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. 2020 Aug 10;106:102830. doi: 10.1016/j.dsp.2020.102830

Novel generalized Fourier representations and phase transforms

Pushpendra Singh 1
PMCID: PMC7416779  PMID: 32834705

Abstract

The Fourier representations (FRs) are indispensable mathematical formulations for modeling and analysis of physical phenomena and engineering systems. This study presents a new set of generalized Fourier representations (GFRs) and phase transforms (PTs). The PTs are special cases of the GFRs and true generalizations of the Hilbert transforms. In particular, the Fourier transform based kernel of the PT is derived and its various properties are discussed. The time derivative and integral, including fractional order, of a signal are obtained using the GFR. It is demonstrated that the general class of time-invariant and time-variant filtering operations, analog and digital modulations can be obtained from the proposed GFR. A narrowband Fourier representation for the time-frequency analysis of a signal is also presented using the GFR. A discrete cosine transform based implementation, to avoid end artifacts due to discontinuities present in the both ends of a signal, is proposed. A fractional-delay in a discrete-time signal using the FR is introduced. The fast Fourier transform implementation of all the proposed representations is developed. Moreover, using the analytic wavelet transform, a wavelet phase transform (WPT) is proposed to obtain a desired phase-shift in a signal under-analysis. A wavelet quadrature transform (WQT) is also presented which is a special case of the WPT with a phase-shift of π/2 radians. Thus, a wavelet analytic signal representation is derived from the WQT. Theoretical analysis and numerical experiments are conducted to evaluate effectiveness of the proposed methods.

Keywords: Generalized Fourier representation, Hilbert transform, Phase transform, Analytic wavelet transform, Wavelet phase transform and wavelet quadrature transform, Discrete cosine transform

1. Introduction

The Fourier representation (FR) of a signal is the most important mathematical formulation for modeling and analysis of physical phenomena, engineering systems and signals in numerous applications. It has been used to obtain solution of problems in almost all fields of mathematics, science, engineering and technology. It is the fundamental of signal processing, analysis, information extraction and interpretation. There are many variants of the FR such as continuous-time Fourier series (FS), Fourier transform (FT), Fourier sine transform (FST) and Fourier cosine transform (FCT), discrete-time FT (DTFT), discrete-time Fourier series (DTFS), discrete FT (DFT), discrete sine transform (DST) and discrete cosine transform (DCT) [1], [2]. All these are the orthogonal transforms which can be efficiently computed using the Cooley–Tukey fast FT (FFT) algorithm [3]. Recently, many studies [4], [5], [6] have been performed using the Fourier theory, and many applications including signal decomposition and time-frequency analysis of a nonlinear and nonstationary time-series have been proposed.

The DCT was proposed in the seminal paper [1] for image processing based pattern recognition and Wiener filtering. The modified DCT (MDCT) [7] is based on the DCT of overlapping data which uses the concept of time-domain aliasing cancellation [8]. The DCT and MDCT are widely-used due to decorrelation and energy compaction properties in many applications like image (e.g., JPEG), video (e.g., Motion JPEG, MPEG, Daala, digital video, Theora) and audio (e.g., MP3, WMA, AC-3, AAC, Vorbis, ATRAC) compression, electrocardiogram data analysis [11], and for numerical solution of partial differential equations by spectral methods. There are eight types of DCTs and eight types of DSTs depending upon the symmetry about a data point and the boundary conditions.

The Hilbert transform (HT) [75], [76], [77], [78], [79] is an inevitable tool which has been studied and used in numerous applications such as quadrature amplitude modulation, analytic signal representation, time-frequency analysis, signal processing and system identification, signal and speech demodulation, and image processing [53], [54], [55], [56], [57], [58], [59], [60], [61], [62]. The Fourier theory based quadrature method was proposed by Gabor [10] in 1946 as a practical approach for obtaining the HT and Gabor analytic signal (GAS) representation of a signal. The GAS has been extensively used in communication engineering, physics, time-frequency-energy (TFE) representation, and signal analysis. The TFE representation of a signal is obtained using the concept of instantaneous frequency (IF) [12], [13], [14], [15], [16], [17], [18], [19], [20], [21], [22], [23], [24], [25], [26], [27], [46] which is an important parameter in many applications. Recently, using eight types of DCTs and eight types of DSTs, sixteen types of Fourier quadrature transforms (FQTs) and corresponding Fourier-Singh analytic signal (FSAS) representations are introduced for a nonlinear and non-stationary time-series analysis [4]. The FQTs and FSAS representations are alternatives to the HT and GAS representation, respectively. The HT and FQTs are π/2 phase shifters. However, there is no general method to provide a desired phase shift to signal under analysis. This study presents a phase transform (PT), which is based on the proposed generalized Fourier representation (GFR) of a signal, to obtain the desired phase shift and time-delay. This work also discusses the various special cases of the GFR, namely the Fourier representation, PT, time-delay including fractional delay of discrete-time signals, time derivative and integral including fractional order, amplitude modulation (AM), frequency modulation (FM) and other digital modulation schemes.

There are numerous applications of wavelet transform (WT) [31], [32], [33], [34] which uses a wavelet function to analyze the signals, and when analyzing wavelet is analytic then corresponding WT is known as the analytic wavelet transform (AWT) [36], [37], [38]. An analytic signal representation of a real-valued and finite energy signal using the AWT is presented in [40], [41], [42], where authors obtained the wavelet analytic signal (WAS) using an analytic wavelet function (AWF) with its real part being an even function (Theorem 2 of [40]) and thus the imaginary part of the AWF is an odd function. The advantages of the WAS in both precision and antinoise performance are also demonstrated. This study eliminates the restriction (that the real part of an AWF has to be an even function to obtain the WAS) and proposes two representations of the wavelet quadrature transform (WQT) and corresponding WAS using any AWF. This work also proposes two representations of the wavelet phase transform (WPT) and shows that the proposed WQT is a special case of the WPT where phase-shift is π/2 radians.

There are many generalizations of the FT such as Laplace transform, Gabor transform [80], fractional FT (FrFT) [81], [82], acoustic scattering and Schrödinger's equation based FT [83], short-time FT (STFT), wavelet transform [84], quaternion FT [85], chirplet transform [86], S transform [87], and de Branges theory based FT [88]. Most of these transforms are based on the modification in the forward FT integral (analysis equation) and the original signal is recovered from the corresponding inverse transforms. On the other hand, the proposed GFRs are based on the inverse transform integrals (synthesis equations), e.g., the inverse FT and inverse wavelet transform.

The main contributions of this study are summarized as follows:

  • 1.

    Introduction of the GFRs which are completely based on the Fourier representations of a signal.

  • 2.

    Introduction of the phase transforms (PTs) using Fourier representations (i.e., FS, FT, DFT, FCT, FST, DSTs and DCTs) which can be functions of both the frequency and time (hereafter, unless and until stated, PT means constant or frequency-and-time independent PT). The proposed PT is a special case of the GFR, and a true generalization of the HT. The desired phase-shifts and time-delays can be introduced to a signal under analysis using the PT. In particular, the Fourier transform based kernel of the PT is derived and its various properties are discussed. It is shown that the HT is a special case of the PT when phase-shift is π/2 radians. An extension of the one-denominational PT for the two-dimensional image signals is also provided in Appendix C, which can be easily further extended for multidimensional signals.

  • 3.

    The time derivative and time integral, including fractional order, of a signal can be obtained using the GFR. The DCT based implementation is presented to avoid end artifacts due to discontinuities present at the both ends of a signal.

  • 4.

    Introduction of the fractional delay in a discrete-time signal using the Fourier representation.

  • 5.

    It is demonstrated that the zero-phase, linear and nonlinear phase filters such as low-pass, high-pass, band-pass, and band-stop or band-reject filters, which can be time-invariant or time-varying, are special cases of the proposed GFR.

  • 6.

    Contrary to the perception in the literature [59], it is demonstrated that the continuous-time both aperiodic (19) and periodic (94) Hilbert kernels possess zero rather than pole (singularity/infinity) at the origin.

  • 7.

    Using the proposed GFR, the narrowband Fourier representation (NBFR) is obtained for the time-frequency representation and analysis of a signal.

  • 8.

    The FFT implementations of all the above proposed representations are developed.

  • 9.

    Using the AWT, the WPT is introduced to obtain a desired phase-shift in a signal under-analysis, and two representations of the WQT are derived. Thus, the WAS representation is obtained using the WQT which is a special case of the WPT where phase-shift is π/2 radians.

Moreover, using the PT, It is observed that (i) a constant phase shift (e.g., HT as π/2 phase shift) in a signal corresponds to variable time-delays in all the harmonics, (ii) to obtain a constant time-delay in a signal, one needs to provide variable phase-shifts in all the harmonics, (iii) a constant phase-shift is same as the constant time-delay only for a single frequency sinusoid.

All the acronyms and symbols used in this work are summarized in Appendix A. Rest of the study is organized as follows: The GFR and its various special cases are presented using Fourier series in Section 2. The PT using FT is presented in Section 3.1. The PT using Fourier sine and cosine transforms is presented in Section 3.2. The WPT and WQT using AWT are presented in Section 3.3. The implementations of the GFR using the DFT and DCT are presented in Section 3.4 and Section 3.5, respectively. Simulation results and discussions are presented in Section 4. Section 5 presents conclusion and future scope of the study.

2. The generalized Fourier representation

This section proposes the GFR, presents its various special cases, and provides convergence of the GFR.

2.1. The GFR using Fourier series representation

Let xT(t) be a real valued periodic signal (i.e., xT(t+T)=xT(t),t) which follows the Dirichlet conditions. The Fourier series expansion of xT(t) is given by

xT(t)=a0+k=1[akcos(kω0t)+bksin(kω0t)],ω0=2πT=2πf0 rad/s,  (1)

where a0=1Tt1t1+TxT(t)dt; ak=2Tt1t1+TxT(t)cos(kω0t)dt and bk=2Tt1t1+TxT(t)sin(kω0t)dt for k=1,2,,. Using ak=Xkcos(ϕk), bk=Xksin(ϕk), where Xk=±ak2+bk2, |Xk|=ak2+bk2 and ϕk=tan1(bk/ak), [i.e., |Xk|ejϕk=akjbk, X0=a0], one can write

xT(t)=a0+k=1|Xk|cos(kω0t+ϕk). (2)

Using the Fourier series representation (2), the GFR is hereby proposed as

xT(t,Hk(t),αk(t))=a0H0(t)cos(α0(t))+k=1Hk(t)|Xk|cos(kω0t+ϕkαk(t)), (3)

where 0Hk(t)M< and 0αk(t)<2π (for k=0,1,2,,) are introduced as amplitude and phase scaling/modulating functions of both frequency (k) and time (t). Now, the various cases of the GFR are presented as follows:

Case 1: The GFR (3) is the Fourier series representation of a signal when Hk(t)=1 and αk(t)=0, for all t and k=0,1,2,,.

Case 2: Using the GFR (3) with Hk(t)=1, the four types of phase transforms are hereby proposed as follows:

(i) Frequency- and time-dependent (FDTD) PT as

xT(t,αk(t))=a0cos(α0(t))+k=1|Xk|cos(kω0t+ϕkαk(t)), (4)

where introduced phase, αk(t), is a function of time and frequency.

(ii) Frequency-independent and time-dependent (FITD) PT as

xT(t,α(t))=a0cos(α(t))+k=1|Xk|cos(kω0t+ϕkα(t)), (5)

which can be written as

xT(t,α(t))=xT(t)cos(α(t))+xT(t,π/2)sin(α(t)), (6)

where xT(t,π/2)=k=1|Xk|sin(kω0t+ϕk).

(iii) Frequency-dependent and time-independent (FDTI) PT with αk(t)=αk, t,k as

xT(t,αk)=a0cos(α0)+k=1|Xk|cos(kω0t+ϕkαk), (7)

where αk[0,2π) is the phase shift in k-th harmonics.

(iv) Frequency- and time-independent (FITI) or constant PT as

xT(t,α)=a0cos(α)+k=1|Xk|cos(kω0t+ϕkα)=xT(t)cos(α)+xT(t,π/2)sin(α), (8)

where xT(t,π/2) is the HT, a special case of the PT (8) with constant phase shift of α=π/2 radians (or 90), which is defined as

xˆT(t)=xT(t,π/2)=k=1|Xk|sin(kω0t+ϕk). (9)

Using (2) and (9), one can write the analytic signal (AS) representation as

zT(t)=xT(t)+jxˆT(t)=a0+k=1|Xk|[cos(kω0t+ϕk)+jsin(kω0t+ϕk)]. (10)

The kernel of the proposed PT for the periodic signals, δT(t,α)=cos(α)δT(t)+sin(α)δT(t,π/2), is derived in Example 1. One can easily obtain an arbitrary constant PT of a periodic signal using the periodic convolution with this kernel as, xT(t,α)=xT(t)δT(t,α)=cos(α)xT(t)+sin(α)xT(t,π/2).

Case 3: A time-delay of the periodic signal xT(t) can be defined as

xT(ttk)=a0+k=1|Xk|cos(kω0t+ϕkkω0tk). (11)

From (7), (9) and (11), we observe that (a) the HT is a constant phase shifter which introduces variable time-delays in all the harmonics, i.e., tk=π/(2kω0); (b) to obtain a constant time-delay (say, tk=td) in a signal, one need to provide variable phase shifts in all the harmonics, i.e., αk=kω0td; (c) a constant phase shift is same as constant time-delay only for a single frequency sinusoid (say k=1 and hence α1=ω0td); (d) variable phase shifts are same as variable time-delays only for a zero mean (a0=0) signal with αk=kω0tk, ∀k.

It is interesting to observe that the phase shift in a constant signal (7) is presented as, a0cos(α0), which is valid because (i) for the HT, it is zero, (ii) for the phase shift of π, it is multiplied by minus one, and (iii) there is no change in its value, if phase shift is zero. However, time-delay operation in a constant signal [e.g., a0 in (11)] does not change its value.

