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An Introduction
to FFT andTime
Domain Windows
Part 11 in a series of tutorials in
instrumentation and measurement
Sergio Rapuano and Fred J. Harris

A
student in a digital signal processing class
once asked an insightful set of questions.
They went something like this: “Professor,
can you explain why we spend so much of our time
describing signals in the frequency domain? We first ex-
amined circuits in terms of their sinusoidal steady-state
response. We then went on to study Fouri-
er series and Fourier transforms, and
we are now studying sampled
data Fourier transforms, and
the Discrete Fourier trans-
form, all based on sinusoids
and sampled data sinu-
soids. Why not use other
basis functions? What is so
special about sinusoids?”
Great questions! We wish
we had thought to ask them!
The answers are simple.
Many of the dynamic systems we
analyze, synthesize, design, develop,
and operate can be approximated by linear
time invariant systems modeled by linear, constant-
coefficient, differential equations. So now the question
is, “What does that mean?” It means this: if we dif-
ferentiate a sinusoid, it is still a sinusoid. If we form a
weighted sum of derivatives of a sinusoid, the sum is
still a sinusoid. The sinusoid never stops being a sinu-
soid. No other wave shape can make that claim! The
sinusoids preserve their identity in a linear system; the
system can change the sinusoid’s amplitude and phase
but it cannot change its basic structure. Sinusoids are
eigenfunctions of linear, constant-coefficient, differ-
ential equations. As such, the sinusoids can be used to
analyze and characterize the linear system.
The collection of amplitude and phase
changes experienced by sinusoids
of different frequencies passing
through the system compactly
describes the system. We call
this description its frequency
response. The frequency re-
sponse is intimately tied to
the system’s transfer function
and its differential equation.
The Fourier transform is also
used to describe signals in the time
or spatial domains. We limit our dis-
cussion here to time domain descriptions.
Signals of interest are decomposed into a set of
complete orthonormal basis functions, the real sines and
cosines, or, equivalently, the set of complex exponentials
[3]. The classic Fourier transform is the mechanism that
performs this decomposition, leading to a frequency
domain description of the signal. There are several ad-
vantages to analyzing signals in the frequency domain.

