2D Filters Motivation & Applications As the rapid development of information science and computing technology, the theory of digital filters design and application has achieved leap-forward development.
A derivation of the discrete Fourier transform In mathematics, computer science, and electrical engineering, the discrete Fourier transform, occasionally called the finite Fourier transform, is a transform for Fourier analysis of finite-doma...
Abdul Jerri Dr. Abdul Jabbar Hassoon Jerri (July 20, 1932) is an Iraqi American physicist and mathematician, most recognized for his contributions to information theory in general, in particular to the unde...
Adaptive equalizer An adaptive equalizer is an equalizer that automatically adapts to time-varying properties of the communication channel.
Adaptive filter An adaptive filter is a system with a linear filter that has a transfer function controlled by variable parameters and a means to adjust those parameters according to an optimization algorithm.
Adaptive predictive coding Adaptive predictive coding (APC) is a narrowband analog-to-digital conversion that uses a one-level or multilevel sampling system in which the value of the signal at each sampling instant ...
Adaptive-additive algorithm In order to reconstruct this phase the Adaptive-Additive Algorithm (or AA algorithm), which derives from a group of adaptive (input-output) algorithms, can be used.
Advanced process control In control theory Advanced process control (APC) refers to a broad range of techniques and technologies implemented within industrial process control systems.
Aliasing In signal processing and related disciplines, aliasing refers to an effect that causes different signals to become indistinguishable (or aliases of one another) when sampled.
All-pass filter An all-pass filter is a signal processing filter that passes all frequencies equally in gain, but changes the phase relationship between various frequencies.
Almost periodic function In mathematics, an almost periodic function is, loosely speaking, a function of a real number that is periodic to within any desired level of accuracy, given suitably long, well-distributed "alm...
Analog-to-digital converter An analog-to-digital converter (abbreviated ADC, A/D or A to D) is a device that converts a continuous physical quantity (usually voltage) to a digital number that represents t...
Anti-aliasing filter An anti-aliasing filter is a filter used before a signal sampler, to restrict the bandwidth of a signal to approximately satisfy the sampling theorem.
Anticausal system An anticausal system is a hypothetical system with outputs and internal states that depend solely on future input values.
ASOCS ASOCS Ltd. is a privately held company, develops and markets multi-core processors that enable software to program physical layer functions and algorithms.
Audio forensics Audio forensics is the field of forensic science relating to the acquisition, analysis, and evaluation of sound recordings that may ultimately be presented as admissible evidence in a court of l...
Audio normalization Audio normalization is the application of a constant amount of gain to an audio recording to bring the average or peak amplitude to a target level.
Audio Signal Processor The Audio Signal Processor or ASP (also known as the SoundDroid) is a large-scale digital signal processor developed by James A. Moorer at Lucasfilm's The Droid Works.
Automatic control Automatic control is the application of control theory for regulation of processes without direct human intervention.
Banded waveguide synthesis Banded Waveguides Synthesis is a physical modeling synthesis method to simulate sounds of dispersive sounding objects, or objects with strongly inharmonic resonant frequencies efficiently.
Bandlimiting Bandlimiting is the limiting of a deterministic or stochastic signal's Fourier transform or power spectral density to zero above a certain finite frequency.
Bartlett's method In time series analysis, Bartlett's method (also known as the method of averaged periodograms), is used for estimating power spectra.
Beta encoder A beta encoder is an analog to digital conversion (A/D) system in which a real number in the unit interval is represented by a finite representation of a sequence in base beta, with beta bei...
BIBO stability In signal processing, specifically control theory, BIBO stability is a form of stability for linear signals and systems that take inputs.
Bilinear time-frequency distribution Bilinear time-frequency distributions, or quadratic time-frequency distributions, arise in a sub-field field of signal analysis and signal processing called time-frequency signal processin...
Bilinear time–frequency distribution Bilinear time-frequency distributions, or quadratic time-frequency distributions, arise in a sub-field field of signal analysis and signal processing called time-frequency signal processin...
Bilinear transform The bilinear transform (also known as Tustin's method) is used in digital signal processing and discrete-time control theory to transform continuous-time system representations to discrete...
Bin-centres A bin-centres test signal is one which has been constructed such that it has frequency components at FFT bin-centre frequencies.
Bistritz Criterion The Bistritz criterion refers to a simple method to determine whether a discrete linear time invariant (LTI) system is stable 12.
