Digital signal processing

thumb|right|In The Signal by William Powell Frith, a woman sends a signal by waving a white handkerchief.

system that converts an analog signal, such as a sound picked up by a microphone or light entering a digital camera, into a digital signal; device converting a physical quantity to a digital number

pseudo-function δ such that an integral of δ(x-c)f(x) always takes the value of f(c)

A codec is a computer hardware or software component that encodes or decodes a data stream or signal. Codec is a portmanteau of coder/decoder.

mathematical signal manipulation by computers

electronic device that converts a digital signal into an analog signal (mixed signal device), mostly realized as part of an integrated circuit (IC)

specialized microprocessor optimized for digital signal processing in real time, mainly used for audio and/or video applications

branch of mathematics regarding periodic and continuous signals

measurement of a signal at discrete time intervals

theorem in signal processing describing discrete samples of a continuous signal

process of mapping a continuous set to a countable set

microprocessor with more than one processing unit

In digital signal processing, aliasing is a phenomenon that a reconstructed signal from samples of the original signal contains low frequency components that are not present in the original one. This is caused when, in the original signal, there are components at frequency exceeding a certain frequency called Nyquist frequency, f_s / 2, where f_s is the sampling frequency (undersampling). This is because typical reconstruction methods use low frequency components while there are a number of frequency components, called aliases, which sampling result in the identical sample. It also often refer

technique representing data as sums of cosine functions

𝑂(𝑁 log 𝑁) divide‐and‐conquer algorithm to calculate the discrete Fourier transforms

proces used before a signal sampler to prevent aliasing

technique used in advanced mathematics

type of filter in signal processing

class of parallel computers in Flynn's taxonomy, with multiple processing elements that perform the same operation on multiple data points simultaneously

in signal processing, the frequency whose cycle-length is twice the interval between samples

type of instruction set architecture

mechanism to reduce or enhance aspects of a sampled, discrete-time signal

450px|thumb|right|Image on left is original. Center image is reduced to 16 colors. Right image also 16 colors, but dithered to reduce banding effect.

mathematical model of system that produces an output signal from any input signal subject to the constraints of linearity and time-invariance

property of many linear time-invariant (LTI) systems

effects unit that allows musicians to modify the sound of a musical instrument

function used in signal processing

speech analysis and encoding technique

process of reducing the sampling rate of a signal

method for creating waveforms

signal conversion technique

Fourier analysis technique applied to sequences

Method for converting signals between digital and analog

audio processing technique that changes the original pitch of a sound

system with self-optimizing transfer function

function that "converges" to periodicity

application of gain to a recording to achieve a target level

model of signal reconstruction in digital-to-analog (DAC) converters

process control theorem

line simplification algorithm

or multiplier–accumulator, in digital signal processing

digital control system

method used in digital signal processing and discrete-time control theory to transform continuous-time system representations to discrete-time and vice versa

signal (re-)construction algorithm

transform in numerical harmonic analysis

adaptive filter algorithm for digital signal processing

algorithm used for frequency estimation and radio direction finding

filter that passes signals of all frequencies with same gain, but changes the phase relationship among various frequencies

tool for Digital Signal Processing

idealized filter that removes all signal frequency components above a given frequency

changing the speed or duration of an audio signal without affecting its pitch

system where the output depends only on past and current inputs

twice the bandwidth of a bandlimited function or channel

digital signal generator creating a synchronous (clocked), discrete-time, discrete-valued representation of a waveform, usually sinusoidal

type of analog-digital conversion

technique for calibrating the pitch of an audio recording to match musical notes

algorithm

changing the sampling rate of a discrete signal to obtain a new discrete representation of the underlying continuous signal

In signal processing, oversampling is the process of sampling a signal at a sampling frequency significantly higher than the Nyquist rate. Theoretically, a bandwidth-limited signal can be perfectly reconstructed if sampled at the Nyquist rate or above it. The Nyquist rate is defined as twice the bandwidth of the signal. Oversampling is capable of improving resolution and signal-to-noise ratio, and can be helpful in avoiding aliasing and phase distortion by relaxing anti-aliasing filter performance requirements.

technique used in speech processing in which the presence or absence of human speech is detected