Case 4: It is observed that the μ-th order time derivative of a periodic signal is a special case of the GFR (3) when H0(t)=tμΓ(1μ), Hk(t)=(kω0)μ, α0(t)=0 and αk(t)=μπ/2, t,k. Thus, from (3), we obtain the μ-th order fractional time derivative of the signal xT(t) as

Dμ{xT(t)}=a0tμΓ(1μ)+k=1(kω0)μ|Xk|cos(kω0t+ϕk+μπ/2),μ0, (12)

where Γ(1μ) is a gamma function.

Case 5: It is observed that the ν-th order time integral of a periodic signal is a special case of the GFR (3) when H0(t)=tνΓ(1+ν), Hk(t)=(kω0)ν, α0(t)=0 and αk(t)=νπ/2, t,k. Thus, from (3), we obtain the ν-th order fractional time integral of the signal xT(t) as

Dν{xT(t)}=a0tνΓ(1+ν)+k=1(kω0)ν|Xk|cos(kω0t+ϕkνπ/2),ν0. (13)

Case 6: Using the proposed GFR (3), we can obtain an amplitude modulated (AM) signal with an arbitrary but fixed value of k (say k=1), carrier frequency ωc=ω0, α1(t)=0, H0(t)=0, and H1(t)=(Am+m(t))0, where m(t) is a message signal whose maximum frequency ωm<<ωc. From (3), one can also obtain an angle modulated (which includes both the frequency and phase modulation) signal with a fixed value of k (say k=1), carrier frequency ωc=ω0, H0(t)=0, Hk(t)=1, and α1(t)=m(t).

Similarly, it can be shown that the digital modulation schemes such as phase-shift keying, frequency-shift keying, amplitude-shift keying and quadrature amplitude modulation are special cases of the GFR (3).

Case 7: The GFR (3) represents different types of filtering operations to the periodic signal xT(t), e.g., the zero-phase filtering operations: (i) low-pass filtering if H0(t)=1; Hk(t)=1, for 1kK and Hk(t)=0, for k>K; αk(t)=0, (ii) high-pass filtering if H0(t)=0; Hk(t)=0, for k<K; Hk(t)=1, for kK; αk(t)=0, (iii) band-pass filtering if H0(t)=0; Hk(t)=1, for K1kK2 and Hk(t)=0, otherwise; αk(t)=0, and (iv) band-stop filtering if H0(t)=1; Hk(t)=0, for K1kK2 and Hk(t)=1, otherwise; αk(t)=0. Other than these zero-phase filtering, there are countless possibilities of linear and nonlinear phase filtering operations by properly selecting the Hk(t) and αk(t) based on the applications and requirements.

Moreover, the class of time-varying filters are also possible when Hk(t) or αk(t), or both are functions of time and frequency. For example, the Fourier decomposition method (FDM) proposed in the seminal work [5] maps the FR (2) of possibly infinite constant amplitude-and-frequency components into a set of finite number of amplitude-and-frequency modulated (AM-FM) Fourier intrinsic band functions (FIBFs) as, xT(t)=a0+m=1M[xm(t)cos(ϕm(t))], which is a special case of the proposed GFR (3) and can be used to obtain the time-varying filtering (TVF) as

yT(t)=a0+m=1M[Hm(t)xm(t)cos(ϕm(t))]. (14)

This kind of TVF can be used in many applications such as effective noise removal from the gravitational waves which is presented in the Example 2.

2.2. Convergence of the GFR

We are, generally, interested in the following three types of convergence of the sequence of partial sums

sT,N(t)=k=0N|Xk|cos(kω0t+ϕk), (15)

(i) sT,NxT pointwise on a set [t1,t1+T] if sT,N(t)xT(t) as N for all t[t1,t1+T]; (ii) sT,NxT uniformly on a set [t1,t1+T] if sup|sT,N(t)xT(t)|0 as N for t[t1,t1+T]; and (iii) sT,NxT in L2 norm on a set [t1,t1+T] if t1t1+T(sT,N(t)xT(t))2dt0 as N for t[t1,t1+T]. All these convergences are well defined in the literature [63], [64], [65], [66], [67] and depend on the type of convergence and continuity of the function xT(t) which can be decomposed in terms of sine and cosine basis functions using the Fourier representation (2). The Fourier series convergence theorems are summarized as follows:

Theorem 1. (Dirichlet 1824) Let xT be a T-periodic, piecewise continuous function with piecewise-continuous first-derivative. Then sT,Nx˜T pointwise on R as N , where x˜T(t)=[x˜T(t+)+x˜T(t)]/2 .

Theorem 2. Let xT be a T-periodic, continuous function with piecewise-continuous first-derivative. Then sT,NxT uniformly on R as N .

Theorem 3. Let xT be a T-periodic, piecewise continuous function with piecewise-continuous first-derivative. Then sT,NxT in L2 norm as N .

The convergence of the proposed GFR (3) can be considered similar to the FR (2), provided the phase and amplitude scaling functions are properly selected, as follows: (1) if the FR (2) converges uniformly (i.e., k=0|Xk|<), as k=0Hk(t)|Xk|cos(kω0t+ϕkαk(t))k=0Hk(t)|Xk|, then for the uniform convergence of (3), inner product Hk(t),|Xk| must be finite for each t, i.e., k=0Hk(t)|Xk|Mk=0|Xk|<; (2) if the FR (2) converges in L2 norm (i.e., k=0|Xk|2<), then for the L2 norm convergence of (3), k=0(Hk(t)|Xk|)2M2k=0|Xk|2<; and (3) if the FR (2) converges pointwise, then the GFR (3) would converge pointwise if Hk(t) and αk(t) are bounded, T-periodic, continuous (or piecewise continuous) functions (because the uniform convergence implies both the pointwise and L2 convergences). Therefore, for all the three cases, as long as Hk(t) and αk(t) are bounded, T-periodic, continuous (or piecewise continuous) functions with piecewise-continuous first-derivatives, the proposed GFR (3) would converse to a new desired function.

3. Phase transforms using the FT, FCT and FST

This section presents the GFR and PT to achieve a desired phase-shift in a signal using the FT, FCT, and FST. Moreover, the FFT implementations of the various cases of the proposed GFR are also presented.

3.1. PT using the Fourier transform

The FT and inverse FT (IFT) pairs of a signal, x(t) which satisfies the Dirichlet conditions, are defined as

X(ω)=x(t)exp(jωt)dt,<ω<,x(t)=12πX(ω)exp(jωt)dω,<t<, (16)

subject to the existence of the integrals, and these pairs can be denoted by x(t)X(f), where ω=2πf. From the definitions of FT and IFT (16), one can observe that X(0)=x(t)dt=0, and x(0)=12πX(ω)dω=0, provided that x(t) and X(ω) are zero-mean functions, respectively. Moreover, an odd function is a zero-mean function, however, reverse is not always true.

First, we consider subtle details of the zero-mean function, x1(t)=exp(a|t|)sgn(t),a>0, where sign function is defined [9] as

sgn(t)={1,t>0,0,t=0,1t<0. (17)

The FT of x1(t) is given by X1(ω)=j2ωa2+ω2,a>0, from which we observe that X1(f)=0 at f=0,aR (as x1(t) is an odd function, therefore, X1(0)=exp(a|t|)sgn(t)dt=0,aR; however, if a<0, X1(f) does not exist for f0). It is pertinent to notice that, lima0X1(f)=1jπf, limf01jπf=j, and limf0+1jπf=j which implies that the limf01jπf does not exist. The FT of a zero-mean or an odd function is always constrained to be zero at the origin, and thus, we write the FT S(f) of the zero-mean sign function, s(t)=sgn(t)=lima0x1(t), as

lima0X1(f)=S(f)={0,f=0,1jπf,f0. (18)

Using the duality principle of the FT, i.e., if x(t)X(f) then X(t)x(f), one can obtain, jS(t)jsgn(f)=jsgn(f). We denote jS(t)=h(t), and thus obtain

H(f)=jsgn(f), and h(t)={0,t=0,1πt,t0. (19)

Now, we compute the function h(t) from H(f) using the IFT (16) as: h(t)=j2π[0exp(jωt)dω+0exp(jωt)dω], and obtain

h(t)=1π0sin(ωt)dω, (20)

where, one can clearly observe that, h(0)=0, which is also required by the definition (19). Further, we consider a limiting case of the following integral, with σ>0, and obtain the solution of integration (20) as

1π0eσωsin(ωt)dω=1πtσ2+t2h(t)=limσ01πtσ2+t2={0,t=0,1πt,t0. (21)

The function, 1πt, is the well-known HT kernel which is singular at the origin, t=0 [59]. In the sense of limit, it is not defined at the origin, because, limt01t= and limt0+1t=, therefore, limt01t does not exist. Thus, we have presented a trivial but important modification to the HT kernel in (19) by defining it at the origin, and obtained its integral form in (20). One can observe that if two or more functions are equal, almost everywhere, except on a set of points where Lebesgue measure (e.g., length, area, or volume) is zero, then the FT of these functions is same. Therefore the FT of the original HT kernel 1πt and the modified HT kernel h(t) (19) is same. These definitions, (19), (20) and (21), of the HT kernel are further supported by the kernel of the discrete-time HT [5], [47], [48], [49], [50], defined as h[n]=(1cos(πn))πn, which is also zero at the origin (i.e., h[0]=0), and it can be obtained by using the discrete counter part of (20), i.e., h[n]=1π0πsin(Ωn)dΩ.

For a real-valued signal x(t), the Gabor analytic signal is defined as

z(t)=x(t)+jxˆ(t)=1π0X(ω)exp(jωt)dω,<t<, (22)

where the real part is the original signal and the imaginary part is the HT of the real part. Thus, the signal x(t) and its HT xˆ(t) can be written as

x(t)=1π0|X(ω)|cos(ωt+ϕ(ω))dω,<t<,xˆ(t)=1π0|X(ω)|sin(ωt+ϕ(ω))dω,<t<, (23)

where X(ω)=Xi(ω)+jXr(ω)=|X(ω)|exp(jϕ(ω)), ϕ(ω)=tan1(Xi(ω)/Xr(ω)). The GFR, corresponding to (3), is present here as

x(t,H(ω,t),α(ω,t))=1π0H(ω,t)|X(ω)|cos(ωt+ϕ(ω)α(ω,t))dω. (24)

The four PTs [corresponding to (4), (5), (7) and (8)] of the signal x(t) are here defined as

x(t,α(ω,t))=1π0|X(ω)|cos(ωt+ϕ(ω)α(ω,t))dω,FDTD-PT,x(t,α(t))=1π0|X(ω)|cos(ωt+ϕ(ω)α(t))dω,FITD-PT, (25)
x(t,α(ω))=1π0|X(ω)|cos(ωt+ϕ(ω)α(ω))dω,FDTI-PT,x(t,α)=1π0|X(ω)|cos(ωt+ϕ(ω)α)dω,FITI-PT.

The PT, x(t,α(ω)) defined in (25), is the real part of the PT of analytic signal hereby defined as

z(t,α(ω))=1π0X(ω)exp(j(ωtα(ω)))dω, (26)

and, therefore, the transfer function is given by

H(α(ω))={ejα(ω),ω0,ejα(ω),ω<0. (27)

For example, to obtain a time-delayed signal x(tt0) from the input signal x(t), one can set α(ω)=ωt0, and therefore from (25) and (16), x(t,α(ω))=x(tt0)12πX(ω)exp(jωt)×exp(jωt0)dω=1π0|X(ω)|cos(ωt+ϕ(ω)ωt0)dω.

Thus, we have defined a general phase shifter in (25) which is a generalization of the IFT as well as the HT because it is (i) IFT if α(ω)=0, (ii) HT if α(ω)=π/2. The impulse response of the PT, designated as PT kernel, from (25) with α(ω)=α, is hereby derived as

ħ(t,α)=cos(α)δ(t)+sin(α)h(t)δ(t,α)=cos(α)δ(t)+sin(α)δ(t,π/2), (28)

where δ(t) is the Dirac delta function, ħ(t,π/2)=h(t)=δˆ(t)=δ(t,π/2) is the HT kernel as defined in (19) and (20), ħ(t,α)=δ(t,α) and δ(t,0)=δ(t). The direct derivation of the kernel of the PT (28) the from FITI-PT (25) is elementary. Here, an indirect proof of the PT kernel is presented as follows: one can easily show [using (16) or (22) and setting x(t)=δ(t)X(ω)=1 and ϕ(ω)=0] that, δ(t)=12πcos(ωt)dω=1π0cos(ωt)dω, and h(t)=1π0sin(ωt)dω. Using these facts, from (25) we obtain, ħ(t,α)=1π0cos(ωtα)dω=1π0[cos(α)cos(ωt)+sin(α)sin(ωt)]dω, and thus (28). The kernel of the PT (28) is well-defined for all values of α [e.g., ħ(t,π/4)=12δ(t)+12h(t)] for all time including at the origin, only due to the modification presented in the HT kernel (19) by defining it zero at the origin; otherwise, mathematically it would be invalid to add the delta function with the Hilbert kernel which is so far not defined at the origin in the literature.

We can obtain the kernel of an analytic signal and compute the phase difference between the delta function δ(t) and HT kernel h(t)=δ(t,π/2) (19), as

zδ(t)=δ(t)+jh(t)=aδ(t)ejϕδ(t), where aδ(t)=δ(t)+|h(t)|, and ϕδ(t)=tan1(h(t)δ(t))=π2sgn(t)={π/2,t>0,0,t=0,π/2,t<0. (29)

Fig. 1 shows plots of the delta function δ(t) (top-left), (top-right) HT kernel h(t) (19); amplitude aδ(t) (bottom-left), and phase ϕδ(t) (bottom-right) of the kernel of the analytic signal (29) which is well-defined for all time including at the origin due to the modification presented in the HT kernel (19). It is observed that if the HT kernel (19) is not defined or has singularity at the origin, then phase ϕδ(t) would be undefined at the origin in (29). The kernel of the analytic signal can also be written as zδ(t)=δ(t)+jh(t)=1π0exp(jωt)dω or zδ(t)=limσ0+1π0exp(ω(σjt))dω=limσ0+1π(σσ2+t2+jtσ2+t2), 1πσσ2+t2dt=1, which implies (i) δ(t)=limσ0+1π(σσ2+t2) and h(t)=0, for t=0, (ii) δ(t)=0 and h(t)=1πt, for t0.

