Whittaker–Shannon Interpolation Formula
   HOME

TheInfoList



OR:

The Whittaker–Shannon interpolation formula or sinc interpolation is a method to construct a continuous-time
bandlimited Bandlimiting is the limiting of a signal's frequency domain representation or spectral density to zero above a certain finite frequency. A band-limited signal is one whose Fourier transform or spectral density has bounded support. A bandli ...
function from a sequence of real numbers. The formula dates back to the works of
E. Borel E is the fifth letter of the Latin alphabet. E or e may also refer to: Commerce and transportation * €, the symbol for the euro, the European Union's standard currency unit * ℮, the estimated sign, an EU symbol indicating that the weight ...
in 1898, and
E. T. Whittaker Sir Edmund Taylor Whittaker (24 October 1873 – 24 March 1956) was a British mathematician, physicist, and historian of science. Whittaker was a leading mathematical scholar of the early 20th-century who contributed widely to applied mathema ...
in 1915, and was cited from works of
J. M. Whittaker John Macnaghten Whittaker Fellow of the Royal Society, FRS FRSE LLD (7 March 1905 – 29 January 1984) was a British mathematician and Vice-Chancellor of the University of Sheffield from 1953 to 1965. Life Whittaker was born 7 March 1905 i ...
in 1935, and in the formulation of the
Nyquist–Shannon sampling theorem The Nyquist–Shannon sampling theorem is a theorem in the field of signal processing which serves as a fundamental bridge between continuous-time signals and discrete-time signals. It establishes a sufficient condition for a sample rate that per ...
by
Claude Shannon Claude Elwood Shannon (April 30, 1916 – February 24, 2001) was an American people, American mathematician, electrical engineering, electrical engineer, and cryptography, cryptographer known as a "father of information theory". As a 21-year-o ...
in 1949. It is also commonly called Shannon's interpolation formula and Whittaker's interpolation formula. E. T. Whittaker, who published it in 1915, called it the Cardinal series.


Definition

Given a sequence of real numbers, ''x'' 'n'' the continuous function :x(t) = \sum_^ x \, \left(\frac\right)\, (where "sinc" denotes the
normalized sinc function In mathematics, physics and engineering, the sinc function, denoted by , has two forms, normalized and unnormalized.. In mathematics, the historical unnormalized sinc function is defined for by \operatornamex = \frac. Alternatively, the ...
) has a
Fourier transform A Fourier transform (FT) is a mathematical transform that decomposes functions into frequency components, which are represented by the output of the transform as a function of frequency. Most commonly functions of time or space are transformed, ...
, ''X''(''f''), whose non-zero values are confined to the region , ''f'',  ≤ 1/(2''T'').  When the parameter ''T'' has units of seconds, the bandlimit, 1/(2''T''), has units of cycles/sec (
hertz The hertz (symbol: Hz) is the unit of frequency in the International System of Units (SI), equivalent to one event (or cycle) per second. The hertz is an SI derived unit whose expression in terms of SI base units is s−1, meaning that on ...
). When the ''x'' 'n''sequence represents time samples, at interval ''T'', of a continuous function, the quantity ''f''''s'' = 1/''T'' is known as the
sample rate In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal. A common example is the conversion of a sound wave to a sequence of "samples". A sample is a value of the signal at a point in time and/or spa ...
, and ''f''''s''/2 is the corresponding
Nyquist frequency In signal processing, the Nyquist frequency (or folding frequency), named after Harry Nyquist, is a characteristic of a sampler, which converts a continuous function or signal into a discrete sequence. In units of cycles per second ( Hz), it ...
. When the sampled function has a bandlimit, ''B'', less than the Nyquist frequency, ''x''(''t'') is a perfect reconstruction of the original function. (See
Sampling theorem Sampling may refer to: *Sampling (signal processing), converting a continuous signal into a discrete signal * Sampling (graphics), converting continuous colors into discrete color components *Sampling (music), the reuse of a sound recording in ano ...
.) Otherwise, the frequency components above the Nyquist frequency "fold" into the sub-Nyquist region of ''X''(''f''), resulting in distortion. (See
Aliasing In signal processing and related disciplines, aliasing is an effect that causes different signals to become indistinguishable (or ''aliases'' of one another) when sampled. It also often refers to the distortion or artifact that results when a ...
.)


Equivalent formulation: convolution/lowpass filter

The interpolation formula is derived in the
Nyquist–Shannon sampling theorem The Nyquist–Shannon sampling theorem is a theorem in the field of signal processing which serves as a fundamental bridge between continuous-time signals and discrete-time signals. It establishes a sufficient condition for a sample rate that per ...
article, which points out that it can also be expressed as the
convolution In mathematics (in particular, functional analysis), convolution is a operation (mathematics), mathematical operation on two function (mathematics), functions ( and ) that produces a third function (f*g) that expresses how the shape of one is ...
of an infinite impulse train with a sinc function: : x(t) = \left( \sum_^ T\cdot \underbrace_\cdot \delta \left( t - nT \right) \right) \circledast \left( \frac\left(\frac\right) \right). This is equivalent to filtering the impulse train with an ideal (''brick-wall'')
low-pass filter A low-pass filter is a filter that passes signals with a frequency lower than a selected cutoff frequency and attenuates signals with frequencies higher than the cutoff frequency. The exact frequency response of the filter depends on the filter des ...
with gain of 1 (or 0 dB) in the passband. If the sample rate is sufficiently high, this means that the baseband image (the original signal before sampling) is passed unchanged and the other images are removed by the brick-wall filter.


