Aliasing is a distortion artifact that arises in the sampling and reconstruction of continuous signals or images when the sampling rate is insufficient to capture the signal's highest frequency components, causing those high frequencies to be falsely represented as lower frequencies in the digitized version.[1] This phenomenon, first formally described in the context of communication theory, violates the Nyquist-Shannon sampling theorem, which requires the sampling frequency to be at least twice the bandwidth of the signal (known as the Nyquist rate) to enable accurate reconstruction without distortion. Aliasing manifests across various fields, including signal processing, where it introduces erroneous low-frequency components into measured data, and computer graphics, where it produces visual anomalies like jagged edges or moiré patterns.[2]In digital signal processing, aliasing occurs during analog-to-digital conversion if the input signal contains frequencies above half the sampling rate, leading to spectral folding where aliased frequencies mirror around the Nyquist frequency.[3] For example, a 10 kHz tone sampled at 15 kHz may appear as a 5 kHz tone due to this folding effect, rendering the original signal irrecoverable without prior low-pass filtering.[4] Prevention typically involves anti-aliasing filters to attenuate high frequencies before sampling, ensuring compliance with the sampling theorem and maintaining signal integrity in applications such as audio recording and telecommunications.In computer graphics and imaging, aliasing results from discretizing continuous scenes onto pixel grids, where insufficient samples per pixel cause spatial frequencies to alias into visible artifacts, such as stair-stepping on curves or repetitive interference patterns in fine details.[5] This mimics undersampling in signal processing. Techniques like supersampling, where multiple samples are taken per pixel and averaged, or multisampling anti-aliasing (MSAA), mitigate these effects by increasing effective resolution, improving visual fidelity in video games, animations, and digital photography.[6]
Fundamentals
Definition
Aliasing is a distortion artifact in signal processing that arises when high-frequency components of a signal are misinterpreted as lower frequencies during the sampling or reconstruction process due to undersampling.[7] This phenomenon causes the original signal to be misrepresented in the digital domain, leading to false low-frequency components that were not present in the analog input.[4]The basic mechanism of aliasing occurs when the sampling rate f_s is below the Nyquist rate, defined as twice the highest frequency component in the signal; in such cases, frequencies above the Nyquist frequency f_N = f_s / 2 fold back into the principal frequency band from 0 to f_N, creating aliases. The aliased frequency f_a is given by the equationf_a = \left| f - k f_s \right|,where f is the original frequency, f_s is the sampling frequency, and k is the integer chosen to minimize f_a within the range [0, f_s/2].[8] This folding effect is a direct consequence of the periodic replication of the signal's spectrum in the frequency domain upon sampling.[9]Aliasing manifests in two primary forms: temporal aliasing, which affects time-domain signals like audio or video where sampling occurs at discrete time intervals, and spatial aliasing, which occurs in images or spatial arrays due to insufficient sampling density across space.[10] The Nyquist-Shannon sampling theorem provides the theoretical foundation for preventing aliasing by requiring a sampling rate at least twice the signal's bandwidth.[11]
Nyquist-Shannon Sampling Theorem
The Nyquist-Shannon sampling theorem states that a continuous-time signal x(t) that is bandlimited to a maximum frequency B (meaning its Fourier transform X(f) is zero for all |f| > B) can be perfectly reconstructed from its discrete samples x(nT) taken at uniform intervals T, provided the sampling rate f_s = 1/T satisfies f_s > 2B.[12] This minimum rate $2B is known as the Nyquist rate, ensuring that the original signal's information is fully preserved without distortion or loss.[11] The theorem establishes the fundamental limit for sampling in signal processing, guaranteeing exact recovery under the bandlimited assumption.