Bistritz criterion The Bistritz criterion is a simple method to determine whether a discrete linear time invariant (LTI) system is stable 12.
Bistritz stability criterion In signal processing and control theory, the Bistritz criterion is a simple method to determine whether a discrete linear time invariant (LTI) system is stable proposed in see also.
Cascaded integrator-comb filter In digital signal processing, a cascaded integrator–comb (CIC) is an optimized class of finite impulse response (FIR) filter combined with an interpolator or decimator.
Cheung-Marks theorem In information theory, the Cheung–Marks theorem, named after K. F. Cheung and Robert J. Marks II, specifies conditions where restoration of a signal by the sampling theorem can become ill-...
Cheung–Marks theorem In information theory, the Cheung–Marks theorem, named after K. F. Cheung and Robert J. Marks II, specifies conditions where restoration of a signal by the sampling theorem can become ill-...
Codec A codec is a device or computer program capable of encoding or decoding a digital data stream or signal.
dbx Model 700 Digital Audio Processor The dbx Model 700 Digital Audio Processor was a professional audio ADC/DAC combination unit, which digitized a stereo analog audio input into a bitstream, which was then encoded and encapsulated...
Delay equalization In signal processing, delay equalization corresponds to adjusting the relative phases of different frequencies to achieve a constant group delay, using by adding an all-pass filter in series wit...
Delta modulation Delta modulation (DM or Δ-modulation)is an analog-to-digital and digital-to-analog signal conversion technique used for transmission of voice information where quality is not of primary importance.
Delta-sigma modulation Delta-sigma (ΔΣ; or sigma-delta, ΣΔ) modulation is a digital signal processing, or DSP method for encoding analog signals into digital signals as found in an ADC. It is also us...
DFT matrix A DFT matrix is an expression of a discrete Fourier transform (DFT) as a matrix multiplication.
Differential nonlinearity Differential nonlinearity is a term describing the deviation between two analog values corresponding to adjacent input digital values.
Digital delay line A digital delay line is a discrete element in digital filter theory, which allows a signal to be delayed by a number of samples.
Digital down converter In digital signal processing, a digital down-converter (DDC) converts a digitized real signal centered at an intermediate frequency (IF) to a basebanded complex signal centered at zero fre...
Digital feedback reduction Digital feedback reduction is the application of digital techniques to sound reinforcement in order to reduce audio feedback and increase headroom.
Digital filter In signal processing, a digital filter is a system that performs mathematical operations on a sampled, discrete-time signal to reduce or enhance certain aspects of that signal.
Digital frequency Digital frequency is the analogue for discrete signals as frequency is to continuous signals.
Digital signal A digital signal is a physical signal that is a representation of a sequence of discrete values (a quantified discrete-time signal), for example of an arbitrary bit stream, or of a digitized (sa...
Digital signal controller A digital signal controller (DSC) is a hybrid of microcontrollers and digital signal processors (DSPs).
Digital signal processing Digital signal processing (DSP) is the mathematical manipulation of an information signal to modify or improve it in some way.
Digital signal processor A digital signal processor (DSP) is a specialized microprocessor with an architecture optimized for the operational needs of digital signal processing.
Digital-to-analog converter In electronics, a digital-to-analog converter (DAC, D/A, D2A or D-to-A) is a function that converts digital data (usually binary) into an analog signal (current, voltage,...
Dirac delta function In mathematics, the Dirac delta function, or function, is a generalized function, or distribution, on the real number line that is zero everywhere except at zero, with an integral of one ov...
Direct digital synthesizer Direct Digital Synthesizer (DDS) is a type of frequency synthesizer used for creating arbitrary waveforms from a single, fixed-frequency reference clock.
Discrete cosine transform A discrete cosine transform (DCT) expresses a finite sequence of data points in terms of a sum of cosine functions oscillating at different frequencies.
Discrete Fourier transform In mathematics, the discrete Fourier transform is a specific kind of discrete transform, used in Fourier analysis.
Discrete signal A discrete signal or discrete-time signal is a time series consisting of a sequence of qualities.
Discrete transform In signal processing, discrete transforms are mathematical transforms, often linear transforms, of signals between discrete domains, such as between discrete time and discrete frequency.
Discrete wavelet transform In numerical analysis and functional analysis, a discrete wavelet transform (DWT) is any wavelet transform for which the wavelets are discretely sampled.
