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Signal

In The Signal by William Powell Frith, a woman sends a signal by waving a white handkerchief. A signal is both the process and the result of transmission of data over some media accomplished by embedding some variation. Signals are important in multiple subject fields including signal processing, information theory, and biology. In signal processing, a signal is a function that conveys information about a phenomenon.^[1] Any quantity that can vary over space or time can be used as a signal to share messages between observers.^[2] The IEEE Transactions on Signal Processing includes audio, video, speech, image, sonar, and radar as examples of signals.^[3] A signal may also be defined as any observable change in a quantity over space or time (a time series), even if it does not carry information. In nature, signals can be actions done by an organism to alert other organisms, ranging from the release of plant chemicals to warn nearby plants of a predator, to sounds or motions made by animals to alert other animals of food. Signaling occurs in all organisms even at cellular levels, with cell signaling. Signaling theory, in evolutionary biology, proposes that a substantial driver for evolution is the ability of animals to communicate with each other by developing ways of signaling.) In human engineering, signals are typically provided by a sensor, and often the original form of a signal is converted to another form of energy using a transducer. For example, a microphone converts an acoustic signal to a voltage waveform, and a speaker does the reverse.^[1] Another important property of a signal is its entropy or information content. Information theory serves as the formal study of signals and their content. The information of a signal is often accompanied by noise, which primarily refers to unwanted modifications of signals, but is often extended to include unwanted signals conflicting with desired signals (crosstalk). The reduction of noise is covered in part under the heading of signal integrity. The separation of desired signals from background noise is the field of signal recovery, one branch of which is estimation theory, a probabilistic approach to suppressing random disturbances.)

Definitions and Basic Concepts

Definition of a Signal

A signal is a function that conveys information about a phenomenon, typically by varying with respect to an independent variable such as time or position.^[4] This variation represents physical quantities, including voltage in electrical circuits, sound pressure in acoustics, or light intensity in optics.^[5] In essence, signals serve as carriers of information, encoding data about the state or behavior of a system or process.^[6] Signals appear across diverse fields, adapting to the specific phenomena they describe. In physics, they manifest as electromagnetic waves propagating energy and information through space.^[7] In engineering, particularly electrical and electronics, signals often take the form of measurable electrical voltages or currents that drive devices and systems.^[5] Biological contexts feature signals like neural impulses, which are electrochemical events traveling along neuron membranes to transmit sensory or motor information.^[8] In communications, signals encode messages for transmission over channels, such as radio waves carrying audio data.^[9] The concept of a signal originated in the context of early communication technologies like telegraphy, where electrical pulses were used to send messages over wires in the 19th century.^[9] This evolved with the advent of radio and further formalized through mathematical analysis of waves by Joseph Fourier in 1822, who developed techniques to decompose complex waveforms into simpler components, laying the groundwork for modern signal theory.^[10] To understand signals without prior knowledge, consider the independent variable—often time, denoted as t—as the domain over which the signal is defined, and the dependent variable—such as amplitude, denoted as x(t)—as the value that changes to represent the information.^[4]

Mathematical Representation

In signal processing, a continuous-time signal is mathematically represented as a function x(t), where the independent variable t belongs to the set of real numbers \mathbb{R}, indicating that the signal is defined for all values of time along a continuum.^[11] Conversely, a discrete-time signal is denoted as x, where the independent variable n is an integer from the set \mathbb{Z}, signifying that the signal exists only at discrete, countable instants of time.^[11] These notations provide a formal framework for modeling how signals vary with respect to time, enabling precise analysis and computation.^[12] From a vector space perspective, signals are treated as elements within appropriate function spaces, such as the space of square-integrable functions L^2, which forms an infinite-dimensional vector space over the complex numbers \mathbb{C}.^[13] In this context, the inner product between two signals x(t) and y(t) is defined as \langle x, y \rangle = \int_{-\infty}^{\infty} x(t) y^*(t) \, dt, where y^*(t) is the complex conjugate of y(t); this operation quantifies the similarity or correlation between signals and satisfies properties like linearity and positive-definiteness.^[13] Such a structure allows signals to be manipulated using linear algebra tools, including norms derived from the inner product as \|x\| = \sqrt{\langle x, x \rangle}, which measures the "length" or energy of the signal.^[13] Graphically, signals are commonly depicted in the time domain through plots where the horizontal axis represents time t (or n for discrete cases) and the vertical axis represents the signal's amplitude, providing a visual representation of how the signal evolves over time.^[14] The amplitude units depend on the physical context, such as volts (V) for electrical signals, pascals (Pa) for acoustic pressure signals, or dimensionless quantities for normalized representations, while time is typically measured in seconds (s).^[15]^[14] Basic operations on signals include addition and scaling, which preserve the vector space structure. For continuous-time signals, addition yields z(t) = x(t) + y(t), and scaling produces w(t) = a x(t) for a scalar constant a; analogous operations apply to discrete-time signals as z = x + y and w = a x.^[11] These operations form the foundation for more complex manipulations, such as linear combinations like v(t) = a x(t) + b y(t), without altering the fundamental time-domain representation.^[11]