[13]Mathematically, for a bandlimited signal with bandwidth B, the sampling frequency must exceed f_s > 2B to allow reconstruction. The perfect reconstruction formula, known as the cardinal series or sinc interpolation, expresses the continuous signal as:x(t) = \sum_{n=-\infty}^{\infty} x(nT) \cdot \operatorname{sinc}\left(\frac{t - nT}{T}\right),where \operatorname{sinc}(u) = \sin(\pi u)/(\pi u) is the normalized sinc function, and T = 1/f_s.[12] This interpolation leverages the orthogonality of the sinc basis functions, which span the space of bandlimited signals.[13]The proof relies on properties of the Fourier transform. The spectrum of the sampled signal consists of periodic replicas of the original spectrum X(f), shifted by multiples of f_s. If f_s > 2B, these replicas do not overlap, preserving the basebandspectrum X(f) for |f| < B intact; low-pass filtering then recovers x(t) exactly.[13] Overlap occurs precisely when f_s \leq 2B, leading to irreversible information loss through spectral folding, where higher frequencies masquerade as lower ones—a phenomenon known as aliasing.[14]
Signal Characteristics
Bandlimited Functions
A bandlimited function, or signal, is one whose Fourier transform vanishes outside a finite frequency interval, typically denoted as [-B, B] in hertz, where B is the bandwidth.[15] This property implies that the signal contains no energy at frequencies beyond B, making it a fundamental class in signal processing for avoiding aliasing during sampling.[15]The Paley-Wiener criterion provides a deeper characterization, stating that bandlimited functions are entire functions of exponential type in the complex plane, meaning they are analytic everywhere and grow no faster than exponentially along the real axis.[16] This analyticity arises from the compact support of the Fourier transform and underscores the smooth, non-local nature of such functions, which cannot be strictly time-limited without introducing high-frequency components.[16]Representative examples include ideal low-pass filtered signals, where the frequency spectrum is confined to [-B, B], and the sinc function, whose time-domain form \operatorname{sinc}(t) = \frac{\sin(\pi t)}{\pi t} corresponds exactly to a rectangular spectrum in the frequency domain.[17] In the context of the Nyquist-Shannon sampling theorem, only bandlimited signals permit perfect reconstruction from samples taken at a rate exceeding $2B; practical signals, which are not ideally bandlimited, rely on anti-aliasing filters to approximate this condition and suppress higher frequencies before sampling.[18][19]
Bandpass Signals
Bandpass signals are those whose frequency spectrum is confined to a specific band away from zero frequency, typically centered at a high carrier frequency f_c, with the spectral content limited to the interval [f_c - B/2, f_c + B/2], where B denotes the signal bandwidth and f_c \gg B. This structure distinguishes them from baseband signals by concentrating energy in a narrow high-frequency range, making them common in modulated communication systems.The sampling requirements for bandpass signals differ from those for lowpass signals, allowing a minimum sampling rate of $2B rather than $2(f_c + B/2), through a process known as bandpass sampling or undersampling. This is possible because the information content depends on the bandwidth B, not the absolute frequencies; however, the sampling frequency f_s must be carefully chosen so that the periodic spectral replicas do not overlap with the original band or each other after sampling. Bandpass signals represent a subset of bandlimited functions, but their shifted spectrum enables this relaxed rate when positioned correctly.[20]Aliasing occurs in bandpass sampling if an improper f_s causes overlap between the original spectrum and its images shifted by multiples of f_s, leading to irreversible distortion. To prevent this, f_s must satisfy the condition \frac{2(f_c + B/2)}{k} \geq f_s \geq \frac{2(f_c - B/2)}{k-1} for some positive integer k, where k ranges from 1 up to the largest value ensuring the lower bound does not exceed $2B. These ranges define "safe" bands for f_s, ensuring clean downconversion of the bandpass signal to baseband without interference.The primary advantage of bandpass sampling lies in significantly reducing the required data rate and hardware demands, as lower f_s values enable the use of slower, less expensive analog-to-digital converters while handling high-frequency signals. This is particularly valuable in communication applications, such as software-defined radios, where it facilitates direct digitization of intermediate-frequency signals, improving system efficiency and flexibility.[21]
