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  Acoustic Pulse Reflectometry  [hits=27 votes=1 rating=5.0]
  Adaptor  [hits=16 votes=1 rating=5.0] : Adaptors for Wave Digital Elements
  Alias Theorem  [hits=1 votes=1 rating=5.0]
  Aliasing of Continuous Signal  [hits=1 votes=1 rating=5.0]
  Allpass Filter (2 competing definitions)
  Allpass Interpolator  [hits=47 votes=1 rating=5.0] - A Bessel lowpass filter is defined to have maximally flat group delay at dc. - Allpass bandlimited interpolation is ideal with respect to amplitude, but generally has some delay error at high frequencies (usually inaudible).
  Amplitude Response  [hits=1 votes=1 rating=5.0] - The START_HTML_TAGiEND_HTML_TAGamplitude responseSTART_HTML_TAG/iEND_HTML_TAG of an LTI filter is simply the magnitude of the (complex) frequency response
  Analytic Signal (2 competing definitions)
  Auditory Filter Bank  [hits=28 votes=1 rating=5.0] : Auditory Filter Banks
  Auditory Toolbox  [hits=28 votes=1 rating=5.0]
  Autocorrelation  [hits=1 votes=1 rating=5.0]
  Bandlimited Interpolation (9 competing definitions)
  Bark Bilinear Transform  [hits=51 votes=1 rating=5.0] : Bark and ERB Bilinear Transforms
  Bark Frequency Scale  [hits=17 votes=1 rating=5.0] : The Bark Frequency Scale
  Bartlett Window  [hits=1 votes=1 rating=5.0]
  Bayesian Hypothesis Testing  [hits=1 votes=1 rating=5.0]
  Bayesian Parameter Estimation  [hits=19 votes=1 rating=5.0]
  Bayesian Sequential Analysis  [hits=1 votes=1 rating=5.0]
  Bessel Filter  [hits=30 votes=1 rating=5.0] - A Bessel lowpass filter is defined to have maximally flat group delay at dc. - Allpass bandlimited interpolation is ideal with respect to amplitude, but generally has some delay error at high frequencies (usually inaudible).
  Bilinear Transform (5 competing definitions)
  Binary Integer Fixed-Point Numbers  [hits=41 votes=1 rating=5.0]
  Biquad (2 competing definitions)
  Blackman Window  [hits=34 votes=1 rating=5.0]
  Causal Filter  [hits=1 votes=1 rating=5.0] : Causal Recursive Filters
  Central Limit Theorem  [hits=42 votes=1 rating=5.0] : Central Limit Theorem -- from MathWorld
  Characteristic Polynomial Equation  [hits=1 votes=1 rating=5.0]
  Chebyshev Filter (2 competing definitions)
  Chebyshev Polynomial  [hits=1 votes=1 rating=5.0]
  Chinese Remainder Theorem (3 competing definitions)
  Chorus Effect (2 competing definitions)
  Circular Convolution  [hits=1 votes=1 rating=5.0]
  Coherence Function  [hits=1 votes=1 rating=5.0]
  Comb Filter  [hits=1 votes=1 rating=5.0] : Comb Filters
  Conjugation and Reversal  [hits=1 votes=1 rating=5.0]
  Convolution (2 competing definitions)
  Convolution Representation  [hits=39 votes=1 rating=5.0] : Impulse-Response Representation - Every LTI filter can be implemented by convolving its impulse response with the input signal
  Convolution Theorem  [hits=30 votes=1 rating=5.0]
  Correlation (2 competing definitions)
  Correlation Theorem  [hits=1 votes=1 rating=5.0]
  Critical Band  [hits=20 votes=1 rating=5.0] : The Bark Frequency Scale
  Cross-Correlation  [hits=1 votes=1 rating=5.0]
  Cross-Spectrum  [hits=1 votes=1 rating=5.0] : Cross-Correlation