Classification of Signals

Analog versus Digital Signals

Analog signals are continuous in both time and amplitude, representing physical phenomena such as sound waves or electrical voltages that vary smoothly without discrete steps.^[16] For instance, the grooves on a vinyl record encode audio as a continuous variation in depth, while radio waves propagate electromagnetic signals with continuously varying amplitude and frequency. This continuity allows analog signals to capture infinite resolution in amplitude, theoretically preserving the full fidelity of natural processes.^[17] However, analog signals are highly susceptible to noise and interference, as any distortion introduced during transmission or processing accumulates and degrades the signal quality.^[18] In contrast, digital signals are discrete in amplitude, represented by a finite set of quantized levels, and are typically sampled at discrete time intervals to form sequences of binary data.^[16] For example, audio on a compact disc (CD) is digitized into a binary sequence of 16-bit samples taken at 44.1 kHz, enabling exact replication through error-checking mechanisms.^[19] This quantization approximates the original continuous amplitude with steps, such as 65,536 levels for 16-bit audio, which limits resolution but facilitates robust storage and manipulation in computers.^[20] The conversion between analog and digital domains involves specific processes to bridge their differing natures. Analog-to-digital conversion (ADC) first samples the continuous-time signal at regular intervals and then quantizes each sample's amplitude to the nearest discrete level, producing a digital code.^[21] The reverse, digital-to-analog conversion (DAC), reconstructs an approximate analog waveform from digital samples using interpolation or filtering techniques.^[22] A critical consideration in ADC is the Nyquist rate, which states that to avoid aliasing—where high frequencies masquerade as lower ones—the sampling frequency must be at least twice the highest frequency component in the signal, as established by the Nyquist-Shannon sampling theorem.^[23] Analog signals offer advantages in natural fidelity and simplicity for direct representation of real-world phenomena, such as in high-end audio reproduction where continuous variations avoid quantization artifacts.^[17] However, their disadvantages include vulnerability to noise, lack of built-in error correction, and challenges in long-term storage without degradation.^[18] Digital signals, conversely, excel in error detection and correction through redundancy, ease of processing via algorithms, and reliable storage in binary formats, making them preferable for modern communications and computing despite introducing minor fidelity losses from quantization.^[24] This trade-off is exemplified by Shannon's sampling theorem, which underpins digital signal integrity by ensuring faithful reconstruction when sampling adheres to the Nyquist criterion.^[25]

Continuous-Time versus Discrete-Time Signals

Continuous-time signals are functions defined for all values of time t in a continuous interval, typically the real numbers, representing physical quantities that vary smoothly over time. For instance, a cosine wave x(t) = \cos(2\pi f t), where f is the frequency, exemplifies a continuous-time signal commonly encountered in natural phenomena like sound waves or electrical voltages.^[26]^[27] Analysis and modeling of systems processing these signals often involve solving linear differential equations to describe their dynamic behavior.^[28]^[29] In contrast, discrete-time signals are sequences defined only at discrete instants, typically at integer multiples n of a fixed sampling period T, denoted as x = x(nT). These signals arise primarily from sampling continuous-time signals to convert them into a form suitable for numerical processing, enabling representation as finite or countably infinite sequences of values.^[30]^[31] The transition from continuous to discrete time is governed by the Nyquist-Shannon sampling theorem, which states that a continuous-time signal bandlimited to a maximum frequency f_{\max} can be perfectly reconstructed if sampled at a rate f_s > 2f_{\max}, preventing aliasing where higher frequencies masquerade as lower ones.^[32] This theorem, formalized by Claude Shannon in 1949, ensures no information loss under proper sampling conditions.^[32] Reconstruction of the original continuous-time signal from its discrete samples requires an ideal low-pass filter with a cutoff at f_s/2 to eliminate spectral replicas introduced by sampling, yielding perfect recovery via sinc interpolation when the sampling theorem is satisfied.^[33]^[34] Continuous-time signals find primary application in analog circuits, such as amplifiers and filters in audio equipment, where real-world phenomena are processed without digitization.^[35] Conversely, discrete-time signals are essential in digital computers and processors, facilitating efficient algorithmic manipulation and storage of sampled data.^[36]

Periodic versus Aperiodic Signals

In signal processing, signals are classified as periodic or aperiodic based on whether they exhibit repetition over time. A continuous-time signal x(t) is periodic if there exists a positive constant T > 0, known as the fundamental period, such that x(t + T) = x(t) for all t.^[37]^[38] The fundamental period T is the smallest such value, and the corresponding fundamental angular frequency is \omega = 2\pi / T radians per second.^[37] A classic example of a periodic signal is the sine wave, given by x(t) = \sin(2\pi t / T), which repeats every T seconds and has a discrete line spectrum consisting of impulses at integer multiples of the fundamental frequency $1/T.^[37]^[39] Periodic signals are prevalent in applications like alternating current power systems and musical tones, where the repetition enables efficient analysis using Fourier series representations that yield discrete frequency components.^[40] In contrast, an aperiodic signal does not repeat with any fixed period T, meaning no such T > 0 satisfies x(t + T) = x(t) for all t.^[37] Examples include a single rectangular pulse, which is nonzero only over a finite duration, or an exponential decay x(t) = e^{-t} for t \geq 0 (and zero otherwise), both of which lack ongoing repetition.^[41] Aperiodic signals, common in transient phenomena like impulse responses in systems or one-time events in communications, possess continuous frequency spectra when analyzed via the Fourier transform.^[39] A related class is almost periodic signals, which are limits of periodic signals and can be expressed as sums of periodic components with incommensurate periods (i.e., periods that share no common divisor).^[42] These signals, such as the sum of two sinusoids with irrational frequency ratios, exhibit quasi-repetition but no exact period, appearing in natural processes like planetary motions or certain modulated waves.^[43] Their spectra are discrete but denser than those of strictly periodic signals, bridging the gap between periodic and aperiodic behaviors.^[42]