Sinusoidal Sampling
Frequency Folding
Frequency folding provides a geometric interpretation of aliasing in the sampling of sinusoidal signals, where the frequency axis is repeatedly folded at multiples of half the sampling frequency f_s/2. This folding occurs because the sampling process creates periodic replicas of the signal's spectrum centered at integer multiples of the sampling frequency f_s, causing higher frequencies to overlap with lower ones in the baseband from -f_s/2 to f_s/2. Specifically, a frequency component f > f_s/2 maps to an aliased frequency f_s - f within the principal range, as the periodic nature of the sampled spectrum reflects frequencies across the folding points.[22]The folding diagram illustrates this process as a sawtooth pattern along the frequency axis, starting from zero and extending positively and negatively. The axis folds back upon itself at each odd multiple of f_s/2, such as f_s/2, $3f_s/2, and so on, creating a zigzag path that maps all frequencies to the interval [0, f_s/2]. Frequencies f + k f_s, for any integer k, alias to the same sampled values, as they land on equivalent positions after successive folds. This visual representation highlights how the infinite frequency axis is compressed into the finite Nyquist interval, with the direction of folding alternating to preserve the ambiguity between original and aliased components.[23]Mathematically, this equivalence arises from the periodicity and even symmetry of the cosine function in sampled sinusoids. Consider a continuous-time sinusoid \cos(2\pi f t) sampled at times t = n / f_s, yielding the discrete sequence \cos(2\pi f n / f_s). Due to the $2\pi-periodicity of cosine, \cos(2\pi (f + k f_s) n / f_s) = \cos(2\pi f n / f_s + 2\pi k n) = \cos(2\pi f n / f_s) for any integer k, demonstrating that frequencies differing by multiples of f_s produce identical samples.[8]In this framework, harmonics or frequency components above the Nyquist frequency f_s/2 fold back into the lower band, often appearing as lower-frequency impostors that distort the reconstructed signal. For instance, a high-frequency sinusoid just above f_s/2 folds to a low-frequency alias near zero, creating ambiguity in distinguishing the true origin without prior bandlimiting. This folding mechanism underscores the necessity of sampling above twice the maximum signal frequency to prevent such overlaps and ensure unambiguous recovery.[22]
Complex Sinusoids
A real-valued sinusoid \cos(2\pi f t) can be expressed as the real part of a complex exponential signal using Euler's formula: \cos(2\pi f t) = \Re \left\{ e^{j 2\pi f t} \right\}. This representation decomposes the sinusoid into symmetric positive and negative frequency components, e^{j 2\pi f t} and e^{-j 2\pi f t}, respectively. When sampling such a signal at rate f_s, the positive and negative frequencies alias symmetrically around multiples of the sampling frequency, as the discrete-time samples cannot distinguish between them due to the conjugate symmetry of the real sinusoid.[24]In the complex domain, aliasing manifests directly through the periodicity of the discrete-time complex exponential. A sampled complex sinusoid e^{j 2\pi f n / f_s} for integer n is mathematically identical to e^{j 2\pi (f + k f_s) n / f_s} for any integer k, rendering frequencies differing by integer multiples of f_s indistinguishable in discrete time.[24] This equivalence arises because e^{j 2\pi k n} = 1 for integer k and n, folding higher frequencies into the principal range [-f_s/2, f_s/2).[25]From a frequency-domain perspective, sampling a continuous-time signal multiplies its time-domain representation by a Dirac comb \sum_{n=-\infty}^{\infty} \delta(t - n/f_s), which corresponds to convolving the signal's spectrum with another Dirac comb in the frequency domain at intervals of f_s.[26] This convolution replicates the original spectrum at f + k f_s for all integers k, creating periodic images that overlap if the signal is not bandlimited to below f_s/2, thereby causing aliasing.[27]Unlike the real sinusoid case, which emphasizes symmetric folding due to conjugate pairs, the complex representation preserves phase information across aliases and facilitates analysis in tools like the discrete Fourier transform (DFT), where the periodic spectrum directly reveals replicated components.[28] This approach provides deeper insight into spectral distortions without relying on geometric visualizations of frequency folding.
