Research Journal of Applied Sciences, Engineering and Technology 4(18): 3510-3515, 2012 ISSN: 2040-7467 © Maxwell Scientific Organization, 2012 Submitted: May 08, 2012 Accepted: June 08, 2012 Published: September 15, 2012 Beamforming in Short Time Fractional Fourier Domain (FRFD) Mostafa Rahmani and Hassan Ghassemian Department of Electrical and Computer Engineering, Tarbiat Modares University, Tehran, Iran Abstract: This study aims to generate a model for the problem of beamforming in fractional Fourier domain, through which a general solution for obtaining weighted beamforming in fractional Fourier domain for different criteria might be delivered. In order to generate this model, a description of fractional delay concept is presented, through which the string vector in fractional Fourier domain is calculated and a model for beamforming in fractional Fourier domain is generated which is similar to that used for obtaining weighted beamforming in time domain for narrow-band signals. To complement our solution, a method for obtaining the optimal fractional domain as well as direction of arrival in optimal fractional Fourier domain. Keywords: Array processing, beamforming, direction of arrival, fractional fourier domain, fractional fourier transform, steering vector ordinary Fourier Transform (FT). Therefore, replacing the FT with the FrFT should improve the performance considerably. In this study, a novel Beamforming algorithm of wideband LFM signals based on the FrFT is proposed which can be applied to linear array. Implementations of the proposed algorithm are also developed, which are based on the spectral peak searching. At the end of the study, numerical examples and simulations are presented followed by a discussion of the results. INTRODUCTION Beamforming is array signal processing technique used in sensor arrays for directional signal transmission or reception. This is achieved by combining elements in the array in such a way that signals at particular angle experience constructive interference and while others experience destructive interference. Various beamforming algorithms have been proposed (Monzingo and Miller, 1980). The conventional beamformer data back to the second world war and is a mere application of Fourier-based spectral analysis to spatio-temporally sampled data. Later, adaptive beamformers (Gabriel, 1972) and classical time delay estimation techniques were applied to enhance one's ability to resolve closely spaced signal sources. The extension of the time delay estimation methods to more than one signal and the limited resolution of beamforming together with an increasing number of novel application, renewed interest of researchers in statistical signal processing. However, for parametric techniques, the intuitive notion of resolution is nontrivial to define in precise terms. This in turn, resulted in the emergence of the parameter estimation approach as an active research area. For the first time in 2003 Yetik introduced Beamforming in FRFD (Yetik and Nehorai, 2003), than (Zhou and Tao, 2008) complicated this algorithm (Tao, 2004). In this methods Beamforming has been model as spatial filtering, in the other words their work was an extension Osaka's work (Kutay et al., 1997) on array signal processing. Our study tends to solve the beamforming problem through parametric method in FRFD. The motivation behind the proposed method is the ability of the Fractional Fourier Transform (FrFT) to process the chirp signals better than the METHODOLOGY Fractioanl fourier transform: Fractional Fourier transform represents a powerful method in signal processing which has been used over past two decades. This transform, being a generalization of ordinary Fourier transform, is indeed equal to signal rotation in time-frequency domain in a given angle (Namias, 1980). Letting x(t ) be an arbitrary signal, its pth-order FrFT is defined as: X (u) Fp[ x(t)] x(t) K (t, u) dt (1) where, p(/2); F p [.] is operator signals of FRFT, K (t , u) is kernel function: 1 j cot 2 t 2 u2 cot tu csc , K (t,u) exp j 2 (t u), (t u), n 2n (2n 1) Corresponding Author: Mostafa Rahmani, Department of Electrical and Computer Engineering, Tarbiat Modares University, Tehran, Iran 3510 Res. J. Appl. Sci. Eng. Technol., 4(18): 3510-3515, 2012 The FRFT is a linear transform, the ( , u ) is called the FRFT domain. From the above relationship it is obvious that the time and frequency are the 2n th and 2n / 2th FRFT domains, respectively, meaning that the FRFT is unified time-frequency transform and the FRFT can demonstrate