Effective number of bits Effective number of bits (ENOB) is a measure of the dynamic performance of an analog-to-digital converter (ADC) and its associated circuitry.
Encoding law In digital communications, an encoding law is a (typically non-uniform) allocation of signal quantization levels across the possible analog signal levels in an analog to digital converter system.
ENOB Effective Number of Bits (ENOB) is a measure of the quality of a digitised signal.
eXpressDSP eXpressDSP is a software package produced by Texas Instruments.
Fast Fourier transform A fast Fourier transform (FFT) is an algorithm to compute the discrete Fourier transform (DFT) and its inverse.
Fast Fourier Transform Telescope Fast Fourier Transform Telescope is Tegmark and Zaldarriaga's name for a design for an all-digital synthetic-aperture telescope.
Fast Walsh-Hadamard transform In computational mathematics, the Hadamard ordered fast Walsh–Hadamard transform (FWHTh) is an efficient algorithm to compute the Walsh–Hadamard transform (WHT).
Fast Walsh–Hadamard transform In computational mathematics, the Hadamard ordered fast Walsh–Hadamard transform (FWHTh) is an efficient algorithm to compute the Walsh–Hadamard transform (WHT).
FDOA Frequency difference of arrival (FDOA), also frequently called differential Doppler (DD), is a technique analogous to TDOA for estimating the location of a radio emitter based on observati...
Filter bank In signal processing, a filter bank is an array of band-pass filters that separates the input signal into multiple components, each one carrying a single frequency sub-band of the original signal.
Filter design Filter design is the process of designing a signal processing filter that satisfies a set of requirements, some of which are contradictory.
Finite impulse response In signal processing, a finite impulse response (FIR) filter is a filter whose impulse response (or response to any finite length input) is of finite duration, because it settles to zero in ...
First-order hold The first-order hold (FOH) is a mathematical model of the practical reconstruction of sampled signals that could be done by a conventional digital-to-analog converter (DAC) and an analog circuit...
Folding (DSP implementation) Folding is a transformation technique using in DSP architecture implementation for minimizing the number of functional blocks in synthesizing DSP architecture.
Fourier analysis In mathematics, Fourier analysis is the study of the way general functions may be represented or approximated by sums of simpler trigonometric functions.
Frequency estimation Frequency estimation is the process of estimating the complex frequency components of a signal in the presence of noise.
Full scale In electronics and signal processing, full scale or full code represents the maximum amplitude a system can present.
Generalized distributive law The generalized distributive law (GDL) is a general message passing algorithm devised by Srinivas M. Aji and Robert J. McEliece.
Geometric Arithmetic Parallel Processor The GAPP (Geometric Arithmetic Parallel Processor), invented by Polish mathematician Włodzimierz Holsztyński in 1981, was patented by Martin Marietta and is now owned by Silicon Optix, Inc. The...
Geometric-Arithmetic Parallel Processor The GAPP (Geometric-Arithmetic Parallel Processor), invented by Polish mathematician Włodzimierz Holsztyński in 1981, was patented by Martin Marietta and is now owned by Silicon Optix, Inc. In t...
Gerchberg-Saxton algorithm The Gerchberg-Saxton (GS) algorithm is an iterative algorithm for retrieving the phase of a pair of light distributions (or any other mathematically valid distribution) related via a propagating...
Gerchberg–Saxton algorithm The Gerchberg-Saxton (GS) algorithm is an iterative algorithm for retrieving the phase of a pair of light distributions (or any other mathematically valid distribution) related via a propagating...
Goertzel algorithm The Goertzel algorithm is a Digital Signal Processing (DSP) technique that provides a means for efficient evaluation of individual terms of the Discrete Fourier Transform (DFT), thus making it u...
HADES (software) HADES (Haskins Analysis Display and Experiment System) refers to a family of signal processing computer programs that was developed in the 1980s at Haskins Laboratories by Philip Rubin and colle...
Half-band filter In digital signal processing, such as cell phones, digital receivers, television, CD & DVD players, half-band filters are widely used for their efficiency in multi-rate applications.
High Frequency Content measure The High Frequency Content measure is a simple measure, taken across a signal spectrum (usually a STFT spectrum), which can be used to characterize the amount of high-frequency content in the signal.
Host media processing A telephony system based on host media processing is one that uses a general-purpose computer to process a telephony call’s media stream rather than using digital signal processors to perform th...