Fig. 1.

Fig. 1

Plots of the delta function δ(t) (top-left), HT kernel h(t) (top-right); amplitude aδ(t) (bottom-left), and phase ϕδ(t) (bottom-right) of the kernel of the analytic signal (29).

The arbitrary but fixed phase shifter defined in (28) is a linear time-invariant (LTI) system, thus its output can be written as the convolution of input with impulse response, i.e.,

x(t,α)=Hα{x(t)}=x(t)ħ(t,α)=cos(α)x(t)+sin(α)xˆ(t), (30)

or x(t,α)=cos(α)x(t,0)+sin(α)x(t,π/2), where x(t,0)=x(t) and x(t,π/2)=xˆ(t) as defined in (23). Clearly, the PT of a time-domain signal x(t) is another time-domain and phase shifted signal x(t,α). There are some obvious properties of the PT (30) which follow directly from the definition such as Hα=cos(α)I+sin(α)H, where I and H are the identity and HT operators, respectively, i.e., I{x(t)}=x(t) and H{x(t)}=xˆ(t)=x(t,π/2); inverse PT Hα1=Hα (or Hα1Hα=HαHα1=I); Hαm=Hmα, Hα2Hα1=Hα1Hα2=Hα1+α2. Therefore, Hαm{x(t)}=x(t,mα), Hα2{Hα1{x(t)}}=Hα2{x(t,α1)}=x(t,α1+α2), and x(t,α1+α2)=x(t) if α1+α2=2πm,mZ. Now, we explore and present the some basic properties of the proposed PT (30) as follows:

  • 1.

    Linearity: The PT is a linear operator, i.e., Hα{a1x1(t)+a2x2(t)}=a1x1(t,α)+a2x2(t,α) for arbitrary scalars a1 and a2, functions x1(t) and x2(t).

  • 2.

    The PT of a constant signal: For any constant c, Hα{c}=ccos(α).

  • 3.

    Time-shifting and time-dilation: If x(t,α) is the PT of x(t), i.e., Hα{x(t)}=x(t,α), then Hα{x(tt0)}=x(tt0,α) and Hα{x(at)}=cos(α)x(at)+sin(α)sgn(a)xˆ(at), where a0.

  • 4.
    Relation with the Fourier transform: We obtain the Fourier transform of the kernel of the PT (30) as, H(f,α)=cos(α)+sin(α)(jsgn(f)), which can be written as
    H(f,α)={ejα,f>0,cos(α),f=0,ejα,f<0, (31)
    thus the PT provides, −α and +α phase shifts to the positive and negative frequencies, respectively, and when, α=π/2, it becomes the HT. Further, if X(f) is the Fourier transform x(t), then the Fourier transform of x(t,α) is X(f,α)=X(f)H(f,α).
  • 5.
    Orthogonality: If x(t) is a real-valued energy signal (i.e., E=x(t),x(t)), then the inner product of x(t) with x(t,α) is given by
    x(t),x(t,α)=cos(α)x(t),x(t)+sin(α)x(t),xˆ(t)cos(α)=x(t),x(t,α)x(t),x(t), (32)
    as x(t),xˆ(t)=0, and thus they are orthogonal only for the phase shifts α=π2(2m+1),mZ.
  • 6.
    Energy: If x(t) is a real-valued energy signal, then x(t,α) is also a real-valued energy signal, and its energy (Eα) is computed by the inner product of x(t,α) with itself as
    Eα=x(t,α),x(t,α)=cos2(α)x(t),x(t)+sin2(α)xˆ(t),xˆ(t). (33)
    For a zero mean signal, energy is preserved in the HT, i.e., x(t),x(t)=xˆ(t),xˆ(t), so the energy is preserved in the proposed PT.
  • 7.

    Time-derivative: The PT of the derivative of signal is the derivative of the PT, i.e.,

    Hα{ddtx(t)}=ddtHα{x(t)}.

  • 8.

    PT of product of low-pass and high-pass signals: Let x1(t) be a low-pass signal such that its FT X1(f)=0 for |f|>f0 and let x2(t) be a high-pass signal with X2(f)=0 for |f|<f0. Then, PT Hα{x1(t)x2(t)}=x1(t)Hα{x2(t)}=x1(t)x2(t,α). One can easily show it using the property of HT (i.e., Bedrosian theorem [29], [30]) as H{x1(t)x2(t)}=x1(t)xˆ2(t). Thus, to obtain the PT of product of the low-pass signal and high-pass signal, only the high-pass signal needs to be phase shifted.

  • 9.

    PT of an analytic signal: From (26) and (30), we obtain the PT of analytic signal as Hα{z(t)}=x(t,α)+jxˆ(t,α)=z(t)ejα=[x(t)+jxˆ(t)][cos(α)jsin(α)] which gives Hα{z(t)}=cos(α)x(t)+sin(α)xˆ(t)+j[sin(α)x(t)+cos(α)xˆ(t)]=x(t,α)+jx(t,α+π/2). Thus, x(t,α) can be computed by considering the real part of the Hα{z(t)}.

It is to be noted that, in the literature, a generalized Hilbert transform to obtain a phase shift of α radians is well-defined only in the frequency-domain, e.g., author in study [28] defined H(f,α) as

H(f,α)={ejα,f>0,0,f=0,ejα,f<0, (34)

which is same as (31) for f0 and differs at f=0. However, the time-domain description of (34) is not available so far in the literature. In this paper, we have presented detailed study, both in the time and frequency domains, along with various properties.

Using the proposed GFR (24), the narrowband inverse Fourier transform (NB-IFT) or NBFR xH(t,λ) is hereby defined for the time-frequency representation and analysis of the signal x(t), a special case of (24) with H(ω,t)=H(ωλ) and α(ω,t)=0, as

xH(t,λ)=1π0H(ωλ)|X(ω)|cos(ωt+ϕ(ω))dω. (35)

This is a counter part of the STFT which is obtained using the forward Fourier transform of the time-windowed signal x(t)g(tτ) as

Xg(ω,τ)=x(t)g(tτ)exp(jωt)dt. (36)

Observation 3.1 (a): The proposed PT (30) is a solution of the following partial differential equation

2α2x(t,α)+x(t,α)=0, (37)

which can be considered as (a) the initial value problem with conditions x(t,0)=x(t) and αx(t,0)=xˆ(t), or (b) the boundary value problem with conditions x(t,0)=x(t) and x(t,π)=x(t) or x(t,π/2)=xˆ(t).

Observation 3.1 (b): Here, we consider three examples of the PT property 8, which can be used to obtain a unique PT and thus the HT of 2D and higher dimensional signals, as follows. First, we consider a signal with frequencies ω1,ω20 as x(t)=cos(ω1t)cos(ω2t)=12[cos(ω1t+ω2t)+cos(ω1tω2t)]=12[cos(ω1t+ω2t)+cos(ω2tω1t)]. We obtain the PT of signal x(t) as (i) x(t,α)=12[cos(ω1t+ω2tα)+cos(ω1tω2tα)]=cos(ω1tα)cos(ω2t), if ω1>ω2, (ii) x(t,α)=12[cos(ω1t+ω2tα)+cos(ω2tω1tα)]=cos(ω1t)cos(ω2tα), if ω2>ω1. Next, we consider x(t)=sin(ω1t)sin(ω2t), then we obtain the PT as (i) x(t,α)=sin(ω1tα)sin(ω2t), if ω1>ω2, and (ii) x(t,α)=sin(ω1t)sin(ω2tα), if ω2>ω1. Finally, we consider x(t)=sin(ω1t)cos(ω2t), then we obtain the PT as (i) x(t,α)=sin(ω1tα)cos(ω2t), if ω1>ω2, and (ii) x(t,α)=sin(ω1t)cos(ω2tα), if ω2>ω1. Thus, one can observe that to obtain a unique PT, in the phase argument of sin and cos, before introducing a phase α, one must consider (ω1tω2t) or (ω2tω1t) depending upon whether ω1>ω2 or ω2>ω1, respectively.

Observation 3.1 (c): If the considered signal x(t) is a complex valued function then the GFR, corresponding to the GFR of real valued function (24), can be written as

x(t,H(ω,t),α(ω,t))=12π0H(ω,t)|X(ω)|exp(j(ωt+ϕ(ω)+α(ω,t)))dω+12π0H(ω,t)|X(ω)|exp(j(ωt+ϕ(ω)α(ω,t)))dω, (38)

and thus the PTs corresponding to (25) can be easily obtained by setting H(ω,t)=1,t,ω.

3.2. PT using the Fourier sine and cosine transforms

The FCT and inverse FCT (IFCT) pairs of a signal are defined as

Xc(ω)=2π0x(t)cos(ωt)dt,ω0x(t)=2π0Xc(ω)cos(ωt)dω,t0, (39)

subject to the existence of the integrals, i.e., x(t) is absolutely integrable (0|x(t)|dt<) and its derivative x(t) is piece-wise continuous in each bounded subinterval of [0,).

The Fourier cosine quadrature transform (FCQT) x˜c(t) of the signal x(t) is defined [4] as

x˜c(t)=2π0Xc(ω)sin(ωt)dω, (40)

where

X˜c(ω)=2π0x˜c(t)sin(ωt)dt, (41)

and

X˜c(ω)={0,ω=0,Xc(ω),ω>0. (42)

The FSAS, using the signal x(t) and its FCQT x˜c(t), is defined [4] as

z˜c(t)=x(t)+jx˜c(t)=2π0Xc(ω)exp(jωt)dω, (43)

where the real part is the original signal and the imaginary part is the FQT of the real part.

The PT x(t,α) of signal x(t) using the FCT is hereby defined as

x(t,α(t,ω))=2π0Xc(ω)cos(ωtα(t,ω))dω, (44)

where α(t,ω) is the time and frequency dependent phase shift. If the phase shift is an arbitrary constant, i.e., α(t,ω)=α, then (44) can be written

x(t,α)=2π0Xc(ω)cos(ωtα)dω=cos(α)x(t)+sin(α)x˜c(t), (45)

where α is constrained to its principal value [0,2π) or [π,π) when it represents the wrapped phase, and αR in the case of unwrapped phase. Thus, we have defined the general phase shifter of signal which is a generalization of the IFCT as well as FCQT because it is (i) IFCT if α=0, and (ii) FCQT if α=π/2.

The FST and inverse FST (IFST) pairs of a signal are defined as

Xs(ω)=2π0x(t)sin(ωt)dt,x(t)=2π0Xs(ω)sin(ωt)dω, (46)

subject to the existence of the integrals. The Fourier sine quadrature transform (FSQT) x˜s(t) using the FST of signal x(t) is defined [4] as

x˜s(t)=2π0Xs(ω)cos(ωt)dω, (47)

with

X˜s(ω)=2π0x˜s(t)cos(ωt)dt, (48)

where one can observe that the both representations, defined as FST of x(t) in (46) and FCT of x˜s(t) in (48), are same for all the frequencies, i.e., Xs(ω)=X˜s(ω). The FSAS, using the signal and its FSQT, is defined [4] as

z˜s(t)=x˜s(t)+jx(t)=2π0Xs(ω)exp(jωt)dω, (49)

where the imaginary part is the original signal and the real part is the FQT of the imaginary part.

The PT x(t,α(t,ω)) of signal x(t) using the FST is hereby defined as

x(t,α(t,ω))=2π0Xs(ω)sin(ωtα(t,ω))dω. (50)

If α(t,ω)=α, then (50) can be written as

x(t,α)=2π0Xs(ω)sin(ωtα)dω=cos(α)x(t)sin(α)x˜s(t), (51)

where α[0,2π) is the phase shift introduced in the signal x(t). Thus, we have defined the general phase shifter which is a generalization of the IFST as well as FSQT because it is (i) IFST if α=0, and (ii) FSQT if α=π/2.

The FQTs, presented in (40) and (47), are different from the HT (23) by the definition itself. The proposed Fourier–Singh phase transform (FSPT) representations, defined in (44) and (50), are the effective phase shifters which can be used in various applications such as envelop detection, IF estimation, time-frequency-energy representation and analysis of nonlinear and nonstationary data.

So far, the FS, FT, FCT and FST have been used to define PT. Moreover, the STFT can also be used to define the PT for obtaining a desired phase shift in a signal under consideration. The next section presents the other form of PT using the continuous-time AWT.