Deterministic versus Random Signals

Deterministic signals are those whose values at any given time can be precisely predicted using a mathematical equation or a set of initial conditions, without any uncertainty.^[44] For example, a sinusoidal signal expressed as x(t) = A \sin(\omega t + \phi), where A is the amplitude, \omega is the angular frequency, and \phi is the phase, is deterministic and often periodic.^[45] Another example is a ramp signal, x(t) = k t for t \geq 0, which increases linearly and is fully specified by its slope k.^[44] In contrast, random signals, also known as stochastic signals, exhibit variability such that their exact values cannot be predicted deterministically; instead, they are characterized probabilistically through statistical measures derived from an ensemble of possible realizations.^[44] The mean function, defined as \mu_x(t) = E[x(t)], where E[\cdot] denotes the expected value, provides the average value of the signal at time t.^[44] The autocorrelation function, R_x(t, t+\tau) = E[x(t) x(t+\tau)], quantifies the correlation between the signal's values at times separated by lag \tau.^[45] A subclass of random signals involves stationary processes, where statistical properties remain invariant over time shifts. Wide-sense stationary (WSS) processes are defined by a constant mean, \mu_x(t) = \mu for all t, and an autocorrelation that depends solely on the time difference, R_x(\tau) = E[x(t) x(t+\tau)].^[44] Examples of random signals include Gaussian white noise, which has a zero mean, constant variance, and flat autocorrelation within its bandwidth, modeling thermal noise in electronic systems.^[45] Deterministic signals find primary applications in control systems, where precise predictability enables exact modeling and feedback design.^[44] Random signals are essential in communications for modeling channel noise and interference, allowing probabilistic analysis of error rates and signal detection.^[45]

Energy versus Power Signals

In signal processing, signals are classified as energy signals or power signals based on the finiteness of their total energy and average power, providing a framework for analyzing their magnitude characteristics over time. This classification applies to deterministic signals, where energy and power can be computed directly using integrals.^[46] An energy signal has finite total energy but zero average power. The total energy E of a continuous-time signal x(t) is given by

E = \int_{-\infty}^{\infty} |x(t)|^2 \, dt < \infty,

where the squared magnitude |x(t)|^2 represents the instantaneous power, and the integral sums this over all time. For electrical signals, energy is measured in joules (J). A typical example is a pulse signal, such as a rectangular pulse of finite duration, which has nonzero energy confined to a limited time interval but dissipates to zero thereafter, resulting in an average power of zero.^[46]^[47] In contrast, a power signal has infinite total energy but finite nonzero average power. The average power P is defined as

P = \lim_{T \to \infty} \frac{1}{2T} \int_{-T}^{T} |x(t)|^2 \, dt < \infty,

capturing the long-term average of the instantaneous power. For electrical signals, power is measured in watts (W). Periodic sinusoids, such as x(t) = \cos(\omega_0 t), exemplify power signals because their energy accumulates indefinitely over infinite time, yet the average power remains constant and finite due to the repeating pattern.^[46] Some signals fit neither category or exhibit hybrid properties. A direct current (DC) signal, like x(t) = A (constant amplitude), is a power signal with finite average power P = |A|^2 but infinite energy, as the integral diverges over infinite time. Conversely, a finite-duration periodic signal, such as a sinusoid truncated to a specific interval, qualifies as an energy signal because its total energy is finite, while the average power approaches zero as the observation window expands. Signals like x(t) = t (linearly growing) are neither, possessing both infinite energy and infinite power. No signal can be both an energy signal and a power signal simultaneously, as finite energy implies zero average power.^[46] Parseval's theorem establishes that the total energy of a signal is equivalent whether calculated in the time domain or the frequency domain, linking the L^2 norm in both representations without altering the value.^[47]

Even versus Odd Signals

In signal processing, signals are classified based on their symmetry properties with respect to the origin at t = 0. An even signal x(t) satisfies the condition x(-t) = x(t) for all t, meaning it is symmetric about the vertical axis passing through the origin.^[48] This symmetry implies that the signal values at t and -t are identical, resulting in a mirror image across the y-axis in the time domain.^[30] Conversely, an odd signal x(t) obeys x(-t) = -x(t), exhibiting antisymmetry about the origin, where the signal at -t is the negative of the value at t, and x(0) = 0 if defined.^[37] These properties are fundamental for analyzing signal behavior under reflection.^[49] A classic example of an even signal is the cosine function \cos(\omega t), which remains unchanged when t is replaced by -t since \cos(-\omega t) = \cos(\omega t).^[50] In contrast, the sine function \sin(\omega t) is odd because \sin(-\omega t) = -\sin(\omega t), flipping the sign across the origin.^[51] These trigonometric signals illustrate how even and odd symmetries manifest in periodic waveforms commonly encountered in electrical engineering and physics.^[52] Any arbitrary signal x(t) can be uniquely decomposed into its even and odd components, expressed as x(t) = x_e(t) + x_o(t), where the even part is x_e(t) = \frac{x(t) + x(-t)}{2} and the odd part is x_o(t) = \frac{x(t) - x(-t)}{2}.^[37] This decomposition holds for both continuous- and discrete-time signals and allows for separating symmetric and antisymmetric behaviors, facilitating targeted analysis.^[53] The even component captures all symmetric information, while the odd component isolates the antisymmetric aspects.^[30] In the context of Fourier analysis, even signals possess purely real Fourier coefficients, reflecting their cosine-like symmetry, whereas odd signals yield purely imaginary coefficients due to their sine-like antisymmetry.^[54] This property arises because the Fourier transform of an even function integrates to a real-valued output, while for odd functions, it results in an imaginary spectrum.^[55] Such characteristics simplify the computation of transforms by reducing the number of terms needed. The symmetry of even and odd signals has practical applications in simplifying the analysis of linear systems, such as in electrical circuits where waveform symmetry aids in predicting responses to symmetric inputs, and in mechanical vibrations where even or odd modes of oscillation reduce computational complexity in modeling structural dynamics.^[51] For instance, decomposing vibrations into even and odd parts streamlines eigenvalue problems in modal analysis.