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Probably the most evi-
dent is the relevant in-
crease in dynamic range
What is so special about
in comparison with time sinusoids?…The sinusoid
domain methods. Frequency
domain analysis can resolve: never stops being a
measure and detect the ampli-
tude and phase of overlapped
sinusoid. No other
signals the amplitudes of which differ waveshape can
by orders of magnitude. This powerful
characteristic makes frequency domain make that
measurements important in many fields,
including telecommunication, instrumenta-
claim!
tion, radar, sonar, consumer entertainment, and
other systems requiring signal analysis, signal
detection, modulation, and demodulation.
The Fourier theory had originally been formu-
lated for continuous time and amplitude signals, i.e.,
analog signals and systems. With the advent of the digi-
tal computer, in particular of the microprocessor, spectrum
analysis could be performed in the digital domain along
with many of the other signal processing tasks performed
by digital techniques. The sampling theorem was well es-
tablished prior to the arrival of the digital computer, and the
missing components required to perform signal processing
on sampled data signals were the transducers to pass sig-
nals back and forth between the continuous domain and the
sampled data domain. The need for high-performance, low-
cost analog-to-digital converters (ADCs) and their duals,
the digital-to-analog converters, was recognized early on,
and their development closely paralleled that of the micro-
processor. It is interesting to note that the first applications
of sampled data signal processing occurred in the sampled
data control area because the required sample rates were
low, as were the bandwidths of the mechanical systems being
controlled. The next application was that of processing audio
signals, also with a relatively low bandwidth matched to
the then-improved performance capabilities of early ADCs.
This evolution is seen in the publication history, with the first
venue for signal processing articles being the IEEE Journal on
Audio and Electroacoustics (1965), which morphed into Audio,
Speech, and Signal Processing (1984), on its way to becoming
the Transactions on Signal Processing (1991).
The mathematical tools were ready and waiting as the
enabling technology and applications came together in the
early 1960s. When digital data (the sampled and quantized
representation of an analog signal) were delivered by ADCs to
the digital domain, they found a rich body of signal processing
options ready to manipulate and extract signal parameters.
The Z-transform had already been developed as the sampled
data counterpart to the Laplace transform, and the discrete
Fourier transform (DFT) had been developed as a counterpart
to the Fourier transform (FT). One modification to the DFT was
required to perform machine computation of the sampled data
spectrum. The DFT is defined as a sum over a two-sided inter-
val from −∞ to +∞. In practice we often change the lower limit of
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the sum to 0 to acknowl-
edge causality and to
obtain the one-sided DFT
as a sum from 0 to +∞. The
practical consideration is that
we can never perform an infi-
nite sum in a computer. We have to
limit the range of the sum to a finite
interval, say, 0 to N − 1, a sum contain-
ing N terms. This little detail essentially
turns the data collection process on at in-
dex 0 and then turns it off at index N − 1. This
gating operation means that the sum is always
performed over finite data collection apertures,
with boundary conditions equivalent to an abrupt
turning on and off of the collected data. Turning data
abruptly on and off has an undesired influence on the
spectrum of the collected signal samples. We ameliorate
this effect using a multiplicative weighting term applied to
the data in the collection interval, which slowly and gently
turns the data on and off at the boundaries. This is a common
operation in many signal processing systems facing finite ap-
ertures. In time-series signal processing we call the weighting
function a window, in spatial processing (beam forming) we
call it a shading function, and in photolithography we call it an
apodizing function.
The DFT is implemented in digital systems by a family of
algorithms collectively known as the fast Fourier transform
(FFT). The FFT offers a significant reduction in computational
workload relative to the DFT. The DFT requires on the order
of N2 complex operations (multiplies and adds), while the
FFT can be implemented with workloads between 2 × N and
log2(N) × N/2 complex operations. Some information on FFT
history can be found in [4]. The FFT is widely applied in digital
signal processing–based systems. [5] These include modula-
tion and demodulation applications in telecommunication
systems, with examples being Orthogonal Frequency Domain
Multiplexing (OFDM) and Asymmetric Digital Subscriber
Lines (ADSL) and measurement and instrumentation systems,
the emphasis of this paper. Processing a finite aperture obser-
vation introduces several artefacts into the spectral analysis
process. One important artefact is spectral leakage, the spill-
ing of energy centered at one frequency into the surrounding
spectral regions. This effect limits our ability to reliably detect
low-level signal in the presence of nearby high-level signals.
Windows are designed and applied to suppress this artefact.
A second attribute brought to bear by the finite aperture is the
uncertainty principle. The finite aperture limits spectral re-
solvability, the ability to detect closely spaced similar-strength
tones as individual signal components [3]. What we will learn
here is that when we apply a window to the signal to control
biases due to spectral leakage, we increase the width of the
spectral main lobe, which causes secondary effects related to
spectral resolution (separating nearby signals), processing
gain (separating signals from noise), and window overlap (sat-
isfying Nyquist). We will examine and highlight these effects.
 The paper is intended to give the basic elements of FFT-
based spectrum analysis, starting with the properties of the
DFT and introducing considerations related to its fast imple-
mentation. We examine the default window, the rectangle,
and its contribution to spectral leakage, as well as designed
windows and how they can be used to obtain accurate spec-
tral estimates. Finally, we discuss and compare the main
characteristics of the windows used in spectrum and network
measurements along with information on their secondary
attributes.
The Discrete Fourier Transform
The mathematic tool for analyzing signals and systems
in the frequency domain is the FT. When applied to a real
or complex valued analog signal x a(t), it produces its as-
sociated frequency domain representation, Xa(f), called its
spectrum. The basic idea behind spectrum analysis is that,
subject to easy-to-satisfy restrictions, a signal can be formed
as a weighted sum of complex exponential functions (called
spectral components). The weighting terms at each frequency
are the complex amplitude and phase. Spectral analysis is the
process by which we estimate the amplitude and phase of the
components located at each frequency. For this reason, Xa(f)
is a complex signal, the magnitude of which represents the
sinusoid’s amplitudes versus frequency, which we denote
the magnitude spectrum of xa(t), and whose phase represents
the sinusoid phases versus frequency, which we denote the
phase spectrum of xa(t).
An aperiodic signal has frequency domain representation
defined by the FT integral shown in Equation 1, as follows:
(1)
with an inverse transform integral as shown in Equation 2:
(2)
When xa(t) is periodic in T0 s, the spectral components are
harmonics located at multiples of the fundamental frequency,
1/T0 Hz. The signal xa(t) can be represented, in the mean square
sense, with a countable sum of sine waves by the Fourier Series
(FS) [1] integral shown in Equation 3. The resulting magnitude
and phase spectra reside only at the integer multiples of the
frequency 1/T0, or k/T0.
(3)
with an inverse transform sum as shown in Equation 4:
(4)
Note that within a scale factor, the FS (Equation 3) of a
periodic signal is a sampled FT (Equation 1) of a single cycle
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of the periodic signal, as is shown in Equation 5. What we
realize here is that sampling in the frequency domain induces
periodicity in the time domain. The converse, of course, is also
true: sampling in the time domain induces periodicity in the
frequency domain and results in the sampled data Fourier
transform (SDFT):
(5)
The FT and the FS are applicable to analog signals. That
means that FT and FS describe continuous time signals and
have continuous and discrete spectra, respectively, with
arbitrary amplitude and phase profiles. All mathematics
associated with the Fourier theory is applicable to digital sys-
tems. The SDFT is defined for sequences of arbitrary length
and is the counterpart of the inverse FS. By the application of
appropriate windows, the SDFT can process finite length se-
quences. The SDFT is shown in Equation 6. Here the variable
θ is digital frequency with units of radians/sample, bounded
by the interval −p < q ≤ p, which reflects the mapping of
sampled data spectra to the unit circle; it reflects as well the
Nyquist criterion for the sampling process. Since the empha-
sis here is finite sums, we also present the finite sum version
of Equation 6 in Equation 7. Note that the SDFT is continuous
and periodic in 2p, even though the time series is a sampled
data series:
(6)
(7)
The inverse SDFT (of Equation 6 or Equation 7) is shown in
Equation 8 and is seen to be the sampled data counterpart of
Equation 3:
(8)
Again we return to the reality that we can only perform
finite sums with a digital machine. By similar reasoning, we
can only compute the spectrum shown in Equation 7 at a finite
number of frequencies. We elect to sample the spectrum of
Equation 7 at multiples of the digital frequency 2p/N, where
N is the number of samples in the original sum. This is suffi-
ciently dense sampling to satisfy the Nyquist theorem for this
sampling process. This sampling is shown in Equation 9. The
sampling process converts the SDFT to the DFT.

