In this paper, we expand the idea of spreading the transmitted symbols in OFDM systems by unitary spreading matrices based on the rotated Hadamard or rotated Discrete Fourier Transform (DFT) matrices proposed in the literature to apply to Space-Time Coded Multiple-Input Multiple- Output OFDM (STC-MIMO-OFDM) systems. We refer the resulting systems to as STC-MIMO Block Spread OFDM (STC-MIMO-BOFDM) systems. In the proposed systems, a multi-dimensional diversity, including time, frequency, space and modulation diversities, can be used, resulting in better bit error performance in frequency selective Rayleigh fading channels compared to the conventional OFDM systems with or without STCs. Simulations carried out with the Alamouti code confirm the advantage of the proposed STC-MIMO-BOFDM systems.

A new technique for the blind separation of convolutive mixtures is proposed in this paper. Inspired by the works of Amari, Sabala , and Rahbar, we firstly start from the application of Kullback-Leibler divergence in frequency domain, and then we integrate Kullback-Leibler divergence over the whole frequency range of interest to yield a new objective function which turns out to be time-domain variable dependent. In other words, the objective function is derived in frequency domain which can be optimized with respect to time domain variables. The proposed technique has the advantages of frequency domain approaches and is suitable for very long mixing channels, but does not suffer from the local permutation problem as the separation is achieved in time-domain

Power efficient topologies in an ad-hoc network can reduce battery usage and increase the lifetime of a network. Topology control algorithms including a local minimum spanning tree (LMST), distributed relative neighbourhood graph (DRNG) and K-Neigh graphs are computed by using the location or the distance information of the network nodes. Inaccurate distance or location information can lead to disconnected topology graphs. Furthermore, a distance based topology graph may not necessarily be connected due to wireless signal attenuation and propagation characteristics. In a realistic scenario, a small link distance may correspond to a large transmission power as the signal may need to transverse through obstacles. Hence it is imperative to include the environment characteristics when generating power efficient connected topology graphs. In this paper we identify the topology construction procedures that may lead to disconnected network topologies. We propose a new procedure that can work in conjunction with the neighbour discovery protocol to provide a better connected and power efficient network topology. Simulations indicate that the proposed algorithm provides a significant improvement in the connectivity of power base DRNG graphs.

The paper proposes a new technique for blind source separation (BSS) in the subband domain using an extended lapped transform (ELT) decomposition for nonstationary, convolutively mixed signals. As identified by S. Araki et al. (see Proc. 4th Int. Symp. on Independent Component Analysis and Blind Signal Separation - ICA2003, p.499-504, 2003), the motivation for subband-based BSS is the drawback of frequency domain BSS when dealing with separating mixed speech signals over a few seconds resulting in few samples in individual frequency bins leading to poor separation performance. In the proposed approach, mixed signals are decomposed into subband components by an ELT and within each subband a time domain Newton BSS algorithm is employed based on the nonstationarity property of the input signals and the joint diagonalization of output correlation matrices with time varying second order statistics (SOS). This subband version is compared to a fullband version using the same BSS algorithm.

This paper addresses experimental measurements from an IEEE 802.11 ad hoc network testbed, which indicate a strong signal strength dependence in the ability of a hidden terminal to gain access to the radio channel. We present analytical results investigating the 'hidden terminal jamming' ability of the IEEE 802.11 DSSS physical layer. Results indicate that in a hidden terminal topology, the presence of an interfering transmission with a signal strength marginally greater than the transmission currently being received will result in an intolerable increase in BER, effectively jamming the ongoing transmission. These results confirm previous experimental measurements which show that after a number of MAC layer timeout/retransmission periods, the original (weaker) connection is effectively prevented from gaining access to the channel.

We investigate the performance of common capture models in terms of the fairness properties they reflect across contenting hidden connections. We propose a new capture model, message retraining, as a means of providing an accurate description of experimental data. Using two fairness indices we undertake a quantitative study of the accuracy with which each capture model is able to reflect experimental data. Standard capture models are shown to be unable to accurately reflect the fairness properties of empirical data. The message retraining capture model is shown to provide a good estimate of actual system performance in varying signal strength conditions.

Current label switching protocols can use routing, address, and address hierarchy information to group flows for cut-throughs that bypass IP forwarding. This paper examines a label switching solution that uses the IP version 6 (IPv6) address structure to classify and cut-through flows based on address hierarchy. The performance of this approach is examined using actual backbone traffic traces with associated hierarchical address information obtained from Internet address registries, routing arbiter databases and route servers. This hierarchical address information is used to map a hierarchical address structure over the packet level trace. We investigate the relationship between aggregation bit-mask size versus label switching performance. We show that aggregation greater than IPv6 destination site address does not significantly improve performance. Our trace driven simulation studies show that it is possible to reduce the network layer packet forwarding requirements to below 0.15% of total packets at boundary routers within the core network by using IPv6 address hierarchy linked label switching.