Properties and Analysis

Amplitude and Phase

In signal processing, the amplitude of a signal quantifies its magnitude or strength, often represented as the peak value from the zero axis for sinusoidal waveforms, such as the coefficient A in the expression A \sin(\omega t).^[56] For more general signals, amplitude can refer to the envelope that bounds the signal's oscillations, which varies over time in modulated signals like amplitude-modulated carriers where the envelope follows the modulating waveform.^[57] This peak or envelope measure determines the signal's deviation from its baseline, directly influencing its detectability and power in transmission systems.^[58] The phase of a signal describes its temporal shift or alignment relative to a reference, typically expressed as the argument \phi in \sin(\omega t + \phi), where \phi is measured in radians or degrees and corresponds to a time delay of \phi / \omega.^[59] This shift indicates the starting point of the oscillation cycle, affecting how the signal aligns with others; for instance, a phase of zero aligns the waveform with the reference, while a \pi radian shift inverts it.^[60] In even and odd signal contexts, cosine (even) and sine (odd) represent phase offsets of 0 and \pi/2 radians, respectively, relative to a base sinusoid.^[61] For non-sinusoidal or complex signals, instantaneous amplitude and phase provide time-varying measures derived from the analytic signal, formed by applying the Hilbert transform to shift the signal's negative frequencies to positive ones, yielding a complex representation from which amplitude is the modulus and phase is the argument.^[62] The instantaneous amplitude captures the local envelope, while the instantaneous phase tracks the evolving argument, enabling analysis of signal behavior at each moment without assuming periodicity.^[63] This approach is particularly useful for monocomponent signals, where the phase's derivative further yields instantaneous frequency, though the focus here remains on amplitude and phase extraction.^[64] Signals are often normalized to unit amplitude, scaling the waveform by dividing by its peak or RMS value, to facilitate comparisons of shape or phase without magnitude bias.^[58] Amplitude directly scales a signal's energy, as the total energy E = \int |x(t)|^2 dt increases quadratically with amplitude factor a for a x(t), establishing its role in power budgeting.^[30] Phase, meanwhile, governs interference in superpositions, where aligned phases (\Delta \phi = 0) yield constructive addition amplifying the resultant amplitude, and opposing phases (\Delta \phi = \pi) cause destructive cancellation reducing it to zero.^[65]

Frequency Content

The frequency content of a signal refers to its decomposition into sinusoidal components of varying frequencies, revealing how the signal's energy or power is distributed across the frequency spectrum. This spectral representation provides insight into the signal's characteristics, such as its oscillatory behavior and information-carrying capacity. For instance, the spectrum illustrates the allocation of energy to different frequencies, where the magnitude at each frequency indicates the contribution of that component to the overall signal.^[66] In periodic signals, the spectrum consists of discrete lines at the fundamental frequency and its harmonics, forming a line spectrum where energy is concentrated at specific frequencies. Aperiodic signals, in contrast, exhibit a continuous spectrum, with energy distributed smoothly over a range of frequencies rather than at discrete points. This distinction arises because periodic signals repeat at regular intervals, limiting their frequency components to multiples of the fundamental period, while aperiodic signals lack such repetition, resulting in a broader, continuous energy distribution.^[67]^[68] Harmonics are the integer multiples of the fundamental frequency in periodic signals, such as the second harmonic at 2f, third at 3f, and so on, where f is the fundamental. These components determine the signal's shape; for example, odd harmonics often dominate in symmetric waveforms like square waves, contributing to their sharp transitions. The amplitudes of higher harmonics typically decrease, but their presence shapes the signal's timbre or waveform profile.^[69]^[70] Bandwidth defines the range of frequencies within the spectrum that contain the significant portion of the signal's energy, typically measured as the difference between the highest and lowest frequencies of interest (B = f_max - f_min). Baseband signals occupy a low-frequency range starting near zero, such as audio signals from 20 Hz to 5 kHz, suitable for direct transmission over short distances. Bandpass signals, however, are shifted to a higher frequency band around a carrier, maintaining the same bandwidth but enabling efficient long-range propagation, as in radio communications. Conceptually, the Dirac delta function models impulses in the frequency spectrum, representing idealized point concentrations of energy at specific frequencies, such as the discrete lines in a periodic signal's spectrum. For a time-domain impulse, its spectrum is a constant across all frequencies, indicating equal energy distribution, while periodic impulses yield a comb of deltas at harmonic frequencies. This abstraction aids in understanding spectral sparsity and sampling effects.^[71] Higher frequencies in the spectrum carry information about sharper details and rapid changes in the signal, such as edges in images or transients in time series, where abrupt variations require high-frequency components to represent fidelity. In image processing, for example, edges and contours are encoded in high spatial frequencies, enabling enhancement techniques that boost these components for improved sharpness.^[72]

Time and Frequency Domains

In the time domain, signals are analyzed directly as functions of time, such as x(t) for continuous-time signals, allowing observation of temporal characteristics like duration, amplitude peaks, and overall waveform shape. This perspective is particularly useful for assessing signal similarity through techniques like autocorrelation, which measures how a signal correlates with a time-shifted version of itself, revealing periodicities or redundancies without requiring frequency decomposition. For instance, autocorrelation can detect echoes in audio signals by identifying lags where the function peaks, providing insights into signal structure solely from time-based data.^[73] In contrast, the frequency domain represents the signal as a function of frequency, denoted as X(\omega), obtained via mathematical transforms that decompose the signal into its constituent frequency components. This view elucidates the spectral composition, such as dominant frequencies or harmonic content, facilitating operations like filtering where unwanted frequency bands are attenuated by multiplying the spectrum with a filter's transfer function. Frequency-domain analysis often simplifies the study of linear systems, as it reveals how signals are built from sinusoidal basis functions, enabling efficient manipulation of broadband phenomena that may be obscured in the time domain.^[74] A fundamental trade-off exists between time and frequency resolutions in signal analysis: improving localization in one domain degrades it in the other, analogous to an uncertainty principle where short-duration signals yield broad frequency spreads, and vice versa. This limitation arises because finite observation windows constrain the precision of both temporal pinpointing and spectral detail, impacting applications like real-time processing where balancing the two is essential.^[75] The convolution theorem states that the convolution of two signals in the time domain corresponds to the pointwise multiplication of their frequency-domain representations, bridging the domains for efficient computation in filtering and system analysis.^[76] To address the resolution trade-off, multiresolution analysis using wavelets provides a joint time-frequency representation, allowing variable window sizes that offer high time resolution for transients and high frequency resolution for steady components. Wavelets emerged in the 1980s through contributions from researchers like Stéphane Mallat, who linked multiresolution frameworks to digital signal processing, enabling scalable decomposition of signals into localized basis functions beyond fixed Fourier methods.^[77]