(9)
The inverse DFT, which is the finite sum replacement of
Equation 8, is shown in Equation 10.
(10)
An important observation is that X(k) is a periodic set of
frequency samples due to the original time domain sampling
process. The positive frequencies are located at indices 0 ≤ k ≤
N/2, and since index N is congruent to index 0, the negative
frequencies −1 ≥ k ≥ −N/2 are located at N − 1 ≥ k ≥ N/2. Simi-
larly, x(n) has become a periodic set of time domain samples,
periodic in N samples, due to the sampling of the spectral
function.
Note that quantization of sampled data sequence has no
bearing on the properties of the SDFT. The quantization noise
can be attributed to source coding noise and is treated as addi-
tive noise, which is seen as one of the noise terms in the spectra
when the spectra is computed by finite precision arithmetic
processors.
Summarizing this section, we can say that
◗ Sampling an analog signal at frequency FS induces peri-
odicity of its spectrum with period FS or with normalized
period 2p.
◗ The DFT forms equally spaced samples of the periodic
sampled data spectrum.
◗ Sampling the periodic spectrum of the input sequence
at multiples of FS/N induces periodicity of the input se-
quence with period N.
◗ The samples in the frequency domain are equally spaced
at multiples of FS/N; therefore, spectral values differ-
ent from kFS/N are not available from the N-length
transform. The frequency interval FS/N defines the
transform frequency resolution, and the frequencies
kFS/N are denoted frequency bins.
◗ Additional frequency resolution can be
obtained for a sequence of length N by
performing a 2N or 4N-point transform The
on the zero-extended versions of the main
sequence.
advantage
Applications of the DFT
By using the DFT and its properties, it of the DFT
is possible to do the following: over the FT is that
◗ Compute the spectrum of a
sampled data signal. it permits machine
◗ Perform circular con-
volution of two sam- computation of spectra.
pled signals as the
N-point IDFT of
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the spectral product of their N-point DFTs. One of the
sequences may have to be zero extended to length N.
◗ Perform linear convolution of two sampled signals as the
2N-point IDFT of the spectral product of their 2N-point
DFTs. Both N-point sequences must be zero extended to
length 2N.
◗ Determine the frequency response of a sampled data
system as the ratio of the N-point DFTs of the windowed
output and input series.
◗ Synthesize the time-domain sampled waveform from its
frequency domain representation.
The main advantage of the DFT over the FT is that it permits
machine computation of spectra.
The DFT is a vector process, converting input time vectors
of length N into output frequency vectors of length N. One
matter of concern in applying the DFT is the high computa-
tional burden of N2 complex operations to convert the input
vector to the output vector. A second matter of concern is that
the spectral sampling inherent in the DFT describes the peri-
odic extension of the input signal, which may produce spectral
artefacts as a result of the artificial boundaries.
The first concern has been put to rest by the development of
efficient FFT algorithms, which require significantly reduced
computational resources to perform the transform. The second
concern is handily addressed by the use of windows to sup-
press the boundary conditions, which in turns suppresses the
spectral leakage artefacts.
Efficient Calculation of DFT,
the FFT Algorithm
The DFT is seen to be a set of projections of the input time
series onto the basis vectors W nkN, where WN is the Nth root of
unity, . The projections are performed as N-point inner
products, with each projection requiring N complex multiply
and add operations. Thus, N2 complex multiplications are re-
quired for the direct computation of the DFT. FFT algorithms
exist for any composite length N DFT. By composite we mean
N is factorable into a product of factors: i.e., N = N1 × N2. For
instance, consider the factors of N = 800 with one possible
factoring of 32 and 25.
Many FFT algorithms operate by the divide-and-
conquer method. In this process, the transform is
partitioned into a sequence of reduced-length
transforms that are collectively performed with
reduced workload. For instance, a data vector
of length N = N1 × N2 is mapped to a two-di-
mensional intermediate array of dimen-
sion N1-rows by N2-columns. We per-
form a two-dimensional transform
by N1 transforms over the row vec-
tors of length N2, which requires
N1N22 operations followed by
N2 transforms over the col-
umn vectors of length N1,
which requires N 2 N 12
operations. Depend-
Fig. 1. Log magnitude plots of the FFT of complex and of real noisy sine wave at 1.7 kHz sampled at 10 kHz with spectral cuts at Fs and at Fs  /2.