the characteristics for the signal changing from the time domain to the frequency domain having the order in the range from 0 increasing to 2 . In practice the Discrete FRFT (DFRFT) is used. The commonly used fast algorithm of DFRFT is Osaka's algorithm (Ozaktas et al., 1996). In the fast algorithm of the FRFT digital computation, the original signal s(n) must be normalized. It assume that, the original continuous signal is compact along the axes of time and frequency, the time-domain representation of the signal is confined to the interval [t / 2, t / 2] and its frequency-domain representation is confined to the interval [ f / 2, f / 2 ] . So the signal's timebandwidth product is N t f , which is always greater than unity because of the uncertainty relation. Because time-domain and frequency domain have different dimension. In order to compute the FRFT, time-domain and frequency-domain must be transformed to the domain whose dimension is “1”. It introduces the scaling parameter s with the dimension of time and introduces scaled coordinates: x t / s, v fs (2) (x,v) are clearly The new coordinates dimensionless. The signal is confined to the interval and [t /( 2s ), t /( 2s )] [ f s / 2, f s / 2 ] , respectively under the new coordinates. In order to have the two intervals equal to each other, it chooses s t / f , so that the length of both intervals are equal to the dimensionless quantity x tf . Both intervals are normalized as [x / 2, x / 2]. In the newly defined coordinates, according to the sampling theorem, the signal can be represented in both domains with N x2 samples spaced 1/ x 1/ N apart. According to the algorithm, the transform S ( , m) of the digital signal defined as below: s(n) is DFRFT at p order is A N jt m 2 2mnn 2 /(2x) 2 n s e 2x N 2x cot , csc S ( , m) The FrFT has found many applications in digital signal processing such as filtering (Aldirmaz and Durak, 2010; Erden et al., 1999), signal restoration (Erden et al., 1999), system synthesis (Erden and Ozaktas, 1998), mutual intensity synthesis (Erden et al., 1996), system decomposition (Kutay et al., 1999; Yetik et al., 2000), optimum Wiener filtering (Kutay and Ozaktas, 1998), image restoration (Kutay and Ozaktas, 1998) and perspective projections (Yetik et al., 2000). More complete treatment of the FrFT and many of its optics and digital signal processing applications can be found in Ozaktas et al. (2000). The FrFT suggests a potential improvement in any application where the ordinary FT is used since it provides an extra degree of freedom corresponding to the choice of the order s(n) . We can attempt to improve the solution to any problem that utilizes the FT by carrying the extra parameter throughout the solution and then optimizing over this parameter. The FrFT is most likely to improve the solutions to problems where chirp signals are involved. This is because a chirp signal forms a line in the timefrequency plane and therefore, there exists an order for which such a signal is compact. Array model: Let a uniform linear array as shown in Fig. 1. This array has M sensors with a reference sensor on the coordinate origin point. The i-th sensor coordinate location is (i-1) d, (i = 1, 2, …, M). Assume that wideband signal from far-field impinge on the array. Thus, the observed signal at the i-th sensor can be expressed as: f s ri .u x n s n ( ) ( ) n (t ) s ( n qi ) n (t ) i c q f s ( i 1) d sin i c where, n (t) is the additive white Gauss noise, Which is assumed to be statistically independent with signal sources, qi is the path delay, c is a wave velocity and is source direction. Beamforming in fractional fourier domain: In order for the model to be expressed parametrically, first the concept of fractional delay should be defined and the relations' must be extracted for the discrete state, by use of which the beamforming of wideband signal in the FRFD is modeled as beamforming of narrowband signal in the Fourier domain. (3) 3511 (4) Res. J. Appl. Sci. Eng. Technol., 4(18): 3510-3515, 2012 2 f s q cot n j 2 F p x(n f s q)e N s X m.e C() 1 j / N C( ) f s q2 B( ) f s qm 2 , cot csc , B() s s2 (8) Modelling of beamforming in FRFD: According to the definition fractional delay Eq. (8), we obtain xˆi (n) fractional delay for each sensor as a below: j 2 qi cot n N s2 Fig. 1: Uniform linear array xˆi (n) s(n qi )e Fractional delay: DFRFT for the delay signal and Multiplication by an exponential signal can be expressed as a below: xˆi (n) xi (n)Qi (n) , Qi (n) e X m ( f s q / s) cos e (5) Xˆ i (m) S m.e 2 k j n F x ( n )e N p X m sk sin