necessary in order to ensure that the system can provide the required quality of service (QoS) for each type of service. Previous works on SOM-CDMA systems focused only on interference from mobiles in an isolated cell. In this paper, the capacity of SOM-CDMA cellular system in terms of aggregate data rate, by taking both inter-cell and intra-cell interferences into consideration, is investigated. Our results show that under the standard path loss model and ideal power control assumptions, the maximum aggregate data rate of SOM-CDMA systems will be degraded to approximately 50% of the isolated cell case. We also observed that the system capacity increases with the cell radius, while below a certain radius, spectral overlay becomes impractical. The bound of maximum mobiles admitted in SOM-CDMA with given SNR requirement and data rate of each frequency band can be a guide to system designers who design the call admission scheme.

Multi-Protocol Label Switching is a technology that combines the simplicity of IP routing with the high-speed switching of ATM. Mobile IP is a protocol that allows users to move around and yet maintain continuous IP network connectivity. In this paper, we propose a scheme to integrate the Mobile IP and MPLS protocols. The integration improves the scalability of the Mobile IP data forwarding process by leveraging on the features of MPLS which are fast switching, small state maintenance and high scalability. In addition, we have removed the need for IP-in-IP tunneling from HA to FA under this scheme. This paper covers some issues regarding Mobile IP scalability and also defines the signaling and control mechanisms required to integrate MPLS and Mobile IP. Keywords---Multi-protocol Label Switching (MPLS), Label Distribution Protocol (LDP), Mobile IP (MIP) I.

This paper investigates quality-of-service (QoS) performance of voice services in hybrid fiber/coax (HFC) networks. We focus on the upstream channel and the performance of the system is evaluated in terms of voice packet loss probability. Two different classes of voice services are considered. For voice without speech activity detector (SAD), there is no packet dropping and hence the analysis is trivial. For voice with SAD, we develop an elaborate discrete-time Markov chain model to obtain the voice packet dropping probability. We also describe the voice traffic models and the performance measures, and include an overview of the media access (MAC) protocol. Numerical examples are presented to demonstrate the performance the system.

Affine motion model is widely used in motion segmentation. Accurately estimating and evaluating affine parameters are two key problems in such kind of approaches. This paper tries to address these issues by presenting an effective video object segmentation method based on affine motion similarity measure. The image is firstly segmented into irregularly-shaped intensity homogenous regions. Then, relatively reliable affine parameters for each region are estimated by a robust motion estimator according to the individual coordinate system of each region. Finally, a new motion similarity measure and merge process is applied to obtain meaningful objects. Experimental results demonstrate the effectiveness of the proposed method.

Bimodal speech recognition is a robust technique for automated speech analysis, and has received a lot of attention in the last few decades. In this paper, we analyze the effect of the HMM models on the performance of the bimodal speech recognizer, present a comparative analysis of the different HMM models that can be used in bimodal speech recognition, and finally propose a novel model, which has been experimentally verified to perform better than others. One of the unique characteristic of our HMM model is the novel fusion strategy of the acoustic and the visual features, that takes into account the different sampling rates of these two signals. Compared to audio only, the bimodal speech recognition scheme has a much more improved recognition accuracy, especially in presence of noise.

This paper proposes a novel scheduling discipline for service differentiation in multi-service networks, which is referred to as the probabilistic priority (PP) discipline. The PP is based on the strict priority (SP) discipline with the difference that each priority class is assigned a parameter. The parameter determines the probability with which its corresponding queue is served when it is polled by the server. Service differentiation as well as fairness among traffic classes can be achieved in PP by setting the assigned parameters properly. In addition, PP can be easily reduced to the ordinary SP or to the reverse SP. Moreover, PP can provide service segregation among groups of traffic classes while at the same time provide service differentiation among classes within each group.

