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Analytic signal

The analytic signal is a complex-valued representation of a real-valued signal in which the negative frequency components are suppressed, resulting in a spectrum confined to non-negative frequencies.^[1]^[2] This construction facilitates the analysis of signal characteristics such as instantaneous amplitude and phase, which are otherwise challenging to define precisely for real signals.^[2] For a real-valued signal x(t), the corresponding analytic signal z(t) is formed as z(t) = x(t) + j \hat{x}(t), where \hat{x}(t) denotes the Hilbert transform of x(t).^[1] The Hilbert transform acts as a linear filter that introduces a -\pi/2 phase shift for positive frequencies and a +\pi/2 phase shift for negative frequencies, effectively doubling the positive frequency components while nullifying the negative ones in the analytic representation.^[1]^[3] Key properties of the analytic signal include its magnitude |z(t)|, which yields the instantaneous amplitude or envelope of the original signal, and its argument \arg(z(t)), whose time derivative provides the instantaneous frequency \omega(t) = \frac{d}{dt} \arg(z(t)).^[2] This framework is particularly useful for narrowband signals, where the envelope varies slowly compared to the carrier frequency.^[2] In the frequency domain, the Fourier transform of the analytic signal Z(\omega) satisfies Z(\omega) = 2X(\omega) for \omega > 0, Z(0) = X(0), and Z(\omega) = 0 for \omega < 0.^[1] Applications of analytic signals span multiple domains in engineering and physics, including amplitude modulation demodulation—where the envelope extraction A(t) = |z(t)| recovers the modulating signal—vibration monitoring for fault detection in mechanical systems, and empirical mode decomposition for non-stationary signal analysis.^[1]^[2] They also underpin time-frequency representations, such as the Hilbert-Huang transform, for studying signal energy distribution.^[2] The concept originated in the mid-20th century, with Dennis Gabor introducing it in 1946 as part of his foundational work on communication theory and signal analysis using complex representations.^[4] Independently, Jean-André Ville formalized the analytic signal in 1948, emphasizing its role in defining instantaneous frequency and phase for practical signal processing.^[2]

Fundamentals

Definition

The analytic signal is a complex-valued function derived from a real-valued signal x(t), constructed such that its Fourier transform contains no negative-frequency components.^[1] This representation eliminates the redundancy in the frequency domain inherent to real signals, where the spectrum is symmetric, allowing for a more efficient encoding of the signal's information content.^[5] For a real signal x(t), its Fourier transform X(\omega) exhibits Hermitian symmetry, meaning X(-\omega) = X^*(\omega), where ^* denotes the complex conjugate; this symmetry implies that the negative-frequency components are mirrors of the positive ones.^[6] The analytic signal z(t) is formed by doubling the amplitudes of the positive-frequency components in X(\omega) and setting the negative-frequency components to zero, resulting in a one-sided spectrum that fully captures the original signal without loss.^[1] The imaginary part of the analytic signal is obtained via the Hilbert transform of x(t), yielding z(t) = x(t) + j \mathcal{H}\{x(t)\}, where \mathcal{H}\{\cdot\} denotes the Hilbert transform and j is the imaginary unit; this construction was proposed by Dennis Gabor in his foundational work on communication theory.^[4] The Hilbert transform acts as an all-pass filter that shifts the phase of positive frequencies by -\pi/2 radians, ensuring the resulting complex signal aligns with the desired spectral properties.^[1] The term "analytic" refers to the fact that, for bandlimited signals, z(t) satisfies the Cauchy-Riemann equations in the complex plane when viewed as a function of t + i\tau in the upper half-plane, making it holomorphic there.^[7] This analyticity property underscores the signal's conformity to principles of complex analysis, distinguishing it from general complex representations.^[8]