IBM Mwave Mwave was a technology developed by IBM allowing for the combination of telephony and sound card features on a single adapter card.
Ideal sampler In signal processing, an ideal sampler is a theoretical operation whose input is a continuous signal and whose output is a sequence of instantaneous values of the signal at discrete moments of t...
Impulse invariance Impulse invariance is a technique for designing discrete-time infinite-impulse-response (IIR) filters from continuous-time filters in which the impulse response of the continuous-time system is ...
Instantaneous phase Instantaneous phase and instantaneous frequency are important concepts in signal processing that occur in the context of the representation and analysis of time-varying functions.
Integral nonlinearity Integral nonlinearity (acronym INL) is a term describing the maximum deviation between the ideal output of a DAC and the actual output level (after offset and gain errors have been removed).
James A. Moorer James Andy Moorer is an internationally known figure in digital audio and computer music, with over 40 technical publications and four patents to his credit.
Lapped transform In signal processing, a lapped transform is a type of linear discrete block transformation where the basis functions of the transformation overlap the block boundaries, yet the number of coeffic...
Least mean squares filter Least mean squares (LMS) algorithms are a class of adaptive filter used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean squares of the error ...
Least-squares spectral analysis Least-squares spectral analysis is a method of estimating a frequency spectrum, based on a least squares fit of sinusoids to data samples, similar to Fourier analysis.
Line spectral pairs Line spectral pairs (LSP) or line spectral frequencies (LSF) are used to represent linear prediction coefficients (LPC) for transmission over a channel.
Linear predictive coding Linear predictive coding (LPC) is a tool used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed for...
Logarithmic number system A logarithmic number system (LNS) is an arithmetic system used for representing real numbers in computer and digital hardware, especially for digital signal processing.
Low-pass filter A low-pass filter is a filter that passes low-frequency signals and attenuates (reduces the amplitude of) signals with frequencies higher than the cutoff frequency.
LTI system theory Linear time-invariant system theory, commonly known as LTI system theory, comes from applied mathematics and has direct applications in NMR spectroscopy, seismology, circuits, signal proce...
Machine listening Machine listening is a technique using software and hardware to extract meaningful information from audio signals.
Matched Z-transform method The matched Z-transform method, also called the pole–zero mapping or pole–zero matching method, is a technique for converting a continuous-time filter design to a discrete-time filte...
Media processor A media processor, mostly used as an image / video processor, is a microprocessor-based system-on-a-chip which is designed to deal with digital streaming data in real-time (e.g.
Minimum phase In control theory and signal processing, a linear, time-invariant system is said to be minimum-phase if the system and its inverse are causal and stable.
Multi-core processor A multi-core processor is a single computing component with two or more independent actual central processing units (called "cores"), which are the units that read and execute program instructions.
Multidimensional sampling In digital signal processing, multidimensional sampling is the process of converting a function of a multidimensional variable into a discrete collection of values of the function measured on a ...
Multiply-accumulate In computing, especially digital signal processing, multiply-accumulate is a common operation that computes the product of two numbers and adds that product to an accumulator.
Multiply-accumulate operation In computing, especially digital signal processing, the multiply–accumulate operation is a common step that computes the product of two numbers and adds that product to an accumulator.
NeuroMatrix NeuroMatrix is a digital signal processor series developed by NTC Module.
Noise shaping Noise shaping is a technique typically used in digital audio, image, and video processing, usually in combination with dithering, as part of the process of quantization or bit-depth reduction of...
Non-uniform discrete Fourier transform In applied mathematics, the non-uniform discrete Fourier transform (NDFT) of a signal is a type of Fourier transform, related to a discrete Fourier transform or discrete-time Fourier transform, ...
Numerically-controlled oscillator A numerically controlled oscillator is a digital signal generator which creates a synchronous, discrete-time, discrete-valued representation of a waveform, usually sinusoidal.
Nyquist frequency The Nyquist frequency, named after electronic engineer Harry Nyquist, is ½ of the sampling rate of a discrete signal processing system.
Nyquist ISI criterion In communications, the Nyquist ISI criterion describes the conditions which, when satisfied by a communication channel (including responses of transmit and receive filters), result in no intersy...
Nyquist rate In signal processing, the Nyquist rate, named after Harry Nyquist, is twice the bandwidth of a bandlimited function or a bandlimited channel.