Nyquist Frequency
The Nyquist frequency, denoted f_N, is defined as half the sampling frequency f_s, expressed as f_N = \frac{f_s}{2}. This frequency marks the upper limit of the basebandspectrum that can be faithfully captured in a discrete-time representation without distortion from aliasing.[29]The derivation of the Nyquist frequency stems from the Nyquist-Shannon sampling theorem, which analyzes the frequency-domain effects of sampling a continuous-time signal. Sampling multiplies the signal by a periodic impulse train, resulting in the Fourier transform of the sampled signal being a periodic repetition of the original spectrum, with copies spaced at intervals of f_s. To prevent overlap between the original spectrum and its replicas—which would introduce aliasing—the original signal must be bandlimited such that its highest frequency component does not exceed f_N; otherwise, higher frequencies fold into the baseband, corrupting the representation.[29]When a signal component at frequency f > f_N is present, it aliases to a lower frequency in the sampled domain, given by f_a = |f_s - f| for the primary folding zone between f_N and f_s. This folding exemplifies the boundary behavior where frequencies above the Nyquist limit masquerade as lower ones, leading to irreversible distortion unless mitigated.The Nyquist frequency plays a critical role in practical sampling systems by determining the cutoff for anti-aliasing filters, which are low-pass filters designed to attenuate components above f_N prior to sampling. Exceeding this threshold without such filtering causes irreversible aliasing, compromising signal integrity in applications like digital audio and imaging.[29]
Visual and Spatial Aliasing
Angular Aliasing
Angular aliasing manifests in imaging systems and sensors when spatial frequencies, particularly angular components, exceed the Nyquist limit—defined as half the sampling rate in pixels per cycle—resulting in their representation as lower frequencies or reversed motion directions in the captured image.[30] This occurs because the discrete sampling process cannot distinguish between the true high-frequency signal and its aliases, leading to distortions in perceived angular dynamics, such as in rotating objects viewed through pixelated sensors.[31]A prominent illustration of this phenomenon is the wagon-wheel effect, observed in film and video where the spokes of a rapidly rotating wheel appear to move backwards or halt. This illusion arises from the undersampling of the wheel's rotational motion by the frame rate, causing the true angular velocity to alias into a lower or negative apparent velocity. The aliased angular frequency is given by\omega_a = \omega - k \cdot f_swhere \omega is the true angular frequency, f_s is the sampling frame rate, and k is an integer that maps the frequency into the observable range below the Nyquist frequency.[32]The underlying mathematical model for angular aliasing in two-dimensional spatial sampling treats the image as a continuous function discretized on a rectangular grid, equivalent to multiplication by a comb function in the spatial domain. In the Fourier domain, this sampling produces infinite periodic replicas of the original spectrum centered at multiples of the sampling frequencies in both dimensions. Overlap between these replicas—known as spectral folding—causes high angular spatial frequencies to alias into the principal low-frequency band, distorting the reconstructed image's angular content.[33]Representative examples include stroboscopic illusions in video footage of rotating machinery, where undersampled angular motion creates apparent reversals akin to the wagon-wheel effect, and moiré patterns on digital displays, where the interaction between the display's pixel lattice and high-frequency angular patterns in the content generates spurious low-frequency waves.[34] This mirrors frequency folding in one-dimensional temporal sampling but applies to spatial angular domains.[31]
Imaging and Graphics Examples
Spatial aliasing in digital images arises when the spatial frequencies present in a scene exceed the Nyquist frequency of the imagingsensor, causing high-frequency details to be misrepresented as lower-frequency artifacts, such as jagged lines or "jaggies" along edges. This occurs because the sensor samples the continuous light field at discrete pixel locations, and if the sampling rate is insufficient, fine details like sharp edges or patterns are undersampled, leading to distortion where the edge appears stepped or wavy rather than smooth. For instance, in photography, capturing a scene with high-contrast linear features, such as fabric weaves or architectural lines, can produce these jaggies if the lens projects frequencies beyond the sensor's capability onto the pixel array.[35][36]The spatial Nyquist frequency, which defines the maximum resolvable spatial frequency without aliasing, is given by f_{N,spatial} = \frac{1}{2p}, where p is the pixel pitch (the distance between adjacent pixel centers). For a typical sensor with a pixel pitch of 5 μm, this yields f_{N,spatial} = 100 cycles per millimeter, meaning any scene detail finer than this limit will alias. In practice, sensors with smaller pitches, such as 18 μm in mid-infrared arrays, have a Nyquist frequency of approximately 27.8 cycles per millimeter, highlighting how hardware design directly impacts aliasing susceptibility in imaging applications.