Examples of Signals

Everyday and Natural Signals

Everyday and natural signals encompass a wide array of phenomena encountered in daily life and the environment, serving as fundamental examples of how information is transmitted through physical variations. These signals often arise without human intervention, manifesting as continuous fluctuations that convey essential data about their sources, such as vibrations, energy propagation, or biological rhythms.^[78] Sound waves represent one of the most ubiquitous natural signals, originating from sources like speech and music through rapid variations in air pressure. In speech, the human voice produces these waves by modulating airflow from the lungs, creating pressure changes that propagate as longitudinal waves detectable by the ear. Similarly, musical instruments generate sound waves via mechanical vibrations—such as a guitar string's oscillation—that translate into pressure variations in the surrounding medium, allowing the transmission of harmonic patterns over distances. These acoustic signals are inherently analog and continuous in time, capturing the nuanced dynamics of natural sound production.^[79]^[80]^[81] Light serves as another prevalent natural signal, primarily through the visible spectrum of electromagnetic waves emitted or reflected by objects in the environment. Sunlight, for instance, consists of a continuous range of wavelengths from approximately 400 to 700 nanometers, which the human eye perceives as colors from violet to red, enabling visual recognition of surroundings. This electromagnetic propagation occurs at the speed of light in vacuum, carrying information about the source's temperature and composition without requiring a medium.^[82]^[83] Seismic waves provide a powerful example of natural signals generated by geological events, such as earthquakes, where energy release in the Earth's crust produces propagating disturbances. These waves travel through the planet's interior and surface, manifesting as body waves (P-waves compressing material longitudinally and S-waves shearing transversely) and surface waves (Love and Rayleigh waves causing ground rolling). Detected by seismographs, they reveal details about subsurface structures and event magnitudes, often exhibiting complex, irregular patterns due to environmental interactions.^[84] In everyday contexts, bioelectric signals like the electrocardiogram (ECG) from the human heartbeat illustrate periodic natural phenomena, where electrical impulses from cardiac muscle cells generate measurable voltage variations across the body. The ECG waveform typically features repeating P-QRS-T complexes corresponding to atrial depolarization, ventricular contraction, and repolarization, occurring at rates of 60-100 beats per minute in a healthy adult, thus embodying a rhythmic, bioelectric signal essential for monitoring cardiovascular health.^[85] Radio reception involves everyday exposure to modulated electromagnetic waves, where broadcast signals from distant transmitters are captured by antennas as varying electric fields inducing currents. For example, amplitude-modulated (AM) radio waves carry audio information by altering the wave's strength, while frequency-modulated (FM) variants adjust the wave's oscillation rate, allowing listeners to receive news or music through portable devices. These signals blend natural electromagnetic propagation with incidental modulation from atmospheric conditions.^[86] Traffic light timing exemplifies discrete signals in urban environments, where sequences of on-off states (red, yellow, green) are programmed to alternate at fixed intervals, such as 30-60 seconds per phase, to regulate vehicle flow at intersections. This step-like variation creates a digital-like pattern of binary illumination changes, synchronized across roads to prevent collisions and facilitate orderly movement.^[87] Most natural and everyday signals share key characteristics: they are predominantly analog, varying continuously over time to reflect real-world fidelity, yet often aperiodic—lacking strict repetition—and interspersed with noise from environmental factors like wind or interference. This inherent irregularity underscores their organic origins, contrasting with more controlled forms, while their continuous-time nature allows for infinite resolution in amplitude, capturing subtle variations.^[78]^[88] Historically, the telegraph signals of the 1830s marked an early intersection of natural and engineered communication, using discrete electrical pulses to transmit Morse code—short "dots" and long "dashes"—over wires, as developed by Samuel F. B. Morse. These on-off impulses, generated by a key and battery, enabled rapid long-distance messaging, such as the 1844 transmission of "What hath God wrought," revolutionizing information exchange before widespread telephony.^[89]^[90]