Whilst linear prediction is the cornerstone of most modern speech coders, few of these coders incorporate the perceptual characteristics of hearing into the calculation of the linear predictor coefficients (LPCs). This paper proposes a method of incorporating simultaneous masking into the calculation of the LPCs. This modification requires only a modest increase in computational complexity and results in the linear predictor removing more perceptually important information from the input speech signal. This results in a filter that better models the formants of the input speech spectrum. The net effect is that an improvement in quality is achieved for a given bit rate or alternately a bit rate reduction can be achieved while maintaining perceived quality. These results have been confirmed through subjective listening tests.

Optical switching will enable core Internet packet switching to scale with future transmission rate increases. Currently proposed optical ATM switches do not allow packet reassembly, which is necessary for packet level forwarding. This results in the requirement to create end to end ATM virtual connections for flows even if they contain only one packet. In electronically switched networks MPOA and MPLS allow both cell and packet level forwarding to overcome this problem. This paper examines the feasibility of implementing such protocols over an optically switched network. Two different architectures are examined: use of an adjunct electrical router; and native optical packet reassembly. An examination of the optical reassembly buffer requirements show that the use of MPLS will require significantly more buffering than MPOA.

We address issues with the performance of IEEE 802.11, when used in the adhoc mode, in the presence of hidden terminals. We present results illustrating the strong dependence of channel capture behavior on the SNR observed on contending hidden connections. Experimental work has illustrated that in a hidden terminal scenario, the connection having the strongest SNR is able to capture the channel, despite the use of the RTS-CTS-DATA-ACK 4-way handshake designed to alleviate this problem. Our results indicate that the near-far SNR problem may have a significant effect on the performance of an adhoc 802.11 network.

Integrated services IP networks are expected to provide a variety of services with differentiated QoS. This requires the implementation of mechanisms that can discriminate service classes in terms of QoS. The IETF has recently proposed a differentiated services (Diffserv) framework for provision of QoS. In this paper we analyse the performance of two Diffserv mechanisms: threshold dropping and priority scheduling in terms of packet loss and mean packet delay. A comparison of the two mechanisms is carried out with the requirement that both mechanisms provide the same level of packet loss for the preferred flow. This comparison extends the results reported in the literature for these two mechanisms. In particular, in this paper we determine the impact of buffer threshold and buffer size on packet loss and mean packet delay in these mechanisms.

Differentiated service (DiffServ) has been proposed as an alternative for Integrated service. It aims to provide the same service to a group of flows that have similar quality of service requirements. Assured forwarding (AF) and expedited forwarding (EF) are two proposals for DiffServ provision. We present a performance analysis of an N drop-precedences threshold dropping (TD) queue, which is one of the proposed mechanisms for AF. In this analysis, traffic flows are assumed Poisson with exponentially distributed service time. We present simulation results that verify the analysis. This paper is an extension of the work attempted by Bolot et al. (see Proceedings of IEEE INFOCOM, 1999) and Sahu (see Umass CMPCSI Technical Report 99-09, University of Massachusetts) since it considers the general case with multiple classes of flow. We also show that the Poisson based analysis can be shown to hold for aggregation of bursty Markov sources in some cases and not to hold in others.

This paper investigates the use of fixed bandwidth reservation for multi-service mobile networks. In particular it extends previous fixed multi-service network results to the mobile scenario and presents analytical and simulation results for new and handover call blocking probabilities. It also investigates the performance sensitivity to different traffic load ratios as well as the cell dwell time distribution.

This correspondence uses a pitch synchronous wavelet transform (PSWT) as an alternative characteristic waveform decomposition method for the waveform interpolation (WI) paradigm. The proposed method has the benefit of providing additional scalability in quantization than the existing WI decomposition to meet desired quality requirements. The PSWT is implemented as a quadrature mirror filter bank and decomposes the characteristic waveform surface into a series of reduced time resolution surfaces. Efficient quantization of these surfaces is achieved by exploiting their perceptual importance and inherent transmission rate requirements. The multiresolution representation has the additional benefit of more flexible parameter quantization, allowing a more accurate description of perceptually important scales, especially at higher coding rates. The proposed PSWT-WI coder is very well suited to high quality speech storage applications.

For efficient coding of speech, it is desirable to separate the slowly and rapidly evolving spectral components to take advantage of their different perceptual qualities. We present a multi-level wavelet decomposition mechanism, using low delay FIR filters, applied to waveform interpolation (WI) coding. The technique overcomes the substantial delay problems encountered by Chong, Burnett, Chicharo and Thomson (see Proc. Int Conf. Acoustics, Speech, Sig. Processing, Seattle, USA, vol. 1., p.513-16, 1998) and identifies a preferred technique for the quantisation of the decomposed surfaces. The phase is shown to be particularly sensitive to the compounding of quantisation errors within the tree-structured transform. The proposed solution involves the use of variable dimension vector quantisation (VDVQ) on separately decomposed magnitude/phase surfaces. This approach provides for coarse or no phase quantisation while maintaining high speech quality. The techniques discussed may also be applied to other transforms and to the quantisation of surfaces in the standard waveform interpolation coder.

