Discrete-time Fourier transform

In mathematics, the discrete-time Fourier transform (DTFT), also called the finite Fourier transform, is a form of Fourier analysis that is applicable to a
sequence of values.

The DTFT is often used to analyze samples of a continuous function. The term discrete-time refers to the fact that the transform operates on discrete data, often
samples whose interval has units of time. From uniformly spaced samples it produces a function of frequency that is a periodic summation of the continuous
Fourier transform of the original continuous function. Under certain theoretical conditions, described by the sampling theorem, the original continuous function can
be recovered perfectly from the DTFT and thus from the original discrete samples. The DTFT itself is a continuous function of frequency, but discrete samples of
it can be readily calculated via the discrete Fourier transform (DFT) (see § Sampling the DTFT), which is by far the most common method of modern Fourier
analysis.

Both transforms are invertible. The inverse DTFT is the original sampled data sequence. The inverse DFT is a periodic summation of the original sequence. The
fast Fourier transform (FFT) is an algorithm for computing one cycle of the DFT, and its inverse produces one cycle of the inverse DFT.

Definition
The discrete-time Fourier transform of a discrete sequence of real or complex numbers x[n], for all integers n , is a Trigonometric series, which produces a periodic
function of a frequency variable. When the frequency variable, ω, has normalized units of radians/sample, the periodicity is 2π, and the DTFT series is:[1]: p .147 

(Eq.1)
 

   
 

The discrete-time Fourier transform is analogous to a Fourier series, except instead of starting with a periodic function of time and producing discrete sequence
over frequency, it starts with a discrete sequence in time and produces a periodic function in frequency. The utility of this frequency domain function is rooted in
the Poisson summation formula. Let X(f) be the Fourier transform of any function, x(t), whose samples at some interval T (seconds) are equal (or proportional) to
the x[n] sequence, i.e. T⋅x(nT) = x[n].[2]  Then the periodic function represented by the Fourier series is a periodic summation of X(f) in terms of frequency f in
hertz (cycles/sec):[a][A]

     

The integer k has units of cycles/sample, and 1/T is the sample-rate, fs (samples/sec). So
X1/T(f) comprises exact copies of X(f) that are shifted by multiples of fs hertz and combined
by addition. For sufficiently large fs the k = 0 term can be observed in the region
[−fs/2, fs/2] with little or no distortion (aliasing) from the other terms. In Fig.1, the
extremities of the distribution in the upper left corner are masked by aliasing in the periodic
summation (lower left).

We also note that e−i2πfTn is the Fourier transform of δ(t − nT). Therefore, an alternative
definition of DTFT is:[B]

Fig 1. Depiction of a Fourier transform (upper left) and its periodic

summation (DTFT) in the lower left corner. The lower right corner
depicts samples of the DTFT that are computed by a discrete
Fourier transform (DFT).

(Eq.3)
 

   
 

The modulated Dirac comb function is a mathematical abstraction sometimes referred to as impulse sampling.[4]

Inverse transform
An operation that recovers the discrete data sequence from the DTFT function is called an inverse DTFT. For instance, the inverse continuous Fourier transform of
both sides of Eq.3 produces the sequence in the form of a modulated Dirac comb function:

However, noting that X1/T(f) is periodic, all the necessary information is contained within any interval of length 1/T . In both Eq.1 and Eq.2, the summations over
n are a Fourier series, with coefficients x[n]. The standard formulas for the Fourier coefficients are also the inverse transforms:
  
 

 
 
(Eq

Periodic data
When the input data sequence x[n] is N-periodic, Eq.2 can be computationally reduced to a discrete Fourier transform (DFT), because:

All the available information is contained within N samples.

X1/T(f) converges to zero everywhere except at integer multiples of 1/(NT), known as harmonic frequencies. At those frequencies, the DTFT
diverges at different frequency-dependent rates. And those rates are given by the DFT of one cycle of the x[n] sequence.
The DTFT is periodic, so the maximum number of unique harmonic amplitudes is (1/T) / (1/(NT)) = N

The DFT coefficients are given by:

    and the DTFT is:

      [b]

Substituting this expression into the inverse transform formula confirms:

(all integers)

as expected. The inverse DFT in the line above is sometimes referred to as a Discrete Fourier series (DFS).[1]: p 542 

Sampling the DTFT

When the DTFT is continuous, a common practice is to compute an arbitrary number of samples (N) of one cycle of the periodic function
X1/T: [1]: p p 557–559 & 703 

where is a periodic summation:

    (see Discrete Fourier series)

The sequence is the inverse DFT. Thus, our sampling of the DTFT causes the inverse transform to become periodic. The array of | Xk |2 values is known as a
periodogram, and the parameter N is called NFFT in the Matlab function of the same name.[5]

In order to evaluate one cycle of numerically, we require a finite-length x[n] sequence. For instance, a long sequence might be truncated by a window
function of length L resulting in three cases worthy of special mention. For notational simplicity, consider the x[n] values below to represent the values modified
by the window function.