3.3. PT using the continuous analytic wavelet transform

In this subsection, the continuous AWT is used to define the WPT of a signal, and it is shown that the WQT is a special case of the WPT when phase-shift is π/2 radians. The continuous wavelet transform (CWT) of a signal x(t)L2(R) is defined as [35], [36], [37]

Xψ(s,τ)=x(t),ψs,τ(t)=x(t)ψs,τ(t)dt, (52)

where asterisk denotes the complex conjugate operation. The CWT (52) can be represented in the Fourier domain, using the Parseval's relation x(t)y(t)dt=12πX(ω)Y(ω)dω, as

Xψ(s,τ)=|s|2πX(ω)Ψ(sω)ejωτdω, (53)

where ψs,τ(t)=1|s|ψ(tτs)|s|Ψ(sω)ejωτ with scaling (sR,s0) and translation (τR) parameters is a family of daughter wavelets, and ψ(t)L2(R) is a mother wavelet function which has finite energy and zero mean. The wavelet ψ(t)Ψ(ω) satisfies the admissibility condition [38]

Cψ=|Ψ(ω)|2|ω|dω<, (54)

and ensures reconstruction of the original signal x(t) using the inverse CWT defined as [35], [36], [37]

x(t)=1CψXψ(s,τ)ψs,τ(t)dτ1s2ds. (55)

If ψ(t) is an analytic wavelet, then Ψ(ω)=0 for ω<0, and (52) represents the AWT [36], [37], moreover (53) and (54) can be re-presented as

Xψ(s,τ)=|s|2π0X(ω)Ψ(sω)ejωτdω,Cψ=0|Ψ(ω)|2|ω|dω<. (56)

Now, using the analytic wavelet, we consider the real and imaginary parts of the AWT separately as follows: let ψ(t)=ψr(t)+jψi(t) be an analytic wavelet, where the imaginary part is the HT or FQT of the real part, Ψ(ω)=2U(ω)Ψr(ω), U(ω) is the unit step function, and ψs,τ(t)=ψrs,τ(t)+jψis,τ(t). Therefore, the AWT (52) for a real-valued energy signal x(t) can be written as Xψ(s,τ)=Xrψ(s,τ)jXiψ(s,τ), where Xrψ(s,τ)=x(t),ψrs,τ(t) and Xiψ(s,τ)=x(t),ψis,τ(t). Thus, we obtain

x(t)=2CψXrψ(s,τ)ψrs,τ(t)dτ1s2ds,x(t)=2CψXiψ(s,τ)ψis,τ(t)dτ1s2ds, (57)

and define the following two representations of the WQT as

xw(t)=2CψXrψ(s,τ)ψis,τ(t)dτ1s2ds,xw(t)=2CψXiψ(s,τ)ψrs,τ(t)dτ1s2ds, (58)

where xw(t)=x(t,π/2) and the subscript w indicates that the phase-shift has been obtained using the wavelet transform. Moreover, using (57) and (58), the original signal x(t) (55) can be written as

x(t)=1Cψ([Xrψ(s,τ)ψrs,τ(t)+Xiψ(s,τ)ψis,τ(t)]+j[Xrψ(s,τ)ψis,τ(t)Xiψ(s,τ)ψrs,τ(t)])dτ1s2ds,=1Cψ[Xrψ(s,τ)ψrs,τ(t)+Xiψ(s,τ)ψis,τ(t)]dτ1s2ds. (59)

Thus, using the two signal representations (57) and the two quadrature components (58), the following four representations of the WAS are here defined as

zw(t)=x(t)+jxw(t)=2CψXrψ(s,τ)ψs,τ(t)dτ1s2ds,zw(t)=x(t)+jxw(t)=2CψXψ(s,τ)ψrs,τ(t)dτ1s2dszw(t)=x(t)+jxw(t)=2CψjXψ(s,τ)ψis,τ(t)dτ1s2dszw(t)=x(t)+jxw(t)=2CψjXiψ(s,τ)ψs,τ(t)dτ1s2ds. (60)

The desired phase in the AWT (56) is introduced as

Xψ(s,τ,α(ω))=|s|2π0Ψ(sω)X(ω)ejωτejα(ω)dω. (61)

If the introduced phase in the AWT (61) is independent of frequency, i.e., α(ω)=α, then we can write Xψ(s,τ,α)=Xψ(s,τ)ejα=[cos(α)Xrψ(s,τ)sin(α)Xiψ(s,τ)]j[cos(α)Xiψ(s,τ)+sin(α)Xrψ(s,τ)], and we obtain the following two representations of an arbitrary constant WPT of the signal x(t) as

x(t,α)=2CψXrψ(s,τ,α)ψrs,τ(t)dτ1s2ds,x(t,α)=2CψXiψ(s,τ,α)ψis,τ(t)dτ1s2ds. (62)

The WPT can also be obtained from the WAS (60) by considering the real part of the following representation

zw(t,α)=[x(t)+jxw(t)]ejα=x(t,α)+jxw(t,α), (63)

where xw(t,α)=[cos(α)xw(t)sin(α)x(t)], and the WPT

x(t,α)=cos(α)x(t)+sin(α)xw(t). (64)

3.3.1. A single-integral representation of the WPT

The CWT is a redundant transform which reveals in-depth structural characteristics of the signal, and due to redundancy there may exist many ways to define the inverse. A single-integral can be used for the inverse CWT, if the signal x(t) and wavelet ψ(t) are satisfying the following two conditions [39]: (1) the signal x(t) is a real-valued function, and (2) either the wavelet ψ(t) is an even function which has a real-valued FT, or the wavelet ψ(t) is an analytic wavelet which has FT that supports only for the positive frequencies, i.e., Ψ(ω)=0 for ω<0.

If x(t) and y(t) are two finite energy signals, ψ1(t) and ψ2(t) are two wavelet functions which satisfy the two-wavelet admissibility condition, the following equality holds:

Cψ1,ψ2x(t),y(t)=Xψ1(s,τ)Xψ2(s,τ)dτ1sds, (65)

where Xψ1(s,τ)=x(t),ψ1(s,τ)(t), Xψ2(s,τ)=y(t),ψ2(s,τ)(t) and

Cψ1,ψ2=Ψ1(ω)Ψ2(ω)|ω|dω. (66)

Therefore, the two-wavelet admissibility condition is defined as

|Ψ1(ω)||Ψ2(ω)||ω|dω<. (67)

The main idea of a single-integral inverse CWT is that the admissibility condition of two-wavelet (67) can be satisfied even if either one of the wavelets is not admissible, and it can be further simplified by allowing one of the signals and wavelets to be distributions. Thus, by considering the analytic wavelet ψ1(t), real-valued signal x(t), both signal y(t) and wavelet ψ2(t) to be the delta function, one can obtain the WAS representation using the following single-integral inverse CWT as

z(t)=x(t)+jxw(t)=2Cψ1,δ0Xψ1(t,s)1sds, (68)

where the imaginary part xw(t) is the WQT of signal x(t). Therefore, if Xψ1(s,τ,α)=Xψ1(s,τ)ejα as defined in (61), then we can define

z(t,α)=x(t,α)+jxw(t,α)=2Cψ1,δ0Xψ1(t,s,α)1sds, (69)

where x(t,α) is the proposed WPT which introduces a desired phase-shift α in the signal x(t) using AWT, x(t,0)=x(t), x(t,π/2)=xw(t) and xw(t,α)=x(t,α+π/2).

Observation 3.3: We can represent a real signal as, x(t)=a(t)cos(ϕ(t)), and obtain its quadrature component, xq(t)=a(t)sin(ϕ(t)), using anyone of the AS representations (10), (22), (43), (49), (60) and (68), and obtain the following AM-FM transform, which is a special case of the proposed GFR (24), as

x(t,H(t),α(t))=H(t)a(t)cos(ϕ(t)α(t)),=H(t)a(t)cos(ϕ(t))cos(α(t))+H(t)a(t)sin(ϕ(t))sin(α(t)),=H(t)x(t)cos(α(t))+H(t)xq(t)sin(α(t)), (70)

where H(t) and α(t) are the functions of time. If H(t)=1 then, like FDTI-PT and FDTD-PT (25), (70) is the FITD-PT, and thus time-varying filtering (14) can be written as

xT(t,Hm(t),αm(t))=a0+m=1M[Hm(t)xm(t)cos(ϕm(t)αm(t))]. (71)

3.4. Implementation of the GFR using DFT

The DFT and inverse DFT (IDFT) pairs, of a signal x[n] of length N, are defined as

X[k]=1Nn=0N1x[n]exp(j2πkn/N),0kN1,x[n]=k=0N1X[k]exp(j2πkn/N),0nN1. (72)

The DFT and IDFT are computed efficiently using the FFT algorithm. Unless otherwise mentioned, we consider x[n] as a real valued signal. We obtain the following kernel of a discrete-time PT corresponding to the kernel of the continuous-time PT (28) as

ħ[n,α]=cos(α)δ[n]+sin(α)h[n], (73)

where ħ[n,α]=δ[n,α], δ[n,0]=δ[n], and ħ[n,π/2]=h[n]=δˆ[n]=δ[n,π/2]=(1cos(πn))πn is the discrete-time HT kernel. Thus, a constant phase shift in the signal x[n] is represented as

x[n,α]=cos(α)x[n]+sin(α)xˆ[n], (74)

where x[n]=x[n,0] and xˆ[n]=x[n,π/2]. The PT defined in (7) and (25) can be computed using the FFT by considering the real part of the following analytic signal

z[n,αk]=x[n,αk]+jx[n,αk+π/2]=IFFT{X[k]H[k]}, (75)

where x[n,αk]=Re{z[n,αk]}, and H[k] is defined as

H[k]={cos(αk),k=0,N/2,2exp(jαk),1kN/21,0,(N/2+1)kN1, (76)

if N is an even number, and

H[k]={cos(αk),k=0,2exp(jαk),1k(N1)/2,0,((N1)/2+1)kN1, (77)

if N is an odd number. For a constant phase shift, the imaginary part is the HT of the real part in (75), and if αk=0,k, the real part is the original signal x[n]. In (76) and (77), we can remove the multiplication factor of 2 and define H[k]=exp(jαk) for positive frequencies (1kN/21), and H[k]=exp(jαk) for negative frequencies ((N/2+1)kN1).

Observation 3.3.1: The phase shift of a sinusoidal signal does not change its amplitude (or energy), except for the lowest (i.e., DC) and highest frequency components. For example, if x[n]=c+cos(πn), then x[n,α]=cos(α)(c+cos(πn)); and for α=π/2, it becomes zero which cannot be recovered by further phase shift. Therefore, to overcome this issue, we can define the phase shift of a constant signal and highest frequency component as z[n,α]=(c+cos(πn))ejα=cejα+ej(nπα), which preserves the energy in these cases as well. Moreover, in the case of π/2 phase shift, the complete signal is getting transferred to the imaginary signal, and one can recover the original signal by further phase shift of (α+2mπ), and considering x[n,α]=Re{z[n,α]}. This is also consistent with the complex plane representation, where multiplication of ±j with a complex number (z=a+jb) introduces ±π/2 phase shift (e.g., jz=jab) without any change in the amplitude. Therefore, one can use H[k]=ejα for k=0,N/2 in (76) and (77).

We observe that the delayed signal x[nn0] can be obtained using the IDFT (72) as

x[nn0]=X[0]+k=1N1X[k]exp(j2πk(nn0)/N), (78)

which is valid only for an integer value of n0, because the complex conjugate of exp(j2πkn0N) is exp(j2π(Nk)n0N) only for some integer value of n0, and it is not valid for a fractional delay. Therefore, in order to obtain both the integer and fractional delays nkR in the signal x[n], corresponding to (11) which can be computed by (75), we define H[k] as

H[k]={1,k=0,2exp(j2πknk/N),1kN/21,exp(jπnk),k=N/2,0,(N/2+1)kN1, (79)

where N is an even number (similarly it can also be defined when N is an odd number), and for some constant delay n0 one can set nk=n0,k.

The μ-th order derivative of the signal x[n], denoted as xμ[n], corresponding to (12), can be estimated by

xμ[n]=a0(n/Fs)μΓ(1μ)+Re[IFFT{X[k]H[k]}] (80)

using H[k] which is defined as

H[k]={0,k=0,2(2πk/N)μexp(jμπ/2),1kN/21,(π)μexp(jμπ/2),k=N/2,0,(N/2+1)kN1, (81)

where mean-value a0=n=0N1x[n]/N, μ0, and N is an even number (similarly it can also be defined when N is an odd number); and when μ0 then it is the μ-th order integral of a signal as defined in (13).

The HT of discrete signal x[n] can be obtained by the inverse DTFT as

xˆ[n]=12πππjsgn(Ω)X(Ω)exp(jΩn)dΩ, (82)

where X(Ω)=n=x[n]exp(jΩn) is the DTFT of signal x[n]. Practically, the HT (82) is implemented by discretization in frequency with the DFT and inverse DFT using the FFT algorithm. This approach provides an error convergence that is polynomially decreasing with the size of the DFT points. To obtain better error convergence, there are many HT approximation methods such as (i) the HT based on the sinc basis functions expansion [47], [48], [49], [50], and (ii) the HT approximation based on an expansion in rational eigenfunctions of the HT operator [89]. An approximation of the HT in terms of the sinc functions can be easily obtained as follows. The HT (82) can be written using the convolution operation as xˆ[n]=x[n]h[n], and thus

xˆ[n]=m=x[m](1cos(π(nm))π(nm))=π2m=x[m](nm)sinc2(nm2), (83)

where sinc(x)=sin(πx)/(πx). If the signal is time limited with length N, i.e., x[n] is non-zero for n[0,N1] and zero otherwise, then (83) can be written as

xˆ[n]=π2m=0N1x[m](nm)sinc2(nm2). (84)

Because an accurate estimation of the PT depends on the accuracy of the HT, therefore these HT approximation methods can be used in (74) to compute the PT of a signal.

3.5. Implementation of the GFR using DCTs/DSTs

Let x[n] be a finite energy signal of a length N. The DCT type-2 (DCT-2) of the signal x[n] is defined as [1]

Xc2[k]=2Nσkn=0N1x[n]cos(πk(2n+1)2N),0kN1, (85)

and the inverse DCT (IDCT) is obtained by

x[n]=2Nk=0N1σkXc2[k]cos(πk(2n+1)2N),0nN1, (86)

where normalization factors σk=12 for k=0, and σk=1 for k0. If the consecutive samples of the sequence x[n] are correlated, then DCT concentrates energy in a few Xc2[k] and decorrelates them. The DCT basis sequences, cos(πk(2n+1)2N), which are a class of discrete Chebyshev polynomials [1], form an orthogonal set as inner product cos(πk(2n+1)2N),cos(πm(2n+1)2N)=0 for km.

The discrete Fourier cosine quadrature transform (FCQT), x˜c2[n], of the signal x[n] is defined as [4]

x˜c2[n]=2Nk=0N1σkXc2[k]sin(πk(2n+1)2N),0nN1, (87)

where Xc2[k] is the DCT-2 as defined in (85). The PT using the DCT-2 is hereby defined as

x[n,α(k)]=2Nk=0N1σkXc2[k]cos(πk(2n+1)2Nα(k)),0nN1, (88)

and if α(k)=α, then (88) can be written as

x[n,α]=cos(α)x[n]+sin(α)x˜c2[n]. (89)

Thus, using the PT (88), variable fractional time-delays in the signal x[n] can be introduced as x[nnk]=x[n,α(k)], where α(k)=πknk/N, nkR, and for a constant fractional time-delay nk=n0,k.

