System Maintenance:
There may be intermittent impact on performance while updates are in progress. We apologize for the inconvenience.
By Topic

Signal Processing, IET

Issue 2 • Date June 2008

Filter Results

Displaying Results 1 - 16 of 16
  • Mathematics in signal processing [editorial]

    Publication Year: 2008 , Page(s): 47 - 48
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (82 KB)  

    First Page of the Article
    View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Modelling room transfer functions using the decimated Pade approximant

    Publication Year: 2008 , Page(s): 49 - 58
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (283 KB)  

    The numerical issues involved in modelling measured room transfer functions (RTFs) are examined. This is explored using a nonlinear parametric estimation technique known as the decimated Pade' approximant (DPA). The DPA combines the well-known methodology of Pade' rational polynomial approximation with beamspace windowing. This combination helps to overcome the severe numerical instabilities encountered in calculations with large data records. The aim is to accurately extract the parameters that reliably quantify the analytic structure of signals composed of decaying sinusoidal oscillations. DPA parameter estimation provides the ability to construct a high-resolution spectral estimate of such signals for either specific spectral regions or the entire Nyquist interval. As demonstrated in the authors' previous work (O'Sullivan and Cowan, 2006), this technique, developed in quantum chemistry, readily cross-fertilises to the field of acoustics, where it can fully reconstruct the complicated spectra of experimental RTFs. A noise-filtering technique using the removal of Froissart doublets to obtain an irreducible rational model is investigated. This noise filtering can be used to find an order of the parametric model that is inherent in the data. Additionally, an example is shown suggesting that such a process may be useful in room spatialisation problems. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Parametric modelling for single-channel blind dereverberation of speech from a moving speaker

    Publication Year: 2008 , Page(s): 59 - 74
    Cited by:  Papers (2)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (659 KB)  

    Single-channel blind dereverberation for the enhancement of speech acquired in acoustic environments is essential in applications where microphone arrays prove impractical. In many scenarios, the source-sensor geometry is not varying rapidly, but in most applications the geometry is subject to change, for example when a user wishes to move around a room. A previous model-based approach to blind dereverberation by representing the channel as a linear time-varying all-pole filter is extended, in which the parameters of the filter are modelled as a linear combination of known basis functions with unknown weightings. Moreover, an improved block-based time-varying autoregressive model is proposed for the speech signal, which aims to reflect the underlying signal statistics more accurately on both a local and global level. Given these parametric models, their coefficients are estimated using Bayesian inference, so that the channel estimate can then be used for dereverberation. An in-depth discussion is also presented about the applicability of these models to real speech and a real acoustic environment. Results are presented to demonstrate the performance of the Bayesian inference algorithms. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Dynamical hysteresis in communications: a volterra functional approach

    Publication Year: 2008 , Page(s): 75 - 86
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (620 KB)  

    A formalism to characterise nonlinear dynamical hysteresis is described for multi-channel input-output physical systems that can have multi-valued solutions. The formalism presented is based on an extension to the Volterra functional representation of nonlinear dynamical input-output processes, an extension that overcomes the single-valued limitation of both the Taylor and Volterra series expansions. One important attribute of the formalism is that the coefficients can be physically significant and another is that the response function values can be determined directly from noisy data thereby offering potential insight into the underlying physics of the observed phenomena. The estimated response function values are the empirical coefficients required by a phenomenological theory of the system and can be used to predict likely behaviour, and to design and precisely control improved systems. The response function values that characterise the multi-channel nonlinear and dynamical hysteresis behaviour are estimated using the simultaneous moment equation method. The coefficients that characterise the hysteretic behaviour are obtained by solving a tractable system of simultaneous moment equations that are generated by operating on a suitably truncated Volterra functional expansion. These simultaneous moment equations enable the unknown constant coefficient values to be determined from noisy multi-channel input-output data. In order to demonstrate the attributes of the formalism under controlled conditions, a numerical example is presented which illustrates how to accurately estimate the coefficients of multi-channel nonlinear dynamical hysteresis phenomena corrupted by additive noise. The problems of time-dependent coefficients and the analysis of real data are to be considered elsewhere. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • MIMO channel orthogonalisations applying universal eigenbasis

    Publication Year: 2008 , Page(s): 87 - 96
    Cited by:  Papers (5)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (289 KB)  

    Some mathematical fundamentals of the multiple input multiple output (MIMO) channel orthogonalisations using prolate spheroidal wave functions (PSWFs) as a universal basis are presented. In contrast to the Karhunen-Loeve expansion or singular value decomposition, the PSWF basis is -invariant- to the properties of the MIMO channel covariance matrix and provides an almost minimum set of eigenfunctions that allow the channel to be represented with a predefined accuracy. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Unified approach to adaptive filters and their performance

    Publication Year: 2008 , Page(s): 97 - 109
    Cited by:  Papers (8)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (296 KB)  

