T.D. Abhayapala

Australian National University, Canberra, Australian Capital Territory, Australia

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Publications (226)201.98 Total impact

  • Source
    Mengqiu Zhang, Rodney A. Kennedy, Thushara D. Abhayapala, Wen Zhang
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    ABSTRACT: Principal component analysis (PCA) is known to be a powerful linear technique for data set dimensionality reduction. This paper focuses on revealing the essence of PCA to interpret the data, which is to identify the internal structure of the random process from a large experimental data set. We give an explanation of the PCA procedure performed on a generated data set to demonstrate the exact meaning of the dimensionality reduction. Especially, a method is proposed to precisely determine the number of significant principal components for a random process. Then, the internal structure of the random process can be modeled by analyzing the relation between the PCA results and the original data set. This is vital in the efficient random process modeling, which is finally applied to an application in HRTF Modeling.
    01/2010;
  • S.M.A. Salehin, T.D. Abhayapala
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    ABSTRACT: Photoacoustic imaging is a high resolution, biomedical imaging modality used for cancer detection and imaging biological tissues. Exact photoacoustic reconstruction methods were proposed for simple geometries in both time and frequency domain. However, for arbitrary geometries only time reversal methods were previously proposed. In this paper, we propose a frequency domain algorithm for photoacoustic image reconstruction given an arbitrary sensor geometry. The spatial source distribution is expanded using a 2D Fourier Bessel expansion and the proposed method estimates Fourier Bessel coefficients of this expansion using a least squares method at frequencies corresponding to the zeros of Bessel functions. The effectiveness of the proposed method was corroborated by numerical experiments.
    Signal Processing and Communication Systems (ICSPCS), 2010 4th International Conference on; 01/2010
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    ABSTRACT: A three-dimensional single-input single-output Mobile-to-Mobile wireless channel model is developed in this paper by considering the underlying physics of free space wave propagation. Based on this channel model, the temporal correlation function for a general three dimensional scattering environment is derived. The temporal correlation function is characterized by the joint angular power distribution at the transmitter and receiver antennas and the speed of transmitter and receiver antennas. Using a von Mises-Fisher distribution as the angular power distribution, the usefulness of the derived temporal correlation function is discussed.
    01/2010;
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    Y.J. Wu, T.D. Abhayapala
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    ABSTRACT: Any attempt to create multiple independent soundfields in separate zones over an extended region of open space results in unintended interference in a given zones from other zones. In this paper, we design spatial band stop filters to suppress interzone interference in the regions of interests and pass the desired soundfields with no distortion. This is achieved by using the higher order spatial harmonics of one zone to cancel the undesirable effects of the lower order harmonics of the same zone on the other zones. We illustrate the work by designing and simulating a 2D two-zone soundfield.
    Applications of Signal Processing to Audio and Acoustics, 2009. WASPAA '09. IEEE Workshop on; 11/2009
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    ABSTRACT: This paper proposes an efficient rate one space-frequency block code (SFBC) for multiple-input multiple-output orthogonal frequency division multiplexing (MIMO-OFDM) systems. The proposed SFBC incorporates concept of matched rotation precoding (MRP) to achieve full transmit diversity and optimal system performance for arbitrary number of transmit antennas, subcarrier interval and subcarrier grouping. The MRP exploits the inherent rotation property of SFBC and has relaxed restrictions on subcarrier interval and subcarrier grouping, making it ideal for adaptive/time varying systems. The lowerbound of the coding gain for MRP is derived and shown that it is useful when designing a SFBC for practical scenarios, e.g. when transmitters have only partial knowledge of power delay profile or when the power delay profile has only a few dominant delayed paths. Simulation results show that the MRP can achieve a similar or better performance than existing SFBCs.
    Communications, 2009. ICC '09. IEEE International Conference on; 07/2009
  • Source
    R. Iqbal, P. Sadeghi, T.D. Abhayapala