  DFT Matrix  [hits=1 votes=1 rating=5.0] : Matrix Formulation of the DFT
  DFT Matrix  [hits=1 votes=1 rating=5.0]
  DFT Matrix  [hits=1 votes=1 rating=5.0] : Matrix Formulation of the DFT
  DFT Sinusoid (2 competing definitions)
  Dashpot  [hits=1 votes=1 rating=5.0]
  Decimation Operator  [hits=24 votes=1 rating=5.0] : Downsampling Operator
  Decimation Theorem  [hits=12 votes=1 rating=5.0] : Downsampling Theorem (Aliasing Theorem) - The START_HTML_TAGiEND_HTML_TAGdecimation theoremSTART_HTML_TAG/iEND_HTML_TAG quantifies START_HTML_TAGiEND_HTML_TAGaliasingSTART_HTML_TAG/iEND_HTML_TAG due to downsampling
  Delay-Free Loop  [hits=1 votes=1 rating=5.0] : Delay-Free Loops
  Delay Effects (2 competing definitions)
  Delay Line  [hits=81 votes=1 rating=5.0] : Harmony Central®: Effects Explained: Delay
  Delay Line Interpolation (7 competing definitions)
  Delay Loop Expansion  [hits=1 votes=1 rating=5.0]
  Design of Fractional Delay Filters Using Convex Optimization (2 competing definitions)
  Difference Equation  [hits=1 votes=1 rating=5.0]
  Differentiation Theorem  [hits=17 votes=1 rating=5.0]
  Differentiation Theorem for Laplace Transform  [hits=24 votes=1 rating=5.0]
  Digital Audio Resampling Home Page  [hits=1 votes=1 rating=5.0] - Digital Audio Resampling Home Page: Theory, software, and references pertaining to bandlimited interpolation and sampling-rate conversion
  Digital Filter (2 competing definitions)
  Digital Filter Design (2 competing definitions)
  Digital Reverberation (2 competing definitions)
  Digital Signal Interpolation (4 competing definitions)
  Digital Signal Processing (3 competing definitions)
  Digital Waveguide  [hits=1 votes=1 rating=5.0] : Digital Waveguide Model
  Digital Waveguide Interpolation  [hits=1 votes=1 rating=5.0] : Delay-Line Interpolation
  Digital Waveguide Mesh  [hits=39 votes=1 rating=5.0] : MUS421/EE367B Supplement to Lecture 7B: Wave Digital Filters and Waveguide Networks for Numerical Integration of Time-Dependent PDEs
  Digital Waveguide Network  [hits=30 votes=1 rating=5.0] : Digital Waveguide Networks
  Digital Waveguide Simulation  [hits=1 votes=1 rating=5.0] : Digital Waveguide Model
  Digital Waveguide Theory  [hits=1 votes=1 rating=5.0]
  Direct Form I  [hits=1 votes=1 rating=5.0] : Direct-Form I
  Direct Form II  [hits=1 votes=1 rating=5.0]
  Direct Form Implementation  [hits=1 votes=1 rating=5.0] : Time Domain Filter Representations
  Discrete-Time Lumped Model (2 competing definitions)
  Discrete-Time Sampling-Rate Conversion  [hits=41 votes=1 rating=5.0] : Downsampling Theorem (Aliasing Theorem) - The START_HTML_TAGiEND_HTML_TAGdecimation theoremSTART_HTML_TAG/iEND_HTML_TAG quantifies START_HTML_TAGiEND_HTML_TAGaliasingSTART_HTML_TAG/iEND_HTML_TAG due to downsampling
  Discrete Fourier Transform (9 competing definitions)
  Discrete Time Fourier Transform  [hits=1 votes=1 rating=5.0] : Discrete Time Fourier Transform (DTFT) - The Discrete Time Fourier Transform (DTFT) is the appropriate Fourier transform for discrete-time signals of arbitrary length. It can be obtained as the limit of a Discrete Fourier Transform (DFT) as its length goes to infinity.
