The
Whittaker–Shannon interpolation formula or
sinc interpolation is a method to reconstruct a continuous-time
bandlimitedBandlimiting is the limiting of a deterministic or stochastic signal's Fourier transform or power spectral density to zero above a certain finite frequency...
signal from a set of equally spaced samples.
Definition
The
interpolation formula, as it is commonly called, dates back to the works of E. Borel in 1898, and
E. T. WhittakerEdmund Taylor Whittaker FRS FRSE was an English mathematician who contributed widely to applied mathematics, mathematical physics and the theory of special functions. He had a particular interest in numerical analysis, but also worked on celestial mechanics and the history of physics...
in 1915, and was cited from works of J. M. Whittaker in 1935, and in the formulation of the
Nyquist–Shannon sampling theoremThe 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. Sampling is the process of converting a signal into a numeric sequence...
by Claude Shannon 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.
The sampling theorem states that, under certain
limiting conditions, a function
x(
t) can be recovered exactly from its samples,
x[
n] =
x(
nT), by the Whittaker–Shannon interpolation formula
:
where
T = 1/
fs is the
sampling interval,
fs is the
sampling rateThe 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. For time-domain signals, the unit for sampling rate is hertz , sometimes noted as Sa/s...
, and sinc(
x) is the normalized
sinc function.
Validity condition
If the function
x(
t) is bandlimited, and sampled at a high enough rate, the interpolation formula is guaranteed to reconstruct it exactly. Formally, if there exists some
B ≥ 0 such that
- the function x(t) is bandlimited to bandwidth B; that is, it has a Fourier transform
In mathematics, Fourier analysis is a subject area which grew from the study of Fourier series. The subject began with the study of the way general functions may be represented by sums of simpler trigonometric functions...
for |f| > B; and
- the 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. For time-domain signals, the unit for sampling rate is hertz , sometimes noted as Sa/s...
, fs, exceeds the Nyquist rateIn signal processing, the Nyquist rate, named after Harry Nyquist, is two times the bandwidth of a bandlimited signal or a bandlimited channel...
, twice the bandwidth: fs > 2B. Equivalently:
-

then the interpolation formula will exactly reconstruct the original
x(
t) from its samples. Otherwise, aliasing may occur; that is, frequencies at or above
fs/2 may be erroneously reconstructed.
See AliasingIn signal processing and related disciplines, aliasing refers to an effect that causes different signals to become indistinguishable when sampled...
for further discussion on this point.
Interpolation as convolution sum
The interpolation formula is derived in the
Nyquist–Shannon sampling theoremThe 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. Sampling is the process of converting a signal into a numeric sequence...
article, which points out that it can also be expressed as the
convolutionIn mathematics and, in particular, functional analysis, convolution is a mathematical operation on two functions f and g, producing a third function that is typically viewed as a modified version of one of the original functions. Convolution is similar to cross-correlation...
of an
infinite impulse trainIn mathematics, a Dirac comb is a periodic Schwartz distribution constructed from Dirac delta functions...
with a sinc function
:
This is equivalent to filtering the impulse train with an ideal (
brick-wall)
low-pass filterA low-pass filter is an electronic filter that passes low-frequency signals but attenuates signals with frequencies higher than the cutoff frequency. The actual amount of attenuation for each frequency varies from filter to filter. It is sometimes called a high-cut filter, or treble cut filter...
.
Convergence
The interpolation formula always converges absolutely and
locally uniform as long as
By the Hölder inequality this is satisfied if the sequence

belongs to any of the
spacesIn mathematics, the Lp spaces are function spaces defined using a natural generalization of the p-norm for finite-dimensional vector 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 processIn the mathematical sciences, a stationary process is a stochastic process whose joint probability distribution does not change when shifted in time or space...
, 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 processIn the mathematical sciences, a stationary process is a stochastic process whose joint probability distribution does not change when shifted in time or space...
, then it is not a member of any

or
Lp spaceIn mathematics, the Lp spaces are function spaces defined using a natural generalization of the p-norm for finite-dimensional vector spaces...
, 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 densityIn statistical signal processing and physics, the spectral density, power spectral density , or energy spectral density , is a positive real function of a frequency variable associated with a stationary stochastic process, or a deterministic function of time, which has dimensions of power per hertz...
according to the
Wiener–Khinchin theoremThe Wiener–Khinchin theorem states that the power spectral density of a wide–sense stationary random process is the Fourier transform of the corresponding autocorrelation function.-History:Norbert Wiener first published the result in...
. 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 refers to an effect that causes different signals to become indistinguishable when sampled...
, Anti-aliasing filterAn anti-aliasing filter is a filter used before a signal sampler, to restrict the bandwidth of a signal to approximately satisfy the sampling theorem....
, Spatial anti-aliasing
- Fourier transform
In mathematics, Fourier analysis is a subject area which grew from the study of Fourier series. The subject began with the study of the way general functions may be represented by sums of simpler trigonometric functions...
- Rectangular function
- Sampling (signal processing)
In signal processing, sampling is the reduction of a continuous signal to a discrete signal. A common example is the conversion of a sound wave to a sequence of samples ....
- Signal (electronics)
- Sinc function, Sinc filter
In signal processing, a sinc filter is an idealized filter that removes all frequency components above a given bandwidth, leaves the low frequencies alone, and has linear phase...