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    ABSTRACT: In this paper, we consider a wireless communication scenario in which the channel output is marginally Gaussian, but not jointly Gaussian. In particular, we study the joint probability distribution of channel outputs in correlated Rayleigh fading channels in response to constant power signaling, such as M-ary phase shift keying (MPSK). We show that the distribution of the channel output at any given sampling time is marginally Gaussian. However, the joint distribution of a sequence of channel outputs cannot be jointly Gaussian. A consequence of this result is that the information rates stated to be exact in two recent contributions, are strict upper bounds to the achievable data rates. We examine the tightness of these upper bounds by comparing them with the MPSK upper bound under perfect channel state information (CSI) assumption. We find that the CSI upper bound is considerably tighter in slow fading channels, high signal-to-noise ratios, and low-dimension (such as binary) PSK signaling.
    Computer Science and Information Engineering, 2009 WRI World Congress on; 05/2009
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    R. Iqbal, T.D. Abhayapala, T.A. Lamahewa
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    ABSTRACT: Clarke's classical model of mobile-radio reception assumes isotropic-rich scattering around the mobile receiver antenna. The assumption of isotropic scattering is valid only in limited circumstances. Here, a generalised Clarke model is developed, which is applicable to mobile-radio reception in general scattering environments. The authors give expressions for the autocorrelation and power spectral density of the channel fading process and demonstrate the generality of the model by applying it to different non-isotropic scattering scenarios. Using the generalised model, the effect of mobile direction of travel and the non-isotropicity on the statistics of the channel fading process is analysed. It is also shown that if the mobile direction of travel is equiprobable in all directions, a non-isotropic scattering environment on average is as good as an isotropic scattering environment.
    IET Communications 05/2009; · 0.72 Impact Factor
  • Source
    S. Kodituwakku, R.A. Kennedy, T.D. Abhayapala
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    ABSTRACT: A novel sinhc kernel, which generates a Kaiser window based time-frequency distribution for non-stationary signal analysis is introduced. The Kaiser distribution belongs to the Cohen class of time-frequency distributions, and satisfies desirable distribution properties. By controlling the shape of the Kaiser window, auto-term resolutions and cross-term magnitudes can be successfully traded.
    Electronics Letters 03/2009; · 1.04 Impact Factor
  • Source
    Y.J. Wu, T.D. Abhayapala
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    ABSTRACT: Reproduction of a soundfield is a fundamental problem in acoustic signal processing. A common approach is to use an array of loudspeakers to reproduce the desired field where the least-squares method is used to calculate the loudspeaker weights. However, the least-squares method involves matrix inversion which may lead to errors if the matrix is poorly conditioned. In this paper, we use the concept of theoretical continuous loudspeaker on a circle to derive the discrete loudspeaker aperture functions by avoiding matrix inversion. In addition, the aperture function obtained through continuous loudspeaker method reveals the underlying structure of the solution as a function of the desired soundfield, the loudspeaker positions, and the frequency. This concept can also be applied for the 3-D soundfield reproduction using spherical harmonics analysis with a spherical array. Results are verified through computer simulations.
    IEEE Transactions on Audio Speech and Language Processing 02/2009; · 1.68 Impact Factor
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    ABSTRACT: This paper presents a novel matched rotation precoding (MRP) scheme to design a rate one space-frequency block code (SFBC) and a multirate SFBC for MIMO-OFDM systems with limited feedback. The proposed rate one MRP and multirate MRP can always achieve full transmit diversity and optimal system performance for arbitrary number of antennas, subcarrier intervals, and subcarrier groupings, with limited channel knowledge required by the transmit antennas. The optimization process of the rate one MRP is simple and easily visualized so that the optimal rotation angle can be derived explicitly, or even intuitively for some cases. The multirate MRP has a complex optimization process, but it has a better spectral efficiency and provides a relatively smooth balance between system performance and transmission rate. Simulations show that the proposed SFBC with MRP can overcome the diversity loss for specific propagation scenarios, always improve the system performance, and demonstrate flexible performance with large performance gain. Therefore the proposed SFBCs with MRP demonstrate flexibility and feasibility so that it is more suitable for a practical MIMO-OFDM system with dynamic parameters.
    02/2009;
  • Source
    R.A. Kennedy, Wen Zhang, T.D. Abhayapala