Mathematical formulation

The Fourier transform of a real-valued signal x(t) is defined as

X(f) = \int_{-\infty}^{\infty} x(t) \, e^{-j 2\pi f t} \, dt,

where f denotes frequency in hertz. Because x(t) is real-valued, X(f) possesses Hermitian symmetry, satisfying X(-f) = X^*(f).^[9] The analytic signal z(t) is obtained by modifying the spectrum to eliminate negative frequencies. Its Fourier transform is

Z(f) = \begin{cases} 2 X(f) & f > 0, \\ X(0) & f = 0, \\ 0 & f < 0, \end{cases}

with the time-domain representation given by the inverse Fourier transform

z(t) = \int_{-\infty}^{\infty} Z(f) \, e^{j 2\pi f t} \, df.

The factor of 2 for positive frequencies ensures that the real part of z(t) recovers x(t), as the original spectrum's Hermitian symmetry contributes equally from positive and negative components.^[1] Equivalently, the analytic signal can be expressed using the Hilbert transform operator \mathcal{H}, defined for a signal x(t) as the Cauchy principal value integral

\mathcal{H}\{x\}(t) = \frac{1}{\pi} \, \PV \int_{-\infty}^{\infty} \frac{x(\tau)}{t - \tau} \, d\tau.

Thus, z(t) = x(t) + j \, \mathcal{H}\{x\}(t). To verify this form yields no negative frequencies, note that the Fourier transform of the Hilbert transform is \hat{X}(f) = -j \, \sgn(f) \, X(f), where \sgn(f) is the sign function (+1 for f > 0, -1 for f < 0, and $0 at f = 0). The spectrum of z(t) is then

Z(f) = X(f) + j \, \hat{X}(f) = X(f) \left[1 + j \cdot (-j \, \sgn(f))\right] = X(f) \left[1 + \sgn(f)\right].

For f > 0, this simplifies to Z(f) = 2 X(f); for f < 0, Z(f) = 0; and at f = 0, Z(0) = X(0). This confirms that the negative-frequency components vanish, as the contributions from x(t) and j \, \mathcal{H}\{x\}(t) cancel for f < 0.^[10] Regarding bandwidth, the analytic signal z(t) has its spectrum supported solely on non-negative frequencies up to the maximum frequency present in the positive-frequency portion of X(f). If the original signal x(t) has a total bandwidth of $2B (spanning -B to B), the positive-frequency part spans $0 to B with bandwidth B, and the analytic signal inherits this bandwidth B. The doubling of the spectral amplitude for f > 0 preserves the energy associated with the positive frequencies without altering the bandwidth.^[1]

Examples

Sinusoidal signals

The analytic signal construction is particularly straightforward for sinusoidal inputs, as it leverages the Hilbert transform to eliminate negative frequency components while preserving the positive ones. Consider a real-valued cosine signal x(t) = \cos(\omega t), where \omega > 0 is the angular frequency. The Hilbert transform of this signal is \hat{x}(t) = \sin(\omega t), yielding the analytic signal z(t) = x(t) + j \hat{x}(t) = \cos(\omega t) + j \sin(\omega t) = e^{j \omega t}.^[1]^[11] In the frequency domain, the Fourier transform of x(t) = \cos(\omega t) consists of impulses at \pm \omega with equal amplitude \pi (using the convention where the transform of e^{j \omega t} is $2\pi \delta(\Omega - \omega)). The analytic signal z(t) has a spectrum that suppresses the negative-frequency component at -\omega, resulting in a single impulse at +\omega with doubled amplitude $2\pi. This frequency suppression is a defining feature of the analytic representation, ensuring Z(\Omega) = 0 for \Omega < 0.^[1]^[5] For a more general single-tone signal x(t) = A \cos(\omega t + \phi), with constant amplitude A > 0 and phase \phi, the Hilbert transform is \hat{x}(t) = A \sin(\omega t + \phi). Thus, the analytic signal becomes z(t) = A e^{j (\omega t + \phi)}, which faithfully preserves both the amplitude A = |z(t)| and the phase information in the argument of the complex exponential.^[5]^[12] This construction extends naturally to a sum of sinusoids with distinct positive frequencies. For x(t) = \sum_{k=1}^N A_k \cos(\omega_k t + \phi_k), where \omega_k > 0 and all frequencies are distinct, the Hilbert transform is \hat{x}(t) = \sum_{k=1}^N A_k \sin(\omega_k t + \phi_k), and the analytic signal is z(t) = \sum_{k=1}^N A_k e^{j (\omega_k t + \phi_k)}. The spectrum of z(t) thus contains only the positive-frequency components, each doubled in amplitude relative to the original signal's one-sided spectrum.^[1]^[5]