ing on the details of the one-dimensional to two-dimensional
mapping, there may be a set of phase rotations, called twiddle
factors, applied to the output of the first set of transforms prior
to application of the second set. The total workload for the
two-dimensional transform is seen to be N1N22 + N2N12 = (N1N2)
(N1 + N2), which, simply stated, is the sum of the factors times
the product of the factors. The workload for the original, un-
factored DFT is N2 = (N1N2)(N1N2). The ratio of the workloads
is seen to be the sum of the factors divided by the product of
the factors, R = [(N1N2)(N1 + N2)/(N1N2)(N1N2)] or R = (N1 +
N2)/(N1N2). With the factors for N = 1,000 of 32 and 25, this
ratio is seen to be (32 + 25)/(32 × 25) = 0.071. Here we see that
the factored form requires only 7.1% of the workload of the
unfactored form, a savings of nearly 93%. The factoring pro-
cess is nested, with the 25-point transforms performed as two-
dimensional transforms over 5-by-5 arrays and the 32-point
transforms performed as 2-D transforms over 4 × 8 arrays.
A common-length FFT is known as the radix-2 DFT, which
performs successive halving of the length N DFT, with N being
a power of 2, such as 210 = 1,024. The partition described above
starts with a two-dimensional array of length N/2 by 2, which
results in a savings ratio of (512 + 2)/(512 × 2) = 0.502, a sav-
ings of nearly 50%. Each successive iterated partition results
in successive 50% workload savings. The radix-2 transform
requires twiddle factors between successive partitions. The
iterative partitioning operation can be repeated until the proc-
ess performs 512 two-point transforms, which are combined to
obtain 256 four-point transforms, which in turn are combined
to obtain 128 eight-point transforms, continuing until two
512-point transforms are combined to form the 1,024-point
transform [1]. Each successive combining operation requires
N/2 complex multiplications. The number of combining steps
is equal to the number of times the array length can be parti-
tioned 1 to 2 (or halved), which is log2(N). Consequently, the
total number of complex multiplications required to calculate
a radix-2 FFT is N/2 log2(N).
To give an idea of the computational efficiency of the FFT
algorithm over the direct DFT calculation, consider the DFT
of length 1,024. The savings ratio is [N/2 log2(N)]/[N2] or
log2(N)/(2N) = 10/2,048 or 0.0049, a savings of 99.5%. The
actual numbers are 5,120 operations for the FFT, compared to
1,048,576 operations for the direct DFT computation.
The output of an FFT is a vector of length N, obtained by
sampling the periodic spectrum of the sampled data signal at
the frequencies kFs/N. It is common to call these spectral lo-
cations frequency bins. The periodic spectrum is visualized on

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 Display,
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Fig. 2. Top; analog 2 kHz sine wave and spectrum; middle: 3 ms rectangular window and spectrum; bottom: windowed sine wave and spectrum.

a circle of circumference Fs or in normalized units of 2pf/Fs. As
we unwrap the circle to map it to a line, we traditionally cut
the circle at angle 0 so that the line extends over the interval
[0 to 2p), closed on the left, open on the right. The spectrum
obtained by this cut has frequency 0 at the far left, positive
frequencies extending to the right until the mid-point, at
which point we have the half sample rate, beyond which we
have the negative frequencies. The FFT computes samples of
the spectrum at increments of 2p/N, indexing the samples
from 0 to N − 1. Sample N is congruent to sample 0 and is, in
fact, the start of the next spectral period. Thus, sample −k is
the same as sample N − k and the frequencies with negative
index; the negative frequencies are located in the second half
of the interval. It is common in spectral displays to redefine
the cut on the circle to be located at angle p to redefine the
interval as (−p to p). In this form the center of the display is 0
frequency, with negative frequencies located to the left and
positive frequencies located to the right. MATLAB uses the
command fftshift to redefine the spectral cut. The result of
FFT calculation is a double-sided spectrum mapped to the
frequency bins from 0 to N − 1. In the redefined spectral cut
the indices are mapped to –N/2 to +N/2 − 1, or in normal-
ized coordinates, (−0.5 to +0.5). For complex signals the DFT
is asymmetrical and the entire span (−0.5 to +0.5) is pre-
sented, and for real input signals, the DFT is Hermetian sym-
metric, so a reduced span of (0 to +0.5) is presented, since the
positive frequencies, the first N/2 + 1 samples, are sufficient
to represent the signal spectrum. Figure 1 presents the log
magnitude spectrum formed by an FFT of a complex and of a
real 1.7 kHz sine wave with AWGN (Additive White Gauss-
ian Noise) sampled at 10 kHz. Here we see the spectra with
and without the redefined spectral cuts. As can be seen, the
spectrum of the complex signal is asymmetric and the spec-
trum of the real signal is symmetric about both Fs/2 and 0.
Spectral Leakage
As described earlier, the DFT computes samples of the pe-
riodic spectrum X(q) associated with the N-point sampled
data sequence {x(n)}. Sampling in the time domain causes
replication (or periodic extension) of the spectrum, with rep-

