Discrete-time Fourier transform

Last updated

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

Contents

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. In simpler terms, when you take the DTFT of regularly-spaced samples of a continuous signal, you get repeating (and possibly overlapping) copies of the signal's frequency spectrum, spaced at intervals corresponding to the sampling frequency. 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 reconstructs the original sampled data sequence, while the inverse DFT produces 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.

Introduction

Relation to Fourier Transform

Let be a continuous function in the time domain. We begin with a common definition of the continuous Fourier transform, where represents frequency in hertz and represents time in seconds:

We can reduce the integral into a summation by sampling at intervals of seconds (see Fourier transform § Numerical integration of a series of ordered pairs). Specifically, we can replace with a discrete sequence of its samples, , for integer values of , and replace the differential element with the sampling period . Thus, we obtain one formulation for the discrete-time Fourier transform (DTFT):

This Fourier series (in frequency) is a continuous periodic function, whose periodicity is the sampling frequency . The subscript distinguishes it from the continuous Fourier transform , and from the angular frequency form of the DTFT. The latter is obtained by defining an angular frequency variable, (which has normalized units of radians/sample), giving us a periodic function of angular frequency, with periodicity : [a]

  (Eq.1)
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). Fourier transform, Fourier series, DTFT, DFT.svg
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).

The utility of the DTFT is rooted in the Poisson summation formula, which tells us that the periodic function represented by the Fourier series is a periodic summation of the continuous Fourier transform: [b]

Poisson summation
  (Eq.2)

The components of the periodic summation are centered at integer values (denoted by ) of a normalized frequency (cycles per sample). Ordinary/physical frequency (cycles per second) is the product of and the sample-rate,   For sufficiently large the term can be observed in the region with little or no distortion (aliasing) from the other terms. Fig.1 depicts an example where is not large enough to prevent aliasing.

We also note that is the Fourier transform of Therefore, an alternative definition of DTFT is: [A]

  (Eq.3)

The modulated Dirac comb function is a mathematical abstraction sometimes referred to as impulse sampling. [3]

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 is periodic, all the necessary information is contained within any interval of length   In both Eq.1 and Eq.2 , the summations over are a Fourier series, with coefficients   The standard formulas for the Fourier coefficients are also the inverse transforms:

  (Eq.4)

Periodic data

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

The DFT of one cycle of the sequence is:

And can be expressed in terms of the inverse transform, which is sometimes referred to as a Discrete Fourier series (DFS): [1] :p 542

With these definitions, we can demonstrate the relationship between the DTFT and the DFT:

    [c] [B]

Due to the -periodicity of both functions of this can be simplified to:

which satisfies the inverse transform requirement:

Sampling the DTFT

When the DTFT is continuous, a common practice is to compute an arbitrary number of samples of one cycle of the periodic function :  [1] :pp 557–559 & 703  [2] :p 76

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 values is known as a periodogram , and the parameter is called NFFT in the Matlab function of the same name. [4]

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

Case: Frequency decimation. for some integer (typically 6 or 8)

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

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 -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 scalloping loss would be unacceptable. So multi-block windows are created using FIR filter design tools. [14] [15]   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 the better the potential performance.

Case:

When a symmetric, -length window function () is truncated by 1 coefficient it is called periodic or DFT-even. That is a common practice, but the truncation affects the DTFT (spectral leakage) by a small amount. It is at least of academic interest to characterize that effect.  An -length DFT of the truncated window produces frequency samples at intervals of instead of   The samples are real-valued, [16] :p.52  but their values do not exactly match the DTFT of the symmetric window. The periodic summation, along with an -length DFT, can also be used to sample the DTFT at intervals of   Those samples are also real-valued and do exactly match the DTFT (example: File:Sampling the Discrete-time Fourier transform.svg). To use the full symmetric window for spectral analysis at the spacing, one would combine the and data samples (by addition, because the symmetrical window weights them equally) and then apply the truncated symmetric window and the -length DFT.