Modulated signals

In amplitude modulation (AM), a carrier signal is modulated by a low-frequency message signal that varies its amplitude. Consider an AM signal given by x(t) = [1 + m \cos(\omega_m t)] \cos(\omega_c t), where m is the modulation index (|m| < 1), \omega_m is the modulating frequency, and \omega_c is the carrier frequency with \omega_c \gg \omega_m. The analytic signal for this narrowband case approximates to z(t) \approx [1 + m \cos(\omega_m t)] e^{j \omega_c t}, obtained by suppressing the negative-frequency components around the carrier in the spectrum.^[1]^[13] This representation arises because the Hilbert transform of x(t) yields approximately [1 + m \cos(\omega_m t)] \sin(\omega_c t) under the narrowband assumption, forming the imaginary part of z(t). The spectrum of the AM signal features symmetric sidebands around \pm \omega_c; the analytic signal eliminates the lower sideband and the negative carrier, retaining only the positive-frequency components, which simplifies analysis of the modulation envelope.^[1]^[13] For frequency modulation (FM), the carrier phase is varied by the message signal. An FM signal can be expressed as x(t) = A \cos(\omega_c t + \beta \sin(\omega_m t)), where A is the constant amplitude, \beta is the modulation index, and the instantaneous frequency deviates around \omega_c. The corresponding analytic signal is z(t) = A e^{j (\omega_c t + \beta \sin(\omega_m t))}, with the Hilbert transform providing the quadrature component A \sin(\omega_c t + \beta \sin(\omega_m t)).^[13]^[14] The FM spectrum consists of sidebands at \omega_c \pm k \omega_m for integer k, governed by Bessel functions; the analytic representation suppresses the negative-frequency sidebands, ensuring a one-sided spectrum centered at positive frequencies. This holds exactly when the modulation does not generate significant energy below zero frequency, a condition met in narrowband FM where \beta \ll 1 and \omega_c \gg \omega_m. However, for wideband FM, approximations may be needed as higher-order sidebands could extend into negative frequencies, potentially violating the analytic property.^[13]^[14]

Properties

Instantaneous amplitude and phase

The analytic signal z(t) = x(t) + j \hat{x}(t), where \hat{x}(t) denotes the Hilbert transform of the real-valued signal x(t), enables the extraction of time-varying attributes that describe the signal's local behavior.^[15] These attributes include the instantaneous amplitude and phase, which provide a complex representation of the signal's envelope and oscillation at each instant.^[5] The instantaneous amplitude A(t) is defined as the modulus of the analytic signal:

A(t) = |z(t)| = \sqrt{x(t)^2 + \hat{x}(t)^2}.

This quantity captures the local strength or envelope of the signal, varying smoothly for signals without abrupt changes.^[15] The instantaneous phase \phi(t) is the argument of the analytic signal:

\phi(t) = \arg(z(t)) = \atantwo(\hat{x}(t), x(t)),

where \atantwo ensures the correct quadrant by considering the signs of both real and imaginary parts.^[5] This phase function is typically wrapped within (-\pi, \pi], but for deriving frequency, it requires unwrapping to produce a continuous, monotonically increasing trajectory. The instantaneous frequency \omega(t) is obtained as the time derivative of the unwrapped phase:

\omega(t) = \frac{d\phi(t)}{dt}.