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licas positioned at multiples of 1/T, the reciprocal of spacing
between time samples (T =1/Fs). Similarly, sampling in the
frequency domain causes replication (or periodic extension)
of the time series, with replicas positioned at multiples of NT,
the reciprocal of the spacing between the frequency samples
(Fs/N =1/NT). This property can be illustrated by computing
the N-point DFT of a sampled sequence that contains an inte-
ger number of cycles in the sequence of length N. A complex
sinusoidal sequence containing an integer number of cycles in
N samples is defined in Equation 11.
interval), indicating a single sinusoid in the time series.
The infinite extent sinusoid has a FS at a single frequency
k/(NT). By including distribution functions in the FT, we rep-
resent the spectrum of a complex sine wave by a delta function
δ(f − k/NT). When the complex sinusoid is time limited to a
finite support of length NT seconds, its spectrum is modified.
The altered spectrum is obtained by convolving the transform
of the infinite extent version, δ(f − k/NT), with the transform of
the truncating function, WR(f), defined as follows:

(12)

This function is called rectangular window of length NT,

and its FT WR(f) is the ubiquitous function sin(pt/NT)/(pt/
NT), commonly denoted sinc(t/NT) [1]. Figure 2 shows a
segment of the input sinusoid xa(t), the window wR(t), and
(11)
the windowed signal xa(t) × WR(t), as well as their spectra
For this sequence the periodic extension of the N-sample se- Xa(f), WR(f), and Xa(f)*WR(f), where the symbol * denotes a
quence is the same as the samples obtained by sampling the convolution.
original sinusoid, the original signal. The DFT of the sequence The same results will be observed in the sample data
has a non-zero spectral component at index k (for k cycles per domain. The process of acquiring a finite-length record of
a signal can be modeled as
sampling and analog-to-digi-
tal conversion of an infinite
number of samples followed
by the multiplication of the re-
sulting infinite sequence {x(n)}
by a discrete window function
{w(n)}, the values of which are
non-zero within a specified
span of N samples. If, in the
example considered above, the
(a)
sampling period T = 1/Fs and
the acquisition time interval Ts,
and the signal period, T0, are
chosen to satisfy the relation

(13)
the FFT will have the appear-
ance shown in Figure 3. In this
(b) example, the sampled signal is
a 2 kHz sine wave with a period
equal to 0.5 ms that is sampled
at a 20 kHz rate. The sampling
period is 50 μs and covers a 3
ms time interval spanning six
cycles of the sine wave to col-
lect a total of N = 60 samples.
The FFT values will be zero
for all frequency bins except
(c) for the one corresponding to 2
kHz, because a sample is taken
Fig. 3. Top: sampled and windowed 2 kHz sine wave; middle: mid: single-sided spectrum of signal; bottom: DFT
at the peak of the main lobe of
samples of sampled sine wave. Xa(f) *WR(f), while all the others

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 Display,
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are taken at the zeros of the same

spectrum. Note that in this case,
with the sinusoid containing
an integer number of cycles per
interval, there is no uncertainty
in estimating the sine wave fre-
quency and amplitude.
Almost assuredly, we will not
see signals in which all the spec-
tral components have an integer
(a)
number of cycles per interval of
length N. More likely, the periods
T 0 are arbitrary and unknown
and there isn’t an integer k that
satisfies the condition (Equation
11) for a given N and a selected
sampling period T. In these cases
the frequency components cor-
responding to the FFT bins will
not correctly represent the am- (b)
plitude and spectral position of
the underlying spectral peaks
and further will observe the side
lobe spectral terms related to the
offset spectral peaks, which spill
spectral components over all the
frequency bins.
For example, let us consider
the case of the sine wave xa(t) and
suppose that T 0 is not known. (c)
Let us suppose that NT = 6.5 T0, Fig. 4. Top: sampled and windowed 2.2-kHz sine wave; middle, single-sided spectrum of signal; bottom: DFT
as shown in Figure 4. The peri- samples of sampled sine wave. Note samples of spectral side lobes.
odically extended version of this
sequence will have a discontinu-
ity at each N sample not present in the original non-windowed the spectra of the high-level signals could mask or cover
sequence. The FT of this sequence is the same shifted sinc the spectra of the low-level components with its side lobes.
function as in the previous example, but the zero crossings no Closely spaced spectral components may have overlapping
longer correspond to the DFT sample positions. Consequently, main lobes, which may reduce our ability to resolve them as
the DFT will now compute samples of the side lobe rather than separate components.
samples of the zero crossings. This phenomenon is called spec- Spectral leakage of window spectral side lobes is respon-
tral leakage. As with the continuous FT, the original spectrum is sible for (i) errors in estimating the frequency and amplitude
the convolution of Xa(f) and WR(f), with the sample positions of the frequency components from the FFT bins and (ii) a
fixed but the zero crossings translated with the center of limitation of system dynamic range as a result of the mask-
the sinc. Note that the sample values in the main lobe do ing affect of strong components on weak components.
not coincide with the peak of the main lobe. Using this Spectral width of the window’s spectral main lobe is
result to estimate the signal parameters will lead to responsible for the minimum separation required
an error in frequency and amplitude. A typical peak between two frequency components of similar
detection algorithm will find a sine wave with amplitude to assure reliable detection of distinct
frequency 2 kHz and magnitude 0.35, instead A signal components. This separation, called
of 1.1 kHz and 0.50, respectively. minimum resolution bandwidth, also reduces
The errors in measuring the signal fre-
common- the capability of detecting weak frequency
quency and amplitude are related to the length FFT components near strong components.
frequency resolution and the main lobe Spectral leakage cannot be avoided
width. If the signal is composed of
is known as the when the signal components are un-
many frequency components with radix-2 DFT…. known. Proper design trades be-
wide variation in amplitudes, tween spectral resolvability and