Case: Frequency decimation. L = N ⋅ I, for some integer I (typically 6 or 8)

A cycle of reduces to a summation of I segments of length N.  The DFT then goes by various names, such as:

window-presum FFT[6]
Weight, overlap, add (WOLA)[7][8][9][10][11][12][C][D]

polyphase DFT[10][11]
polyphase filter bank[13]
multiple block windowing and time-aliasing.[14]
Recall that decimation of sampled data in one domain (time or frequency) produces overlap (sometimes known as aliasing) in the other, and vice versa. Compared
to an L -length DFT, the summation/overlap causes decimation in frequency,[1]: p .558  leaving only DTFT samples least affected by spectral leakage. That is
usually a priority when implementing an FFT filter-bank (channelizer). With a conventional window function of length L , scalloping loss would be unacceptable.
So multi-block windows are created using FIR filter design tools.[15][16]  Their frequency profile is flat at the highest point and falls off quickly at the midpoint
between the remaining DTFT samples. The larger the value of parameter I, the better the potential performance.

Case: L = N+1 .

When a symmetric, L -length window function ( ) is truncated by 1 coefficient it is called periodic or DFT-even. The truncation affects the DTFT.  A DFT of the
truncated sequence samples the DTFT at frequency intervals of 1/N. To sample at the same frequencies, for comparison, the DFT is computed for one cycle of
the periodic summation, [E]

Case: Frequency interpolation. L ≤ N

In this case, the DFT simplifies to a more familiar form:

In order to take advantage of a fast Fourier transform algorithm for computing the DFT, the summation
is usually performed over all N terms, even though N − L of them are zeros. Therefore, the case
L < N is often referred to as zero-padding.
Spectral leakage, which increases as L decreases, is detrimental to certain important performance
metrics, such as resolution of multiple frequency components and the amount of noise measured by
each DTFT sample. But those things don't always matter, for instance when the x[n] sequence is a
Fig 2. DFT of ei2πn/8 for L = 64 and N = 256
noiseless sinusoid (or a constant), shaped by a window function. Then it is a common practice to use
zero-padding to graphically display and compare the detailed leakage patterns of window functions.
To illustrate that for a rectangular window, consider the sequence:

and

Figures 2 and 3 are plots of the magnitude of two different sized DFTs, as indicated in their labels. In
both cases, the dominant component is at the signal frequency: f = 1/8 = 0.125 . Also visible in Fig
2 is the spectral leakage pattern of the L = 64 rectangular window. The illusion in Fig 3 is a result of
sampling the DTFT at just its zero-crossings. Rather than the DTFT of a finite-length sequence, it
gives the impression of an infinitely long sinusoidal sequence. Contributing factors to the illusion are
the use of a rectangular window, and the choice of a frequency (1/8 = 8/64) with exactly 8 (an integer)
cycles per 64 samples. A Hann window would produce a similar result, except the peak would be
widened to 3 samples (see DFT-even Hann window (https://commons.wikimedia.org/wiki/File:DFT-e
ven_Hann_window_&_spectral_leakage.png)).
Fig 3. DFT of ei2πn/8 for L = 64 and N = 64
Convolution
The convolution theorem for sequences is:
[18]: p.297 [ c]

An important special case is the circular convolution of sequences x and y defined by where is a periodic summation. The discrete-frequency nature
of means that the product with the continuous function is also discrete, which results in considerable simplification of the inverse transform:

[19][1]: p.548 

For x and y sequences whose non-zero duration is less than or equal to N, a final simplification is:

The significance of this result is explained at Circular convolution and Fast convolution algorithms.

Symmetry properties
When the real and imaginary parts of a complex function are decomposed into their even and odd parts, there are four components, denoted below by the
subscripts RE, RO, IE, and IO. And there is a one-to-one mapping between the four components of a complex time function and the four components of its
complex frequency transform:[18]: p .291 

From this, various relationships are apparent, for example:

The transform of a real-valued function (x + x ) is the even symmetric function X + i X . Conversely, an even-symmetric transform
RE RO RE IO
implies a real-valued time-domain.
 The transform of an imaginary-valued function (i x + i x ) is the odd symmetric function X + i XIE, and the converse is true.
IE IO RO
The transform of an even-symmetric function (x + i x ) is the real-valued function X + X , and the converse is true.
RE IO RE RO
The transform of an odd-symmetric function (x + i xIE) is the imaginary-valued function i XIE+ i XIO, and the converse is true.
RO

Relationship to the Z-transform

is a Fourier series that can also be expressed in terms of the bilateral Z-transform.  I.e.:

where the notation distinguishes the Z-transform from the Fourier transform. Therefore, we can also express a portion of the Z-transform in terms of the Fourier
transform:

Note that when parameter T changes, the terms of remain a constant separation apart, and their width scales up or down. The terms of X1/T(f) remain
a constant width and their separation 1/T scales up or down.