Engineered and Synthetic Signals

Engineered and synthetic signals are purposefully designed by humans to serve specific functions in technology, science, and engineering applications, often prioritizing precision, repeatability, and control over natural variability. Synthetic signals, in particular, are commonly used as test stimuli in signal processing systems to evaluate performance characteristics such as frequency response, transient behavior, and impulse response. These signals are typically deterministic, ensuring predictable outcomes for reliable testing and calibration.^[91] Among synthetic signals, the sine wave is a fundamental periodic signal employed to assess linear system responses at specific frequencies, as its single-frequency content simplifies analysis of amplitude and phase shifts. Square waves, with their abrupt transitions, are utilized to examine transient responses and harmonic content in systems like amplifiers and filters. The impulse signal, often approximated by a very narrow rectangular pulse, serves as a probe to determine the full impulse response of a system, providing insight into its overall dynamic behavior. Chirp signals, which linearly or exponentially sweep through frequencies, are valuable for broadband testing, such as measuring frequency responses over a wide range without multiple discrete tests.^[91]^[92]^[93] Engineered signals extend these concepts into practical implementations, where they are tailored for operational efficiency in devices and networks. In radar systems, short, high-power pulses are transmitted to detect and locate objects, with pulse width determining the range resolution—typically on the order of microseconds to achieve meter-level accuracy. Digital bitstreams, consisting of binary sequences like non-return-to-zero (NRZ) encodings, form the basis of data transmission in communication links, enabling high-speed, error-resistant transfer of information through channels such as optical fibers or wireless media. Pulse-width modulation (PWM) signals, which vary the duty cycle of rectangular pulses, are widely applied in motor control to regulate speed and torque by adjusting average power delivery, offering efficiency advantages over linear methods in applications like electric vehicles and robotics.^[94]^[95]^[96] A prominent example of an engineered signal is the Global Positioning System (GPS) signal, developed by the U.S. Department of Defense in the 1970s, which employs spread-spectrum techniques using pseudorandom noise (PRN) codes to enable precise ranging and anti-jamming capabilities. These codes, such as the coarse/acquisition (C/A) code at 1.023 MHz, are modulated onto carrier signals at L1 (1575.42 MHz) and L2 frequencies, allowing receivers to correlate the incoming signal with a locally generated replica for accurate time-of-arrival measurements.^[97]^[98] Recent advancements in software-defined signals have transformed engineered applications, particularly in 5G networks post-2020, where software-defined radios (SDRs) enable dynamic generation and adaptation of waveforms for enhanced flexibility and performance. SDR platforms facilitate the implementation of 5G physical layer protocols, such as new radio (NR) cell search and beamforming, by processing signals in software rather than fixed hardware, supporting features like massive MIMO and low-latency communications.^[99]^[100]

Signal Processing Techniques

Basic Operations

Basic operations on signals involve simple manipulations in the time domain that preserve the fundamental structure of the signal while altering its position, amplitude, or orientation. These operations form the foundation for analyzing and processing signals in linear systems, enabling the decomposition of complex signals into simpler components. They are particularly useful in applications such as audio engineering and communications, where signals must be combined or adjusted without introducing nonlinear distortions.^[30] Addition and subtraction of signals are performed pointwise, yielding a new signal z(t) = x(t) \pm y(t) for continuous-time signals x(t) and y(t). This operation underlies the principle of superposition in linear systems, allowing the response to a sum of inputs to be the sum of individual responses. For instance, in audio mixing, multiple sound tracks are added sample by sample to create a composite waveform, such as combining vocals and instrumentation in music production. Subtraction similarly isolates components, like removing noise from a recorded signal by subtracting an estimated noise waveform.^[101]^[102] Amplitude scaling multiplies the signal by a constant a, resulting in y(t) = a x(t), which proportionally adjusts the signal's magnitude without changing its shape or duration. If |a| > 1, the amplitude increases (amplification); if $0 < |a| < 1, it decreases (attenuation); and if a < 0, it inverts the signal. This operation is essential for normalizing signal levels or emphasizing certain frequency components in preliminary processing stages. For example, scaling an audio signal by 2 doubles its volume, directly affecting perceived loudness in playback systems.^[30]^[101] Time-shifting translates the signal along the time axis, producing y(t) = x(t - t_0) for a delay of t_0 > 0 or an advance if t_0 < 0. This operation models delays in transmission channels or synchronization in multi-signal environments, such as aligning audio tracks in recording software. The shape and amplitude remain unchanged, only the temporal position shifts, which is critical for studying causality and timing in system responses.^[30]^[101] Time-reversal flips the signal about the vertical axis, defined as y(t) = x(-t), effectively reversing the direction of time progression. This is useful for checking signal symmetry or simulating backward playback in audio analysis. For an asymmetric signal like a unit step function u(t), reversal yields u(-t), which steps down at t = 0. Combined with shifting, it facilitates operations like reflection in signal design.^[30]^[101] These operations—addition, scaling, shifting, and reversal—exhibit linearity when applied to signals, meaning they satisfy homogeneity (scaling inputs scales outputs proportionally) and additivity (superposition of inputs yields superposition of outputs). A transformation is linear if, for inputs x_1(t) and x_2(t) with outputs y_1(t) and y_2(t), and constants a and b, the output to a x_1(t) + b x_2(t) is a y_1(t) + b y_2(t). This property ensures that basic manipulations do not introduce interactions between signal components, preserving analyzability in linear time-invariant systems. For example, in audio superposition, adding scaled and shifted tracks maintains independent processing through filters.^[103]^[102]

Transformation Methods

Transformation methods in signal analysis involve mathematical operations that convert signals from one domain to another, typically from the time domain to a frequency or complex domain, to facilitate easier examination of their properties such as frequency content and stability. These transforms enable the decomposition of signals into components that reveal underlying structures, with inverse transforms allowing reconstruction of the original signal. Among the most fundamental are the Fourier and Laplace transforms for continuous-time signals, and their discrete counterparts, the discrete Fourier transform (DFT) and Z-transform, for digital signals and systems. The Fourier transform is a cornerstone technique for representing aperiodic continuous-time signals in the frequency domain. It decomposes a signal x(t) into a continuum of complex exponentials, providing the frequency spectrum X(\omega). The forward transform is defined as

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

where \omega is the angular frequency and j = \sqrt{-1}. The inverse Fourier transform reconstructs the signal as

x(t) = \frac{1}{2\pi} \int_{-\infty}^{\infty} X(\omega) e^{j \omega t} \, d\omega.

This bidirectional mapping supports synthesis and analysis of signals by highlighting their sinusoidal components, originally introduced by Joseph Fourier in his 1822 treatise on heat conduction. The transform is particularly suited for stationary signals, extending the Fourier series to non-periodic cases. The Laplace transform extends the Fourier transform by incorporating exponential damping, making it ideal for analyzing system stability in the s-domain, where s = \sigma + j\omega with \sigma representing growth or decay rates. For a causal signal x(t) (zero for t < 0), the unilateral Laplace transform is

X(s) = \int_{0}^{\infty} x(t) e^{-s t} \, dt.