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 the sine wave. The known
bias in amplitude due to the
reduced-amplitude main
lobe of the Hann window
relative to the rectangle
window is easily removed.
The amplitude of each win-
dow is simply the sum of
the window coefficients
and can be properly scaled
out after the measurement.
The residual amplitude er-
ror resulting from position
of the center lobe relative to
the DFT fixed sample posi-
tions is known as scalloping
loss. This loss is smaller
than the loss obtained with
the rectangle window.
This loss can also be to-
tally removed by passing
a second-order polynomial
through the three maxi-
mum-amplitude log-mag-
nitude samples in the main
lobe and by determining
the peak position and am-
plitude of the parabola.
This works amazingly well
because in log amplitude,
all spectral windows are
approximately parabolas.
Fig. 5. Effects of the Hann window. Top: windowed sine wave; middle: single-sided spectrum of windowed signal; bottom: To reiterate, we comment
DFT samples of windowed and sampled sine wave. that good windows reduce
the side lobe levels, thus
detection capability are required to reduce errors in amplitude improving the detectability of weak frequency components,
and frequency estimates. and the errors made by estimating each component frequency
and amplitude can be easily eliminated by polynomial in-
Windows terpolation. An alternate interpolation option is obtained by
Instead of the rectangular window, other window functions simply zero extending the length-N–windowed time series to
are generally adopted to control the limitations listed above. length 2N or 4N and performing the increased length DFT.
All well-designed windows have the characteristic of going In Figure 6, we show examples of the time and frequency re-
gently and smoothly to near-zero values at the boundaries sponses of windows most often used in digital spectrum analysis
of their support intervals. Consequently, the discontinuities [1], [6]. The choice of the window may vary for a specific applica-
at the boundaries of the periodically extended sequence are tion, as each has different effects on the FFT output. Generally, a
suppressed as the amplitudes and many order derivatives window with a very narrow main lobe will have a high spectral
are matched at the boundaries. As illustrated in Figure 5, the resolvability and a lower uncertainty in measuring the frequency
window modifies the input signal by smoothly and gently of a spectral component. In most cases, a narrow main lobe implies
bringing the signal envelope to near-zero values at the bound- high side lobes causing low detectability of weak spectral compo-
aries of the acquisition interval. The result will be a controlled nents. In addition, a narrow main lobe will cause a commensu-
insertion of leakage. rate uncertainty in the measurement of the spectral component
Examining the DFT samples shown in Figure 5 we see that amplitudes as a result of high scallop loss. For example, the Hann
the samples far removed from the main lobe are quite small window is the window of choice in applications requiring high
and will reduce the masking effects due to high side lobes. We resolvability. As a result of its narrow main lobe, the frequency
also see the multiple samples of the main lobe, which reduces resolution is maximized and the frequency measurement uncer-
our ability to estimate the center frequency and amplitude of tainty is minimized. However, because of the higher side lobes,

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 Display,
actuators,
Sensor Input Processing Output signals,
control

Fig. 6. Time and spectral response of common window functions: rectangular, Hann, Kaiser-Bessel, and Harris Flat-Top. The width of the rectangular window
main lobe is overlapped to the other window spectra as dashed red lines to easily compare them.

the detectability of low-level nearby spectral terms is reduced in characteristics of the window frequency response. In [3], [7], and
comparison with other windows. The uncertainty of the ampli- [8], a complete description of them is given in mathematical and
tude measurement is greater than for windows designed for low practical terms. Here some parameters are recalled and briefly
scallop loss, such as the harris Flat Top window. As a result of the introduced in order to give some indications on how to interpret
wide flat spectral response, the Flat Top has small scallop loss and them for finding the right trade-off for a given application.
hence exhibits small amplitude errors, but the same flatness leads
to significant frequency uncertainty, which must be resolved by Minimum Resolution Bandwidth
one of the spectral interpolation options. As described above, as a result of the convolution of the sig-
The Flat Top window, on the other hand, has a flat but very nal and the window spectra, the windowed signal spectrum
wide main lobe and lower side lobes. The Kaiser-Bessel win- includes a replica of the window frequency response located
dow, with a time-bandwidth parameter, β, offers a selectable at each frequency component of the signal, resulting in an
trade between side lobe levels and main lobe width. Some
digital signal processing systems offer the option to select a
window from a set to give the user more degrees of freedom
in the analysis. In all cases the user should be aware of the ef-
fect of the windows on the measurement results. A complete
introduction to the windowing techniques as well as a review
of more window characteristics can be found in [3].

Windows Characterization
There are several figures of merit used for classifying the win- Fig. 7. Overlapped components that cannot be distinguished (one peak) and
dows used in digital spectrum analysis. All parameters refer to that can be distinguished (two peaks) [3].

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 of spectral decay is also
1/f but has even lower side
lobes because of its even-
smaller discontinuity.