  Discrete Wavelet Filterbank  [hits=1 votes=1 rating=5.0] : Dyadic Wavelet Transform
  Downsampling (2 competing definitions)
  Dual of the Convolution Theorem  [hits=24 votes=1 rating=5.0]
  Dynamic Range  [hits=1 votes=1 rating=5.0]
  Energy Decay Curve  [hits=20 votes=1 rating=5.0]
  Energy Decay Relief  [hits=130 votes=1 rating=5.0]
  Equalizer  [hits=1 votes=1 rating=5.0]
  Equation-Error Digital Filter Design  [hits=1 votes=1 rating=5.0]
  Equation-Error Method for Digital Filter Design (2 competing definitions)
  Equivalent Circuit (2 competing definitions)
  Exponent Zero  [hits=1 votes=1 rating=5.0] : The Exponent Zero
  Exponential Function (2 competing definitions)
  FFTW  [hits=1 votes=1 rating=5.0] : FFTW Home Page - FFTW is a C subroutine library for computing the Discrete Fourier Transform (DFT) in one or more dimensions, of both real and complex data, and of arbitrary input size. We believe that FFTW, which is free software, should become the FFT library of choice for most applications. Our benchmarks, performed on on a variety of platforms, show that FFTW's performance is typically superior to that of other publicly available FFT software. Moreover, FFTW's performance is portable: the program will perform well on most architectures without modification.
  FFT Source Code  [hits=20 votes=1 rating=5.0]
  FIR Digital Filter Design (5 competing definitions)
  FIR Filter (2 competing definitions)
  Farrow Structure  [hits=1 votes=1 rating=5.0]
  Fast Fourier Transform (3 competing definitions)
  Feedback Comb Filter  [hits=1 votes=1 rating=5.0] : Feedback Comb Filters
  Feedforward Comb Filter  [hits=1 votes=1 rating=5.0] : Feedforward Comb Filters
  Filter Design  [hits=1 votes=1 rating=5.0] : Circuit Sage: Filter Design and Analysis
  Filter Order  [hits=1 votes=1 rating=5.0]
  Filter Stability  [hits=1 votes=1 rating=5.0] : Stability - A filter is said to be START_HTML_TAGiEND_HTML_TAGstableSTART_HTML_TAG/iEND_HTML_TAG if its impulse response decays to zero as time goes to infinity
  Filter Transposition  [hits=1 votes=1 rating=5.0]
  Finite Difference Approximation (2 competing definitions)
  Finite Difference Scheme  [hits=1 votes=1 rating=5.0] : Finite Difference Schemes
  Flanger (2 competing definitions)
  Flip Operator  [hits=28 votes=1 rating=5.0]
  Flip Theorem (2 competing definitions)
  Force Wave  [hits=1 votes=1 rating=5.0] : Force Waves
  Fourier Series Expansion  [hits=1 votes=1 rating=5.0] : Fourier Series (FS) - A Fourier Series (FS) expansion is the appropriate Fourier transform for START_HTML_TAGiEND_HTML_TAGperiodicSTART_HTML_TAG/iEND_HTML_TAG signals.
  Fourier Superposition Integral  [hits=1 votes=1 rating=5.0] : Fourier Transform (FT) and Inverse - The Fourier Transform (FT) gives the (complex) spectrum of a continuous-time signal of any length. - The Fourier superposition integral expresses a continuous function as a sum of sinusoids. The amplitude and phase of each sinusoid is obtained using the Fourier transform. Thus, a Fourier superposition integral simply inverts a corresponding Fourier transform.
  Fourier Theorem (2 competing definitions)
  Fourier Transform  [hits=1 votes=1 rating=5.0] : Fourier Transform (FT) and Inverse - The Fourier Transform (FT) gives the (complex) spectrum of a continuous-time signal of any length. - The Fourier superposition integral expresses a continuous function as a sum of sinusoids. The amplitude and phase of each sinusoid is obtained using the Fourier transform. Thus, a Fourier superposition integral simply inverts a corresponding Fourier transform.