Fig 2. DFT of e for L = 64 and N = 256 Zeropad.png
Fig 2. DFT of e for L = 64 and N = 256
Fig 3. DFT of e for L = 64 and N = 64 No-zeropad.png
Fig 3. DFT of e for L = 64 and N = 64

Case: Frequency interpolation.

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 terms, even though of them are zeros. Therefore, the case is often referred to as zero-padding.

Spectral leakage, which increases as 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 sequence is a 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: . Also visible in Fig 2 is the spectral leakage pattern of the 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).

Convolution

The convolution theorem for sequences is:

[17] :p.297 [d]

An important special case is the circular convolution of sequences s 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:

[18] [1] :p.548

For s 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: [17] :p.291

From this, various relationships are apparent, for example:

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 S1/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:

Time domain
s[n]
Frequency domain
S2π(ω)
RemarksReference
[17] :p.305
integer

   odd M
   even M

integer

The term must be interpreted as a distribution in the sense of a Cauchy principal value around its poles at .
[17] :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
Trapezoid signal.svg real numbers
complex

Properties

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

PropertyTime domain
s[n]
Frequency domain
RemarksReference
Linearitycomplex numbers [17] :p.294
Time reversal / Frequency reversal [17] :p.297
Time conjugation [17] :p.291
Time reversal & conjugation [17] :p.291
Real part in time [17] :p.291
Imaginary part in time [17] :p.291
Real part in frequency [17] :p.291
Imaginary part in frequency [17] :p.291
Shift in time / Modulation in frequencyinteger k [17] :p.296
Shift in frequency / Modulation in timereal number [17] :p.300
Decimation  [E] integer
Time Expansioninteger [1] :p.172
Derivative in frequency [17] :p.303
Integration in frequency
Differencing in time
Summation in time
Convolution in time / Multiplication in frequency [17] :p.297
Multiplication in time / Convolution in frequency Periodic convolution [17] :p.302
Cross correlation
Parseval's theorem [17] :p.302

See also

Notes

  1. In fact Eq.2 is often justified as follows: [1] :p.143,eq 4.6
  2. From § Table of discrete-time Fourier transforms we have:
  3. WOLA should not be confused with the Overlap-add method of piecewise convolution.
  4. WOLA example: File:WOLA channelizer example.png
  5. This expression is derived as follows: [1] :p.168

Page citations

  1. Oppenheim and Schafer, [1] p 147 (4.17), where:   therefore
  2. Oppenheim and Schafer, [1] p 147 (4.20), p 694 (10.1), and Prandoni and Vetterli, [2] p 255, (9.33), where:    and  
  3. Oppenheim and Schafer, [1] p 551 (8.35), and Prandoni and Vetterli, [2] p 82, (4.43). With definitions:     and   this expression differs from the references by a factor of because they lost it in going from the 3rd step to the 4th. Specifically, the DTFT of at § Table of discrete-time Fourier transforms has a factor that the references omitted.
  4. Oppenheim and Schafer, [1] p 60, (2.169), and Prandoni and Vetterli, [2] p 122, (5.21)

Related Research Articles

<span class="mw-page-title-main">Discrete Fourier transform</span> Type of Fourier transform in discrete mathematics

In mathematics, the discrete Fourier transform (DFT) converts a finite sequence of equally-spaced samples of a function into a same-length sequence of equally-spaced samples of the discrete-time Fourier transform (DTFT), which is a complex-valued function of frequency. The interval at which the DTFT is sampled is the reciprocal of the duration of the input sequence. An inverse DFT (IDFT) is a Fourier series, using the DTFT samples as coefficients of complex sinusoids at the corresponding DTFT frequencies. It has the same sample-values as the original input sequence. The DFT is therefore said to be a frequency domain representation of the original input sequence. If the original sequence spans all the non-zero values of a function, its DTFT is continuous, and the DFT provides discrete samples of one cycle. If the original sequence is one cycle of a periodic function, the DFT provides all the non-zero values of one DTFT cycle.