Processing Gain/Processing
Loss
The amplitude estimation
of a frequency component
is affected by the broad-
band noise passed by the
bandwidth of its spectral
main lobe. In this sense, the
window behaves as a filter,
gathering contributions for
its estimate over its band-
width [3]. Remember, to re-
duce side lobes we increase
main lobe width, which
permits more noise into the
spectral measurement and
at the same time reduces
the amplitude of the main
Fig. 8. Scalloping loss is 3.9 dB for a rectangular window and 1.2 dB for a Kaiser-Bessel window measuring a spectral lobe, which reduces the
peak at position 4.5 cycles per interval, midway between bins 4 and 5.
amplitude of the desired
sine wave measurement.
overlap of the main lobes corresponding to the nearest compo- The window reduces the signal-to-noise ratio (SNR) relative
nents. As a consequence, even if the frequency resolution of the to the SNR of the default rectangle window. A measure of how
DFT is very high, it isn’t possible to resolve near components much SNR improvement obtained from a windowed FFT can
with similar amplitude whose distance is lower than the width be found as the ratio between the SNR before and after the
of the window main lobe (Figure 7). To this aim, the width of calculation, called Processing Gain (PG):
each lobe is defined to be the frequency interval correspond-
ing to a 6-dB attenuation to the DC gain and is called minimum
Resolution Bandwidth (RBW) [3], [6]. As indicated in [3], the
minimum RBW goes from 1.2 to 2.6 bins, depending on the
specific window. (14)

Side Lobes
The detectability of weak components is mainly affected by
two characteristics of the window frequency response: the
highest side lobe level and the rate of side lobe roll-off. The lev-
el of the highest side lobe is given as a ratio in dB normalized
to the main lobe level. The maximum side lobe level usually
falls inversely with the main lobe width. The side lobe roll-off
is usually given in terms of the frequency response decrease in
dB per frequency octave or decade. This rate is1/f (m+1), where m
is the order of the derivative of the window envelope in which
the first discontinuity resides. For instance, a rectangle is dis-
continuous in the zero-th derivative, and its rate of spectral
decay is 1/f 1 or 6 dB/octave, and a Hann window is discon-
tinuous in the second derivative and its rate of spectral decay
is 1/f 3 [1], [3]. Note that the Hann window has a discontinuity
in its zero-th derivative, and its rate of spectral decay is also
1/f, but it has lower side lobes than the rectangle because it has
a smaller discontinuity. Similarly, the Kaiser-Bessel window
also has a discontinuity in its zero-th derivative, and its rate
where, So/No is the output SNR, Si/Ni is the input SNR, and w(k)
is the window sample. The PG of a rectangle window is N (sum
squared/sum of squares), the PG of a Hann window is N/1.5
and the PG for a 60-dB Kaiser Window is N/1.7. Usually the PG is
normalized to the PG of a rectangle of the same length N. Interest-
ingly, the reciprocal of the PG is the equivalent noise bandwidth
ENBW, the width of a spectral rectangle of unit amplitude that
passes the same noise power as the window being described.
Thus, the ENBW of a rectangle is 1/N and of a Hann window is
1.5/N. The Hann window’s equivalent filter passes half again
as much noise as does the rectangle window’s equivalent filter.
Incidentally, the reduction in SNR or PG due to use of a good
window is exactly cancelled by the variance reduction obtained
when averaging overlapped windowed transforms [8], [9].
Scalloping Loss
When the frequency being analyzed by the DFT is bin-cen-
tered, the DFT sample coincides with the peak of the window’s
main lobe response. When the frequency is positioned offset