Considering eight-types of the DCTs and eight-types of the DSTs, sixteen-types of the Fourier quadrature transforms (FQTs) and corresponding Fourier-Singh analytic signal (FSAS) representations are well defined [4]. Therefore, sixteen-types of the PTs can be derived from these FQTs and FSAS representations. So far, we have presented the PT, in (88) and (89), using only the DCT-2 and corresponding FQT. Accordingly, we next consider and define all sixteen-types of the PTs as follows. Using the standard DCTs and DSTs transform matrices, presented in Appendix B (97) and (98), the following sixteen FQTs (i.e., eight FCQTs, x˜ci, and eight FSQTs, x˜si) and corresponding sixteen FSAS representations (FSASRs), z˜ci and z˜si, for i=1,2,,8, are defined in [4] as

Xci=Cix;x=CiTXci;(DCTs and IDCTs)x˜ci=S˜iTXci=S˜iTCix;z˜ci=x+jx˜ci;(FCQTs and FSASRs)Xsi=Six;x=SiTXsi;(DSTs and IDSTs)x˜si=C˜iTXsi=C˜iTSix;z˜si=x˜si+jx;(FCQTs and FSASRs), (90)

where Xci=[Xci[0]Xci[1]Xci[N1]]T and Xsi=[Xsi[0]Xsi[1]Xsi[N1]]T are the DCT and DST of i-th type of the signal x=[x[0]x[1]x[N1]]T, respectively. Hence, the linear transformations of x into x˜ci, and x into x˜si are defined with the transformation matrices S˜iTCi and C˜iTSi, respectively. Thus, the following sixteen PTs, using the sixteen types of FQTs and FSAS representations, are defined as

xci(α)=Re{ejαz˜ci}=cos(α)x+sin(α)x˜ci,1i8,xsi(α)=Im{ejαz˜si}=cos(α)xsin(α)x˜si,1i8. (91)

4. Results and discussions

This section presents simulation results to demonstrate efficacy of the proposed GFRs and PTs. We mainly consider those signals which have been widely used in the literature for performance evaluation and comparison of the results among the proposed and other existing methods.

Example 1: First, in this example, the kernel of the PT for a continuous-time periodic signal is derived as follows. If the periodic signal is the Dirac comb (impulse train) function, from (1) we can write

δT(t)=k=δ(tkT)=1T+2Tk=1cos(2πkf0t),<t<, (92)

and, therefore, obtain the phase transform as

δT(t,α)=cos(α)1T+2Tk=1cos(2πkf0tα),<t<,δT(t,α)=cos(α)δT(t)+sin(α)δT(t,π/2),<t<, (93)

where δT(t,π/2) is the periodic HT of the impulse train and defined as

δT(t,π/2)=2Tk=1sin(2πkf0t)={0,t=±m2f0,mZ,1Tcot(πf0t),t±m2f0, (94)

where m is an integer and Z denotes the set of all integers.

The impulse train function (92) and its HT (94) are defined in the sense of distributions only because sum of these series does not converge which can be easily shown as:

1T+2Tk=1Nexp(j2πkf0t)=1T+2Texp(jπ(N+1)f0t)sin(πNf0t)sin(πf0t), and thus

δT(t)=1T+limN2Tk=1Ncos(2πkf0t)=limN1Tsin(π(2N+1)f0t)sin(πf0t)=limN(2N+1)Tsinc(π(2N+1)f0t)sinc(πf0t),δT(t,π/2)=limN2Tk=1Nsin(2πkf0t)=limN1T[cot(πf0t)cos(π(2N+1)f0t)sin(πf0t)]={0,t=±m2f0,mZ,1Tcot(πf0t),otherwise. (95)

By considering the sum of the series, 2Tk=1exp[(σj2πf0t)k]=2Texp[(σj2πf0t)]1exp[(σj2πf0t)], for σ>0, and then taking the imaginary part of the sum as limiting case, one can obtain (95) as, Im{limσ02Texp[(σj2πf0t)]1exp[(σj2πf0t)]}=1Tcot(πf0t).

We can easily obtain the HT kernel (19) for the non-periodic signals from the periodic one (95), as expected, by contemplating the limits, limTlimf00, as follows: limT1Tcot(πf0t)=limf00f0cos(πf0t)sin(πf0t) which can be written as limf001πt×cos(πf0t)[sin(πf0t)/(πf0t)]=1πt,t0, and therefore,limTδT(t,π/2)=h(t), as defined in (19). Using the series expansion of the cotangent function (i.e., cot(x)=1xx3x3452x5945x74725, for 0<x<π/2), we obtain the inequality as, 1Tcot(πf0t)<1πt, for 0<t<T2.

It is pertinent to note, contrary to perception in the literature, that the Hilbert kernels, presented in (19), (20), and (95), are well-defined and possess zero rather than pole at the origin, which is consistent with (i) the Hilbert kernel in discrete-time domain, and (ii) the definition of an odd function which is zero at the origin, if it is defined or limit exists at the origin.

Example 2: Next, in this example, we consider the Gravitational wave event GW150914 data and perform analysis, i.e., noise removal, IF and the Hilbert spectrum (HS) or TFE estimation using the proposed methodology. Albert Einstein predicted the Gravitational waves (GWs) in 1916 which are ripples in the space-time continuum and travel outward at the speed of light from a source of origin. A binary black hole merger event, nearly 1.3 billion light years away, generated the GW event GW150914 [43], marks one of the most-important scientific inventions in the history of human life. Analysis of the GWs reveals information about source or cosmic-event that produces ripples in the space-time continuum.

Using the proposed time-varying filtering (TVF) (14), we consider the noise removal, IF estimation and HS representation of the GW150914 data, recorded by the laser interferometer Gravitational-wave observatory (LIGO) with sampling rate Fs=16384 Hz, which is publicly available for downloading at [44]. Frequency of this GW150914 signal sweeps upwards from 35 Hz to 250 Hz and amplitude-strain increases to peak GW strain of 1.0×1021 [43]. As the IF of a GW signal can be used to estimate many parameters such as primary mass and secondary mass, luminosity distance, total mass, chirp mass, separation, effective spin and velocity of binary black hole merger [43], therefore, noise removal and an accurate IF estimation from the GW time-series is of paramount importance.

Fig. 2 (a) presents the GW150914 H1 strain time-series (top figure), recorded at the LIGO Hanford, that is heavily corrupted with noise. The Fourier spectrum, shown in the bottom of Fig. 2(a), of this GW time-series is not able to reveal the non-stationarity (i.e., upwards sweep or increase in the frequency and amplitude with time) inherently present in the signal. The TFE representation of the GW150914 H1 time-series without decomposition using the FDM is shown in Fig. 2(b) which clearly shows the signal frequency is increasing with time, however, due to noise there are lots of spurious fluctuations in the time-frequency plane. The GW time-series is decomposed using the FDM into a set of six FIBFs [FIBF1 (25–50) Hz, FIBF2 (50–100), FIBF3 (100–200), FIBF4 (200–400), FIBF5 (400–800), FIBF6 (800–8192) Hz] and a low frequency component (LFC) of band 0–25 Hz which are presented in Fig. 2(c). In Fig. 2(d), the top five FG1-FG5 components are obtained by multiplying the time-domain Gaussian window function with corresponding FIBFs (FIBF1-FIBF5), and the bottom graph presents the reconstructed GW waveform which is obtained by the addition of the five FG1-FG5 components. In reconstruction of the GW waveform, the LFC and highest frequency FIBF6 have not been considered as they represent out of band noise components present in the GW150914 H1 time-series. The following time-domain Gaussian windows have been used for the TVF

H1(t)={1,0t<μ1,exp((tμ1)2/2σ12),μ1tT,Hm(t)=exp((tμm)2/2σm2),0tT and m=2,3,4,H5(t)={exp((tμ5)2/2σ52),0t<μ5,1,μ5tT, (96)

where μm is time corresponding to the peak value of FIBF and σm2 is variance of the corresponding FIBF denoted by xm(t) in (14).

Fig. 2.

Fig. 2

The event GW150914 H1 strain time-series, recorded at LIGO Hanford, analysis using the proposed TVF (Example 2): (a) The GW H1 strain time-series (top figure) [43], and its Fourier spectrum (bottom figure), (b) the Hilbert spectrum (TFE) without any decomposition, (c) Decomposition of the GW time-series into a set of FIBFs (FIBF1-FIBF6) and low frequency component (LFC), (d) the proposed TVF produces top five components, FG1-FG5, by multiplication of the FIBF1-FIBF6 with corresponding Gaussian time-windows, and the reconstructed GW (bottom) by sum of FG1-FG5. (For interpretation of the colors in the figure(s), the reader is referred to the web version of this article.)

Fig. 3 shows the further analysis, comparison, residue and HS estimation of the GW150914 H1 strain time-series using the proposed TVF (Example 2): (a) the GW150914 H1 strain time-series (top red dotted line), the proposed reconstructed waveform (top blue solid line), and an estimated difference between them, i.e., residue waveform (bottom); (b) the numerical relativity (NR) time-series [43] (top red dash-dot line), the reconstructed waveform (top blue solid line), and difference between the reconstructed and NR time-series (bottom); the HS estimates of the reconstructed and NR time-series are shown in (c) and (d), respectively. The exactly same analysis of the event GW150914 L1 strain time-series, recorded at LIGO Livingston, using the proposed TVF is also presented in Fig. 4 .

Fig. 3.

Fig. 3

The event GW150914 H1 strain time-series analysis using the proposed TVF (Example 2): (a) the GW150914 H1 strain time-series (top red-dashed line), the reconstructed time-series (top blue solid line), and difference between the original and reconstructed (i.e., residue) signal (bottom), (b) Reconstructed time-series (top blue solid line) and numerical relativity (NR) time-series (top red dashed line), and difference between the reconstructed and NR time-series (bottom), (c) HS (TFE) estimates of the reconstructed time-series, and (d) HS (TFE) estimates of the NR time-series.

Fig. 4.

Fig. 4

The event GW150914 L1 strain time-series, recorded at LIGO Livingston, analysis using the proposed TVF (Example 2): (a) GW150914 L1 strain time-series (top red dotted line), reconstructed signal (top blue solid line), and difference (i.e., residue) signal (bottom), (b) Reconstructed time-series (top blue solid line) numerical relativity (NR) time-series (top red dash-dot line), and difference between the reconstructed and NR time-series (bottom), (c) HS (TFE) estimates of the reconstructed time-series, and (d) HS (TFE) estimates of the NR time-series.

The two detectors at Hanford and Livingston are about 3000 km apart which corresponds to the maximum time-delay of 10 ms and phase-shift of π radians. The time-delay and phase-shift estimation using the cross-correlation between the Hanford and Livingston waveforms are shown in Fig. 5 : (a) the cross-correlation between the H1 and L1 NR signals, (b) the cross-correlation between the reconstructed H1 and L1 NR signals. The cross-correlations, in the both cases, are almost symmetric with respect to negative peak. The estimated time-delays are 7.5 ms and 7.4 ms, which are very close and differs only by 0.1 ms, from the NR signals and reconstructed signals, respectively, and there is a phase-shift of π radians. These time-delay and phase-shift are due to relative positioning of the detectors and propagation delay of the observed GW signals at the two observatories. As the GWs are propagating at the speed of light, the time-delay of 7.5 ms corresponds to an effective distance of ≈2250 km between two observatories, which presents the first detection in Livingston and followed by Hanford, and suggests that the GW150914 event originated from the direction of Southern Hemisphere [45]. The negative peaks in the cross-correlations (in both cases) and very beginning of the signals in reconstructed wave-forms in Fig. 5(b) are the evidences of the phase shift of π radians which is expected in between the observed GWs due to the relative positioning of the two detectors. Fig. 5(c) presents the H1 reconstructed GW advanced by 7.5 ms and the L1 reconstructed GW shifted by π radians (top), and corresponding symmetric cross-correlation (bottom). Fig. 5(d) shows the H1 reconstructed GW, L1 reconstructed GW delayed by 7.5 ms and shifted by π radians (top), and corresponding cross-correlation (bottom) which is symmetric around the origin with positive peak. The presented example clearly demonstrates the efficacy of the proposed TVF for the analysis of real-life non-stationary signals, data and other time-series.

Fig. 5.

Fig. 5

Time-delay and phase-shift estimation using the cross-correlation between Hanford and Livingston waveforms (Example 2): (a) the cross-correlation between the H1 and L1 NR signals, (b) the cross-correlation between the reconstructed H1 and L1 NR signals. The cross-correlations, in the both cases, are almost symmetric with respect to negative peak. The estimated time-shift of 7.5 ms (7.4 ms) and phase-shift of π radians are due to relative positioning of the detectors and propagation delay of the observed GW signals at the two observatories, (c) H1 reconstructed GW advanced by 7.5 ms and L1 reconstructed GW shifted by π radians (top), and corresponding cross-correlation (bottom), (d) H1 reconstructed GW, L1 reconstructed GW delayed by 7.5 ms and shifted by π radians (top), and corresponding cross-correlation (bottom).

Example 3: In this example, the phase shift analysis is considered for a Gaussian function, x(t)=e(t2.5)2, 0t<5 with sampling frequency Fs=1000 Hz, which is shown in Fig. 6 , where in the direction of arrow (a) Phase in the range of [0,π] radians is increasing in step of π/20 radians, first plot is original Gaussian function and last one corresponds to π radians phase-shift, plot corresponding to the tip of arrow is the HT of original signal, i.e., π/2 radian phase shift, (b) Phase in the range of [π,2π] radians is increasing in step of π/20 radians, first plot corresponds to π radians phase-shift and the last one is the original Gaussian function obtained with 2π phase shift, plot corresponding to the tip of arrow is the HT of original signal with minus sign, i.e., 3π/2 radians phase shift, (c) Phase shift in the range of [0,2π] using the DFT which is obtained by combining (a) and (b); (d) Phase shift in the range of [0,2π] using the DCT. It is observed that there is no difference, in phase shift obtained by the DFT (30) and DCT (45) approaches, for a set of signals which represent same underlying periodic extension that inherently present in the DFT (N-sample periodicity) and DCT (2N-sample periodicity with even symmetry) representations.

Fig. 6.