This measures the local rate of phase accumulation, interpreted as the signal's frequency at time t, and is positive for analytic signals due to the suppression of negative frequency components. Unwrapping mitigates discontinuities of $2\pi jumps inherent in the principal phase value, ensuring \omega(t) reflects genuine oscillations rather than artifacts. Geometrically, the analytic signal z(t) traces a phasor in the complex plane, with its tip rotating counterclockwise around the origin at a speed given by \omega(t) while the distance from the origin varies as A(t).^[5] This representation aligns with intuitive notions of amplitude modulation and frequency modulation, portraying the signal as a rotating vector whose magnitude and angular velocity evolve over time.^[15] For bandlimited signals—those with spectra confined to positive frequencies—these instantaneous attributes are uniquely defined and correspond closely to physical interpretations of local amplitude and frequency variations. This uniqueness stems from the analytic signal's construction, which eliminates ambiguity in phase and amplitude assignments for such signals, as established through physical and mathematical constraints.

Bedrosian identities

Bedrosian's theorem provides a key algebraic identity for the Hilbert transform applied to products of signals with disjoint frequency spectra. Specifically, if x(t) is a low-pass signal whose Fourier transform \hat{x}(\omega) has support in [-a, a] and y(t) is a high-pass signal whose Fourier transform \hat{y}(\omega) has support outside (-a, a), then the Hilbert transform satisfies

\mathcal{H}\{x(t) y(t)\} = x(t) \mathcal{H}\{y(t)\},

where \mathcal{H} denotes the Hilbert transform.^[16] This condition ensures no spectral overlap between the components.^[17] A direct consequence of this theorem applies to amplitude-modulated signals. For a signal of the form x(t) = a(t) \cos(\phi(t)), where a(t) is a low-frequency amplitude envelope and \cos(\phi(t)) is a high-frequency carrier with \phi'(t) \gg |a'(t)/a(t)|, the theorem implies that the Hilbert transform yields \mathcal{H}\{x(t)\} = a(t) \sin(\phi(t)).^[16] This justifies the analytic signal representation z(t) = x(t) + i \mathcal{H}\{x(t)\} = a(t) e^{i \phi(t)}, enabling extraction of the instantaneous amplitude |z(t)| = a(t).^[18] The theorem extends to frequency-modulated (FM) signals, where x(t) = \cos(\phi(t)) with \phi(t) incorporating slow frequency variations around a high carrier. Under the Bedrosian condition, the instantaneous frequency \phi'(t) can be reliably estimated from the analytic signal as the derivative of the phase \arg(z(t)), provided the modulation bandwidth remains narrow relative to the carrier.^[18] The proof of Bedrosian's theorem relies on the frequency-domain definition of the Hilbert transform, which multiplies the Fourier transform by -i \sgn(\omega). The Fourier transform of the product x(t)y(t) is the convolution \hat{x} * \hat{y}; due to disjoint supports, this convolution places the result entirely in the high-frequency regime where \sgn(\omega) = 1 (or -1 consistently), so applying the Hilbert multiplier affects only \hat{y}, yielding the identity upon inverse transform.^[17] The theorem's validity depends strictly on spectral separation; it fails for wideband signals with overlapping spectra, such as in high-modulation-index FM where sidebands encroach on low frequencies, leading to distortions in the extracted amplitude and phase.^[18]