Table of discrete-time Fourier transforms

Some common transform pairs are shown in the table below. The following notation applies:

is a real number representing continuous angular frequency (in radians per sample). ( is in cycles/sec, and is in sec/sample.) In
all cases in the table, the DTFT is 2π-periodic (in ).
designates a function defined on .
designates a function defined on , and zero elsewhere. Then:

is the Dirac delta function

is the normalized sinc function

is the triangle function

n is an integer representing the discrete-time domain (in samples)
is the discrete-time unit step function
is the Kronecker delta
 Time domain Frequency domain
X2π (ω) Remarks Reference
x[n]
[18]: p.305 

integer

   
integer
odd M

   

even M

The term must be interpreted as a

distribution in the sense of a Cauchy principal value
around its poles at .

[18]: p.305 

    -π < a < π

real number

real number with

real number with

integer and odd integer

real numbers with

real number ,

it works as a differentiator filter

real numbers with

Hilbert transform

real numbers
complex

Properties
This table shows some mathematical operations in the time domain and the corresponding effects in the frequency domain.

is the discrete convolution of two sequences

is the complex conjugate of x[n].
 Time domain Frequency domain
Property Remarks Reference
x[n]
Linearity complex numbers [18]: p.294 

Time reversal / Frequency reversal [18]: p.297 

Time conjugation [18]: p.291 

Time reversal & conjugation [18]: p.291 

Real part in time [18]: p.291 

Imaginary part in time [18]: p.291 

Real part in frequency [18]: p.291 

Imaginary part in frequency [18]: p.291 

Shift in time / Modulation in frequency integer k [18]: p.296 

Shift in frequency / Modulation in time real number [18]: p.300 

[F]
Decimation   integer

Time Expansion integer [1]: p.172 

Derivative in frequency [18]: p.303 

Integration in frequency

Differencing in time

Summation in time

Convolution in time / Multiplication in frequency [18]: p.297 

Multiplication in time / Convolution in frequency Periodic convolution [18]: p.302 

Cross correlation

Parseval's theorem [18]: p.302 

See also
Least-squares spectral analysis
Multidimensional transform
Zak transform

Notes
A. When the dependency on T is unimportant, a common practice is to replace it with Then f  has units of (cycles/sample), called normalized
frequency.
B. In fact Eq.2 is often justified as follows:[1]: p.143 

C. WOLA should not be confused with the Overlap-add method of piecewise convolution.
D. WOLA example: File:WOLA channelizer example.png
E. An example is figure Sampling the DTFT (https://upload.wikimedia.org/wikipedia/commons/9/91/Sampling_the_Discrete-time_Fourier_transfo
rm.svg). The real-valued DFT samples are a result of DFT-even symmetry[17]: p.52 
 F. This expression is derived as follows:[1]: p.168 

Page citations
a. Oppenheim and Schafer,[1] p 147 (4.20), p 694 (10.1), and Prandoni and Vetterli,[3] p 255, (9.33), where:      
and  
b. Oppenheim and Schafer,[1] p 551 (8.35), and Prandoni and Vetterli,[3] p 82, (4.43), where:        
and  

c. Oppenheim and Schafer,[1] p 60, (2.169), and Prandoni and Vetterli,[3] p 122, (5.21)

