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.

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...

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...

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.

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.

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 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.

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.

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.

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.

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...

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 signal to discrete time interval converter

Analog-signal-to-discrete-time-interval converter (written with or without hyphens; abbreviated as ASDTIC) is a specialized kind of an analog-to-digital converter, which converts the analo...

Analog-signal-to-discrete-time-interval converter (written with or without hyphens; abbreviated as ASDTIC) is a specialized kind of an analog-to-digital converter, which converts the analo...

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...

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.

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.

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.

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 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 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.

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.

Audio time-scale/pitch modification

Time stretching is the process of changing the speed or duration of an audio signal without affecting its pitch.

Time stretching is the process of changing the speed or duration of an audio signal without affecting its pitch.

Audio timescale-pitch modification

Time stretching is the process of changing the speed or duration of an audio signal without affecting its pitch.

Time stretching is the process of changing the speed or duration of an audio signal without affecting its pitch.

Automatic control

Automatic control is the application of control theory for regulation of processes without direct human intervention.

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.

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.

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.

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...

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.

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 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 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...

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.

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.

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.

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.

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.

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-...

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-...

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.

A codec is a device or computer program capable of encoding or decoding a digital data stream or signal.

Computational auditory scene analysis

Computational auditory scene analysis is the study of auditory scene analysis by computational means.

Computational auditory scene analysis is the study of auditory scene analysis by computational means.

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...

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...

Decimation (signal processing)

In digital signal processing, decimation is a technique for reducing the number of samples in a discrete-time signal.

In digital signal processing, decimation is a technique for reducing the number of samples in a discrete-time signal.

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...

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 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...

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.

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.

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.

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...

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 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.

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 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...

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).

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 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.

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,...

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...

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.

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.

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.

In mathematics, the discrete Fourier transform is a specific kind of discrete transform, used in Fourier analysis.

Discrete frequency domain

A discrete frequency domain is a frequency domain that is discrete rather than continuous.

A discrete frequency domain is a frequency domain that is discrete rather than continuous.

Discrete signal

A discrete signal or discrete-time signal is a time series consisting of a sequence of qualities.

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.

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.

In numerical analysis and functional analysis, a discrete wavelet transform (DWT) is any wavelet transform for which the wavelets are discretely sampled.

Discrete-time Fourier transform

In mathematics, the discrete-time Fourier transform (DTFT) is one of the specific forms of Fourier analysis.

In mathematics, the discrete-time Fourier transform (DTFT) is one of the specific forms of Fourier analysis.

Discrete-time signal

A discrete signal or discrete-time signal is a time series consisting of a sequence of quantities.

A discrete signal or discrete-time signal is a time series consisting of a sequence of quantities.

Dither

Dither is an intentionally applied form of noise used to randomize quantization error, preventing large-scale patterns such as color banding in images.

Dither is an intentionally applied form of noise used to randomize quantization error, preventing large-scale patterns such as color banding in images.

Downsampling

In signal processing, downsampling is the process of reducing the sampling rate of a signal.

In signal processing, downsampling is the process of reducing the sampling rate of a signal.

DSSP (imaging)

DSSP stands for digital shape sampling and processing.

DSSP stands for digital shape sampling and processing.

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.

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.

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.

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.

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.

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 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 (FWHT

In computational mathematics, the Hadamard ordered fast Walsh–Hadamard transform (FWHT

_{h}) 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 (FWHT

In computational mathematics, the Hadamard ordered fast Walsh–Hadamard transform (FWHT

_{h}) 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...

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.

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.

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 ...

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...

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.

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.

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.

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.

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.

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...

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...

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...

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...

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...

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...

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.

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.

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...

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.

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...

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 ...

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 ...

Infinite impulse response

Infinite impulse response (IIR) is a property of signal processing systems.

Infinite impulse response (IIR) is a property of signal processing systems.

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.

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).

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.

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.

Kernel adaptive filter

In signal processing, a kernel adaptive filter is a type of nonlinear adaptive filter.

In signal processing, a kernel adaptive filter is a type of nonlinear adaptive filter.

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...

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 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.

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.

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...

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.

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.

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...

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...

Lu-Wu–Iyengar detection algorithm for spectrum sensing

The Lu–Wu–Iyengar detection algorithm for spectrum sensing is an approach to sensing radio and such signals.

The Lu–Wu–Iyengar detection algorithm for spectrum sensing is an approach to sensing radio and such signals.

Machine listening

Machine listening is a technique using software and hardware to extract meaningful information from audio signals.

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...

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.

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.

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.

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.

Multi-rate digital signal processing

Multi-rate signal processing studies digital signal processing systems which include sample rate conversion.

Multi-rate signal processing studies digital signal processing systems which include sample rate conversion.

Multidelay block frequency domain adaptive filter

The Multidelay block frequency domain adaptive filter algorithm is a block-based frequency domain implementation of the Least mean squares filter algorithm.

The Multidelay block frequency domain adaptive filter algorithm is a block-based frequency domain implementation of the Least mean squares filter algorithm.

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 ...

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 ...

Multiple signal classification

MUltiple SIgnal Classification (MUSIC) is an algorithm used for frequency estimation and emitter location.

MUltiple SIgnal Classification (MUSIC) is an algorithm used for frequency estimation and emitter location.