    A streamlined theory is presented for adaptive filters within which the major adaptive filter algorithms can be seen as special cases. The algorithm development part of the theory involves three ingredients: a preconditioned Wiener Hopf equation, its simplest possible iterative solution through the Richardson iteration, and an estimation strategy for the autocorrelation matrix, the cross-correlation vector and a preconditioning matrix. This results in a generalised adaptive filter in which intuitively plausible parameter selections give the major adaptive filter algorithms as special cases. This provides a setting where the similarities and differences between the many different adaptive filter algorithms are clearly and explicitly exposed. Based on the authors' generalised adaptive filter, expressions for the learning curve, the excess mean square error and the mean square coefficient deviation are developed. These are general performance results that are directly applicable to the major families of adaptive filter algorithms through the selection of a few parameters. Finally, the authors demonstrate through simulations that these results are useful in predicting adaptive filter performance. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Design of two-dimensional recursive filters by using Taguchi-based immune algorithm

    Publication Year: 2008 , Page(s): 110 - 117
    Cited by:  Papers (1)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (479 KB)  

    The Taguchi-based immune algorithm (TBIA), based on both the features of a biological immune system and the systematic reasoning ability of the Taguchi method, is applied in this work to solve the design problem of two-dimensional (2D) recursive digital filters. In the TBIA, the clonal proliferation within hypermutation for several antibody diversifications and the recombination by using the Taguchi method for the local search are integrated to improve the capabilities of exploration and exploitation. The systematic reasoning ability of the Taguchi method is implemented in the recombination operation to select the better antibody genes to achieve the potential recombination, and consequently enhance the TBIA. The design of the 2D filter is reduced to a constrained minimisation problem, the solution of which is achieved by the convergence of the presented TBIA. The computational experiments show that the presented TBIA approach can obtain better results than some previous design methods. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Reduced ROM-based architecture for sine/cosine generator

    Publication Year: 2008 , Page(s): 118 - 124
    Cited by:  Papers (2)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (261 KB)  

    A reduced ROM-based architecture blending the concepts of domain folding and angle recoding to implement the coordinate rotation digital computer (CORDIC) algorithm is proposed. Domain folding restricts the domain of the sine/cosine functions in [0, pi/8] instead of [0, 2pi]. The addition identities of the trigonometric functions are adopted to determine the range of sine/cosine functions mapped from the domain [0, pi/4] based on that restricted in the domain [0, pi/8]. The quarter-wave symmetry property is then applied to obtain the range mapped from the full domain [0, 2pi]. Applying the angle recoding process to the angle in the domain [0, pi/8] affords two benefits. One is a reduction of about 50% in the size of the ROM lookup table storing the information of the coarse angles, except for the fact that the number of stages (N) equals 3k+2 where k is an integer; the other benefit is a 1-bit improvement in the precision in the CORDIC implementation. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Joint transform-domain-translated algorithm for MPEG-2 to H.264 downscaling intraframe transcoding

    Publication Year: 2008 , Page(s): 125 - 132
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (471 KB)  

    A joint transform-domain-translated (JTT) algorithm, which can achieve fast MPEG-2 to H.264 downscaling intraframe transcoding, is proposed. The proposed JTT algorithm effectively combines the 8 x 8 inverse discrete cosine transform, the 2:1 downscaler and the 4 x 4 integer transform without involving any complex pixel processes in the spatial domain. The authors also suggest a fast realisation of the JTT algorithm to further reduce its computation. The proposed JTT-transcoding method saves a lot of computation compared with some of the existing approaches in transcoding processing. Experimental results show that the proposed JTT algorithm effectively converts the MPEG-2 intraframe to H.264 intraframe to achieve almost identical PSNR performance compared with the pixel-domain decode-downsize-encode transcoder, which involves MPEG-2 decoding, spatial-domain downsizing and H.264 encoding processes. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Error reduction technique for four-quadrant arctangent approximations

    Publication Year: 2008 , Page(s): 133 - 138
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (277 KB)  

    A general technique of error reduction to improve the accuracy of four-quadrant arctangent approximations through the Mobius transformation is proposed. Simple identities connecting the regular arctangent function and the four-quadrant arctangent function are derived and used to obtain novel four- quadrant arctangent approximations. The new approximations can provide improved accuracy at the expense of a moderate increase in computational cost. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Improved blind-spreading sequence estimation algorithm for direct sequence spread spectrum signals

    Publication Year: 2008 , Page(s): 139 - 146
    Cited by:  Papers (8)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (342 KB)  

    Direct sequence spread spectrum (DSSS) signals are now widely used for communications. DSSS transmitters use a spreading sequence to modulate the baseband signal before transmission. A receiver which does not know the spreading sequence cannot demodulate the signal. Burel and Bouder introduced an eigenanalysis-based blind-spreading sequence estimation algorithm, which performs well even when the received signal is far below the noise level. However, this algorithm does not applied to the long-code DSSS signals. An improved blind-spreading sequence estimation algorithm is presented. This algorithm is based on segmentation. The received signal is divided into K collections of temporal windows, from which K covariance matrices can be computed. The authors prove that K short-time segments of the spreading waveform can be recovered from these matrices using the eigenanalysis technique. Then, the spreading sequence can be reconstructed by concatenating these short-time segments. Simulations show that the proposed algorithm can provide a good estimation for long- or short-code DSSS signals in non-cooperative context, even with low signal-to-noise ratio. Furthermore, for short-code DSSS signals, the computational cost of the proposed algorithm is much lower than that of the original algorithm. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Excess MSE analysis of the concurrent constant modulus algorithm and soft decision-directed scheme for blind equalisation