Convergence

The interpolation formula always converges absolutely and locally uniformly as long as :\sum_\left, \fracn\<\infty. By the
Hölder inequality Hölder: * ''Hölder, Hoelder'' as surname * Hölder condition * Hölder's inequality * Hölder mean * Jordan–Hölder theorem In abstract algebra, a composition series provides a way to break up an algebraic structure, such as a group or a modul ...
this is satisfied if the sequence (x _ belongs to any of the \ell^p(\Z,\mathbb C) spaces with 1 ≤ ''p'' < ∞, that is :\sum_\left, x ^p<\infty. This condition is sufficient, but not necessary. For example, the sum will generally converge if the sample sequence comes from sampling almost any
stationary process In mathematics and statistics, a stationary process (or a strict/strictly stationary process or strong/strongly stationary process) is a stochastic process whose unconditional joint probability distribution does not change when shifted in time. Con ...
, in which case the sample sequence is not square summable, and is not in any \ell^p(\Z,\mathbb C) space.


Stationary random processes

If ''x'' 'n''is an infinite sequence of samples of a sample function of a wide-sense
stationary process In mathematics and statistics, a stationary process (or a strict/strictly stationary process or strong/strongly stationary process) is a stochastic process whose unconditional joint probability distribution does not change when shifted in time. Con ...
, then it is not a member of any \ell^p or Lp space, with probability 1; that is, the infinite sum of samples raised to a power ''p'' does not have a finite expected value. Nevertheless, the interpolation formula converges with probability 1. Convergence can readily be shown by computing the variances of truncated terms of the summation, and showing that the variance can be made arbitrarily small by choosing a sufficient number of terms. If the process mean is nonzero, then pairs of terms need to be considered to also show that the expected value of the truncated terms converges to zero. Since a random process does not have a Fourier transform, the condition under which the sum converges to the original function must also be different. A stationary random process does have an
autocorrelation function Autocorrelation, sometimes known as serial correlation in the discrete time case, is the correlation of a signal with a delayed copy of itself as a function of delay. Informally, it is the similarity between observations of a random variabl ...
and hence a
spectral density The power spectrum S_(f) of a time series x(t) describes the distribution of power into frequency components composing that signal. According to Fourier analysis, any physical signal can be decomposed into a number of discrete frequencies, ...
according to the
Wiener–Khinchin theorem In applied mathematics, the Wiener–Khinchin theorem or Wiener–Khintchine theorem, also known as the Wiener–Khinchin–Einstein theorem or the Khinchin–Kolmogorov theorem, states that the autocorrelation function of a wide-sense-stationary ...
. A suitable condition for convergence to a sample function from the process is that the spectral density of the process be zero at all frequencies equal to and above half the sample rate.


See also

*
Aliasing In signal processing and related disciplines, aliasing is an effect that causes different signals to become indistinguishable (or ''aliases'' of one another) when sampled. It also often refers to the distortion or artifact that results when a ...
,
Anti-aliasing filter An anti-aliasing filter (AAF) is a filter used before a signal sampler to restrict the bandwidth of a signal to satisfy the Nyquist–Shannon sampling theorem over the band of interest. Since the theorem states that unambiguous reconstruct ...
,
Spatial anti-aliasing In digital signal processing, spatial anti-aliasing is a technique for minimizing the distortion artifacts ( aliasing) when representing a high-resolution image at a lower resolution. Anti-aliasing is used in digital photography, computer graphi ...
*
Rectangular function The rectangular function (also known as the rectangle function, rect function, Pi function, Heaviside Pi function, gate function, unit pulse, or the normalized boxcar function) is defined as \operatorname(t) = \Pi(t) = \left\{\begin{array}{rl ...
*
Sampling (signal processing) In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal. A common example is the conversion of a sound wave to a sequence of "samples". A sample is a value of the signal at a point in time and/or sp ...
*
Signal (electronics) In signal processing, a signal is a function that conveys information about a phenomenon. Any quantity that can vary over space or time can be used as a signal to share messages between observers. The ''IEEE Transactions on Signal Processing'' ...
* Sinc function,
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. The filter's impulse response is a sinc functi ...
*
Lanczos resampling filtering and Lanczos resampling are two applications of a mathematical formula. It can be used as a low-pass filter or used to smoothly interpolate the value of a digital signal between its samples. In the latter case it maps each sample of ...
{{DEFAULTSORT:Whittaker-Shannon interpolation formula Digital signal processing Signal processing Fourier analysis E. T. Whittaker