Fig. 6

Phase shift analysis of a Gaussian function of Example 3: (a) Phase in the range of [0,π] is increasing in a step of π/20, in the direction of arrow, first plot is original Gaussian function and last one corresponds to π phase-shift, plot corresponding to the tip of arrow is the HT of original signal, i.e., π/2 phase shift, (b) Phase in the range of [π,2π] is increasing in a step of π/20, in the direction of arrow, first plot corresponds to π phase-shift and last one is the original Gaussian function obtained by 2π phase shift, plot corresponding to the tip of arrow is the HT of original signal with minus sign, i.e., 3π/2 phase shift. Phase shift in the range of [0,2π] using the DFT (c), and using the DCT (d).

Example 4: Fig. 7 presents the phase shift analysis of a sine function, x(t)=sin(2πt), with 0t<1 and Fs=1000 Hz, where phase is increasing in step of π/10 radians: (a) using the DFT and (c) using the DCT with phase in the range of [0,π], in the direction of arrow, first plot is original sine function and last one corresponds to π phase-shift, plot corresponding to the tip of arrow is the π/2 phase shift; (b) using the DFT and (d) using the DCT with phase in the range of [π,2π], in the direction of arrow, first plot is π phase shifted sine wave and last one corresponds to 2π phase-shift, plot corresponding to the tip of arrow is 3π/2 phase-shift; (e) using the DFT which is obtained by superimposing (a) and (b); and (f) using the DCT which is obtained by superimposing (c) and (d). We observe clear differences, in phase shift obtained by the DFT (30) and DCT (45) approaches, for a set of signals which yield one periodic signal for the DFT (N-sample periodicity) representation and another periodic signal for the DCT (2N-sample periodicity with even symmetry) representation.

Fig. 7.

Fig. 7

Phase shift analysis of the sine function of Example 4 where phase is increasing in step of π/10: using DFT (a) and DCT (c) with phase in the range of [0,π], in the direction of arrow, first plot is original sine function and last one corresponds to π phase-shift, plot corresponding to the tip of arrow is π/2 phase shift; using the DFT (b) and DCT (d) with phase in the range of [π,2π], in the direction of arrow, first plot is π phase shifted sine function and last one corresponds to 2π phase-shift, plot corresponding to the tip of arrow is 3π/2 phase shift; using the DFT (e) and DCT (f) with phase in the range of [π,2π].

Example 5: This example considers a Gaussian function x(t)=e(t5)2, with 0t<10, Fs=1/T=10 Hz, and thus x[n]=e(nT5)2. We computed true delayed signal as x(tt0)=e(tt05)2, and delayed signal x[nn0] using the proposed method (79) with fractional delay t0=n0T sec, where n0=0.9. Fig. 8 shows the fractional delay estimation of this Gaussian function: (upper) the original Gaussian function and its delayed version obtained theoretically using the expression x(tt0)=e(tt05)2, (middle) the original Gaussian function and its delayed version obtained by the proposed method (79), and (lower) an estimated error, which is really small in the order of 1011, by taking the difference between truly delayed signal and delayed signal obtained by the proposed method.

Fig. 8.

Fig. 8

Fractional delay analysis of a Gaussian function of Example 5: (upper) the original Gaussian function and its delayed version obtained theoretically, (middle) the original Gaussian function and its delayed version obtained by the proposed method, and (lower) an error estimated by taking the difference between truly delayed signal and delayed signal obtained by the proposed method.

We consider a cos function x(t)=cos(πt), with 0t<100, Fs=1/T=1 Hz, and thus x[n]=cos(πn). We computed true delayed signal as x(tt0)=cos(π(tt0)), and delayed signal x[nn0] using the proposed method (79) with the fractional delay t0=n0T sec, where n0=0.7. Fig. 9 shows the fractional delay estimation of this cos function: (upper) the original cos function and its delayed version obtained theoretically using expression x(tt0)=cos(π(tt0)), (middle) the original cos function and its delayed version obtained by the proposed method (79), and (lower) an estimated error, which is really small in the order of 1014, by taking the difference between truly delayed signal and delayed signal obtained by the proposed method.

Fig. 9.

Fig. 9

Fractional delay analysis of cos(πt) function of Example 5: (upper) the original cos function and its delayed version obtained theoretically, (middle) the original cos function and its delayed version obtained by the proposed method, and (lower) an error estimated by taking the difference between truly delayed signal and delayed signal obtained by the proposed method.

Example 6: This example presents the fractional order derivative (FOD) and fractional order integral (FOI) of a sine function, x(t)=sin(2πt), with 0t<10 and Fs=1000 Hz. Fig. 10 and Fig. 11 present the FOD and FOI, respectively, of the sine wave where fractional order μ{0.0,0.25,0.5,0.75,1.0}; FOD and FOI are estimated using the proposed method (80).

Fig. 10.

Fig. 10

Estimation of the fractional derivative of order 0.0,0.25,0.5,0.75 and 1.0 of the sine function of Example 6 by the proposed method.

Fig. 11.

Fig. 11

Estimation of the fractional integral of order 0.0,0.25,0.5,0.75 and 1.0 of the sine function of Example 6 by the proposed method.

Example 7: In this example, a cosine wave is considered and desired phase shift is obtained using the proposed WPT and WQT. Fig. 12 presents the phase shift analysis of a cosine function, x(t)=cos(2πt), with 0t<5 and Fs=1000 Hz, where phase is increasing in step of π/10 radians using the proposed WPT with the Morse wavelet: (a) top plot with phase in the range of [0,π], in the direction of arrow, first plot is the original cosine function and last one corresponds to π phase-shift, plot corresponding to the tip of arrow is the π/2 phase shift, i.e., the WQT; (b) bottom plot has phase in the range of [π,2π], in the direction of arrow, first plot is π phase shifted cosine wave and last one corresponds to 2π phase-shift, plot corresponding to the tip of arrow is 3π/2 phase-shift in the original signal.

Fig. 12.

Fig. 12

Phase shift analysis of the cosine function of Example 7 where phase is increasing in step of π/10: using the proposed WPT with the Morse wavelet (a) top plot has phase in the range of [0,π], in the direction of arrow, first plot is the original cosine function and last one corresponds to π phase-shift, plot corresponding to the tip of arrow is π/2 phase shift; (b) bottom plot has phase in the range of [π,2π], in the direction of arrow, first plot is π phase shifted cosine function and last one corresponds to 2π phase-shift, plot corresponding to the tip of arrow is 3π/2 phase shift.

In this study, we have considered many examples and demonstrated the efficacy of the proposed methodologies. However, it may be noted that Fourier methods provide better error convergence when function is smooth, error increases when there are discontinuities in the considered function, and error also changes with respect to the size of the DFT implementation [51], [52]. More specifically, error for smooth functions reduces exponentially in the L2 and L norms. However, the most significant issue is that non-smooth functions suffer from spurious high frequency oscillations near discontinuities, error convergence is polynomial and, in the worst case, error converses slower than the linear function. In particular, the L2-norm of the error for a discontinuous function converges at a sub-linear rate and the L-norm does not converge at all [68] due to manifestation of the Gibbs phenomenon [69]. Because of the maximal efficacy and efficiency of Fourier methods, fortunately many studies such as filtering [70], reprojection [71], [72], [73] and mollification [68], [74] have been proposed to resolve the Gibbs phenomenon in the non-smooth scenario and recover the exponential error convergence which makes Fourier methods very useful. Thus, these recovery methods and the proposed one may be combined to obtain better error convergence in the case of non-smooth functions.

5. Conclusion and future scope

This work introduced the generalized Fourier representation (GFR), which is completely based on the Fourier representation of a signal, and presented seven special cases of the GFR, namely the Fourier representation, phase transform (PT), time-delay including fractional delay of discrete-time signals, time derivative and integral including fractional order, analog and digital modulations, and filtering operations. The most important and fundamental contribution of this study is the PT which is a special case of the GFR and a true generalization of the Hilbert transform. Using the proposed PT, the desired phase-shift and time-delay can be obtained in a signal under analysis. The kernel of the PT is derived to obtain any constant phase shift, the various properties of the PT are discussed, and it is shown that the HT is a special case of the PT when phase-shift is π/2 radians. An extension of the one-denominational PT is also provided for the two-dimensional image signals, which can easily be extended for higher dimensional signals. Using the PT, it is demonstrated that (i) a constant phase shift (e.g., π/2 phase shift) in a signal corresponds to variable time-delays in all the harmonics, (ii) a frequency dependent phase shift in all the harmonics of the Fourier representation can be used to obtain a constant time-delay in a signal, (iii) a constant phase shift is same as the constant time-delay only for a single frequency sinusoid. Contrary to perception in the literature, it is demonstrated that the kernel of the Hilbert transform in continuous time-domain has a zero rather than a pole at the origin. The narrowband Fourier representation, for the time-frequency representation and analysis, of a signal is also obtained using the proposed GFR.

The time derivative and time integral, including fractional order, of a signal are obtained using the GFR. The DCT based implementation is proposed to avoid end artifacts due to discontinuities present in the both ends of a signal. A new method is proposed to obtain a fractional delay in a discrete-time signal using the Fourier representations, i.e., DFT, DSTs and DCTs. The FFT implementations of all the proposed representations are also developed. Using the analytic wavelet transform (AWT), the wavelet phase transform (WPT) is proposed to introduce a desired phase-shift in a signal under-analysis, and two representations of wavelet quadrature transform (WQT) are presented as special cases of the WPT where phase-shift is π/2 radians.

In this study, we have considered the limited number of examples for error estimations, noise removal, real-time data analysis and processing using the proposed methods. The future directions of research would be to consider and apply the proposed methods in various applications such as biomedical signals electrocardiogram and electroencephalogram processing, seismic data analysis, image processing, audio and other sound signal representation and noise removal. Other interesting future scopes of the study are (1) to perform a detailed mathematical (e.g., pointwise and Lp norm, 1<p<) convergence analysis of the proposed GFRs, (2) to consider and study error convergence in non-smooth functions with the filtering and mollification methods, (3) to investigate and define the generalized fractional Fourier representations using the fractional Fourier transform (FrFT) and inverse FrFT, and explore its various special cases.

Ethics statement

This study did not involve any active collection of human data.

Data accessibility statement

Gravitational wave data is publicly available, and no other data is used in this study which cannot be generated by MATLAB/Python or any other programming language.

Funding

There is no funding to support this research.

Permission to carry out fieldwork

No permissions were required prior to conducting this research.

Declaration of Competing Interest

The author declare that they have no known competing financial interests or personal relationships that could have appeared to influence the work reported in this paper.

Acknowledgements

Author would like to express his sincere appreciation to the anonymous reviewers for their valuable suggestions. Author also would like to show his gratitude to the Prof. SD Joshi (IIT Delhi) for discussions and sharing wisdom and expertise during this research.

Biography

graphic file with name fx001_lrg.jpg

Pushpendra Singh earned B.E. (Hons.) from Govt. Engineering College Rewa, MP (India), M.Tech. from Indian Institute of Technology Kanpur (IITK), and Ph.D. from Indian Institute of Technology Delhi (IITD). Currently, he is working as an Assistant Professor at Department of Electronics and Communication Engineering, National Institute of Technology Hamirpur, HP (India). He has published many papers in reputed international journals such as “Journal of The Franklin Institute”, “Proceedings of the Royal Society A”, “Royal Society Open Science” and “IEEE Transactions on Neural Systems and Rehabilitation Engineering”. His main areas of research include signal/data modeling, simulation and analysis; Machine learning, deep learning and AI; Image processing; Time-Frequency analysis; Signal processing applications; Biomedical signal processing; Non-linear and non-stationary data analysis; Numerical methods; Modeling and prediction of COVID-19 pandemic; and Fourier decomposition methods.

Appendix A. Acronyms and symbols

The list of acronyms, symbols and parameters used in this study is presented in Table 1 and Table 2 , respectively.

Table 1.

List of acronyms.

FR Fourier representation GFR Generalized FR
PT/FSPT Phase transform/Fourier–Singh phase transform WPT Wavelet phase transform
AWT Analytic wavelet transform HT Hilbert transform
FS Fourier series FT Fourier transform
FST Fourier sine transform FCT Fourier cosine transform
DTFT Discrete-time Fourier transform DTFS Discrete-time Fourier series
DFT Discrete Fourier transform DST Discrete sine transform
STFT Short-time Fourier transform NBFR Narrowband Fourier representation
DCT Discrete cosine transform FFT Fast Fourier transform
MDCT Modified DCT GAS Gabor analytic signal
TFE Time-frequency-energy IF Instantaneous frequency
FQT Fourier quadrature transform FSAS Fourier-Singh analytic signal
AM Amplitude modulation FM Frequency modulation
WT Wavelet transform WAS Wavelet analytic signal
AWF Analytic wavelet function FDTD Frequency- and time-dependent
WQT Wavelet quadrature transform FITI Frequency- and time-independent
AS Analytic signal FDTI Frequency-dependent and time-independent
LFC Low frequency component FITD Frequency-independent and time-dependent
FDM Fourier decomposition method FIBFs Fourier intrinsic band functions
TVF Time-varying filtering IDCT Inverse DCT
IFT Inverse FT IFCT Inverse FCT
FCQT Fourier cosine quadrature transform IFST inverse FST
FSQT Fourier sine quadrature transform CWT Continuous wavelet transform
IDFT Inverse DFT IFFT Inverse FFT
FSASRs FSAS representations IDST Inverse DST
GW Gravitational wave NR Numerical relativity
FOD Fractional order derivative FOI Fractional integral
HS Hilbert spectrum LIGO Laser interferometer gravitational-wave observatory

Table 2.

