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Signal Processing, IET

Issue 2 • Date April 2010

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Displaying Results 1 - 7 of 7
  • Signal detection performance in Rayleigh fading environments with a moving antenna

    Publication Year: 2010 , Page(s): 117 - 129
    Cited by:  Papers (5)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (747 KB)  

    The performances of a single-antenna handheld receiver in detecting a narrowband signal in a Rayleigh fading environment that is temporally static but decorrelates spatially are analysed. Of interest is comparing the detection performance of a static antenna with that of a moving antenna subject to constant processing time. It is shown that the net processing gain resulting from randomly moving the antenna relative to keeping it static can be large, namely over 11 dB in some cases, which is significant for numerous indoor applications. It is further demonstrated that, for a given utilisation scenario, there is an optimum number of spatial samples that maximise the processing gain advantage of the moving antenna. Generally, if the spatial trajectory of the antenna becomes too large, then the loss associated with the signal decorrelation dominates and undermines the gains achieved by the increased spatial diversity. Practical implementation issues including the sensitivity of the proposed method to trajectory estimation are investigated. An extensive set of measurements based on CDMA 2000 signals propagated from outdoor terrestrial base stations and captured in indoor multipath environments using static and moving antennas are utilised to experimentally substantiate these theoretical findings. View full abstract»

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  • New time--frequency distribution based on the polynomial Wigner--Ville distribution and L class of Wigner--Ville distribution

    Publication Year: 2010 , Page(s): 130 - 136
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (325 KB)  

    A new kind of time-frequency distribution based on the polynomial Wigner-Ville distribution (PWVD) and L class of Wigner-Ville distribution (LWVD) is presented in this study, and it is named as LPWVD. The LPWVD can be implemented based on the convolution of LWVD in frequency domain. Compared with the PWVD, the LPWVD is cross-term free for multi-component signals, and the time-frequency convergence is higher than the PWVD. The asymptotic statistical performance of LPWVD for estimating the instantaneous frequency (IF) of a polynomial phase signal (PPS) is studied. The simulated results demonstrate the validity of the LPWVD. View full abstract»

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  • Approach based on combination of vector neural networks for emitter identification

    Publication Year: 2010 , Page(s): 137 - 148
    Cited by:  Papers (1)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (415 KB)  

    To deal with the problem of emitter identification caused by the measurement uncertainty of emitter feature parameters, this study proposes a new identification algorithm based on combination of vector neural networks (CVNN), which is deduced from the backpropagation vector neural network and can realise the non-linear mapping between the interval-value input data and the interval-value output emitter types. The key idea of CVNN is to adopt a combination of multiple multi-input/single-output neural networks to construct an identification system; each of the networks can only realise the identification function between two emitter types. Through quantitative analysis, it can be concluded that the proposed algorithm requires less computational load in the training stage. A number of simulations are presented to demonstrate the identification capability of the CVNN algorithm for emitter signals with and without additive noise. Simulation results show that the proposed algorithm not only has better identification capability, but also is relatively more insensitive to noise. View full abstract»

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  • Least square identification of alias components of linear periodically time-varying systems and optimal training signal design

    Publication Year: 2010 , Page(s): 149 - 157
    Cited by:  Papers (6)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (374 KB)  

    A least-squares (LS) method for identifying alias components of discrete linear periodically time-varying (LPTV) systems is proposed. The authors apply a periodic input signal to a finite impulse response (FIR) - LPTV system and measure the noise-contaminated output. The output of this LPTV system has the same period as the input when the period of the input signal is a multiple of the period of the LPTV system. The authors show that the input and the output can be related by using the discrete Fourier transform. In the frequency domain, an LS method can be used to identify the alias components. A lower bound on the mean square error (MSE) of the estimated alias components is given for FIR-LPTV systems. The optimal training signal achieving this lower MSE bound is designed subsequently. The algorithm is extended to the identification of infinite impulse response (IIR) - LPTV systems as well. Simulation results show the accuracy of the estimation and the efficiency of the optimal training signal design. View full abstract»

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  • Fixed-point design of generalised comb filters: a statistical approach

    Publication Year: 2010 , Page(s): 158 - 167
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (531 KB)  

    This study is concerned with the problem of designing computationally efficient generalised comb (GC) filters. Basically, GC filters are anti-aliasing filters that guarantee superior performance in terms of selectivity and quantisation noise rejection compared to classical comb filters, when used as decimation filters in multistage architectures. Upon employing a partial polyphase (PP) architecture proposed in a companion study, the authors develop a sensitivity analysis in order to investigate the effects of the coefficients' quantisation on the frequency response of the designed filters. The authors show that the sensitivity of the filter response to errors in the coefficients is dependent on the particular split of the decimation factor between the two sub-filters constituting the PP architecture. The sensitivity analysis is then used for developing a fixed-point implementation of a sample filter from the class of GC filters, used as reference filter throughout the study. Finally, the authors present computer simulations in order to evaluate the performance of the designed fixed-point filters. View full abstract»

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  • Block filtered-s least mean square algorithm for active control of non-linear noise systems

    Publication Year: 2010 , Page(s): 168 - 180
    Cited by:  Papers (3)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (1110 KB)  

    In this study, the authors propose a block formulation of an algorithm, called the block-filtered-s LMS (BFSLMS) algorithm, for active control of non-linear noise processes for a multichannel setup. A reduced structure of the fast Fourier transform (FFT)-based BFSLMS-M (FBFSLMS-M) algorithm has also been studied. From this, the multichannel block filtered-x LMS (FBFXLMS-M) algorithm has been derived as a special case. The simulation results show that these algorithms have a matching performance with the already existing algorithms, with a relatively low computational complexity. A reduced structure delayless FBFSLMS algorithm for the multichannel case has also been developed which has a lesser computational complexity than its time-domain counterpart. Apart from this, it has no delay which, in general, is inherent in the block adaptive algorithms. View full abstract»

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  • Three-channel variable filter-bank for digital hearing aids

    Publication Year: 2010 , Page(s): 181 - 196
    Cited by:  Papers (4)
    Save to Project icon | Click to expandQuick Abstract | PDF file iconPDF (1052 KB)  

    This study proposes a three-channel (3-channel) variable filter-bank (VFB) that consists of variable lowpass, variable bandpass and variable highpass digital filters. The three variable digital filters are obtained from a normalised analog prototype Chebyshev type-I lowpass filter using analog frequency transformations along with a modified bilinear transformation. Since both the magnitudes (gains) and band edge frequencies of the three variable digital filters are independently adjustable, the 3-channel VFB is considerably flexible, and can be successfully applied for compensating various hearing loss patterns in digital hearing aids. Various audiograms have been used to verify that high-accuracy fittings can be achieved with low order variable filters. Moreover, the authors reveal and theoretically prove the numerator coefficient-symmetries of the variable lowpass, variable bandpass and variable highpass filters, and show that each variable filter requires only one multiplication for its numerator filtering operations, so the total number of multiplications can be significantly reduced. More specifically, only 11 multiplications and 14 additions are required in the whole 3-channel VFB. Therefore the 3-channel VFB has extremely simple structure and high tuning flexibility for hearing aids. View full abstract»

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