  Fourier symmetry (2 competing definitions)
  Fractional Delay Filter Design  [hits=1 votes=1 rating=5.0] : MUS420/EE367A Lecture 4A Interpolated Delay Lines, Ideal Bandlimited Interpolation, and Fractional Delay Filter Design
  Frequency Domain  [hits=1 votes=1 rating=5.0] : MATHEMATICS OF THE DISCRETE FOURIER TRANSFORM (DFT) - The Discrete Fourier Transform (DFT) computes a discrete-frequency spectrum from a discrete-time signal of finite length. - The Fourier transform of a signal is a frequency-domain function
  Frequency Response  [hits=1 votes=1 rating=5.0] - The START_HTML_TAGiEND_HTML_TAGfrequency responseSTART_HTML_TAG/iEND_HTML_TAG is defined for LTI filters as the Fourier transform of the filter output signal divided by the Fourier transform of the filter input signal
  Gammatone Filter Bank  [hits=1 votes=1 rating=5.0]
  General Reflectance  [hits=1 votes=1 rating=5.0]
  Graphical Convolution  [hits=25 votes=1 rating=5.0]
  Graphical Frequency Response  [hits=1 votes=1 rating=5.0]
  Graphical Phase Response  [hits=1 votes=1 rating=5.0]
  Group Delay  [hits=1 votes=1 rating=5.0] - The START_HTML_TAGiEND_HTML_TAGgroup delaySTART_HTML_TAG/iEND_HTML_TAG of an LTI filter is minus the phase response divided by frequency
  Hamming Window  [hits=46 votes=1 rating=5.0]
  Hann Window  [hits=36 votes=1 rating=5.0]
  Heisenberg  [hits=34 votes=1 rating=5.0] : Heisenberg, Werner (1901-1976) -- from Eric Weisstein's World of Scientific Biography
  Heisenberg Uncertainty Principle  [hits=1 votes=1 rating=5.0] : Uncertainty Principle -- from Eric Weisstein's World of Physics
  Hermitian Operator (2 competing definitions)
  Heterodyne Conversion  [hits=1 votes=1 rating=5.0] : Heterodyne Conversion -- from Eric Weisstein's World of Physics
  Higher Order Statistics  [hits=22 votes=1 rating=5.0] : The Worldwide Homepage on Higher Order Statistics (HOS) - This page is a starting point for finding information about Higher Order Statistics. It includes links to directories of research groups and researchers around the world who are interested in HOS, as well as a HOS library including a list of FAQs and online papers. Furthermore, you will find information on HOS-related conferences etc.
  IIR Filter  [hits=1 votes=1 rating=5.0] : Difference Equation
  Ideal Mass  [hits=1 votes=1 rating=5.0]
  Image Method  [hits=1 votes=1 rating=5.0]
  Impedance Diagram  [hits=1 votes=1 rating=5.0]
  Impulse Response  [hits=60 votes=1 rating=5.0] : Impulse-Response Representation - Every LTI filter can be implemented by convolving its impulse response with the input signal
  In-Phase Sinusoid  [hits=1 votes=1 rating=5.0] : In-Phase & Quadrature Sinusoidal Components
  Induction  [hits=47 votes=1 rating=5.0] : Induction -- from Eric Weisstein's World of Physics
  Inductor  [hits=1 votes=1 rating=5.0] : Inductor -- from Eric Weisstein's World of Physics
  Intermodulation Distortion (2 competing definitions)
  Introduction to Digital Filter Theory (2 competing definitions)
  Inverse Filter (2 competing definitions)
  Kaiser Window  [hits=1 votes=1 rating=5.0]
  Kalman Filter (2 competing definitions)
  Karhunen-Loeve Expansion  [hits=1 votes=1 rating=5.0]
  Karhunen-Loeve Expansion  [hits=1 votes=1 rating=5.0] : Karhunen-Loeve Transform
  Kelly-Lochbaum Scatter Junction  [hits=25 votes=1 rating=5.0] : Kelly-Lochbaum Scattering Junctions
  Ladder Filter  [hits=28 votes=1 rating=5.0]
  Lagrange Interpolation  [hits=1 votes=1 rating=5.0] - Lagrange interpolation is interpolation using a polynomial
  Laplace Transform  [hits=1 votes=1 rating=5.0] : Laplace Transform Analysis
  Length 2 DFT  [hits=1 votes=1 rating=5.0] : The Length 2 DFT
  Leslie Effect  [hits=1 votes=1 rating=5.0] : The Leslie
  Levinson-Durbin  [hits=18 votes=1 rating=5.0] : Computing Reflection Coefficients
  Linear Prediction (2 competing definitions)
  Linear Programming  [hits=13 votes=1 rating=5.0] : Linear Programming FAQ - Excellent FAQ on linear programming, including many pointers to both free and commercial software.