[37][38]In computer graphics, aliasing manifests similarly during rendering processes like rasterization or ray tracing, where scene geometry or textures are sampled onto a discretepixel grid without adequate oversampling, resulting in moiré patterns or texture warping. During rasterization, infinite-frequency components of polygons, such as edges, alias into stairstep jaggies or crawling artifacts when projected onto the pixel grid, as the sampling fails to capture the continuous nature of the geometry. Ray tracing encounters comparable issues when rays sample high-frequency details, like specular highlights or fine geometry, leading to visible distortions if the ray density per pixel is too low. Texture warping, a specific form of aliasing, occurs in minification scenarios where distant or angled textures are undersampled; without proper level-of-detail selection, such as in mipmapping failure, a single screen pixel may integrate multiple mismatched texels, producing shimmering or incorrect patterns.[39][40]Mitigation of these artifacts often involves anti-aliasing techniques like spatial filtering or supersampling, but the root cause remains the violation of the spatial Nyquist criterion, emphasizing the need for pre-filtering high frequencies in both imaging sensors and graphics pipelines to bandlimit the input signal. For example, optical low-pass filters in cameras blur fine details to prevent aliasing at the sensor level, while in graphics, techniques like multisample anti-aliasing average multiple samples per pixel to approximate continuous integration.[36][39]
Audio and Directional Applications
Audio Aliasing
In digital audio processing, aliasing arises when audio signals containing frequencies higher than half the sampling rate, known as the Nyquist frequency f_s/2, are sampled without adequate low-pass filtering. These high frequencies fold back into the lower audible spectrum through the process of frequency folding, producing false tones or artifacts that were not present in the original signal.[41] For instance, in compact disc (CD) audio, the standard sampling rate of 44.1 kHz establishes a Nyquist frequency of 22.05 kHz, which exceeds the typical upper limit of human hearing at 20 kHz to prevent such high-frequency content from aliasing into the audible range below 20 kHz.[42][43]A concrete example illustrates this mechanism: sampling a 10 kHz sine wave at a 15 kHz rate, where the Nyquist frequency is 7.5 kHz, results in the 10 kHz component aliasing to 5 kHz because the sampled waveform points coincide exactly with those of a true 5 kHz sine wave, making the alias indistinguishable from a genuine low-frequency signal without additional context.[44] This folding can manifest audibly as whistles, beats, or inharmonic tones when high-pitched sounds exceed the Nyquist limit, as the aliased frequencies interfere with the intended audio content.[41]In practical applications like digital guitar amplifier modeling, nonlinear distortion processes generate high-frequency harmonics that, if not properly filtered, alias into the audible band due to the system's sampling rate limitations, often from clock-related artifacts or oversampling deficiencies.[45] These aliases appear as unwanted harshness or metallic ringing, detectable as non-harmonic overtones that degrade the natural warmth of analog-style distortion.[41] To mitigate this, audio engineers employ anti-aliasing filters before sampling and oversampling techniques during processing to shift potential aliases beyond the audible range.[41]
Direction Finding
In direction finding applications, such as radar and sonar systems employing uniform linear arrays (ULAs) of sensors, spatial aliasing occurs when the inter-element spacing exceeds half the signal wavelength, leading to ambiguities in estimating the direction of arrival (DOA) of incoming plane waves. This phenomenon, analogous to temporal aliasing in sampling, causes the array's spatial frequency response to fold, resulting in multiple possible angles that produce identical phase measurements across the array.[46]The phase difference between signals received at adjacent elements in a ULA is \Delta \phi = \frac{2\pi d \sin\theta}{\lambda}, where d is the spacing between elements, \theta is the true angle of arrival relative to the array broadside, and \lambda is the wavelength. When d > \lambda/2, this phase difference can wrap around multiples of $2\pi, causing the array factor to exhibit periodic replicas known as grating lobes, which represent spatial aliases of the main lobe.These grating lobes introduce DOA ambiguity, as the true angle \theta can alias to erroneous directions \theta_a satisfying \sin\theta_a = \sin\theta + k \cdot \frac{\lambda}{d}, where k is a nonzero integer. For example, with d = \lambda, a signal from \theta = 0^\circ (broadside) aliases to \theta_a = \pm 90^\circ, or more generally, the array confuses directions separated by approximately $180^\circ / (d/\lambda). In practical antenna arrays for radio direction finding, this effect can cause undersampled systems to misinterpret signals from rear lobes as originating from the front, severely degrading localization accuracy.[47]To prevent such aliasing and ensure unambiguous DOA estimation within the visible angular range (-90^\circ \leq \theta \leq 90^\circ), array designs constrain d < \lambda/2, though this trades off against aperture size and resolution; the focus here remains on how violations lead to grating lobe-induced errors.[48]
Historical Context
Early Developments