Audio-visual speech recognition is a novel extension of acoustic speech recognition and has received a lot of attention in the last few decades. The main motivation behind bimodal speech recognition is the bimodal characteristics of speech perception and production systems of human beings. The effect of the modeling parameters of hidden Markov models (HMM) on the recognition accuracy of the bimodal speech recognizer is analyzed, a comparative analysis of the different HMMs that can be used in bimodal speech recognition is presented, and finally a novel model, which has been experimentally verified to perform better than others is proposed. Also, the geometric visual features are compared and analyzed for their importance in bimodal speech recognition. One of the unique characteristics of our bimodal speech recognition system is the novel fusion strategy of the acoustic and the visual features, which takes into account the different sampling rates of these two signals. Compared to acoustic only, the audio-visual speech recognition scheme has a much more improved recognition accuracy, especially in the presence of noise.

With the rapid proliferation of Internet technologies, accessing and operating engineering instruments remotely anytime anywhere is fast becoming a reality. This paper presents a new Web-based virtual laboratory on a frequency modulation experiment for the teaching of an undergraduate course on communication principles in the National University of Singapore (NUS). The laboratory requires only a common Web browser to access and incorporate schemes for reducing data traffic and authenticating users. It enables students to have a natural hands-on experience of using an expensive spectrum analyzer on a one-to-one basis and provides a solution for distant engineering education. The system uses a double client-server structure where access to the experiment is via two rounds of client-server processing. The virtual laboratory can be accessed at the Web site http://vlab.ee.nus.edu.sg/vlab/freqmod/index.html.

We present the theory of modified nonparametric regression for estimating the 3D face structure of a human from a monocular image sequence. In the preprocessing stage, the face region is segmented from the background using both color and motion information, by using a hierarchical block motion estimation method. By using the affine camera projection geometry, and a given choice of an image frame pair in the sequence, we adopt the KvD model to express the depth at each point on the face region as a function of the unknown out-of-plane rotation, and some measurable quantities computed directly from the optical flow. This is repeated for multiple image pairs (keeping one fixed image frame which we formally call the "base" image, and choosing another frame from the sequence). The true depth map is next estimated from these equations using a modified nonparametric regression technique, and this forms the core contribution of this paper. We conducted experiments on various image sequences to verify the effectiveness of the technique, and propose to extend it for photorealistic modeling of arbitrary (non-face) objects from image sequences.

This paper proposes a new handover management scheme for low Earth orbit (LEO) satellite mobile communication systems-elastic channel locking (ECL) scheme. Based on the feature that the mobile terminal's entry into and departure from a cell is predictable in LEO systems, this scheme provides a flexible way to control quality of service in terms of blocking probability and handover failure probability by adjusting the time to send out channel reservation request during the handover process. A mathematical model is established to analyze this scheme. To gain a comprehensive insight into this scheme, call incompletion probability is employed to evaluate the traffic capacity of a communication system. It is shown that ECL can maximize the channel utilization and enable the system to accommodate the most number of customers.

The medium access control (MAC) protocol through which mobile stations can share a common broadcast channel is essential in an ad-hoc network. Due to the existence of the hidden terminal problem, partially-connected network topology and lack of central administration, existing popular MAC protocols like IEEE 802.11 Distributed Foundation Wireless Medium Access Control (DFWMAC) may lead to "capture" effects which means that some stations grab the shared channel and other stations suffer from starvation. This is also known as the "fairness problem". This paper reviews some related work in the literature and proposes a general approach to address the problem. This paper borrows the idea of fair queueing from wireline networks and defines the "fairness index" for ad-hoc network to quantify the fairness, so that the goal of achieving fairness becomes equivalent to minimizing the fairness index. Then this paper proposes a different backoff scheme for IEEE 802.11 DFWMAC, instead of the original binary exponential backoff scheme. Simulation results show that the new backoff scheme can achieve far better fairness without loss of simplicity.