e C ( ) s2k 2 j 2 / N sin cos sk cos m 2 j j 2 k n N 2 k n N call is fractional delay: .e and if k fsq cot then: s2 s2 Xˆ i (m) Xˆ i , opt (9) , B( ) , csc opt s (10) , s(n), p (m) F xˆi (n) popt opt (11) If Xˆ [ Xˆ 1 (m), Xˆ 2 (m),..., Xˆ M (m)] the output of array can be compactly represented as a matrix form: s2k 2 j / N sin cos sk cosm 2 (12) where, 1 2 j / N 1 C( )q12 B( )q1m j / N C ( ) qM B( ) qM m A e 2 ,...,e 2 (13) (7) is the steering vector in FrFD. The output of beamformer is: 3512 cot opt Xˆ AS ( m) N , ( f q)2 fq j / N s 2 sin cos s sin mkfs q sin2 2s s 2 qi cot n N s2 1 j / N C ( ) qi2 B ( ) qi m 2 S (m) Sopt (m) F 2k j n F p x(n f s q)e N X m sk sin f s q / s cos.e 2 qi cot n N s2 where, (6) Now obtain DFRFT x ( n f s q )e , then we gain k in term of q so that DRFRT it obtain in term of X (m ) , then x ( n f s q ) e j j Then according to the definition DFRFT Eq. (3), obtain DFRFT Eq. (9) and extract the model for beamforming in DFRFT: F p x(n f s q) ( f q)2 f q j 2 / N s sin cos s sin m 2S 2 S xi (n)e Res. J. Appl. Sci. Eng. Technol., 4(18): 3510-3515, 2012 Fig. 2: Output of conventional beamformer in FRFD an ordinary domain for signal along noise M M i 1 i 1 y ( n) x i ( n q i ) a i xˆ i ( n q i ) e and the DFrFT of Yopt (m) F popt j 2 qi cot n N s2 SIMULATION (14) We demonstrate that the proposed method yields improved results, that is smaller MSE. Assume a ULA of M = 12 and the observed signal contain wideband LFM component as a below: y(n) can be obtain: 2 qi cot M n j 2 ai xˆi (n qi )e N s i1 N/2 e jt m 2mn/ sn 2 T 0 . 01 , f s 4 10 4 . 2 q cot M a xˆ (n q )e j N s n ii i i 2 / s2 / N N nN / 2 2 i1 (15) when Eq. (15) is a simplified: M Y opt ( m) ai Xˆ i (m)e i 1 1 j / N C ( ) qi2 B ( ) qi m 2 (16) In the FrFD, Original array model can be degenerated into: Xˆ AS ( m ) N , Y W Xˆ (17) Conventional beamformer: The received signal often include some noise. The noise can be modeled as complex, Gaussian distributed random number. We assume that 3 dB SNR at each sensor. Figure 2 show the output of conventional beamformer in Time domain and conventional beamformer in optimum fractional Fourier domain. The value of the SNR varies from -20 dB to 5dB with an interval of 1dB. At each level of the SNR, we run 800 Monte Carlo simulations and the RMSE obtained is shown in Fig. 3 and Compare of the MSE for the Conventional FrFT Beamformer and Conventional Beamformer Adaptive Beamformer (MVDR): In the presence of strong interference, the target signal may be masked by 3513 f 0 5 10 3 , 6 10 5. Yopt (m) A s ( t ) exp j 2 f 0 t 0 . 5 t 2 Res. J. Appl. Sci. Eng. Technol., 4(18): 3510-3515, 2012 MSE vs SNR Output of MVDR Beamformer in Fractional Fourier Domain 1.5 Proposed Orignal 16 1 12 0.5 10 Magnitud Mean square error 14 8 6 4 0 -0.5 2 0 -20 -15 -10 -5 SNR (dB) -1 5 0 -1.5 Fig. 3: Comparison of the MSE for the conventional FrFT beamformer and conventional beamformer 50 100 150 200 Time (s) 250 300 350 400 Fig. 5: Output of MVDR beamformer in FrFD for signal along noise and interference Output of Conventional Beamformer in Fractional Fourier Domain 1 0.5 Magnitud 0 Conventional in MVDER in Fractional FourierDomain MVDR in Fourier Domain 25 0 FourierDomain Conventional in Fractional FourierDomain 20 -0.5 15 -1 0 50 100 150 200 Time (s) 250 300 350 400 10 Output of Conventional Beamformer 1.5 5 Magnitud 1 0.5 0 -20 0 -15 -10 -5 0 5 10 15 -0.5 -1 0 50 100 150 200 Time (s) 250 300 350 Fig. 6: Comparison of the MSE for the conventional FrFT beamformer, conventional beamformer, MVDR and MVDR in FRFD 400 Fig. 4: Output of conventional beamformer in FrFD and ordinary domain for signal along noise and interference the interference signal. Such problems are very challenging for a conventional beamformer and therefore, adaptive beamformer are introduced to address this problem. To overcome the interference problem we can use the MVDR beamformer, a popular adaptive beamformer. The MVDR Beamformer preserve the signal arriving along a desired direction, while trying to suppress signal coming from other direction. Figure 4 show Output of Conventional beamformer in FRFD and Ordinary Domain for signal along noise. Figure 5 show output of MVDR bean former in FrFD for signal along noise and interference. The value of the SNR varies from -20 to 5 dB with an interval of 1dB, at each level of the SNR, we run 800 Monte Carlo simulations and the RMSE for the Conventional FrFT Beamformer, Conventional Beamformer, MVDR and MVDR in FrFD obtained is shown in Fig. 6. 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