<span class="mw-page-title-main">Fourier analysis</span> Branch of mathematics

In mathematics, Fourier analysis is the study of the way general functions may be represented or approximated by sums of simpler trigonometric functions. Fourier analysis grew from the study of Fourier series, and is named after Joseph Fourier, who showed that representing a function as a sum of trigonometric functions greatly simplifies the study of heat transfer.

<span class="mw-page-title-main">Fourier transform</span> Mathematical transform that expresses a function of time as a function of frequency

In physics, engineering and mathematics, the Fourier transform (FT) is an integral transform that takes a function as input and outputs another function that describes the extent to which various frequencies are present in the original function. The output of the transform is a complex-valued function of frequency. The term Fourier transform refers to both this complex-valued function and the mathematical operation. When a distinction needs to be made, the output of the operation is sometimes called the frequency domain representation of the original function. The Fourier transform is analogous to decomposing the sound of a musical chord into the intensities of its constituent pitches.

In mathematics, the convolution theorem states that under suitable conditions the Fourier transform of a convolution of two functions is the product of their Fourier transforms. More generally, convolution in one domain equals point-wise multiplication in the other domain. Other versions of the convolution theorem are applicable to various Fourier-related transforms.

<span class="mw-page-title-main">Fourier series</span> Decomposition of periodic functions into sums of simpler sinusoidal forms

A Fourier series is an expansion of a periodic function into a sum of trigonometric functions. The Fourier series is an example of a trigonometric series, but not all trigonometric series are Fourier series. By expressing a function as a sum of sines and cosines, many problems involving the function become easier to analyze because trigonometric functions are well understood. For example, Fourier series were first used by Joseph Fourier to find solutions to the heat equation. This application is possible because the derivatives of trigonometric functions fall into simple patterns. Fourier series cannot be used to approximate arbitrary functions, because most functions have infinitely many terms in their Fourier series, and the series do not always converge. Well-behaved functions, for example smooth functions, have Fourier series that converge to the original function. The coefficients of the Fourier series are determined by integrals of the function multiplied by trigonometric functions, described in Common forms of the Fourier series below.

In mathematics and signal processing, the Z-transform converts a discrete-time signal, which is a sequence of real or complex numbers, into a complex valued frequency-domain representation.

In mathematics, the Gibbs phenomenon is the oscillatory behavior of the Fourier series of a piecewise continuously differentiable periodic function around a jump discontinuity. The th partial Fourier series of the function produces large peaks around the jump which overshoot and undershoot the function values. As more sinusoids are used, this approximation error approaches a limit of about 9% of the jump, though the infinite Fourier series sum does eventually converge almost everywhere except points of discontinuity.

<span class="mw-page-title-main">Short-time Fourier transform</span> Fourier-related transform suited to signals that change rather quickly in time

The short-time Fourier transform (STFT) is a Fourier-related transform used to determine the sinusoidal frequency and phase content of local sections of a signal as it changes over time. In practice, the procedure for computing STFTs is to divide a longer time signal into shorter segments of equal length and then compute the Fourier transform separately on each shorter segment. This reveals the Fourier spectrum on each shorter segment. One then usually plots the changing spectra as a function of time, known as a spectrogram or waterfall plot, such as commonly used in software defined radio (SDR) based spectrum displays. Full bandwidth displays covering the whole range of an SDR commonly use fast Fourier transforms (FFTs) with 2^24 points on desktop computers.

In signal processing, a finite impulse response (FIR) filter is a filter whose impulse response is of finite duration, because it settles to zero in finite time. This is in contrast to infinite impulse response (IIR) filters, which may have internal feedback and may continue to respond indefinitely.

In signal processing, a periodogram is an estimate of the spectral density of a signal. The term was coined by Arthur Schuster in 1898. Today, the periodogram is a component of more sophisticated methods. It is the most common tool for examining the amplitude vs frequency characteristics of FIR filters and window functions. FFT spectrum analyzers are also implemented as a time-sequence of periodograms.