  Linear System Theory (2 competing definitions)
  Linearly Interpolated Delay Line (3 competing definitions)
  Loaded Junction  [hits=10 votes=1 rating=5.0]
  Lossless Scatter Matrix  [hits=26 votes=1 rating=5.0] : Lossless Scattering
  Lumped Acoustic Element  [hits=1 votes=1 rating=5.0] : Lumped vs. Distributed Systems
  Lumped Reflectance  [hits=1 votes=1 rating=5.0]
  MXR Phase 45  [hits=29 votes=1 rating=5.0] - Schematic for the MXR Phase 45 and DOD Phasor 201
  Mass-Spring-Dashpot  [hits=1 votes=1 rating=5.0]
  Mass-Spring-Dashpot System  [hits=1 votes=1 rating=5.0]
  Mass-Spring Chain  [hits=1 votes=1 rating=5.0]
  Mass-Spring System  [hits=1 votes=1 rating=5.0]
  Matched Filter  [hits=1 votes=1 rating=5.0] : Cross-Correlation
  Matched Z Transformation  [hits=1 votes=1 rating=5.0] : Finite Differences
  Mathematical Induction  [hits=1 votes=1 rating=5.0] : Induction -- from Eric Weisstein's World of Physics
  Mathematics of the DFT  [hits=1 votes=1 rating=5.0] : MATHEMATICS OF THE DISCRETE FOURIER TRANSFORM (DFT) - The Discrete Fourier Transform (DFT) computes a discrete-frequency spectrum from a discrete-time signal of finite length. - The Fourier transform of a signal is a frequency-domain function
  Minimum Phase  [hits=1 votes=1 rating=5.0] : The Minimum Phase Page - Minimum-phase filters have all their transfer-function zeros inside the unit circle of the z plane
  Mu-Law Compand  [hits=27 votes=1 rating=5.0] : Mu-Law Companding
  Multitaper Method  [hits=42 votes=1 rating=5.0] : SSA-MTM Toolkit User's Guide: MTM Theory
  Mutual Inductance  [hits=1 votes=1 rating=5.0] : Mutual Inductance -- from Eric Weisstein's World of Physics
  Negative Exponent  [hits=22 votes=1 rating=5.0] : Negative Exponents
  Negative Frequency  [hits=1 votes=1 rating=5.0] : Positive and Negative Frequencies
  Nested Allpass Filter  [hits=1 votes=1 rating=5.0] : Nested Allpass Filters
  Normal Distribution  [hits=25 votes=1 rating=5.0] : Mathematical formula for height of a normal curve - The Gaussian probability density function is, in Matlab syntax, exp(-(x-mu)^2/(2*sigma^2))/sqrt(2*pi*sigma^2), where mu is the mean and sigma is the standard deviation.