Computation

Hilbert transform via convolution

The Hilbert transform of a real-valued signal x(t) can be computed in the time domain as a convolution with the Hilbert kernel h(t) = \frac{1}{\pi t}, yielding H\{x(t)\} = x(t) * h(t) = \frac{1}{\pi} \int_{-\infty}^{\infty} \frac{x(\tau)}{t - \tau} d\tau, where the integral is interpreted in the Cauchy principal value sense to handle the singularity at \tau = t.^[19] This formulation arises from the inverse Fourier transform of the Hilbert transform's frequency response, which is -j \operatorname{sgn}(\omega), and represents the output of a linear time-invariant filter with impulse response h(t).^[20] In practice, direct implementation of this convolution faces significant challenges due to the kernel's non-causal nature—extending infinitely in both positive and negative time directions—and its infinite impulse response, which requires integration over the entire signal duration. To address these issues, the kernel is typically truncated to a finite window, such as a rectangular or windowed approximation centered at t=0, introducing some distortion but enabling real-time or finite-data processing; the principal value ensures convergence despite the pole at t=0.^[20]^[19] The analytic signal z(t) is then assembled from the original signal and its Hilbert transform as z(t) = x(t) + j H\{x(t)\} = x(t) + j (x(t) * h(t)), where the imaginary part provides the quadrature component, effectively suppressing negative frequencies in the spectrum.^[20] For discrete-time signals x sampled at rate f_s, the convolution becomes H\{x\} = x * h, where h = \frac{1}{\pi n} for integer n \neq 0 and

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. Practical computation often employs finite impulse response (FIR) filters that approximate the ideal kernel over a finite length N, designed via methods like windowing the sinc-like response, or circular convolution for periodic extensions of finite-length sequences to avoid edge effects.^[19] In the context of modulated signals, such as x(t) = a(t) \cos(\omega_c t) where a(t) is low-pass and \cos(\omega_c t) is high-pass, the Bedrosian identity simplifies the convolution: H\{x(t)\} = a(t) \sin(\omega_c t), avoiding full numerical integration by exploiting frequency separation and enabling efficient quadrature approximation for analytic signal formation.^[21]

Frequency-domain methods

In the frequency domain, the analytic signal for a continuous-time real-valued signal x(t) is computed by first obtaining its Fourier transform X(\omega). The transform of the analytic signal is then defined as X_a(\omega) = 2X(\omega) for \omega > 0, X_a(0) = X(0), and X_a(\omega) = 0 for \omega < 0, followed by the inverse Fourier transform to yield z(t) = x(t) + j \hat{x}(t), where \hat{x}(t) is the Hilbert transform of x(t).^[22] This operation effectively suppresses negative frequencies while preserving and scaling the positive-frequency content, equivalent to applying a -90° phase shift to all positive-frequency components.^[22] For discrete-time signals x of length N, the analytic signal z is obtained using the discrete Fourier transform (DFT). The DFT X of x is computed, after which the coefficients corresponding to negative frequencies are set to zero, those for positive frequencies (indices 1 to \lfloor (N-1)/2 \rfloor) are doubled, the DC component (index 0) is retained unchanged, and for even-length sequences, the Nyquist frequency component (index N/2) is also retained unchanged.^[23] The modified DFT is then inverse-transformed to produce z.^[23] This approach ensures the resulting z has no negative-frequency components in its DFT representation.^[23] The frequency-domain method benefits from the efficiency of the fast Fourier transform (FFT) algorithm, which computes the DFT and inverse DFT in O(N \log N) time complexity, making it suitable for large datasets.^[23] It yields exact results when the input signal is periodic with period N, as the DFT inherently assumes periodicity, and the phase shift property aligns precisely with the Hilbert transform's -90° shift for positive frequencies.^[22] For non-periodic signals, spectral leakage can introduce artifacts in the FFT-based computation; applying a window function, such as the Hann or Hamming window, to x prior to the DFT mitigates this by tapering the signal edges and reducing discontinuities at the boundaries.