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Fig. 9. Time domain and spectra of windows with −90-dB side lobes and main lobe width of Fs /N.
from the bin center, the DFT sample is offset on the main
lobe, and the measurement is less than the peak value. The
worst-case reduction in measurement value occurs when the
signal frequency resides midway between two bin centers.
An example would be a sine wave with 4.5 cycles per interval
of length N, its spectral peak residing midway between bins 4
and 5. Here both adjacent bins offer a reduced-level sample of
the offset main lobe spectral response. Windows with wider
main lobe width have smaller loss as a result of the frequency
offset. As seen in Figure 8, the amplitude error, called scalloping
loss, depends on the window’s main lobe bandwidth. Usually
the scalloping loss is reported in dB and goes from 0.1 to 3.92,
depending on the window [3].
Misconceptions
Common misconceptions to keep in mind when estimating the
spectrum of a signal with an FFT are the following:
◗ The transform length N does not have to be a power of 2.
FFTs come in all sizes, and any (non-prime) length FFT is
available. A 500-point transform is as efficient as a 512 -
point transform. Transforms with lengths that are powers
of 2 have very simple coding structure, which influences
hardware implementations but has little bearing on soft-
ware-based spectral estimation.
December 2007 IEEE Instrumentation & Measurement Magazine 43
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Display,
actuators,
Sensor Input Processing Output signals,
control
◗ A paraphrase of the Nyquist criterion tells us that the
sample rate should exceed the two-sided bandwidth of
the signal. For real baseband signals this is interpreted as
such: the sample rate must be at least twice the highest
frequency component in the signal. This is a very restric-
tive interpretation and should not limit your options. The
transform of a real signal exhibits Hermetian symmetry,
H(k) = H*(−k). As such, without loss of information, the
FFT can be computed and displayed for positive frequen-
cies only. Complex signals, on the other hand, formed, for
instance, by a quadrature downconversion of a span of
positive frequencies, does not exhibit spectral symmetry.
As such, the FFT must be computed and displayed for
both positive and negative frequencies. Bear in mind that
the analog anti-aliasing filters have a transition band-
width, and not all spectral components are alias free. It is
typical, for instance, to allow the aliased transition band
to corrupt 10–20% of the spectrum so that a 1,024-point
FFT can present 400 spectral lines for real signals, or 800
lines for complex signals [2].
◗ A number of pre- and post-processing algorithms aid
the FFT in the spectral estimation task. We discuss one
here to catch the reader’s interest. The spectral resolu-
tion of an FFT is initially defined by the spacing between
 spectral bins Fs/N. A rectangle window has a main lobe [7] F.J. Harris, Multirate Signal Processing for Communication Systems,
spectral width of F s/N, which matches this spacing. Englewood Cliffs, NJ: Prentice Hall, 2004.
When a good window is applied to the data to suppress [8] D. Elliot, “Time domain signal processing with the DFT,” Chapter
the spectral side lobes, the main lobe width increases 8 in Handbook of Digital Signal Processing; Engineering Applications,”
by a factor of 4 to 4Fs/N. This increased width reduces Orlando, FL: Academic Press, 1987.
the spectral resolution by the same factor. A response to [9] F.J. Harris, “On detecting white space spectra for spectral
this reduced resolution is to increase the data record and scavenging in cognitive radios,” in Proc. Wireless Personal
window length from N to 4N and to thus return the reso- Multimedia Communications, 2007, Jaipur, India, publication in Dec.
lution to 4Fs/(4N). This appears to also increase the FFT 2007.
length, which we don’t want to do. We keep the same- [10] F.J. Harris, “Spectral analysis windowing,” in Wiley
length FFT by computing every fourth transform point, Encyclopedia of Electrical and Electronics Engineering, vol. 20,
which matches the original spectra spacing of Fs/N. The J.G. Webster, ed., New York: John Wiley & Sons, Inc., 1999, pp.
4-to-1 spectral downsampling induces a fourfold time 88–105.
domain aliasing of the length 4N windowed signal to the [11] F.J. Harris, “On Overlapped Fast Fourier Transforms”, Int.
N length series that is processed by the N-point trans- Telemetering Conf. (ITC-78), Los Angeles, 1978, 301-306.
form. The folding of windowed signals is performed by
a pre-processor, prior to the FFT, known as a polyphase Sergio Rapuano (rapuano@
partition [7]–[10]. The combination of folded window unisannio.it) received the M.S.
and FFT is referred to as a polyphase channelizer. Figure degree in electronic engineering
9 illustrates the morphing of the window’s spectral main and the Ph.D. degree in com-
lobe width from Fs/N to 4Fs/N and then back to 4Fs/(4N) puter science, telecommunica-
and 6Fs/(6N). tions, and applied electromag-
netism from the University of
Conclusions Salerno. Since 2002, he has been
This paper includes a brief tutorial on digital spectrum analy- with the faculty of engineering
sis and FFT-related issues to form spectral estimates on digi- at the University of Sannio as
tized signals. Some review of the DFT has been presented, and an assistant professor in electric
some discussion on the computational advantages of the FFT and electronic measurement. Dr. Rapuano is a member of the
calculation has also been presented. Finally, the main consid- IEEE I&M Society TC-10 and the secretary of the TC-23 Work-
erations on windowing and window characteristics have been ing Group on “e-tools for Education in Instrumentation and
briefly discussed. Measurement.” He is currently developing his research activi-
ties in the fields of data converters, distributed measurement
References systems, and digital signal processing for measurement and
[1] A.V. Oppenheim and R.W. Schafer, “Digital Signal Processing,” medical measurements.
Englewood Cliffs, NJ: Prentice Hall, 1975.
[2] S. Rapuano, P. Daponte, E. Balestrieri, L. De Vito, S.J. Tilden, S. Fredric J. Harris (fred.harris@
Max, and J. Blair, “ADC parameters and characteristics,” IEEE sdsu.edu) is at San Diego State
Instrument. Meas. Mag., vol. 8, (no. 5), pp. 44–54, Dec 2005. University, where he teaches
[3] F. J. Harris, “On the use of Windows for harmonic analysis with courses in “Digital Signal Pro-
the Discrete Fourier Transform,” Proc. IEEE, vol. 66, (no. 1), pp. cessing and Communication
51–83, Jan 1978. Systems.” He is a fellow of the
[4] M.T. Heideman, D.H. Johnson, and C.S. Burrus, “Gauss and the IEEE and author of the text
history of the fast fourier transform”, IEEE ASSP Magazine, Vol. 1, Multirate Signal Processing for
(no.4), part.1, pp. 14-21, Oct. 1984. Communication Systems (Pren-
[5] E. Brigham, Fast Fourier Transform and Its Applications, Englewood tice-Hall). He roams the world
Cliffs, NJ: Prentice Hall, 1988. collecting old toys and slide
[6] R.A. Witte, Spectrum and Network Measurements, Englewood Cliffs, rules and riding old railways.
NJ: Prentice Hall, 1993.

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