List of symbols and parameters.

xT(t) A periodic signal with period T ω0 (f0) Fundamental frequency in rad/s (Hz)
ak, bk k-th coefficients in FS representation Xk k-th coefficient in FS representation
ϕk k-th phase of complex coefficient Hk(t) k-th amplitude scaling function
αk(t) k-th phase scaling function X(ω) Fourier transform of a signal x(t)
H(ω,t) Amplitude scaling function α(ω,t) Phase scaling function
sgn(t) Signum (sign) function δ(t) Dirac delta function
h(t) Kernel of Hilbert transform δ(t,α) α phase shift in delta function
ħ(t,α) Kernel of phase transform δˆ(t) Hilbert transform of delta function
z(t) Analytic signal x(t,α) α phase shift in a function x(t)
xH(t,λ) NBFR of a function x(t) Xg(ω,τ) STFT of a function x(t)
ψ(t) Mother wavelet function Hα Phase transform operator
zδ(t) Kernel of analytic signal H(f,α) Fourier transform of the PT kernel
Eα Energy of a signal x(t,α) Xc(ω) Fourier cosine transform of a signal x(t)
x˜c(t) FCQT of a signal x(t) Xs(ω) Fourier sine transform of a signal x(t)
x˜s(t) FSQT of a signal x(t) Xψ(s,τ) Wavelet transform of a signal x(t)
s,τ Scaling and translation parameters ψs,τ(t) A family of wavelet daughters
xw(t) WQT of a signal x(t) zw(t) Wavelet analytic signal
X[k] DFT of a signal x[n] ħ[n,α] Kernel of discrete-time phase transform
Xc2[k] DCT-2 of a signal x[n] x˜c2[n] FCQT of a signal x[n]
Ci i-th DCT matrix S˜i i-th FQT matrix corresponding to Ci
Si i-th DST matrix C˜i i-th FQT matrix corresponding to Si

Appendix B. Transform matrices of DCTs and DSTs

The standard sixteen DCTs and DSTs transform matrices (Ci and Si for i=1,2,,8, with their nk-th element, denoted by (Ci)nk and (Si)nk, respectively) are defined in [2] as

(C1)nk=aγnγkcos(nkπN1),(C2)nk=bσkcos[(n+12)kπN],(C3)nk=bσncos[(k+12)nπN],(C4)nk=bcos[(n+12)(k+12)πN],(C5)nk=cσnσkcos(nk2π2N1),(C6)nk=cεnσkcos[(n+12)k2π2N1],(C7)nk=cεkσncos[(k+12)n2π2N1],(C8)nk=dcos[(n+12)(k+12)2π2N+1],(S1)nk=bsin(nkπN),(S2)nk=bεksin[(n+12)(k+1)πN],(S3)nk=bεnsin[(k+12)(n+1)πN],(S4)nk=bsin[(n+12)(k+12)πN],(S5)nk=csin(nk2π2N1),(S6)nk=csin[(n+12)(k+12)2π2N1],(S7)nk=csin[(k+12)(n+1)2π2N1],(S8)nk=cεnεksin[(n+12)(k+12)2π2N1], (97)

where a=2N1, b=2N, c=22N1, and d=22N+1; normalization factors are unity except for γn=γk=12 for n=k=0 or N1, σn=σk=12 for n=k=0, and εn=εk=12 for n=k=N1; 0n,kN1 for all the N-th-order DCTs/DSTs except for the (N1)-th-order DST-1 and DST-5 where 1n,kN1. Inverses of the DCTs and DSTs are computed by transpose relation (as they are unitary transform) Ci1=CiT and Si1=SiT, respectively. The elements of other sixteen transform matrices, using (97), are defined in [4] as follows

(S˜1)nk=aγnγksin(nkπN1),(S˜2)nk=bσksin[(n+12)kπN],(S˜3)nk=bσnsin[(k+12)nπN],(S˜4)nk=bsin[(n+12)(k+12)πN],(S˜5)nk=cσnσksin(nk2π2N1),(S˜6)nk=cεnσksin[(n+12)k2π2N1],(S˜7)nk=cεkσnsin[(k+12)n2π2N1],(S˜8)nk=dsin[(n+12)(k+12)2π2N+1],(C˜1)nk=bcos(nkπN),(C˜2)nk=bεkcos[(n+12)(k+1)πN],(C˜3)nk=bεncos[(k+12)(n+1)πN],(C˜4)nk=bcos[(n+12)(k+12)πN],(C˜5)nk=ccos(nk2π2N1),(C˜6)nk=ccos[(n+12)(k+12)2π2N1],(C˜7)nk=ccos[(k+12)(n+1)2π2N1],(C˜8)nk=cεnεkcos[(n+12)(k+12)2π2N1], (98)

where matrices S˜1, S˜2, S˜3, S˜5, S˜7, C˜1 and C˜5 are of (N1)-th-order matrices, and rest are of N-th-order.

Appendix C. Multidimensional PT

In this appendix, we consider the PT of a 2D signal (e.g., image) which can be easily extended for multidimensional signals. Let g(x,y) be a non-periodic and real valued function, then the 2D-FT is defined as

G(ω1,ω2)=g(x,y)ej(ω1x+ω2y)dxdy, (99)

and the inverse 2D-FT is defined as

g(x,y)=14π2G(ω1,ω2)ej(ω1x+ω2y)dω1dω2. (100)

The 2D PT transfer function corresponding to 1D counterpart (27) can be written as

H(α(ω1,ω2))={ejα(ω1,ω2),0ω1<,<ω2<ejα(ω1,ω2),<ω10,<ω2<, (101)

where the 2D analytic signal (2D-AS) is defined by considering the first and fourth quadrants of the 2D-FT plane as [6]

z(x,y)=g(x,y)+jgˆ(x,y)=12π20G(ω1,ω2)ej(ω1x+ω2y)dω1dω2, (102)

where gˆ(x,y) is the HT of g(x,y). The 2D-PT can be computed by considering the real part of the 2D-PT of 2D-AS which we defined as

z(x,y,α(ω1,ω2))=12π20G(ω1,ω2)ej(ω1x+ω2y)ejα(ω1,ω2)dω1dω2, (103)

and if α(ω1,ω2)=α, then z(x,y,α)=z(x,y)ejα, therefore 2D counter part of 1D PT (30) can be defined as

g(x,y,α)=Re{z(x,y,α)}=cos(α)g(x,y)+sin(α)gˆ(x,y), (104)

where Re{z(x,y,α)} denotes real part of the function z(x,y,α). As the 2D-AS (102) is not unique and it can also be defined by considering the first and second quadrants of the 2D-FT plane [6], so corresponding modifications (i.e., integration limits would be <ω1<,0ω2<) can be easily applied to equations (101), (102) and (103).

Now, using the Observation 3.1(b), to obtain the unique 2D-PT and 2D-AS, we define the 2D phase shifter in frequency domain as H(α(ω1,ω2))=ejsign(ω1+ω2)α(ω1,ω2), i.e.

H(α(ω1,ω2))={ejα(ω1,ω2),ω1+ω20ω2ω1<,<ω2<,ejα(ω1,ω2),ω1+ω2<0<ω1<ω2,<ω2<. (105)

Thus, we obtain the 2D analytic signal (2D-AS) by considering of the 2D-FT plane with ω1+ω20 as

z(x,y)=g(x,y)+jgˆ(x,y)=12π2ω2G(ω1,ω2)ej(ω1x+ω2y)dω1dω2, (106)

where gˆ(x,y) is the HT of g(x,y). The 2D-PT can be computed by considering the real part of the 2D-PT of 2D-AS which we defined as

z(x,y,α(ω1,ω2))=12π2ω2G(ω1,ω2)ej(ω1x+ω2y)ejα(ω1,ω2)dω1dω2, (107)

and if α(ω1,ω2)=α, then z(x,y,α)=z(x,y)ejα. Therefore, the 2D counter part of the 1D PT (30) can be defined as

g(x,y,α)=Re{z(x,y,α)}=cos(α)g(x,y)+sin(α)gˆ(x,y), (108)

where Re{z(x,y,α)} denotes real part of the function z(x,y,α). Thus, we have obtained the 2D-AS (106) and 2D-PT (108) which are uniquely defined. To obtain the 2D-PT (108), the 1D-HT kernel (19) can also be extended to the conventional 2D-HT kernel as

h(x,y)={1π2xy,x0,y0,0, otherwise,  (109)

where gˆ(x,y) is obtained by the 2D convolution of g(x,y) with h(x,y).

C.1. Phase transform of an image signal

The 2D discrete-time Fourier transform (2D-DTFT) and inverse 2D-DTFT, for a non-periodic and real-valued signal g[m,n], are defined as

G(Ω1,Ω2)=m=n=g[m,n]exp(j[Ω1m+Ω2n]) (110)
g[m,n]=12π12πππππG(Ω1,Ω2)exp(j[Ω1m+Ω2n])dΩ1dΩ2. (111)

Let g[m,n]=δ[m,n], then G(Ω1,Ω2)=1 and from (111), we can write

δ[m,n]=12π12πππππexp(j[Ω1m+Ω2n])dΩ1dΩ2, (112)

which can be further simplified as

δ[m,n]=12π12πππππcos([Ω1m+Ω2n])dΩ1dΩ2, (113)

because the imaginary part of (112) is zero, i.e.,

12π12πππππsin([Ω1m+Ω2n])dΩ1dΩ2=0.

Now, from Observation 3.1 which is based on the Bedrosian theorem, we conclude that the 2D Hilbert transform (2D-HT) as a true extension of 1D-HT can be derived by considering the regions of 2D-DTFT as shown in Fig. 13 , thus we hereby define the kernel of the 2D-PT in frequency domain as

H(Ω1,Ω2,α)={ejα,Ω1+Ω20,ejα,Ω1+Ω20. (114)

Using (114), we write (113) as

δ[m,n]=1π12πππ[Ω2πcos([Ω1m+Ω2n])dΩ1]dΩ2, (115)

and define the HT of (115) as

h[m,n]=δ[m,n,π/2]=1π12πππ[Ω2πsin([Ω1m+Ω2n])dΩ1]dΩ2, (116)

and obtain

h[m,n]={0,m=0,n=0,δ[nm]πm,m=n,m0,n0,cos(πm)πm,m0,n=0,cos(πn)πn,m=0,n0,0,mn0. (117)

Moreover, the kernel of the 2D PT can be written as

δ[m,n,α]=cos(α)δ[m,n]+sin(α)δ[m,n,π/2], (118)

and thus

g[m,n,α]=cos(α)g[m,n]+sin(α)g[m,n,π/2], (119)

where δ[m,n,α]=1π12πππ[Ω2πcos([Ω1m+Ω2n]α)dΩ1]dΩ2 is the 2D-PT of unit impulse sequence δ[m,n]; g[m,n,α]=δ[m,n,π/2]g[m,n] is the 2D-PT of signal g[m,n] and (119) is the 2D-HT of signal g[m,n] when α=π/2. Using (111) and above discussion, we obtain the 2D analytic signal (2D-AS) as

z[m,n]=12π1πππΩ2π[G(Ω1,Ω2)exp(j[Ω1m+Ω2n])dΩ1d]Ω2. (120)

and its PT as

z[m,n,α]=z[m,n]ejα=g[m,n,α]+jg[m,n,α+π/2]. (121)

Fig. 13.

Fig. 13

Phase transform regions with four quadrants (I, II, III and IV) in Fourier domain, II quadrant is divided in two parts (a) Ω2 > |Ω1| and (b) |Ω1| > Ω2; IV quadrant is divided in two parts (a) Ω1 > |Ω2| and (b) |Ω2| > Ω1; complete region can be divided in two parts by line Ω1 + Ω2 = 0.

Similar to (29), one can obtain the kernel of 2D discrete-time analytic signal and compute the phase difference between δ[m,n] and its HT h[m,n] (117) as

zδ,h[m,n]=δ[m,n]+jh[m,n],ϕδ,h[m,n]=tan1(h[m,n]δ[m,n])={0,m=0,n=0,π/2,m=n,m>0,n>0,π/2,m=n,m<0,n<0,π/2,(m>0 &even) or (m<0 &odd),n=0,π/2,(m>0 &odd) or (m<0 &even),n=0,π/2,m=0,(n>0 &even) or (n<0 &odd),π/2,m=0,(n>0 &odd) or (n<0 &even). (122)