Applications

Signal demodulation

In communication systems, the analytic signal facilitates the extraction of modulation parameters from bandpass signals by providing a complex representation that isolates positive frequency components, enabling precise recovery of amplitude, phase, and frequency information. This approach is particularly effective for narrowband signals where the Bedrosian theorem ensures the validity of the envelope and phase decompositions.^[1] For amplitude modulation (AM) demodulation, the analytic signal z(t) of a received signal x(t) = A(t) \cos(\omega_c t + \phi(t)) is formed as z(t) = x(t) + j \hat{x}(t), where \hat{x}(t) is the Hilbert transform of x(t). The instantaneous amplitude A(t), which carries the message signal, is then recovered directly as the magnitude A(t) = |z(t)|, bypassing traditional envelope detectors and offering robustness to noise in narrowband scenarios.^[1]^[11] In frequency modulation (FM) demodulation, the instantaneous frequency deviation is obtained from the phase of the analytic signal. Specifically, for an FM signal, the instantaneous angular frequency \omega(t) is computed as the time derivative of the argument: \omega(t) = \frac{d}{dt} [\arg(z(t))], allowing extraction of the modulating signal from the frequency variations around the carrier. This method is widely applied in non-stationary signal analysis, such as in radar and audio processing, where accurate tracking of time-varying frequencies is essential.^[24] Quadrature demodulation leverages the real and imaginary parts of the analytic signal as the in-phase (I) and quadrature (Q) components, respectively. The Hilbert transform generates the Q channel from the I channel, forming z(t) = I(t) + j Q(t), which represents the signal in the complex plane and enables digital downconversion or modulation scheme identification in software-defined radios. This I/Q decomposition simplifies the processing of complex modulation formats like QAM by providing orthogonal channels for baseband recovery.^[1]^[25] The pre-envelope, equivalent to the analytic signal z(t), shifts the bandpass signal to baseband via multiplication by z(t) e^{-j \omega_c t}, yielding the complex envelope \tilde{z}(t) centered at zero frequency. This transformation is crucial for demodulation in passband systems, as it converts the modulated signal into a lowpass equivalent for efficient digital processing, such as in wireless receivers where carrier synchronization is required.^[11] The concept of the analytic signal was introduced by Dennis Gabor in his 1946 paper on communication theory, where he proposed the complex representation using the Hilbert transform to analyze signal information in both time and frequency domains, laying the foundation for its use in modulation and demodulation techniques.^[26]

Biomedical signal analysis

In biomedical signal analysis, the analytic signal, derived via the Hilbert transform, plays a crucial role in processing electrocardiogram (ECG) signals by enabling the extraction of instantaneous phase, which facilitates the detection of QRS complexes. The instantaneous phase highlights abrupt changes corresponding to the R-wave peaks, allowing robust identification even in noisy environments, as demonstrated in algorithms that apply the Hilbert transform to compute the phase derivative for peak localization.^[27]^[28] For non-stationary ECG analysis, the Hilbert-Huang transform (HHT), which combines empirical mode decomposition (EMD) with the Hilbert transform, decomposes the signal into intrinsic mode functions (IMFs) and yields instantaneous frequency and amplitude, providing a time-varying representation suited to irregular heart rhythms. This approach excels in capturing transient events like arrhythmias, where traditional methods falter due to varying heart rates.^[29]^[30] In electroencephalogram (EEG) processing, the analytic signal's amplitude envelope, obtained as the modulus of the complex representation, aids in sleep stage detection by quantifying rhythmic oscillations in frequency bands such as delta and theta. The Hilbert spectrum, generated by applying the Hilbert transform to EMD-derived IMFs, offers a time-frequency representation that reveals energy distribution over time in biomedical signals, outperforming the fast Fourier transform (FFT) in handling non-stationarity, such as in heart rate variability (HRV) analysis where it better resolves fluctuating autonomic modulations.^[31]^[32] For instance, in respiratory signals, the instantaneous frequency computed from the analytic signal tracks breathing rate variations, enabling detection of irregularities like apnea through phase-based metrics derived from the Hilbert transform.^[33]

