The Whittaker–Shannon interpolation formula or sinc interpolation is a method to construct a
continuous-time
In mathematical dynamics, discrete time and continuous time are two alternative frameworks within which variables that evolve over time are modeled.
Discrete time
Discrete time views values of variables as occurring at distinct, separate "poi ...
bandlimited function from a sequence of real numbers. The formula dates back to the works of
E. Borel in 1898, and
E. T. Whittaker in 1915, and was cited from works of
J. M. Whittaker in 1935, and in the formulation of the
Nyquist–Shannon sampling theorem
The Nyquist–Shannon sampling theorem is an essential principle for digital signal processing linking the frequency range of a signal and the sample rate required to avoid a type of distortion called aliasing. The theorem states that the sample r ...
by
Claude Shannon
Claude Elwood Shannon (April 30, 1916 – February 24, 2001) was an American mathematician, electrical engineer, computer scientist, cryptographer and inventor known as the "father of information theory" and the man who laid the foundations of th ...
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''''x''(''nT''), the continuous function
:
(where "sinc" denotes the
normalized sinc function) has a
Fourier transform
In mathematics, the Fourier transform (FT) is an integral transform that takes a function as input then outputs another function that describes the extent to which various frequencies are present in the original function. The output of the tr ...
, ''X''(''f''), whose non-zero values are confined to the region :
. 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), often described as being equivalent to one event (or Cycle per second, cycle) per second. The hertz is an SI derived unit whose formal expression in ter ...
). 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 ...
, 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 Sampling (signal processing), sampler, which converts a continuous function or signal into a discrete sequence. For a given S ...
. 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
*Sample (graphics), Sampling (graphics), converting continuous colors into discrete color components
*Sampling (music), the reuse of a soun ...
.) 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 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 ori ...
.)
Equivalent formulation: convolution/lowpass filter
The interpolation formula is derived in the
Nyquist–Shannon sampling theorem
The Nyquist–Shannon sampling theorem is an essential principle for digital signal processing linking the frequency range of a signal and the sample rate required to avoid a type of distortion called aliasing. The theorem states that the sample r ...
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 f and g that produces a third function f*g, as the integral of the product of the two ...
of an
infinite impulse train with a
sinc function:
:
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 filt ...
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
:
By the
Hölder inequality this is satisfied if the sequence
belongs to any of the
spaces with 1 ≤ ''p'' < ∞, that is
:
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 which case the sample sequence is not square summable, and is not in any
space.
Stationary random processes
If ''x''
'n''is an infinite sequence of samples of a sample function of a wide-sense
stationary process, then it is not a member of any
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 and hence a
spectral density
In signal processing, the power spectrum S_(f) of a continuous time signal x(t) describes the distribution of power into frequency components f composing that signal. According to Fourier analysis, any physical signal can be decomposed into ...
according to the
Wiener–Khinchin theorem. 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 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 ori ...
,
Anti-aliasing filter,
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 graphics ...
*
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\left(\frac\right) = \Pi\left(\frac\ri ...
*
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 ...
*
Signal (electronics)
A signal is both the process and the result of transmission of data over some media accomplished by embedding some variation. Signals are important in multiple subject fields including signal processing, information theory and biology.
In ...
*
Sinc function,
Sinc filter
In signal processing, a sinc filter can refer to either a sinc-in-time filter whose impulse response is a sinc function and whose frequency response is rectangular, or to a sinc-in-frequency filter whose impulse response is rectangular and who ...
*
Lanczos resampling
Lanczos filtering and Lanczos resampling are two applications of a certain 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 ...
{{DEFAULTSORT:Whittaker-Shannon interpolation formula
Digital signal processing
Signal processing
Fourier analysis
E. T. Whittaker