References

  • 1.Ahmed N., Natarajan T., Rao K.R. Discrete cosine transform. IEEE Trans. Comput. 1974:90–93. [Google Scholar]
  • 2.Britanak V., Yip P.C., Rao K.R. February 2006. Discrete Cosine and Sine Transforms: General Properties, Fast Algorithms and Integer Approximations. [Google Scholar]
  • 3.Cooley J.W., Tukey J.W. An algorithm for the machine calculation of complex Fourier series. Math. Comput. 1965;19:297–301. doi: 10.1090/S0025-5718-1965-0178586-1. [DOI] [Google Scholar]
  • 4.Singh P. Novel Fourier quadrature transforms and analytic signal representations for nonlinear and non-stationary time series analysis. R. Soc. Open Sci. 2018;5(1) doi: 10.1098/rsos.181131. [DOI] [PMC free article] [PubMed] [Google Scholar]
  • 5.Singh P., Joshi S.D., Patney R.K., Saha K. The Fourier decomposition method for nonlinear and non-stationary time series analysis. Proc. R. Soc. A. 2017;20160871 doi: 10.1098/rspa.2016.0871. [DOI] [PMC free article] [PubMed] [Google Scholar]
  • 6.Singh P., Joshi S.D. Some studies on multidimensional Fourier theory for Hilbert transform, analytic signal and AM–FM representation. Circuits Syst. Signal Process. 2019;38:5623–5650. doi: 10.1007/s00034-019-01133-x. [DOI] [Google Scholar]
  • 7.Princen J.P., Johnson A.W., Bradley A.B. IEEE Proc. Intl. Conference on Acoustics, Speech, and Signal Processing (ICASSP) 1987. Subband/transform coding using filter bank designs based on time domain aliasing cancellation; pp. 2161–2164. [Google Scholar]
  • 8.Princen J.P., Bradley A.B. Analysis/synthesis filter bank design based on time domain aliasing cancellation. IEEE Trans. Acoust. Speech Signal Process. 1986;34(5):1153–1161. [Google Scholar]
  • 9.Haykin S. third edition. John Wiley & Sons (Asia); Singapore: 1995. Communication Systems. [Google Scholar]
  • 10.Gabor D. Theory of communication, electrical engineers-part III. J. Inst. Radio Commun. Eng. 1946;93(26):429–441. [Google Scholar]
  • 11.Gupta A., Singh P., Karlekar M. A novel signal modeling approach for classification of seizure and seizure-free EEG signals. IEEE Trans. Neural Syst. Rehabil. Eng. 2018;26(5):925–935. doi: 10.1109/TNSRE.2018.2818123. [DOI] [PubMed] [Google Scholar]
  • 12.Daubechies I., Lu J., Wu H.T. Synchrosqueezed wavelet transforms: an empirical mode decomposition-like tool. Appl. Comput. Harmon. Anal. 2011;30(2):243–261. [Google Scholar]
  • 13.Holighaus N., Wiesmeyr C., Balazs P. Continuous warped time-frequency representations–Coorbit spaces and discretization. Appl. Comput. Harmon. Anal. 2019;47(3):975–1013. [Google Scholar]
  • 14.Wu H. Instantaneous frequency and wave shape functions (I) Appl. Comput. Harmon. Anal. 2013;35(2):181–199. [Google Scholar]
  • 15.Singh P. Breaking the limits: redefining the instantaneous frequency. Circuits Syst. Signal Process. 2018;37:3515–3536. doi: 10.1007/s00034-017-0719-y. [DOI] [Google Scholar]
  • 16.Singh P., Joshi S.D., Patney R.K., Saha K. Some studies on nonpolynomial interpolation and error analysis. Appl. Math. Comput. 2014;244:809–821. [Google Scholar]
  • 17.Sandoval S., De Leon P.L. Theory of the Hilbert spectrum. 2015. arXiv:1504.07554 [math.CV]
  • 18.Carson J., Fry T. Variable frequency electric circuit theory with application to the theory of frequency modulation. Bell Syst. Tech. J. 1937;16:513–540. [Google Scholar]
  • 19.Cohen L. Prentice Hall; 1995. Time-Frequency Analysis. [Google Scholar]
  • 20.Gabor D. Theory of communication. Proc. IEE. 1946;93(III):429–457. [Google Scholar]
  • 21.Shekel J. ‘Instantaneous’ frequency. Proc. IRE. 1953;41(4):548. [Google Scholar]
  • 22.Vakman D.E. On the definition of concepts of amplitude, phase and instantaneous frequency. Radio Eng. Electron. Phys. 1972;17:754–759. [Google Scholar]
  • 23.Boashash B. Estimating and interpreting the instantaneous frequency of a signal–part 1: fundamentals. Proc. IEEE. 1992;80(4):520–538. [Google Scholar]
  • 24.Boashash B. Estimating and interpreting the instantaneous frequency of a signal–part 2: algorithms and applications. Proc. IEEE. 1992;80(4):540–568. [Google Scholar]
  • 25.Loughlin P.J., Tacer B. Comments on the interpretation of instantaneous frequency. IEEE Signal Process. Lett. 1997;4(5):123–125. [Google Scholar]
  • 26.Boashash B. Elsevier; Boston: 2003. Time Frequency Signal Analysis and Processing: A Comprehensive Reference. [Google Scholar]
  • 27.Van der Pol B. The fundamental principles of frequency modulation. Proc. IEE. 1946;93(111):153–158. [Google Scholar]
  • 28.Purves S. Phase and the Hilbert transform. Lead. Edge. 2014;33(10):1164–1166. doi: 10.1190/tle33101164.1. [DOI] [Google Scholar]
  • 29.Bedrosian E. A product theorem for Hilbert transforms. Proc. IEEE. 1963;51:868–869. [Google Scholar]
  • 30.Brown J.L. A Hilbert transform product theorem. Proc. IEEE. 1986;74:520–521. [Google Scholar]
  • 31.Bhattacharyya A., Singh L., Pachori R.B. Fourier–Bessel series expansion based empirical wavelet transform for analysis of non-stationary signals. Digit. Signal Process. 2018;78:185–196. [Google Scholar]
  • 32.Stocchi M., Marchesi M. Fast wavelet transform assisted predictors of streaming time series. Digit. Signal Process. 2018;77:5–12. [Google Scholar]
  • 33.Hu J., Chai L., Xiong D., Wang W. A novel method of realizing stochastic chaotic secure communication by synchrosqueezed wavelet transform. Digit. Signal Process. 2018;82:194–202. [Google Scholar]
  • 34.Hu Y., Li F., Li H., Liu C. An enhanced empirical wavelet transform for noisy and non-stationary signal processing. Digit. Signal Process. 2017;60:220–229. [Google Scholar]
  • 35.Daubechies I. 1992. Ten Lectures on Wavelets. (CBMS-NSF Regional Conference Series in Applied Mathematics). [Google Scholar]
  • 36.Lilly J.M., Olhede S.C. On the analytic wavelet transform. IEEE Trans. Inf. Theory. 2010;56(8):4136–4156. [Google Scholar]
  • 37.Lilly J.M., Olhede S.C. Generalized Morse wavelets as a superfamily of analytic wavelets. IEEE Trans. Signal Process. 2012;60(11):6036–6041. [Google Scholar]
  • 38.Holschneider M. Oxford Univ. Press; Oxford: 1998. Wavelets: An Analysis Tool. [Google Scholar]
  • 39.https://www.mathworks.com/help/wavelet/gs/inverse-continuous-wavelet-transform.html
  • 40.Gao J., Dong X., Wang W.B., Li Y., Pan C. Instantaneous parameters extraction via wavelet transform. IEEE Trans. Geosci. Remote Sens. 1999;37(2):867–870. [Google Scholar]
  • 41.Taner M., Koehler F., Sheriff R.E. Complex seismic trace analysis. Geophysics. 1979;44:1041–1063. [Google Scholar]
  • 42.Jinghuai G., Wenbing W., Guangming Z. Wavelet transform and instantaneous attributes analysis of a signal. Chin. J. Geophys. 1997;40:821–832. [Google Scholar]
  • 43.Abbott B.P. Observation of gravitational waves from a binary black hole merger. Phys. Rev. Lett. 2016;116(061102) doi: 10.1103/PhysRevLett.116.061102. [DOI] [PubMed] [Google Scholar]
  • 44.https://losc.ligo.org/events/GW150914/
  • 45.Flandrin P. Ecole normale superieure de Lyon; 2016. A note on the time-frequency analysis of GW150914. Research report. ensl-01370441. [Google Scholar]
  • 46.Gupta A., Joshi S.D., Singh P. On the approximate discrete KLT of fractional Brownian motion and applications. J. Franklin Inst. 2018;355(17):8989–9016. [Google Scholar]
  • 47.Stenger F. Springer; Berlin: 1993. Numerical Methods Based on Sinc and Analytic Functions. [Google Scholar]
  • 48.Stenger F. CRC Press; Boca Raton: 2011. Handbook of Sinc Numerical Methods. [Google Scholar]
  • 49.Feng L., Linetsky V. Pricing discretely monitored barrier options and defaultable bonds in Levy process models: a fast Hilbert transform approach. Math. Finance. 2008;18(3):337–384. doi: 10.1111/j.1467-9965.2008.00338.x. [DOI] [Google Scholar]
  • 50.Fusai G., Germano G., Marazzina D. Spitzer identity, Wiener-Hopf factorisation and pricing of discretely monitored exotic options. Eur. J. Oper. Res. 2016;251(4):124–134. doi: 10.1016/j.ejor.2015.11.027. [DOI] [Google Scholar]
  • 51.Phelan C., Marazzina D., Fusai G., Germano G. Hilbert transform, spectral filters and option pricing. Ann. Oper. Res. 2019;282:273–298. doi: 10.1007/s10479-018-2881-4. [DOI] [Google Scholar]
  • 52.Ruijter M.J., Versteegh M., Oosterlee C.W. On the application of spectral filters in a Fourier option pricing technique. J. Comput. Finance. 2015;19(1):75–106. doi: 10.21314/JCF.2015.306. [DOI] [Google Scholar]
  • 53.Hardy G.H. On Hilbert transforms. Q. J. Math. (Oxford) 1932;3:102–112. [Google Scholar]
  • 54.Hahn S.L. Artech House, Inc.; Norwood, MA, USA: 1996. Hilbert Transforms in Signal Processing. [Google Scholar]
  • 55.Scott A. Wiley-Interscience; New York: 1970. Active and Nonlinear Wave Propagation in Electronics. [Google Scholar]
  • 56.Potamianos A., Maragos P. A comparison of the energy operator and the Hilbert transform approach to signal and speech demodulation. Signal Process. 1994;37(1):95–120. [Google Scholar]
  • 57.Zayed A.I. Hilbert transform associated with the fractional Fourier transform. IEEE Signal Process. Lett. 1998;5(8):206–208. [Google Scholar]
  • 58.Tao R., Li X-M., Wang Y. Generalization of the fractional Hilbert transform. IEEE Signal Process. Lett. 2008;15:365–368. [Google Scholar]
  • 59.King F.W. Cambridge University Press; Cambridge, UK: 2009. Hilbert Transforms, vol. 1. [Google Scholar]
  • 60.King F.W. Cambridge University Press; Cambridge, UK: 2009. Hilbert Transforms, vol. 2. [Google Scholar]
  • 61.Oppenheim A.V., Schafer R.W. 3rd edition. Pearson; Boston: 2010. Discrete-Time Signal Processing. [Google Scholar]
  • 62.Liu Y-W. IntechOpen; 2012. Hilbert Transform and Applications; pp. 291–300. [Google Scholar]
  • 63.Carleson L. On convergence and growth of partial sums of Fourier series. Acta Math. 1966;116(1):135–157. [Google Scholar]
  • 64.Hunt R.A. On the convergence of Fourier series, orthogonal expansions and their continuous analogues. Proc. Conf.; Edwardsville, III. (1967); Carbondale, III: Southern Illinois Univ. Press; 1968. pp. 235–255. [Google Scholar]
  • 65.Antonov N.Y. Convergence of Fourier series. East J. Approx. 1996;2:187–196. [Google Scholar]
  • 66.Konyagin S.V. On the divergence everywhere of trigonometric Fourier series. Sb. Math. 2000;191:97–120. [Google Scholar]
  • 67.Korner T.W. Cambridge University Press; Great Britain: 1989. Fourier Analysis. [Google Scholar]
  • 68.Piotrowska J., Miller J.M., Schnetter E. Spectral methods in the presence of discontinuities. J. Comput. Phys. 2019;390:527–547. [Google Scholar]
  • 69.Gibbs J.W. Fourier's series. Nature. 1898;59(1522):200. [Google Scholar]
  • 70.Vandeven H. Family of spectral filters for discontinuous problems. J. Sci. Comput. 1991;6(2):159–192. [Google Scholar]
  • 71.Gottlieb D., Shu C.-W., Solomonoff A., Vandeven H. On the Gibbs phenomenon 1: recovering exponential accuracy from the Fourier partial sum of a non-periodic analytic function. J. Comput. Appl. Math. 1992;43:81–98. [Google Scholar]
  • 72.Gelb A., Tanner J. Robust reprojection methods for the resolution of the Gibbs phenomenon. Appl. Comput. Harmon. Anal. 2006;20(1):3–25. [Google Scholar]
  • 73.Gottlieb S., Jung J.H., Kim S. A review of David Gottlieb's work on the resolution of the Gibbs phenomenon. Commun. Comput. Phys. 2011;9(3):497–519. [Google Scholar]
  • 74.Gottlieb D., Tadmor E. Recovering pointwise values of discontinuous data within spectral accuracy. In: Murman E.M., Abarbanel S.S., editors. Progress and Supercomputing in Computational Fluid Dynamics, Progress in Scientific Computing. Springer; Birkhäuser, Boston: 1985. [Google Scholar]
  • 75.Tauber A. Über den zusammenhang des reellen und imaginären theiles einer potenzreihe. Monatshefte Math. Phys. 1891;2:79–118. [Google Scholar]
  • 76.Hardy G.H. The theory of Cauchy's principal values. (Third paper: differentiation and integration of principal values) Proc. Lond. Math. Soc. 1902;35:81–107. [Google Scholar]
  • 77.Hardy G.H. The theory of Cauchy's principal values. fourth paper: the integration of principal values–continued–with applications to the inversion of definite integrals. Proc. Lond. Math. Soc. 1908;7(2):181–208. [Google Scholar]
  • 78.Hilbert D. Grundzüge einer allgemeinen theorie der linearen integralgleichungen. Nachr. Akad. Wiss. Gott., Math.-phys. Klasse. 1904;3:213–259. [Google Scholar]
  • 79.Hilbert D. B. G. Teubner; Leipzig: 1912. Grundzüge einer Allgemeinen Theorie der Linearen Integralgleichungen. [Google Scholar]
  • 80.Gabor D. Theory of communication. J. Inst. Elec. Eng. 1946;93(3):429–457. [Google Scholar]
  • 81.Namias V. The fractional order Fourier transform and its application to quantum mechanics. J. Inst. Math. Appl. 1980;25:241–265. [Google Scholar]
  • 82.McBride A.C., Kerr F.H. On Namias' fractional Fourier transforms. IMA J. Appl. Math. 1987;39:159–175. [Google Scholar]
  • 83.Cheney M., Rose J.H. Generalization of the Fourier transform: implications for inverse scattering theory. Phys. Rev. Lett. 1988;60(13):1221–1224. doi: 10.1103/PhysRevLett.60.1221. [DOI] [PubMed] [Google Scholar]
  • 84.Mallat S.G. A theory for multiresolution signal decomposition: the wavelet representation. IEEE Trans. Pattern Anal. Mach. Intell. 1989;11(7):674–693. [Google Scholar]
  • 85.Ell T.A. Proc. 32nd Conf. Decision Contr. Dec. 1993. Quaternion-Fourier transforms for analysis of two-dimensional linear time-invariant partial differential systems; pp. 1830–1841. [Google Scholar]
  • 86.Mann S., Haykin S. The chirplet transform: physical considerations. IEEE Trans. Signal Process. 1995;43(11):2745–2761. [Google Scholar]
  • 87.Stockwell R.G., Mansinha L., Lowe R.P. Localization of the complex spectrum: the S transform. IEEE Trans. Signal Process. 1996;44(4):998–1001. [Google Scholar]
  • 88.Mboupa M., Adali T. A generalization of the Fourier transform and its application to spectral analysis of chirp-like signals. Appl. Comput. Harmon. Anal. 2012;32(2):305–312. [Google Scholar]
  • 89.Weideman J.A.C. Computing the Hilbert transform on the real line. Math. Comput. 1995;64(210):745–762. [Google Scholar]

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