References
1. Oppenheim, Alan V.; Schafer, Ronald W.; Buck, John R. (1999). "4.2, 8.4". Discrete-time signal processing (https://archive.org/details/discreteti
mesign00alan) (2nd ed.). Upper Saddle River, N.J.: Prentice Hall. ISBN 0-13-754920-2. "samples of the Fourier transform of an aperiodic
sequence x[n] can be thought of as DFS coefficients of a periodic sequence obtained through summing periodic replicas of x[n]." 
2. Ahmed, N.; Rao, K.R. (July 10, 1975). Orthogonal Transforms for Digital Signal Processing (https://books.google.com/books?id=F-nvCAAAQ
BAJ) (1 ed.). Berlin Heidelberg New York: Springer-Verlag. doi:10.1007/978-3-642-45450-9 (https://doi.org/10.1007%2F978-3-642-45450-9).
ISBN 9783540065562.
3. Prandoni, Paolo; Vetterli, Martin (2008). Signal Processing for Communications (https://www.sp4comm.org/docs/sp4comm.pdf) (PDF) (1 ed.).
Boca Raton, FL: CRC Press. pp. 72, 76. ISBN 978-1-4200-7046-0. Retrieved 4 October 2020. "the DFS coefficients for the periodized signal
are a discrete set of values for its DTFT"
4. Rao, R. (2008). Signals and Systems (https://books.google.com/books?id=4z3BrI717sMC). Prentice-Hall Of India Pvt. Limited.
ISBN 9788120338593.
5. "Periodogram power spectral density estimate - MATLAB periodogram" (https://www.mathworks.com/help/signal/ref/periodogram.html).
6. Gumas, Charles Constantine (July 1997). "Window-presum FFT achieves high-dynamic range, resolution" (https://web.archive.org/web/2001
0210052902/http://www.chipcenter.com/dsp/DSP000315F1.html). Personal Engineering & Instrumentation News: 58–64. Archived from the
original on 2001-02-10.
7. Crochiere, R.E.; Rabiner, L.R. (1983). "7.2". Multirate Digital Signal Processing (https://kupdf.net/download/multirate-digital-signal-processing-
crochiere-rabiner_58a7065b6454a7e80bb1e993_pdf). Englewood Cliffs, NJ: Prentice-Hall. pp. 313–326. ISBN 0136051626.
8. Wang, Hong; Lu, Youxin; Wang, Xuegang (16 October 2006). "Channelized Receiver with WOLA Filterbank". 2006 CIE International
Conference on Radar. Shanghai, China: IEEE. pp. 1–3. doi:10.1109/ICR.2006.343463 (https://doi.org/10.1109%2FICR.2006.343463).
ISBN 0-7803-9582-4. S2CID 42688070 (https://api.semanticscholar.org/CorpusID:42688070).
9. Lyons, Richard G. (June 2008). "DSP Tricks: Building a practical spectrum analyzer" (https://www.embedded.com/design/real-time-and-perfor
mance/4007611/DSP-Tricks-Building-a-practical-spectrum-analyzer). EE Times. Retrieved 2020-02-20.   Note however, that it contains a link
labeled weighted overlap-add structure which incorrectly goes to Overlap-add method.
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dfs.semanticscholar.org/f3af/7acf516685ad04b3ebe19eb79db4dc3feb39.pdf) (PDF). Dallas: International Signal Processing Conference. p. 4
(fig 7). S2CID 31525301 (https://api.semanticscholar.org/CorpusID:31525301). Archived from the original (http://pdfs.semanticscholar.org/f3af/
7acf516685ad04b3ebe19eb79db4dc3feb39.pdf) (PDF) on 2019-03-08. Retrieved 2020-09-06. "The "Weight Overlap and Add" or WOLA or its
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11. Lillington, John. "A Review of Filter Bank Techniques - RF and Digital" (https://www.armms.org/media/uploads/16_lillington_a-review-of-filter-
bank-techniques--.pdf) (PDF). armms.org. Isle of Wight, UK: Libra Design Associates Ltd. p. 11. Retrieved 2020-09-06. "Fortunately, there is a
much more elegant solution, as shown in Figure 20 below, known as the Polyphase or WOLA (Weight, Overlap and Add) FFT."
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Galactic Center line survey" (http://hss.ulb.uni-bonn.de/2013/3251/3251.pdf) (PDF). hss.ulb.uni-bonn.de. Bonn: Rhenish Friedrich Wilhelms
University of Bonn. pp. 26–27. Retrieved 2020-09-06. "To perform M-fold WOLA for an N-point DFT, M·N real input samples aj first multiplied
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Public/windows.pdf) (PDF). Proceedings of the IEEE. 66 (1): 51–83. Bibcode:1978IEEEP..66...51H (https://ui.adsabs.harvard.edu/abs/1978IE
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 doi:10.1109/PROC.1978.10837 (https://doi.org/10.1109%2FPROC.1978.10837). S2CID 426548 (https://api.semanticscholar.org/CorpusID:42
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gitalsignalpro00proa) (3 ed.). New Jersey: Prentice-Hall International. Bibcode:1996dspp.book.....P (https://ui.adsabs.harvard.edu/abs/1996ds
pp.book.....P). ISBN 9780133942897. sAcfAQAAIAAJ.
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rabi). Englewood Cliffs, NJ: Prentice-Hall, Inc. p. 59 (2.163). ISBN 978-0139141010.

Further reading
Porat, Boaz (1996). A Course in Digital Signal Processing. John Wiley and Sons. pp. 27–29 and 104–105. ISBN 0-471-14961-6.
Siebert, William M. (1986). Circuits, Signals, and Systems. MIT Electrical Engineering and Computer Science Series. Cambridge, MA: MIT
Press. ISBN 0262690950.
Lyons, Richard G. (2010). Understanding Digital Signal Processing (3rd ed.). Prentice Hall. ISBN 978-0137027415.

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