In mathematics, the Poisson summation formula is an equation that relates the Fourier series coefficients of the periodic summation of a function to values of the function's continuous Fourier transform. Consequently, the periodic summation of a function is completely defined by discrete samples of the original function's Fourier transform. And conversely, the periodic summation of a function's Fourier transform is completely defined by discrete samples of the original function. The Poisson summation formula was discovered by Siméon Denis Poisson and is sometimes called Poisson resummation.

In mathematics and signal processing, the Hilbert transform is a specific singular integral that takes a function, u(t) of a real variable and produces another function of a real variable H(u)(t). The Hilbert transform is given by the Cauchy principal value of the convolution with the function (see § Definition). The Hilbert transform has a particularly simple representation in the frequency domain: It imparts a phase shift of ±90° (π/2 radians) to every frequency component of a function, the sign of the shift depending on the sign of the frequency (see § Relationship with the Fourier transform). The Hilbert transform is important in signal processing, where it is a component of the analytic representation of a real-valued signal u(t). The Hilbert transform was first introduced by David Hilbert in this setting, to solve a special case of the Riemann–Hilbert problem for analytic functions.

In mathematics, Parseval's theorem usually refers to the result that the Fourier transform is unitary; loosely, that the sum of the square of a function is equal to the sum of the square of its transform. It originates from a 1799 theorem about series by Marc-Antoine Parseval, which was later applied to the Fourier series. It is also known as Rayleigh's energy theorem, or Rayleigh's identity, after John William Strutt, Lord Rayleigh.

<span class="mw-page-title-main">Dirac comb</span> Periodic distribution ("function") of "point-mass" Dirac delta sampling

In mathematics, a Dirac comb is a periodic function with the formula for some given period . Here t is a real variable and the sum extends over all integers k. The Dirac delta function and the Dirac comb are tempered distributions. The graph of the function resembles a comb, hence its name and the use of the comb-like Cyrillic letter sha (Ш) to denote the function.

In digital signal processing, downsampling, compression, and decimation are terms associated with the process of resampling in a multi-rate digital signal processing system. Both downsampling and decimation can be synonymous with compression, or they can describe an entire process of bandwidth reduction (filtering) and sample-rate reduction. When the process is performed on a sequence of samples of a signal or a continuous function, it produces an approximation of the sequence that would have been obtained by sampling the signal at a lower rate.

<span class="mw-page-title-main">Hann function</span> Mathematical function used in signal processing

The Hann function is named after the Austrian meteorologist Julius von Hann. It is a window function used to perform Hann smoothing. The function, with length and amplitude is given by:

<span class="mw-page-title-main">Gabor transform</span> Special case of the short-time Fourier transform

The Gabor transform, named after Dennis Gabor, is a special case of the short-time Fourier transform. It is used to determine the sinusoidal frequency and phase content of local sections of a signal as it changes over time. The function to be transformed is first multiplied by a Gaussian function, which can be regarded as a window function, and the resulting function is then transformed with a Fourier transform to derive the time-frequency analysis. The window function means that the signal near the time being analyzed will have higher weight. The Gabor transform of a signal x(t) is defined by this formula:

In digital signal processing, a discrete Fourier series (DFS) is a Fourier series whose sinusoidal components are functions of discrete time instead of continuous time. A specific example is the inverse discrete Fourier transform.

<span class="mw-page-title-main">Dirichlet kernel</span> Concept in mathematical analysis

In mathematical analysis, the Dirichlet kernel, named after the German mathematician Peter Gustav Lejeune Dirichlet, is the collection of periodic functions defined as

In mathematical analysis and applications, multidimensional transforms are used to analyze the frequency content of signals in a domain of two or more dimensions.