Nyquist-Shannon sampling theorem The Nyquist–Shannon sampling theorem, after Harry Nyquist and Claude Shannon, is a fundamental result in the field of information theory, in particular telecommunications and signal processing.
Nyquist–Shannon sampling theorem The Nyquist–Shannon sampling theorem, after Harry Nyquist and Claude Shannon, is a fundamental result in the field of information theory, in particular telecommunications and signal processing.
Outboard gear Musical "outboard equipment" or "gear" is used to alter how a musical instrument sounds.
Parks-McClellan filter design algorithm The Parks-McClellan algorithm, published by James McClellan and Thomas Parks in 1972, is an iterative algorithm for finding the optimal Chebyshev finite impulse response filter.
Pipelining (DSP implementation) Pipelining is an important technique used in several applications such as digital signal processing (DSP) systems, microprocessors, etc.
Pitch correction Pitch correction is the process of correcting the intonation of an audio signal without affecting other aspects of its sound.
Pitch detection algorithm A pitch detection algorithm (PDA) is an algorithm designed to estimate the pitch or fundamental frequency of a quasiperiodic or virtually periodic signal, usually a digital recording of speech o...
Pitch shift Pitch shifting is a sound recording technique in which the original pitch of a sound is raised or lowered.
PLL multibit A PLL multibit or multibit PLL is a phase-locked loop which achieves improved performance compared to a unibit PLL by using more bits.
Polyphase matrix A polyphase matrix is a matrix whose elements are filter masks.
Polyphase quadrature filter A polyphase quadrature filter, or PQF, is a filter bank which splits an input signal into a given number N of equidistant sub-bands.
Quadrature mirror filter In digital signal processing, a quadrature mirror filter is a filter most commonly used to implement a filter bank that splits an input signal into two bands.
Quantization (signal processing) Quantization, in mathematics and digital signal processing, is the process of mapping a large set of input values to a (countable) smaller set – such as rounding values to some unit of precision.
Ramer-Douglas-Peucker algorithm The Douglas–Peucker algorithm is an algorithm for reducing the number of points in a curve that is approximated by a series of points.
Ramer-Douglas–Peucker algorithm The Douglas–Peucker algorithm is an algorithm for reducing the number of points in a curve that is approximated by a series of points.
Ramer–Douglas–Peucker algorithm The Douglas–Peucker algorithm is an algorithm for reducing the number of points in a curve that is approximated by a series of points.
Reconstruction filter In a mixed-signal system (analog and digital), a reconstruction filter (or anti-imaging filter) is used to construct a smooth analogue signal from a digital input, as in the case of a digi...
Recursive least squares filter The Recursive least squares (RLS) adaptive filter is an algorithm which recursively finds the filter coefficients that minimize a weighted linear least squares cost function relating to the inpu...
Resampling (audio) Resampling is synonymous with several processes commonly used in manipulating audio, through which a segment of sampled audio (often erroneously termed a sample) is manipulated before being stor...
Resampling (bitmap) Resampling is the digital process of changing the sample rate or dimensions of digital imagery by temporally or areally analysing and sampling the original data.
Sample and hold In electronics, a sample and hold (S/H, also "follow-and-hold") circuit is an analog device that samples (captures, grabs) the voltage of a continuously varying analog signal and holds (lo...
Sample rate conversion Sample rate conversion is the process of changing the sampling rate of a discrete-time signal to obtain a new discrete-time representation of the underlying continuous-time signal.
Sampling rate The sampling rate, sample rate, or sampling frequency defines the number of samples per unit of time taken from a continuous signal to make a discrete signal.
Sensor hub A sensor hub is a microcontroller unit/coprocessor/DSP that helps to integrate data from different sensors and process them.
Shapiro polynomials In mathematics, the Shapiro polynomials are a sequence of polynomials which were first studied by Harold S. Shapiro in 1951 when considering the magnitude of specific trigonometric sums.
Signal (electrical engineering) In the fields of communications, signal processing, and in electrical engineering more generally, a signal is any time-varying or spatial-varying quantity.
Signal (electronics) An electronic signal is the embodiment of a signal in electrical form made by a transducer that converts the signal from whatever its original form to a form expressed as voltage or current, or ...
Signal averaging Signal averaging is a signal processing technique applied in the time domain, intended to increase the strength of a signal relative to noise that is obscuring it.