The inverse transform recovers x(t) via a complex contour integral along the Bromwich path. Introduced by Pierre-Simon Laplace in his 1779 work on differential equations, this transform converges for signals with bounded growth, enabling pole-zero analysis for stability assessment in control systems. For discrete-time signals, the discrete Fourier transform (DFT) provides a frequency-domain representation analogous to the continuous Fourier transform, converting a finite sequence x of length N into

X = \sum_{n=0}^{N-1} x e^{-j 2\pi k n / N}, \quad k = 0, 1, \dots, N-1.

The inverse DFT reconstructs x as

x = \frac{1}{N} \sum_{k=0}^{N-1} X e^{j 2\pi k n / N}.

This transform is essential for digital signal processing, approximating the continuous spectrum for sampled data. The Z-transform generalizes the discrete-time Laplace transform for analyzing linear time-invariant discrete systems, defined for a sequence x as

Z\{x\} = X(z) = \sum_{n=-\infty}^{\infty} x z^{-n},

where z is a complex variable, often on the unit circle |z| = 1 for frequency response. The inverse uses contour integration in the z-plane. Formally named and applied to sampled-data systems by John R. Ragazzini and Lotfi A. Zadeh in their 1952 paper, it facilitates stability analysis via region of convergence and pole placement. An efficient algorithm for computing the DFT is the fast Fourier transform (FFT), which reduces the computational complexity from O(N^2) to O(N \log N) for N-point transforms where N is a power of 2. The Cooley-Tukey FFT, introduced in 1965, achieves this through a divide-and-conquer approach, recursively splitting the DFT into smaller DFTs of even and odd indices. This breakthrough enabled practical real-time spectral analysis in applications like audio processing and imaging.

Signals in Systems

System Response to Signals

In signal processing, a system H transforms an input signal x(t) into an output signal y(t) according to the relation y(t) = H\{x(t)\}, where the system's behavior determines how the input is modified or filtered to produce the output.^[104] For linear time-invariant (LTI) systems, this transformation is fully characterized by the impulse response h(t), which is the output produced when the input is a unit impulse \delta(t).^[103] The impulse response encapsulates the system's dynamics, allowing prediction of responses to arbitrary inputs through superposition principles.^[105] Linearity in LTI systems means that the response satisfies the superposition principle: if inputs x_1(t) and x_2(t) produce outputs y_1(t) = H\{x_1(t)\} and y_2(t) = H\{x_2(t)\}, then a linear combination a x_1(t) + b x_2(t) yields a y_1(t) + b y_2(t), where a and b are constants.^[103] Time-invariance ensures that a time shift in the input results in an identical shift in the output; specifically, if y(t) = H\{x(t)\}, then y(t - \tau) = H\{x(t - \tau)\} for any delay \tau.^[106] These properties enable the decomposition of complex signals into simpler components, such as impulses, for analysis.^[107] Causality is a key property of many physical systems, where the output at any time t depends solely on the input values for times less than or equal to t, implying no dependence on future inputs.^[108] For LTI systems, this manifests as the impulse response h(t) = 0 for t < 0.^[109] Stability, particularly bounded-input bounded-output (BIBO) stability, requires that any bounded input |x(t)| \leq M < \infty produces a bounded output |y(t)| \leq K < \infty.^[110] In LTI systems, BIBO stability holds if the impulse response is absolutely integrable, i.e., \int_{-\infty}^{\infty} |h(t)| \, dt < \infty.^[111] The step response provides another characterization of system behavior, defined as the output to a unit step input u(t), which is zero for t < 0 and one for t \geq 0.^[112] It is particularly useful for assessing rise time, settling time, and overshoot in system performance, often derived from the impulse response via integration.^[110] For example, in control systems, the step response reveals how quickly and accurately the system reaches a steady state following a sudden change.^[113]

Convolution and Filtering

In linear time-invariant (LTI) systems, the output signal y(t) is obtained by convolving the input signal x(t) with the system's impulse response h(t), mathematically expressed as y(t) = \int_{-\infty}^{\infty} x(\tau) h(t - \tau) \, d\tau for continuous-time signals.^[114] This operation represents the core mechanism of filtering, where the impulse response acts as a kernel that modifies the input by weighting and summing shifted versions of itself. For discrete-time signals, the convolution becomes a sum: y = \sum_{k=-\infty}^{\infty} x h[n - k], enabling efficient digital implementation in signal processing applications.^[115] Ideal filters are theoretical constructs that achieve perfect frequency selectivity, serving as benchmarks for practical designs. A low-pass ideal filter passes all frequencies below a cutoff \omega_c with unity gain while completely attenuating higher frequencies, effectively removing high-frequency noise or details.^[116] In contrast, an ideal high-pass filter attenuates frequencies below \omega_c and passes higher ones unaltered, useful for isolating edges or rapid changes in signals such as images.^[116] An ideal band-pass filter transmits a specific frequency band between lower cutoff \omega_l and upper cutoff \omega_h with unity gain, rejecting frequencies outside this range, which is common in communications for selecting carrier bands.^[116] Practical filters are classified by their impulse response duration: finite impulse response (FIR) filters have a finite-length h, ensuring inherent stability and the possibility of linear phase response, while infinite impulse response (IIR) filters use feedback, resulting in an infinite-duration h but requiring fewer coefficients for sharp responses.^[117] A simple example of an FIR filter is the moving average, where y = \frac{1}{M} \sum_{k=0}^{M-1} x[n - k], which smooths signals by averaging M consecutive samples, acting as a basic low-pass filter.^[117] IIR filters, often derived from analog prototypes like Butterworth designs, provide steeper roll-offs at lower orders but can introduce phase distortions.^[117] The frequency response of a filter, denoted H(j\omega), is the Fourier transform of its impulse response h(t), capturing how the filter alters signal amplitudes and phases across frequencies.^[118] Magnitude plots of |H(j\omega)| illustrate passband gain and stopband attenuation, while phase plots of \angle H(j\omega) reveal delays, essential for designing filters that preserve waveform timing in applications like audio.^[118] Convolution-based filtering finds widespread use in audio processing, such as noise reduction where low-pass FIR filters suppress high-frequency interference in speech signals, improving clarity in reverberant environments.^[119] Similarly, equalization employs IIR or FIR convolutions to adjust frequency responses, compensating for room acoustics or speaker imbalances to achieve flat playback.^[119]