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    ABSTRACT: The classical problem of extrapolation of a bandlimited signal from limited time domain data is revisited for signals defined on the sphere. That is, given limited or incomplete measurements of a low pass signal on the unit sphere, find the unique extrapolation to the complete unit sphere. Signals defined on the unit sphere arise in a number of applications, such as beam-patterns in azimuth and elevation and head related transfer functions. Our investigations explore the role of integral equation operators in characterizing the extrapolation problem which leads to an iterative algorithm analogous to that obtained in the time-frequency case.
    Signal Processing and Communication Systems, 2008. ICSPCS 2008. 2nd International Conference on; 01/2009
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    S.M.A. Salehin, T.D. Abhayapala
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    ABSTRACT: This paper presents a localization algorithm to detect lung sounds using a circular array of microphones. We use the natural basis functions of propagation waves in height invariant wavefields to form a spatial minimum variance (MV) problem in eigen space. We also derive a Nyquist criteria for localizing sources within a circular region. This Nyquist criteria shows that the radius of the region where sources can be localized is inversely proportional to the frequency of sound. The modified Nyquist criteria can be used for determining the number of sensors required for a given frequency range and radius of region for which sources need to be localized. The results are corroborated by computer simulations.
    Signal Processing and Communication Systems, 2008. ICSPCS 2008. 2nd International Conference on; 01/2009
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    Thushara D. Abhayapala, Aastha Gupta
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    ABSTRACT: We present a novel theory and design for constructing microphone arrays to extract spherical harmonic components from sound fields. The proposed non-spherical array structure provides a flexible and alternative design to the traditional spherical microphone arrays with lesser restriction on sensor locations. We use the properties of the associated Legendre functions and the spherical Bessel functions to develop a systematic approach to place circular microphone arrays in three dimensions for hybrid array geometries. As an illustration, we design and simulate a fifth order spherical harmonic decomposition array using 70 microphones to operate over a frequency band of an octave.
    Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2009, 19-24 April 2009, Taipei, Taiwan; 01/2009
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    Wen Zhang, Rodney A. Kennedy, Thushara D. Abhayapala
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    ABSTRACT: This paper introduces a continuous functional model for head-related transfer functions (HRTFs) in the horizontal auditory scene. The approach uses a separable representation consisting of a Fourier-Bessel series expansion for the spectral components and a conventional Fourier series expansion for the spatial components. Being independent of the data, these two sets of basis functions remain unchanged for all subjects and measurement setups. Hence, the model can transform an individualized HRTF to a subject specific set of coefficients. A continuous functional model is also developed in the time domain. We show the efficient model performance in approximating experimental measurements by using the HRTF measurements from a KEMAR manikin and the synthetic data from the spherical head model. The statistical results are determined from a 50-subject HRTF data set. We also corroborate the predictive capability of the proposed model. The model has near optimal performance, which can be ascertained by comparison with the standard principle component analysis (PCA) and discrete Karhunen-Loeve expansion (KLE) methods at the measurement points and for a given number of parameters.
    IEEE Transactions on Audio Speech and Language Processing 01/2009; 17:819-829. · 1.68 Impact Factor
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    Jian (Andrew) Zhang, Rodney A. Kennedy, Thushara D. Abhayapala
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    ABSTRACT: We investigate reduced-rank shift-invariant technique and its application for synchronization and channel identification in UWB systems. Shift-invariant techniques, such as ESPRIT and the matrix pencil method, have high resolution ability, but the associated high complexity makes them less attractive in real-time implementations. Aiming at reducing the complexity, we developed novel reduced-rank identification of principal components (RIPC) algorithms. These RIPC algorithms can automatically track the principal components and reduce the computational complexity significantly by transforming the generalized eigen-problem in an original high-dimensional space to a lower-dimensional space depending on the number of desired principal signals. We then investigate the application of the proposed RIPC algorithms for joint synchronization and channel estimation in UWB systems, where general correlator-based algorithms confront many limitations. Technical details, including sampling and the capture of synchronization delay, are provided. Experimental results show that the performance of the RIPC algorithms is only slightly inferior to the general full-rank algorithms.