  Normalized DFT  [hits=1 votes=1 rating=5.0]
  Normalized Scatter Junction  [hits=14 votes=1 rating=5.0] : Normalized Scattering Junctions
  Normalized Scatter Matrix  [hits=45 votes=1 rating=5.0] : Normalized Scattering
  Number Theoretic Transform  [hits=1 votes=1 rating=5.0] : Number theoretic transforms - The number theory transform is analogous to the discrete Fourier transform (DFT) in that it has a similar convolution theorem
  One-Multiply Scatter Junction  [hits=32 votes=1 rating=5.0] : Kelly-Lochbaum Scattering Junctions
  One-Port Network Theory  [hits=1 votes=1 rating=5.0]
  Orthonormal Sinusoid  [hits=1 votes=1 rating=5.0] : An Orthonormal Sinusoidal Set
  Pade-Prony  [hits=34 votes=1 rating=5.0] : The Padé-Prony Method
  Parallel Adaptor  [hits=26 votes=1 rating=5.0]
  Parallel Combination  [hits=1 votes=1 rating=5.0] : Parallel Combination of One-Ports
  Partial Fraction Expansion  [hits=1 votes=1 rating=5.0]
  Passive Reflectance  [hits=1 votes=1 rating=5.0] : Passive Reflectances
  Phase Delay  [hits=1 votes=1 rating=5.0] - The START_HTML_TAGiEND_HTML_TAGphase delaySTART_HTML_TAG/iEND_HTML_TAG of an LTI filter is minus the phase response divided by frequency
  Phase Function  [hits=1 votes=1 rating=5.0] : Phase Function -- from Eric Weisstein's World of Physics
  Phase Response  [hits=1 votes=1 rating=5.0] - The START_HTML_TAGiEND_HTML_TAGphase responseSTART_HTML_TAG/iEND_HTML_TAG of an LTI filter is the angle of the (complex) frequency response
  Phase Shifting (4 competing definitions)
  Phase Vocoder (2 competing definitions)
  Phasor Analysis  [hits=1 votes=1 rating=5.0] - The START_HTML_TAGiEND_HTML_TAGphasorSTART_HTML_TAG/iEND_HTML_TAG of a sinusoid is its START_HTML_TAGiEND_HTML_TAGcomplex amplitudeSTART_HTML_TAG/iEND_HTML_TAG, containing both the amplitude and phase of the sinusoid (all but the frequency parameter)
  Pitch Shifter  [hits=1 votes=1 rating=5.0] : Time Stretching and Pitch Shifting of Audio Signals - Time scaling refers to modifying the time-duration of a signal without modifying its audio spectral characteristics. It is often used to slow down music without changing the pitch of any of the instruments. It is not possible to do this exactly in general, but good approximate methods exist using, e.g., the phase vocoder. - As opposed to the process of pitch transposition achieved using a simple sample rate conversion, Pitch Scaling is a way to change the pitch of a signal without changing its length. In practical applications, this is achieved by changing the length of a sound using one of the below methods and then performing a sample rate conversion to change the pitch. There exists a certain confusion in terminology, as Pitch Scaling is often also incorrectly named 'Pitch Shift' (a term coined by the music industry). A true Pitch Shift (as obtainable by modulating an analytic signal by a complex exponential) will shift the spectrum of a sound, while Pitch Scaling will dilate it, upholding the harmonic relationship of the sound. The actual Pitch Shifting yields a metallic, inharmonic sound which may well be an interesting special effect but which is a totally inadequate process for changing the pitch of any harmonic sound except a single sine wave.
  Pole and Zero  [hits=1 votes=1 rating=5.0] : Pole-Zero Analysis - A zero is a point in the complex z plane at which the transfer function goes to zero. A pole is a point in the z-plane at which the transfer function goes to infinity.