In this paper we investigate the spectral density of line spectral frequency (LSF) content in languages. The results show that the phonetic variation of languages is reflected in the LSF space. This leads to an alternative approach to the design of LSF quantisers. A trained LSF codebook, like the phonetic inventory of a language, is a static description of spectral behaviour of speech. As clear relationships exist between phonetic segments and LSFs, the structure of an LSF codebook can be analysed in terms of the phonetic segments. The new approach incorporates phonetic information into the structure of LSF codebooks through combining individual phonetic codebooks. The investigation leads to the conclusion that phonetic information can be usefully employed in codebook training in terms of perceptual performance and bit-rate reductions.

A new characteristic waveform decomposition method based on wavelets is proposed for the waveform interpolation (WI) paradigm. In WI, pitch-cycle waveforms are filtered in the evolution domain to decompose the signal into two waveform surfaces, one characterising voiced speech and a second representing unvoiced speech. The slow roll-off of FIR filters leads, however, to a significant inter-relationship between the decomposed surfaces. We present the pitch synchronous wavelet transform (PSWT) as an alternative decomposition mechanism. Filtering is again performed in the evolutionary waveform domain, producing characteristic surfaces at several resolutions. This multi-scale characterisation leads to more flexible quantisation of parameters, especially at higher rates than WI's 2.4 kb/s. FIR filters are replaced in the wavelet filter bank by causal, stable FIR filters which achieve significant delay reductions over their IIR counterparts. Furthermore, FIR filters track the dynamic aspects of the evolutionary surfaces faster, overcoming problems existing in the current WI decomposition.

This paper discusses how the strict hierarchical address structure of IP version 6 and high levels of IP route aggregation will effect the operation of popular label switching techniques. We predict that the introduction of IP version 6 will cause significant problems with control driven label switching proposals such as MPLS and Tag Switching. We propose an alternate label switching solution that uses information in the IP version 6 address structure to classify and cut-through flows based on destination site address. Our simulation studies shown that over 96% of packets will be switched by ATM bypassing network layer forwarding using this technique.

In this paper, we present an alternative characteristic waveform (CW) decomposition mechanism for the Waveform Interpolation (WI) paradigm based on the Pitch Synchronous Wavelet Transform (PSWT). In this technique, IIR filters replace the conventional FIR filters of the PSWT, offering computational and spectral magnitude performance advantages, in addition to significant delay reductions. Previously, the PSWT has only incorporated filter banks with slowly reacting FIR wavelet filters. While these filters possess the desirable properties of linear phase, and design simplicity, a large delay is incurred which increases exponentially with increasing resolution. The progression to IIR filter banks gives rise to a multi-resolution decomposition mechanism, beneficial for real-time applications, such as speech coding, where delay is an important issue.

In this paper we investigate an alternative approach to the design of low-bit rate (LBR) quantisation. This approach incorporates phonetic information into the structure of Line Spectral Frequency (LSF) codebooks. In prior work vector quantisation (VQ) has been used to quantise stochastic processes. Speech signals can, however, be described in terms of phonetic segments and linguistic rules. A trained LSF codebook, like the phonetic inventory of a language, is a static description of spectral behaviour of speech. As clear relationships exist between phonetic segments and LSFs the structure of an LSF codebook can be analysed in terms of the phonetic segments. The investigation leads to the conclusion that phonetic information can be usefully employed in codebook training in terms of perceptual performance and bit-rate reductions.

Least mean square (LMS)-based computation of block-based discrete orthogonal transforms has been extensively studied in literature. This paper establishes the relationship between the running discrete cosine transform II (DCT-II), discrete sine transform II (DST-II), and the adaptive LMS algorithm. From this analysis a new parallel structure for computing the running DCT-II and DST-II is proposed.

This paper considers the issue of the effect of languages and linguistic perception on low-rate speech coding. Current algorithms exploit the redundancies of speech but these redundancies are not common across all languages. Similarly speech coder evaluation techniques do not take into account the nuances of linguistic perception across languages. This paper illustrates some of the linguistic sensitivities experienced by low-rate coders and explores approaches to low-rate coder design. This is achieved through an evaluation of cross-language spectral distortion measures which account for specific linguistic peculiarities influencing linguistic perception.

This paper compares the parameter estimation accuracy of four adaptive algorithms for frequency estimation when applied to an IIR digital notch filter. All four algorithms were subjected to the same experimental conditions and the variance of parameter estimates are compared to the Cramer Rao Lower Bound. Results show that the RML yielded the most accurate parameter estimates although its computational burden is quite high. The AML produced good parameter estimates and it has the advantages of proven convergence properties as well as lower computational burden over the RML. For applications where the signal to noise ratio is moderate it is shown that the AGB algorithm may be suitable, particularly where minimal computational burden is desired.