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The pulse width (H) determines the minimum range at which targets can be detected. This minimum range is approximately ½ the length of the wave burst. In the ...
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[PDF] Radar Frequencies and Waveforms
Radar uses frequencies like HF, VHF, UHF, and mm. Waveforms include CW, single pulse, and pulse compression (FM and PM). CW can measure Doppler, single pulse ...
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Pulse Width Modulation Used for Motor Control - Electronics Tutorials
Pulse Width Modulation delivers power through a succession of pulses rather than a continuous voltage by increasing or decreasing pulse width.
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[PDF] Introduction to GPS and other Global Navigation Satellite Systems
Jun 7, 2012 · GPS transmitted C/A-code. Receiver replicated C/A-code. Finding Δt for each GPS signal tracked is called “code correlation”. ▫ Δt is ...
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[PDF] Chapter 25 - Global Positioning System
—The satellites transmit their signals using spread-spectrum techniques that employ two different spreading functions: a 1.023-MHZ coarse/ acquisition (C/A) ...
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Software-Defined Radio-Based 5G Physical Layer Experimental ...
Jan 16, 2023 · In this study, we developed a 5th generation mobile communication (5G) physical layer (PHY) experimental platform based on software-defined radio (SDR).
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Implementation of SDR-based 5G NR Cell Search Equipment
Abstract: This paper describes an initial implementation of software defined radio (SDR) based 5G new radio (NR) cell search equipment.
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[PDF] Operations on Continuous-Time Signals
Common operations on continuous-time signals include time reversal, time shifting, amplitude scaling, addition, multiplication, and time scaling.
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Superposition: - Stanford CCRMA
The superposition property of linear systems states that the response of a linear system to a sum of signals is the sum of the responses to each individual ...
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[PDF] Lecture 3 ELE 301: Signals and Systems - Princeton University
Linearity: A system S is linear if it satisfies both Homogeneity: If y = Sx, and a is a constant then ay = S(ax). Superposition: If y1 = Sx1 and y2 = Sx2, then ...
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[PDF] 2 LINEAR SYSTEMS - MIT OpenCourseWare
In the case of LTI systems, the impulse response is a complete definition of the system, in the same way that a differential equation is, with zero initial ...
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[PDF] LINEAR TIME-INVARIANT SYSTEMS AND THEIR FREQUENCY ...
The impulse response h[n] of an LTI system is just the response to an impulse: δ[n] → LTI → h[n]. The significance of h[n] is that we can compute the response ...
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[PDF] Linear Time-invariant Systems - Purdue Engineering
Thus, the response at time n of a linear system is simply the superposition of the responses due to the input value at each point in time.
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[PDF] Frequency Response of LTI Systems - MIT OpenCourseWare
Nov 3, 2012 · The absolute summability of h[·] is the condition for bounded-input bounded-output (BIBO) stability of an LTI system that we obtained in the ...
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[PDF] Lecture Notes 6 - ECEN 314: Signals and Systems
A CT LTI system is causal if and only if its unit impulse response h(t)=0 for all t < 0. Property 6 (Stability). A CT LTI system is BIBO stable if and only if ...
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[PDF] ESE 531: Digital Signal Processing Lecture Outline Discrete-Time ...
▫ 1) Time reverse the impulse response and shift it n time steps ... ❑ An LTI system is causal if its impulse response is causal: ... ❑ An LTI system is BIBO ...
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[PDF] Linear Systems
Linear time invariant systems are characterized by their impulse response ... This result specifies the response of a BIBO stable system to a step input.
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[PDF] eleg 3124 systems and signals - Dr. Jingxian Wu
– A system in which continuous-time input signals are transformed to ... out the output y(t). – Method 1: differential equations. – Methods 2: convolution ...<|separator|>
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[PDF] Continuous-Time Signals and Systems 1 Preliminaries
Example: If the step response of an LTI system is given to be e−tu(t), then the impulse response is h(t) = −e−tu(t) + e−tδ(t).<|control11|><|separator|>
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[PDF] Linear Time-Invariant Dynamical Systems - Duke People
Oct 6, 2020 · The i, j element of H(t) is the response of output i due to a unit impulse at input j. Note that the impulse response is a special case of the ...
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https://ieeexplore.ieee.org/document/8361830
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[PDF] Convolution
Convolution is a mathematical way of combining two signals to form a third signal. It is the single most important technique in Digital Signal Processing.
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[PDF] CHAPTER 8 ANALOG FILTERS
An ideal filter will have an amplitude response that is unity (or at a fixed gain) for the frequencies of interest (called the pass band) and zero everywhere ...
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[PDF] IIR vs. FIR - MIT OpenCourseWare
It is possible that an IIR of lower order actually requires more #MAD than an FIR of higher order, because FIR filters may be implemented using polyphase ...
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[PDF] 6.3000: Signal Processing
The frequency response is Fourier transform of unit-sample response! A causal system is one in which an input at time n = n0 cannot affect the output at times ...
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https://ieeexplore.ieee.org/document/8361830