  Polyphase Filter Bank (2 competing definitions)
  Port Impedance  [hits=1 votes=1 rating=5.0] : Wave Variables
  Port Resistance  [hits=1 votes=1 rating=5.0] : Wave Variables
  Positive Real Function  [hits=1 votes=1 rating=5.0] : Introduction
  Power Normalized Waveguide Filter  [hits=13 votes=1 rating=5.0]
  Power Spectral Density  [hits=1 votes=1 rating=5.0]
  Power Theorem  [hits=1 votes=1 rating=5.0]
  Power Wave (2 competing definitions)
  Pressure Wave  [hits=1 votes=1 rating=5.0] : Pressure Wave -- from Eric Weisstein's World of Physics
  Prime Factor Algorithm  [hits=1 votes=1 rating=5.0]
  Probability Density  [hits=36 votes=1 rating=5.0] : Probability Density -- from Eric Weisstein's World of Physics
  Prony Method  [hits=36 votes=1 rating=5.0] : Prony's Method
  Pulse Code Modulation  [hits=32 votes=1 rating=5.0] : Pulse Code Modulation (PCM)
  Rational Exponent  [hits=16 votes=1 rating=5.0] : Rational Exponents
  Rational Transfer Function  [hits=22 votes=1 rating=5.0] : Transfer Function
  Rayleigh Energy Theorem (2 competing definitions)
  Rayleigh Scatter  [hits=26 votes=1 rating=5.0] : Rayleigh Scattering -- from Eric Weisstein's World of Physics
  Real Exponent  [hits=1 votes=1 rating=5.0] : Real Exponents
  Reflection Coefficient  [hits=1 votes=1 rating=5.0] : Computing Reflection Coefficients
  Reflection Transfer Function (3 competing definitions)
  Repeat Operator  [hits=30 votes=1 rating=5.0]
  Repeat Theorem (2 competing definitions)
  Sampling Interval  [hits=35 votes=1 rating=5.0] : Sampling Theorem - A signal is said to be START_HTML_TAGIEND_HTML_TAGoversampledSTART_HTML_TAG/IEND_HTML_TAG when the sampling frequency exceeds the signal bandwidth by some finite margin. For example, 2x oversampling means that the sampling rate is double what it needs to be to avoid aliasing in the frequency domain. In audio, the 88.2 kHz is often called 2x oversampling, because the typical audio sampling rate (for compact disks [CDs]) is 44.1 kHz. Mathematically, 2x oversampling is twice the START_HTML_TAGiEND_HTML_TAGminimumSTART_HTML_TAG/iEND_HTML_TAG non-aliasing sampling rate for any given signal. - The sampling interval (or sampling period) of a discrete-time signal is defined as the reciprocal of the sampling rate. Since the sampling rate is normally in units of samples per second (Hz), the sampling interval is normally in units of seconds. See also: sampling rate, sampling theorem - Derivation of the START_HTML_TAGiEND_HTML_TAGsampling theoremSTART_HTML_TAG/iEND_HTML_TAG which states that any signal can be perfectly reconstructed, in principle, from uniformly spaced samples of that signal, provided that the sampling rate is higher than twice the highest frequency present in the signal
  Sampling Rate  [hits=1 votes=1 rating=5.0] - The sampling rate of a discrete-time signal is defined as the number of samples per second. Its units are thus in Hertz (Hz). Shannon's Sampling Theorem states that the original continuous-time signal can be recovered exactly from the samples if and only if the sampling rate is higher than twice the highest frequency present in the original signal. Any higher frequencies will alias to frequencies below half the sampling rate.
  Scatter Filter  [hits=14 votes=1 rating=5.0] : Reflection Transfer Function
  Scatter Junction  [hits=24 votes=1 rating=5.0] : Scattering Junctions
  Scatter Matrix (2 competing definitions)
  Scatter Theory (4 competing definitions)
  Schur Function  [hits=1 votes=1 rating=5.0] : Relation to Schur Functions
  Second-Order Cone Programming  [hits=25 votes=1 rating=0.0] : SOCP - Implementation of a primal-dual potential reduction method for solving second-order cone programming problems: minimize f^T x, subject to
  Self Inductance  [hits=1 votes=1 rating=5.0] : Self Inductance -- from Eric Weisstein's World of Physics
  Series Adaptor  [hits=30 votes=1 rating=5.0]
  Series Combination  [hits=1 votes=1 rating=5.0] : Series Combination of One-Ports
  Shannon Sampling Theorem (2 competing definitions)
  Shift Operator  [hits=25 votes=1 rating=5.0]
  Shift Theorem (3 competing definitions)
  Shift Theorem for Z Transform  [hits=13 votes=1 rating=5.0] : Shift Theorem
  Short Time Fourier Transform  [hits=14 votes=1 rating=5.0] : AES Papers - This paper describes a system for audio analysis, modification, and synthesis, based on the Short Time Fourier Transform (STFT). The system is intended both as a tool for sound manipulation, and as a means to reinforce people's intuitions regarding the relationships between timbre and the harmonic structure of music and other audio signals, as conveyed via their spectrograms. This is done by creating a 3D spectrogram which shows a sound's harmonic structure in great detail as it is sampled.