    EURASIP Journal on Wireless Communications and Networking. 01/2009;
  • Source
    Yan Jennifer Wu, Thushara D. Abhayapala
    IEEE Transactions on Audio Speech and Language Processing 01/2009; 17:107-116. · 1.68 Impact Factor
  • Source
    Yan Jennifer Wu, Thushara D. Abhayapala
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    ABSTRACT: Spatial multizone soundfield reproduction is a difficult problem, which has many potential applications. This paper provides a framework to recreate 2D spatial multizone soundfields using an array of loudspeakers. We derive the desired global soundfield by translating individual desired soundfields to a single global co-ordinate system and applying appropriate angular window functions. We reveal some of the fundamental limits of 2D multizone soundfield reproduction. We show that the ability of multizone reproduction is dependent on (i) maximum radius of multizones, (ii) window length (size, and nature), and (iii) radial distance to the furthermost zone. We illustrate the framework by designing and simulating a two dimensional two zone soundfield.
    Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2009, 19-24 April 2009, Taipei, Taiwan; 01/2009
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    Sandun Kodituwakku, Thushara D Abhayapala, Rodney A Kennedy
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    ABSTRACT: We propose a novel algorithm for extracting atrial activity from single lead electrocardiogram (ECG) signal sustained with atrial fibrillation (AF), based on a short-time expansion of an orthogonal basis function set. The method preserves the time variation of spectral content of the underlying AF signal, thus time-frequency analysis of the AF signal can be successfully performed. The new method is compared to the standard average beat subtraction (ABS) method using synthetic AF sustained ECG data. The orthogonal basis expansion method has a higher correlation with the original AF signal compared to the ABS method for a range of signal to noise ratio (SNR) levels, and correlation is improved by 16% at an SNR of 0dB. Time-frequency analysis of the reconstructed AF signal based on Bessel distribution also shows the superiority of the orthogonal basis expansion method over ABS.
    Conference proceedings: ... Annual International Conference of the IEEE Engineering in Medicine and Biology Society. IEEE Engineering in Medicine and Biology Society. Conference 01/2009; 2009:1820-3.
  • Source
    Wen Zhang, Thushara D. Abhayapala, Rodney A. Kennedy, Ramani Duraiswami
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    ABSTRACT: This paper proposes a continuous HRTF representation in both 3D spatial and frequency domains. The method is based on the acoustic reciprocity principle and a modal expansion of the wave equation solution to represent the HRTF variations with different variables in separate basis functions. The derived spatial basis modes can achieve HRTF near-field and far-field representation in one formulation. The HRTF frequency components are expanded using Fourier Spherical Bessel series for compact representation. The proposed model can be used to reconstruct HRTFs at any arbitrary position in space and at any frequency point from a finite number of measurements. Analytical simulated and measured HRTFs from a KEMAR are used to validate the model.
    Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2009, 19-24 April 2009, Taipei, Taiwan; 01/2009
  • Source
    Rauf Iqbal, Thushara Abhayapala, Parastoo Sadeghi
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    ABSTRACT: The so-called peaky signaling has been shown to improve the performance of a communication system over noncoherent fading channels in the low signal-to-noise ratio (SNR) regime. We consider a signaling scheme which is equivalent to peaky signaling over a time-selective Rayleigh fading channel and make use of the unused degrees of freedom to improve the minimum mean square error (MMSE) performance of pilot-aided channel estimation (PACE) by a processing technique incorporating low pass filtering and downsampling. We assume infinite pilot symbols and optimal Wiener smoother at the receiver and give some analytical as well numerical results suggesting significant gains in MMSE performance of PACE.
    01/2009;

Publication Stats

1k Citations
201.98 Total Impact Points

Institutions

  • 1996–2014
    • Australian National University
      • • Research School of Engineering
      • • College of Engineering & Computer Science
      Canberra, Australian Capital Territory, Australia
  • 2011
    • Callaghan Innovation
      Lower Hutt City, Wellington, New Zealand
  • 2008
    • University of Melbourne
      • Department of Electrical and Electronic Engineering
      Melbourne, Victoria, Australia
    • The University of Nottingham Malaysia Campus
      Kuala Lumpor, Kuala Lumpur, Malaysia
  • 2004–2007
    • National ICT Australia Ltd
      Sydney, New South Wales, Australia
  • 2006
    • Queen's University Belfast
      Béal Feirste, N Ireland, United Kingdom
  • 2001–2004
    • Imperial College London
      • Department of Electrical and Electronic Engineering
      London, ENG, United Kingdom