References

  1. 1 2 3 4 5 6 7 8 9 10 11 Oppenheim, Alan V.; Schafer, Ronald W.; Buck, John R. (1999). "4.2, 8.4". Discrete-time signal processing (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. 1 2 3 4 Prandoni, Paolo; Vetterli, Martin (2008). Signal Processing for Communications (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
  3. Rao, R. (2008). Signals and Systems. Prentice-Hall Of India Pvt. Limited. ISBN   9788120338593.
  4. "Periodogram power spectral density estimate - MATLAB periodogram".
  5. Gumas, Charles Constantine (July 1997). "Window-presum FFT achieves high-dynamic range, resolution". Personal Engineering & Instrumentation News: 58–64. Archived from the original on 2001-02-10.{{cite journal}}: CS1 maint: bot: original URL status unknown (link)
  6. Crochiere, R.E.; Rabiner, L.R. (1983). "7.2". Multirate Digital Signal Processing. Englewood Cliffs, NJ: Prentice-Hall. pp. 313–326. ISBN   0136051626.
  7. 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. ISBN   0-7803-9582-4. S2CID   42688070.
  8. Lyons, Richard G. (June 2008). "DSP Tricks: Building a practical spectrum analyzer". EE Times. Retrieved 2024-09-19.  Note however, that it contains a link labeled weighted overlap-add structure which incorrectly goes to Overlap-add method.
  9. 1 2 Lillington, John (March 2003). "Comparison of Wideband Channelisation Architectures" (PDF). Dallas: International Signal Processing Conference. p. 4 (fig 7). S2CID   31525301. Archived from the original (PDF) on 2019-03-08. Retrieved 2020-09-06. The "Weight Overlap and Add" or WOLA or its subset the "Polyphase DFT", is becoming more established and is certainly very efficient where large, high quality filter banks are required.
  10. 1 2 Lillington, John. "A Review of Filter Bank Techniques - RF and Digital" (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.
  11. Hochgürtel, Stefan (2013), "2.5", Efficient implementations of high-resolution wideband FFT-spectrometers and their application to an APEX Galactic Center line survey (PDF), Bonn: Rhenish Friedrich Wilhelms University of Bonn, pp. 26–31, retrieved 2024-09-19, To perform M-fold WOLA for an N-point DFT, M·N real input samples aj first multiplied by a window function wj of same size
  12. Chennamangalam, Jayanth (2016-10-18). "The Polyphase Filter Bank Technique". CASPER Group. Retrieved 2016-10-30.
  13. Dahl, Jason F. (2003-02-06). Time Aliasing Methods of Spectrum Estimation (Ph.D.). Brigham Young University. Retrieved 2016-10-31.
  14. Lin, Yuan-Pei; Vaidyanathan, P.P. (June 1998). "A Kaiser Window Approach for the Design of Prototype Filters of Cosine Modulated Filterbanks" (PDF). IEEE Signal Processing Letters. 5 (6): 132–134. Bibcode:1998ISPL....5..132L. doi:10.1109/97.681427. S2CID   18159105 . Retrieved 2017-03-16.
  15. Harris, Frederic J. (2004-05-24). "9". Multirate Signal Processing for Communication Systems. Upper Saddle River, NJ: Prentice Hall PTR. pp. 226–253. ISBN   0131465112.
  16. Harris, Fredric J. (Jan 1978). "On the use of Windows for Harmonic Analysis with the Discrete Fourier Transform" (PDF). Proceedings of the IEEE. 66 (1): 51–83. Bibcode:1978IEEEP..66...51H. CiteSeerX   10.1.1.649.9880 . doi:10.1109/PROC.1978.10837. S2CID   426548.
  17. 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 Proakis, John G.; Manolakis, Dimitri G. (1996). Digital Signal Processing: Principles, Algorithms and Applications (3 ed.). New Jersey: Prentice-Hall International. Bibcode:1996dspp.book.....P. ISBN   9780133942897. sAcfAQAAIAAJ.
  18. Rabiner, Lawrence R.; Gold, Bernard (1975). Theory and application of digital signal processing . 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.