  Sifting Property  [hits=1 votes=1 rating=5.0] - The integral of a continuous signal times an impulse signal equals the signal value at the time of the impulse.
  Signal Flow Graph  [hits=1 votes=1 rating=5.0]
  Signal Power  [hits=33 votes=1 rating=5.0] : Signal Metrics
  Similarity Theorem  [hits=21 votes=1 rating=5.0] : Scaling Theorem
  Simpson Rule  [hits=30 votes=1 rating=5.0] - Simpson's rule is a Newton-Cotes formula for approximating the integral of a function f using quartic polynomials (i.e., parabolic arcs instead of the straight line segments used in the trapezoidal rule). Simpson's rule can be derived by integrating a third-order Lagrange interpolating polynomial fit to the function at three equally spaced points.
  Simulation Diagram  [hits=1 votes=1 rating=5.0] : Simulation Diagrams
  Sinc Function (3 competing definitions)
  Sinusoid (4 competing definitions)
  Spatial Derivative  [hits=14 votes=1 rating=5.0] : Spatial Derivatives
  Spatial Frequency  [hits=1 votes=1 rating=5.0] : Spatial Frequency -- from Eric Weisstein's World of Physics
  Spectral Phase  [hits=1 votes=1 rating=5.0]
  Spectral Resolution  [hits=1 votes=1 rating=5.0] : Spectral Resolution -- from Eric Weisstein's World of Physics - Spectral resolution refers to the resolving power of a spectrum analysis. In particular, taking the Fourier transform of a longer data segment yields higher spectral resolution. To resolve two sinusoids which are closely spaced in frequency, a spectrum analysis needs to use on the order of two cycles (or more) of the difference frequency.
  Spectrum Analysis  [hits=1 votes=1 rating=5.0] : DFT Applications - Spectrum analysis of sound is analogous to decomposing white light into its component colors by means of a prism
  Stable Finite Difference Scheme  [hits=1 votes=1 rating=5.0] : Stability of a Finite Difference Scheme
  Statistical Signal Processing (3 competing definitions)
  Stretch Operator  [hits=21 votes=1 rating=5.0]
  Stretch Theorem (2 competing definitions)
  Strictly Proper Transfer Function  [hits=39 votes=1 rating=5.0] : Partial Fraction Expansion
  Tapped Delay Line (3 competing definitions)
  Techniques for Digital Filter Design and System Identification with Application to the Violin  [hits=23 votes=1 rating=5.0] - PhD/EE dissertation: Julius O. Smith III
  Time Scale Modification  [hits=19 votes=1 rating=5.0] : Time Stretching and Pitch Shifting of Audio Signals - Time scaling refers to modifying the time-duration of a signal without modifying its audio spectral characteristics. It is often used to slow down music without changing the pitch of any of the instruments. It is not possible to do this exactly in general, but good approximate methods exist using, e.g., the phase vocoder. - As opposed to the process of pitch transposition achieved using a simple sample rate conversion, Pitch Scaling is a way to change the pitch of a signal without changing its length. In practical applications, this is achieved by changing the length of a sound using one of the below methods and then performing a sample rate conversion to change the pitch. There exists a certain confusion in terminology, as Pitch Scaling is often also incorrectly named 'Pitch Shift' (a term coined by the music industry). A true Pitch Shift (as obtainable by modulating an analytic signal by a complex exponential) will shift the spectrum of a sound, while Pitch Scaling will dilate it, upholding the harmonic relationship of the sound. The actual Pitch Shifting yields a metallic, inharmonic sound which may well be an interesting special effect but which is a totally inadequate process for changing the pitch of any harmonic sound except a single sine wave.
  Total Harmonic Distortion  [hits=1 votes=1 rating=5.0] : Definition: total harmonic distortion
  Transfer Function