    Publication Year: 2008 , Page(s): 147 - 155
    Cited by:  Papers (7)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (323 KB)  

    The concurrent constant modulus algorithm (CMA) and soft decision-directed (SDD) scheme (CCMA+SDD) blind equalisation achieves a considerable improvement in equalisation performance over the CMA for high-order quadrature amplitude modulation (QAM) channels. Because of its concurrent adaptive filter structure and the complexity in analysing the estimation error signal, the actual steady-state performance has largely been left uninvestigated. By introducing an equivalent blind equalisation algorithm, the steady-state excess mean square error (EMSE) of the CCMA+SDD algorithm in a noise-free environment is studied. Based on the energy preservation approach, two first-order approximation closed-form solutions for the EMSE are derived for real-valued and complex-valued cases, respectively. The experimental simulation results for 16- and 64-QAMs show that the proposed steady-state theoretical EMSE results are reasonably accurate. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Best linear unbiased estimator approach for time-of-arrival based localisation

    Publication Year: 2008 , Page(s): 156 - 162
    Cited by:  Papers (17)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (179 KB)  

    A common technique for source localisation is to utilise the time-of-arrival (TOA) measurements between the source and several spatially separated sensors. The TOA information defines a set of circular equations from which the source position can be calculated with the knowledge of the sensor positions. Apart from nonlinear optimisation, least squares calibration (LSC) and linear least squares (LLS) are two computationally simple positioning alternatives which reorganise the circular equations into a unique and non-unique set of linear equations, respectively. As the LSC and LLS algorithms employ standard least squares (LS), an obvious improvement is to utilise weighted LS estimation. In the paper, it is proved that the best linear unbiased estimator (BLUE) version of the LLS algorithm will give identical estimation performance as long as the linear equations correspond to the independent set. The equivalence of the BLUE-LLS approach and the BLUE variant of the LSC method is analysed. Simulation results are also included to show the comparative performance of the BLUE-LSC, BLUE-LLS, LSC, LLS and constrained weighted LSC methods with Crame-r-Rao lower bound. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Robust generalised Capon algorithm for estimating the angular parameters of multiple incoherently distributed sources

    Publication Year: 2008 , Page(s): 163 - 168
    Cited by:  Papers (2)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (163 KB)  

    The authors consider the problem of estimating the directions-of-arrival (DOAs) and angular spreads of incoherently distributed sources, using a uniform linear array. The proposed method is based on the generalised Capon principle and enables the estimation of the DOAs decoupled from that of the angular spreads of sources. Compared with the original generalised Capon algorithm, the proposed algorithm improves the robustness to the mismodelling of the spatial distribution of sources. In fact, this method does not need the a priori knowledge of the azimuthal power distribution of sources, at least for the most studied distributions in the literature which are the uniform distribution and the Gaussian distribution. Furthermore, it works even in the case where the different sources have different angular distribution shapes. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Rao-Blackwellised particle filter based trackbefore- detect algorithm

    Publication Year: 2008 , Page(s): 169 - 176
    Cited by:  Papers (4)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (344 KB)  

    A track-before-detect (TBD) algorithm based on the Rao-Blackwellised particle filter (RBPF) is proposed for over-the-horizon radar (OTHR) target detection and tracking. Since the unthresholded measurements are integrated over time in the TBD algorithm, it can potentially detect and track targets with a much lower signal-to-noise ratio than conventional methods. In order to take advantage of the TBD algorithm, both system dynamic and measurement models are defined for the application under consideration. Furthermore, by exploiting the linear sub-structure available in the models, an RBPF based TBD algorithm is derived in detail. The proposed algorithm can provide reduced variance for the estimates with linear sub-structure and a lower computational complexity than the standard particle filter. Simulation results show that the proposed algorithm is capable of detecting and tracking dim targets for OTHR efficiently. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Design of digital differentiator using difference formula and Richardson extrapolation

    Publication Year: 2008 , Page(s): 177 - 188
    Cited by:  Papers (5)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (254 KB)  

    The design of digital differentiators is investigated. First, the backward difference formula in numerical differentiation is applied to derive the transfer function of a digital differentiator. In this design, the Richardson extrapolation is used to generate high-accuracy results while using low-order formulas. Then, conventional Lagrange finite impulse response (FIR) and Thiran infinite impulse response (MR) allpass fractional delay filters are directly applied to implement the designed differentiator. Next, the proposed method is extended to design digital differentiators using central difference formula. Finally, several numerical examples are illustrated to demonstrate the effectiveness of this new design approach. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.

Aims & Scope

IET Signal Processing publishes novel contributions in signal processing.

Full Aims & Scope

Meet Our Editors

IET Research Journals
iet_spr@theiet.org