Extensions

Multidimensional analytic signals

The extension of the analytic signal to multidimensional cases, such as 2D images f(x, y) or 3D volumes, involves applying the Riesz transform in a selected direction to mimic the 1D Hilbert transform's frequency suppression. This directional approach constructs an analytic signal by suppressing negative frequencies along a chosen axis, typically the x- or y-direction for simplicity, while preserving positive frequencies. For a 2D signal, this ad hoc method applies the 1D Hilbert transform along lines parallel to the selected direction, resulting in a complex-valued representation f_a(x, y) = f(x, y) + j \hat{f}(x, y), where \hat{f} is the directional Hilbert transform.^[34] Mathematically, the directional Riesz transform (equivalent to the Hilbert transform in that direction) is defined in the frequency domain via a Fourier multiplier -j \operatorname{sign}(\xi), where \xi represents the frequency component along the chosen direction (e.g., the x-frequency for horizontal processing). For a general direction defined by a unit vector \mathbf{u}, the multiplier becomes -j \operatorname{sign}(\boldsymbol{\xi} \cdot \mathbf{u}), with \boldsymbol{\xi} the full frequency vector; this phase-shifts components by -\pi/2 for positive projections and \pi/2 for negative ones, enabling extraction of local amplitude and phase. In practice, for 2D edge analysis, the transform is computed by convolving f(x, y) with a directional kernel, such as h_\theta(x, y) = \frac{1}{\pi} \frac{y \cos\theta - x \sin\theta}{x^2 + y^2} for angle \theta, yielding the analytic signal through f_a(x, y) = f(x, y) * h_\theta(x, y). This formulation extends naturally to 3D by applying the transform along a directional axis in volumetric data.^[34]^[35]^[36] These multidimensional analytic signals find applications in image processing, particularly for edge detection, where the instantaneous amplitude |f_a(x, y)| highlights edges by amplifying discontinuities, outperforming traditional gradient-based methods in noisy environments. Orientation estimation benefits from the local phase \arg(f_a(x, y)), which indicates edge normals perpendicular to the chosen direction, aiding in structure alignment for tasks like texture analysis. However, the reliance on a fixed direction introduces anisotropy, as the signal's properties vary with rotation, potentially distorting features not aligned with the processing axis. For rotation-invariant alternatives, monogenic signals using full isotropic Riesz transforms offer a complementary approach.^[34]^[37]

Monogenic signals

The monogenic signal represents a quaternion-valued extension of the analytic signal tailored for two-dimensional signals, particularly in image processing. For a 2D signal f, the monogenic signal is defined as m(f) = f + j \mathcal{R}_x\{f\} + k \mathcal{R}_y\{f\}, where \mathcal{R}_x and \mathcal{R}_y denote the Riesz transforms along the x and y directions, respectively, and j, k are the imaginary quaternion units.^[38] This formulation embeds the signal in the quaternion algebra, generalizing the 1D analytic signal's structure to higher dimensions while incorporating isotropic properties essential for rotation-invariant analysis. Introduced by Felsberg and Sommer in 2001, the monogenic signal was developed specifically for computer vision tasks, such as edge detection and feature extraction, where orientation independence is crucial.^[38] A key advantage of the monogenic signal lies in its isotropy, meaning the local amplitude and phase remain independent of the signal's orientation, in contrast to directional analytic signals that vary with the chosen axis. This property arises from the Riesz transforms, which act as multidimensional analogs to the Hilbert transform and suppress negative frequencies isotropically in the Fourier domain. As a result, the monogenic signal provides a unified representation that avoids the artifacts of axis-aligned processing, enabling more robust feature detection in oriented structures like edges or textures.^[38] Computation of the monogenic signal is efficiently performed in the frequency domain using the 2D Fourier transform. The isotropic filter applied is h(\omega_x, \omega_y) = -j \frac{\omega_x + j \omega_y}{|\omega|}, where |\omega| = \sqrt{\omega_x^2 + \omega_y^2}, effectively combining the signal with its Riesz components by zeroing out negative frequencies in a rotationally symmetric manner. The inverse Fourier transform then yields the quaternion-valued monogenic signal.^[38] From the monogenic signal, several invariant properties can be derived, including a single monogenic phase and orientation estimate. The monogenic phase captures the local signal variation in a manner analogous to the instantaneous phase in 1D, while the orientation is computed as \theta = \atan2(\mathcal{R}_y\{f\}, \mathcal{R}_x\{f\}), providing the dominant direction of local features without directional bias. These attributes make the monogenic signal particularly valuable for applications requiring scale-space analysis and phase-based image processing.^[38]