SigSpec SigSpec is an acronym of "SIGnificance SPECtrum" and addresses a statistical technique to provide the reliability of periodicities in a measured (noisy and not necessarily equidistant) time series.
SIMD Single instruction, multiple data (SIMD), is a class of parallel computers in Flynn's taxonomy.
Similarities between Wiener and LMS Sahasra Bhoomi presents regional ring road city, A hmda layout at bibinagar beside regional ringroad allotted according to the master plan-2031 by hmda.
SINADR Signal-to-noise and distortion ratio is a measurement of the purity of a signal.
Sinc filter In signal processing, a sinc filter is an idealized filter that removes all frequency components above a given cutoff frequency, without affecting lower frequencies, and has linear phase response.
Sndr SINAD stands for Signal-to-noise and distortion ratio.
Sogitec 4X The Sogitec 4X was a digital sound processing workstation developed by Giuseppe Di Giugno at IRCAM in the 1980s.
SoundDroid The SoundDroid is an early digital audio workstation designed by a team of engineers led by James A. Moorer at Lucasfilm between the 1980 and 1987.
Source separation Source separation problems in digital signal processing are those in which several signals have been mixed together and the objective is to find out what the original signals were.
Spectral centroid The spectral centroid is a measure used in digital signal processing to characterise a spectrum.
Spectral flatness Spectral flatness or tonality coefficient, also known as Wiener entropy, is a measure used in digital signal processing to characterize an audio spectrum.
Spectral flux Spectral flux is a measure of how quickly the power spectrum of a signal is changing, calculated by comparing the power spectrum for one frame against the power spectrum from the previous frame.
Spectral leakage In Fourier analysis, spectral leakage refers to the misrepresentation of the Fourier components of a signal that are not harmonic to the fundamental frequency.
Spectral slope In astrophysics and planetary science, spectral slope, also called spectral gradient , is a measure of dependence of the reflectance on the wavelength.
Spectrum continuation analysis Spectrum continuation analysis is a generalization of the concept of Fourier series to non-periodic functions of which only a fragment has been sampled in the time domain.
Spurious-free dynamic range Spurious-Free Dynamic Range (SFDR) is the strength ratio of the fundamental signal to the strongest spurious signal in the output.
Successive approximation ADC A successive approximation ADC is a type of analog-to-digital converter that converts a continuous analog waveform into a discrete digital representation via a binary search through all possible...
Super Bit Mapping Super Bit Mapping (SBM) is a noise shaping process, developed by Sony for CD mastering.
Super-resolution Super-resolution or superresolution is a class of techniques that enhance the resolution of an imaging system.
System analysis System analysis in the field of electrical engineering characterizes electrical systems and their properties.
Talk box A talk box or talkbox is an effects unit that allows musicians to modify the sound of a musical instrument.
Time to digital converter In electronic instrumentation and signal processing, a time to digital converter is a device for recognizing events and providing a digital representation of the time they occurred.
Time-to-digital converter In electronic instrumentation and signal processing, a time to digital converter (abbreviated TDC) is a device for recognizing events and providing a digital representation of the time the...
Tricore TriCore™ is a 32-bit microcontroller architecture from Infineon.
Tristimulus timbre model In music, timbre (or) also known as tone color or tone quality from psychoacoustics, is the quality of a musical note or sound or tone that distinguishes different types of sound pro...
Unfolding (DSP implementation) Unfolding is a transformation technique of duplicating the functional blocks to increase the throughput of the DSP program in such a way that preserves its functional behavior at its outputs.
Unity amplitude A sinusoidal waveform is said to have a unity amplitude when the amplitude of the wave is equal to 1.
Upsampling Upsampling is interpolation, applied in the context of digital signal processing and sample rate conversion.
Voice activity detection Voice activity detection, also known as speech activity detection or speech detection, is a technique used in speech processing in which the presence or absence of human speech is de...
Warped linear predictive coding Warped linear predictive coding (warped LPC or WLPC) is a variant of linear predictive coding in which the spectral representation of the system is modified, for example by replacing...
Waveform buffer In computing, a waveform buffer is a technique for digital synthesis of repeating waveforms.
Welch's method In physics, engineering, and applied mathematics, Welch's method, named after P.D. Welch, is used for estimating the power of a signal vs. frequency, reducing noise compared to the methods it is...