The foundations of understanding aliasing trace back to the early 19th century with Joseph Fourier's seminal work on heat conduction, published in 1822 as Théorie analytique de la chaleur. In this text, Fourier introduced the concept of representing arbitrary functions as sums of sinusoidal components through Fourier series, establishing the groundwork for spectral analysis of signals—a prerequisite for recognizing frequency-related distortions like aliasing, though the phenomenon itself was not yet identified or named.[49]A visual manifestation of aliasing emerged in the 1830s through mechanical devices demonstrating temporal sampling effects. In 1832, Austrian mathematician and inventor Simon von Stampfer developed the stroboscope, a rotating disk with slits that created intermittent illumination to observe motion, inadvertently revealing the stroboscopic effect where continuous rotation appeared stationary or reversed due to undersampling of angular frequencies—now recognized as angular aliasing akin to the wagon-wheel illusion.[50] This pre-digital experiment provided one of the earliest empirical observations of aliasing principles, highlighting how discrete sampling rates could misrepresent higher-frequency motions as lower ones.[50]Early ideas of signal sampling appeared in telegraphy during the mid-19th century, where systems like those developed by Émile Baudot in the 1870s multiplexed discrete pulses over wires to transmit multiple messages, foreshadowing sampling limitations without explicit aliasing analysis.[51] By the 1930s, radio engineering encountered aliasing in receiver designs; Karl Jansky's pioneering radio astronomy experiments at Bell Laboratories (1931–1933) involved directional antennas detecting broadband static, contributing to early advancements in radio signal processing.[52]The transition to systematic aliasing issues occurred during World War II with the deployment of pulsed radar systems. These radars, such as the British Chain Home network operational by 1938 and expanded through the war, used pulse repetition frequencies (PRFs) to measure range; choices of PRF involved trade-offs between range and velocity ambiguities, where low PRF ensured unambiguous range for long-range detection while potentially introducing Doppler ambiguities, complicating target discrimination in some applications.[53] This practical challenge in analog pulsed signals prompted early engineering mitigations, building on prior theoretical work such as Nyquist's 1928 contributions to sampling theory and contributing to later formalizations like the Nyquist-Shannon sampling theorem.[54]
Evolution in Digital Signal Processing
The formalization of the sampling theorem in the mid-20th century marked a pivotal milestone in understanding aliasing within digital signal processing. The groundwork for the sampling theorem was laid by Harry Nyquist in his 1928 paper "Certain Topics in Telegraph Transmission Theory," where he determined that a signal must be sampled at least twice its highest frequency to avoid distortion from higher frequencies appearing as lower ones.[54] In his 1949 paper, Claude Shannon rigorously established the conditions under which a continuous-time signal could be perfectly reconstructed from its samples, emphasizing that frequencies above half the sampling rate would overlap and distort the baseband spectrum—a phenomenon he described in terms of spectral folds that prefigure modern aliasing concepts. This work built on earlier analog foundations from the 1930s, providing the theoretical bedrock for digital applications by quantifying how undersampling leads to irreversible information loss through frequency folding.The 1960s and 1970s saw aliasing emerge as a practical concern with the rise of digital computers, enabling widespread computation of discrete signals. The advent of efficient algorithms like the Cooley-Tukey fast Fourier transform (FFT) in 1965 transformed spectral analysis, revealing aliasing artifacts in discrete spectra by computing the periodic nature of the discrete Fourier transform, where high frequencies wrap around and contaminate lower bins. This efficiency, reducing computation from O(N²) to O(N log N), facilitated the visualization and mitigation of aliasing in early DSP systems, such as radar and seismic processing, where discrete spectra highlighted the need for anti-aliasing filters.By the 1980s, aliasing considerations drove standardization in consumer technologies, exemplified by the audio compact disc (CD) format. Developed jointly by Sony and Philips, the CD adopted a sampling rate of 44.1 kHz to accommodate the human auditory bandwidth up to 20 kHz while providing sufficient guardband for practical anti-aliasing filters with feasible transition slopes, ensuring minimal distortion from spectral overlap.[55] This choice balanced reconstruction fidelity against hardware constraints, influencing subsequent digital audio standards and underscoring aliasing avoidance as a core design principle in DSP.[55]In recent developments through 2025, aliasing has posed new challenges in advanced domains like machine learning and quantum computing. In convolutional neural networks for image processing, aliasing arises during downsampling operations, degrading generalization by introducing spurious high-frequency artifacts that mimic low-frequency patterns, as demonstrated in analyses showing up to 10-15% drops in accuracy on shifted test sets without anti-aliasing measures like blur pooling.[56] Similarly, quantum machine learning encounters "quantum aliasing" due to data scarcity and binarization in quantum state sampling, where limited measurements cause overlapping representations that hinder model training, with studies reporting error rates exceeding 20% in low-sample regimes for classification tasks.[57] These issues highlight ongoing efforts to adapt classical aliasing mitigations, such as oversampling, to quantum and neural contexts for robust signal reconstruction.[56][57]