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https://ieeexplore.ieee.org/document/782222
[24]
[PDF] Estimating and Interpreting the Instantaneous Frequency of a Signal ...
There is a great need, then, to develop effective techniques for IF estimation. Section 111 includes a review of currently available IF estimation techniques.
[25]
A Quadrature Signals Tutorial: Complex, But Not Complicated
Apr 12, 2013 · In this tutorial we'll review the fundamentals of complex numbers and get comfortable with how they're used to represent quadrature signals.
[26]
[PDF] THEORY OF COMMUNICATION* By D. GABOR, Dr. Ing., Associate ...
In Part 1, a new method of analysing signals is presented in which time and frequency play symmetrical parts, and which contains "time analysis" and "frequency ...
[27]
New QRS detection algorithm based on the Hilbert transform
A new robust algorithm for locating the R wave peaks in computer-based ECG analysis using the properties of the Hilbert transform is presented in this paper.Missing: instantaneous | Show results with:instantaneous
[28]
https://ieeexplore.ieee.org/document/7322686
[29]
[PDF] Feature Extraction of ECG Signal Using HHT Algorithm - arXiv
This new approach, the Hilbert-Huang transform, is implemented to analyze the non-linear and non- stationary data. It is unique and different from the existing.
[30]
The empirical mode decomposition and the Hilbert spectrum for ...
A new method for analysing nonlinear and non-stationary data has been developed. The key part of the method is the 'empirical mode decomposition' method.
[31]
EEG—Single-Channel Envelope Synchronisation and Classification ...
Apr 19, 2021 · Hilbert Transform. The Hilbert transform (HT) is a common signal-processing technique used in the evaluation of phase synchronisation. The HT ...
[32]
[PDF] An Introduction To The Monogenic Signal - arXiv
Mar 27, 2017 · The monogenic signal is an image analysis methodology, a generalization of the analytic signal to multidimensional signals like images.Missing: EEG | Show results with:EEG
[33]
EMD: Empirical Mode Decomposition and Hilbert-Huang Spectral ...
Mar 31, 2021 · The Hilbert transform is used to construct an energy-frequency or energy-frequency-time spectrum known as the HilbertHuang Transform (HHT).
[34]
https://ieeexplore.ieee.org/document/723573
[35]
[PDF] Estimation of Instantaneous Frequency from Empirical Mode ...
Abstract— Instantaneous frequency (IF) calculated by empirical mode decomposition (EMD) provides a novel approach to analyze respiratory sounds (RS).
[36]
https://www.informatik.uni-kiel.de/inf/Sommer/doc/Publications/tbl/dagm2000.pdf
[37]
[PDF] On the Maximal Directional Hilbert Transform in Three Dimensions
We allow the possibility that span( ) = Rd for some nonnegative integer d ≤ n and write (d) := {( j, k) : 1 ≤ j < k ≤ d}; we will drop the dependence on d and ...
[38]
[PDF] Riesz Transforms for the Isotropic Estimation of the Local Phase of ...
We proposed to replace the Hilbert transform by the Riesz transform in order to con- struct a multidimensional analytic signal for the use with intrinsically 1- ...
[39]
Riesz Transforms for the Isotropic Estimation of the Local Phase of ...
The estimation of the local phase and local amplitude of 1-D signals can be realized by the construction of the analytic signal.
[40]
https://ieeexplore.ieee.org/document/969520