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.

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.

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.

Native processing

Native processing can have a general, wider meaning.

Native processing can have a general, wider meaning.

NeuroMatrix

NeuroMatrix is a digital signal processor series developed by NTC Module.

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...

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...

Noise-Predictive Maximum-Likelihood (NPML) Detection

1. Principles In general, NPML refers to a family of sequence-estimation data detectors, which arise by imbedding a noise prediction/whitening processinto the branch metric computation of the Vi...

1. Principles In general, NPML refers to a family of sequence-estimation data detectors, which arise by imbedding a noise prediction/whitening processinto the branch metric computation of the Vi...

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, ...

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, ...

Nonuniform sampling

Nonuniform sampling is a branch of Nyquist–Shannon sampling theorem.

Nonuniform sampling is a branch of Nyquist–Shannon sampling theorem.

Normalized frequency (digital signal processing)

In digital signal processing (DSP), the continuous time variable, t, with units of seconds, is replaced by the discrete integer variable, n, with units of samples.

In digital signal processing (DSP), the continuous time variable, t, with units of seconds, is replaced by the discrete integer variable, n, with units of samples.

NTC Module

NTC Module is a Russian scientific technological center, founded in 1990 by the two enterprises of Russian military-industrial complex: NPO Vympel and NII Radiopriborostroyeniye.

NTC Module is a Russian scientific technological center, founded in 1990 by the two enterprises of Russian military-industrial complex: NPO Vympel and NII Radiopriborostroyeniye.

Numerically controlled oscillator

A numerically controlled oscillator (NCO) is a digital signal generator which creates a synchronous (i.e.

A numerically controlled oscillator (NCO) is a digital signal generator which creates a synchronous (i.e.

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.

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.

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...

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.

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.

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.

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.

Musical "outboard equipment" or "gear" is used to alter how a musical instrument sounds.

Oversampled binary image sensor

An oversampled binary image sensor is a new image sensor that is reminiscent of traditional photographic film.

An oversampled binary image sensor is a new image sensor that is reminiscent of traditional photographic film.

Oversampling

In signal processing, oversampling is the process of sampling a signal with a sampling frequency significantly higher than the Nyquist frequency.

In signal processing, oversampling is the process of sampling a signal with a sampling frequency significantly higher than the Nyquist frequency.

Parallel Processing (DSP implementation)

Parallel Processing in digital signal processing is a technique duplicating function units to operate different tasks simultaneously.

Parallel Processing in digital signal processing is a technique duplicating function units to operate different tasks simultaneously.

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.

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.

Pipelining is an important technique used in several applications such as digital signal processing (DSP) systems, microprocessors, etc.

Pisarenko harmonic decomposition

Pisarenko harmonic decomposition, also referred to as Pisarenko's method, is a method of frequency estimation.

Pisarenko harmonic decomposition, also referred to as Pisarenko's method, is a method of frequency estimation.

Pitch correction

Pitch correction is the process of correcting the intonation of an audio signal without affecting other aspects of its sound.

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...

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.

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.

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.

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.

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.

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.

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.

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.

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.

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...

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...

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 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.

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...

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.

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 (signal processing)

In signal processing, sampling is the reduction of a continuous signal to a discrete signal.

In signal processing, sampling is the reduction of a continuous signal to a discrete 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.

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.

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.

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.

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 ...

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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 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 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.

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.

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.

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.

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...

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 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.

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.

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.

A talk box or talkbox is an effects unit that allows musicians to modify the sound of a musical instrument.

The Generalized Distributive Law

The generalized distributive law is a general message passing algorithm devised by Srinivas M. Aji and Robert McEliece.

The generalized distributive law is a general message passing algorithm devised by Srinivas M. Aji and Robert McEliece.

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.

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...

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.

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...

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.

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.

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.

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...

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...

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.

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...

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...

Whittaker-Shannon interpolation formula

The Whittaker–Shannon interpolation formula is a method to reconstruct a continuous-time bandlimited signal from a set of equally spaced samples.

The Whittaker–Shannon interpolation formula is a method to reconstruct a continuous-time bandlimited signal from a set of equally spaced samples.

Whittaker–Shannon interpolation formula

The Whittaker–Shannon interpolation formula is a method to reconstruct a continuous-time bandlimited signal from a set of equally spaced samples.

The Whittaker–Shannon interpolation formula is a method to reconstruct a continuous-time bandlimited signal from a set of equally spaced samples.

Window function

In signal processing, a window function (also known as an apodization function or tapering function) is a mathematical function that is zero-valued outside of some chosen interval.

In signal processing, a window function (also known as an apodization function or tapering function) is a mathematical function that is zero-valued outside of some chosen interval.

XDAIS algorithms

XDAIS or eXpressDsp Algorithm Interoperability Standard is a standard for algorithm development by Texas Instruments for the TMS320 DSP family.

XDAIS or eXpressDsp Algorithm Interoperability Standard is a standard for algorithm development by Texas Instruments for the TMS320 DSP family.

Zero-order hold

The zero-order hold (ZOH) is a mathematical model of the practical signal reconstruction done by a conventional digital-to-analog converter (DAC).

The zero-order hold (ZOH) is a mathematical model of the practical signal reconstruction done by a conventional digital-to-analog converter (DAC).