Future integrated services networks will need to provide quality of service (QoS) guarantees to multimedia applications. To ensure that the contracted QoS is sustained, it is not sufficient to just commit resources. QoS monitoring is required to detect and locate the degradation of QoS performance. In addition, the distribution of QoS, instead of simply end-to-end QoS, needs to be monitored. In QoS distribution monitoring, the distribution of QoS experienced by a real-time flow in different network segments is monitored. This paper presents a brief survey of current QoS monitoring-related mechanisms, followed by a discussion of the challenges involved in providing QoS distribution monitoring. Several approaches are then proposed to meet these challenges.

A new scheme is proposed for fusing multisensor images in which one image is regarded as the main image and the other the complementary; based on the evaluation of certain characteristics in the images. In effect, the scheme is used to fuse an image pair in which one image is superior to the other for interpretation in terms of higher resolution, better image quality, or having more recognizable features. Feature information is based on local statistical characteristics, which are extracted using the analysis of variance (ANOVA) method, in the framework of experimental designs. In effect, feature information from one image is used to influence the corresponding pixel values of the other image. The fused image leads to a better human and/or machine interpretation of the area of interest in the images.

Optimal distribution of divisible loads in bus networks is considered in this paper. The problem of minimizing the processing time is investigated by including all the overhead components that could penalize the performance of the system, in addition to the inherent communication and computation delays. These overheads are considered to be constant additive factors to the respective communication and computation components. Closed-form solution for the processing time is derived and the influence of overheads on the optimal processing time is analyzed. We derive a necessary and sufficient condition for the existence of the optimal processing time. We then study the effect of changing the load distribution sequence on the time performance. Through rigorous analysis, an optimal sequence to distribute the load among the processors is identified, whenever it exists. In case such an optimal sequence fails to exist, we present a greedy algorithm to obtain a suboptimal sequence based on some important properties of the overhead factors. Then, the effect of granularity of the data that is divisible is considered in the analysis for the case of homogeneous networks. An integer approximation algorithm capable of generating integer values of the load fractions in time O(m), where m is the number of processors in the network, is proposed. We then show that the upper bound on the suboptimal solution generated by our algorithm lies within a radius given by the sum of the computation and communication delays. Several numerical examples are presented to illustrate the concepts.

The 2-D CAP (carrierless amplitude and phase) modulation system has been described in several ATM Forum documents as a viable modulation scheme for realizing very high bit rates on UTP-3 (unshielded twisted pair) copper loops. Recently, Shalash and Parhi contributed a technique to extend the CAP modulation scheme to 3-D. This paper investigates the properties of the adaptive 3-D CAP digital receiver for VDSL in the face of quantization noise effects due to the ubiquitous ADC circuitry in the receiver. Our simulation results show that the original 3-D CAP receiver is fairly sensitive to the quantization noise. We will explain why and then contribute a solution which involves a re-design of the signature waveforms that offers much better reception properties at the 3-D CAP receiver.

The factorization properties of polynomials defined over finite integer rings and the associated direct sums are analyzed. The objective is to describe new computationally efficient algorithms for computing one and two dimensional convolution of data sequences defined over finite integer rings. The number theoretic transforms (NTTs) are a special case of the general theory presented.

This paper presents a new filtering scheme for splitting speech signals into subbands which results in no aliasing problem when the subband signals are sub-sampled and recombined. The highpass and lowpass filters are an odd-length complementary linear phase filter pair. As a consequence, the highpass filter output can be obtained by subtracting the lowpass filter output from the appropriately delayed input. This produces a factor-of-two reduction in the computation load when compared to the conventional even-length quadrature mirror filter. Another factor-of-two saving in computation load can be gained by using halfband lowpass filters whose every other coefficient is zero. Furthermore, the filters can be designed using well known minimax optimization techniques such as the Remez exchange algorithm and integer programming.

By expressing correlation functions in power series of bandwidth, this paper first studies the structure of the covariance matrix of the broadband tapped delay line array. The elegant structure of the associated eigenvalues is then deduced, giving (number of taps)/2 sets of eigenvalues, the eigenvalues ith set being proportional to (bandwidth)2(i-1). The increase in eigenvalue spread of using tapped delay line processing in adaptive arrays is then derived.