FFT Analyzer I

1. What is FFT?
Fourier transform was discovered by a French mathematician and scientist, Fourier, as a natural progression
of his Fourier series theory. The Fourier series theory states that any waveform, however complicated, can
be expressed as a series of two or more simple sine waves and cosine waves, if the waveform is periodical,
that is, composed of the same repeated waveforms. The mathematical expression of this theory is called
Fourier series. Fourier transform involves expansion of the Fourier series from - to + and developed.
It is not always clear the extent to which a signal can be actually measured and determined as periodical,
especially when measuring the waveform up to infinity, as it becomes fainter and fainter. Thus, in general,
only a part of the observed waveform is cut out and Fourier transform performed on this period of the
waveform assuming that the waveform pattern is infinitely repeated. Originally, this Fourier transform
calculation required a tremendous number of multiplication calculations. However, J.W. Cooley and J.W.
Tukey proposed a method of calculation that reduces the number of individual calculations by taking the
number of data items as equal to 2n. If the number of data items is assumed to be 1024, the number of
multiplications, that is 1024 x 1024 = 1048576, is reduced to 10240. This method was called Fast Fourier
Transform and is often referred to by its acronym, FFT.

1-1 FFT Analyzer

An FFT calculation tangibly indicates how the coefficients of a Fourier series (Fourier coefficients) are
determined. An FFT analyzer stores an input signal waveform as data by digitally (discretely) sampling it,
determines the Fourier coefficients in a short time using FFT, and displays the results of this analysis. Since
FFT basically involves resolving a signal into simple frequencies by expressing the value of frequency
components (spectrum), the FFT analyzer is also called a frequency analyzer or spectrum analyzer.
For example, an analysis of the sound "A"(pronounced as "aah") using an FFT analyzer produces the
spectral waveforms in Figure 1-1, which indicate the sound waveform's frequency f along the X-axis and
amplitude r along the Y-axis. These spectral waveforms indicate that the sound of "A" is composed of waves
whose frequencies are f1, f2, f3.....,fn, and their amplitudes are r1,r2,r3,......,rn. Stated another way, this graph
represents a combination of waves whose frequencies are f1,f2,f3......,fn, and their amplitudes are
r1,r2,r3,......,rn,which compose together to form the sound waveform of "A".
Figure 1-2 shows the time-base waveforms actually measured and their spectrum of sound "A". The lower
graph shows the time-base waveforms and the upper graph indicates the spectral waveforms. Frequencies
appearing as peaks in the left part of the spectral waveforms correspond to f1,f2,f3,....,fn.
Now let's consider FFT for more tangible examples.

Figure 1-2
1-2 Why is FFT Necessary?
Let's observe the waveforms of vibration generated from an actual machine. Install an acceleration pickup
on a bearing as shown in Figure 1-3 and observe the vibration waveforms obtained from the acceleration
pickup. Composite time-axis waveforms can be observed as in the case of sound "A".

Figure 1-3
Why use an FFT analyzer for frequency analysis (i.e., observe the frequency of waveforms)?
The composite waveform shown in Figure 1-3 can be regarded as a combined waveform obtained from the
vibrations at each position of the machine (see figure 1-4).

Figure 1-5 is a conceptual diagram showing the relationship between the result of the analysis of the
composite vibration waveforms from this rotary machine obtained using an FFT analyzer and the positions
of the sources of vibration.

Figure 1-5
The frequencies corresponding to the vibration generated from each position are determined from the
construction of the machine. In the past, when maintaining and controlling facilities and diagnosing
anomalies, a measure of the entire vibration was obtained, that is, an overall value was measured with a
vibration meter. However, as an overall value only discriminates whether the vibration is strong or weak, the
actual location of the anomaly could not be identified.
By observing a waveform frequently using an oscilloscope (in the time domain), we could see the waveform
change over time (time-axis waveforms) but it was difficult to determine the cause of that change. It was not
possible to estimate the cause of the anomaly and the abnormal positions until frequency analysis data
obtained using FFT enabled us to investigate the degree of change that occurred at what level of frequency
and the frequencies generated from a particular position. In the early stages of failure or for slight anomalies,
the overall value of time-axis waveforms shows very little change, making the failure/anomaly very difficult
to detect. Frequency analysis (observing the waveform in the frequency domain) makes detection of even
slight anomalies possible.
In addition to facility control and failure diagnosis using vibration analyses, frequency analysis is now used
in various other fields, such as for the evaluation of quietness and noise analysis of office equipment and
home electric appliances, to investigate the cause of noise and possible countermeasures.

2.Representation of Waveforms (Sine wave and Cosine wave)

To understand the meaning of the data given by an FFT analyzer, it is necessary to understand the concept of
Fourier series -- the fundamentals of FFT and its mathematical background. This chapter explains the
representation of sound as waveforms, which is necessary for understanding the Fourier series and the
characteristics of the sine wave and cosine wave.

2-1 Amplitude, Phase and Frequency

A waveform can be represented with three different parameters: amplitude, frequency (or period) and phase
(or time-difference).
<Amplitude>
The amplitude shows the magnitude of the waveform. Taking sound as an example, a loud sound has a large
amplitude.
Different physical phenomena are detected using various sensors. Vibration is measured using a vibration
sensor, sound with a sound level meter, force with a load cell, and pressure with a pressure sensor. The
signal is output as a voltage amplitude value proportional to the value of the physical quantity.
The time-axis waveform indicated in an FFT analyzer or measured with an oscilloscope or pen recorder, is
the observation of the voltage output over time.

Figure 2-1
<Frequency>
The frequency represents the number of wave periods repeated in one second and is defined in units of Hz.
The following relationship applies, where frequency is represented by f and the period by T.
(Equation 2-1)

In respect to sound, it is recognized that high frequency waveforms correspond to high tones and low
frequency waveforms to low tones.

Figure 2-2

<Phase>

One period of a wave is represented as an electrical angle of 360 or 2p radians. "Degree" is used for
explanation purposes, but in mathematical expressions, "radian" is generally used and expressed simply as
"rad". The position of a peak, if measured at a particular point in time, may differ even between waves of the
same frequency. The difference in the peak position of a given waveform from the reference waveform is
usually expressed as a quantity called a "phase difference." The phase is considered "negative" if the
particular peak lags behind the corresponding peak in the reference waveform, or "positive" if it comes after
the corresponding peak.

Figure 2-3

2-2 Locus of Waves

Let's look at the motion of a ball when it is attached to a string and rotates around the origin of the string.
The locus of the ball is shown in Figure 2-4.

Figure 2-4

In Figure 2-4, the waveform obtained by the locus of the shadow of the ball projected on the Y-axis
represents a sine wave and the waveform obtained by the locus of the shadow of the ball projected on the Xaxis represents a cosine wave. The length of the string (r) corresponds to the amplitude, and the time taken
for one full rotation of the ball to the period (T). If the angle of one rotation of the ball is expressed in
radians, 360=2p radians, and the angle rotated through by the ball in t time can be calculated from equation
2-2,

(Equation 2-2)

where the frequency f indicates the number of rotations the ball makes in one second, and is given by the
equation 2-3,

(Equation 2-3)

Then, the loci of each shadow after t time are expressed in the following mathematical equations.

(Equation 2-4)

These two equations show the position of the ball after t time as X-Y coordinates (x,y).

Now, if we assume that the ball at the position achieved by angle f as shown in Figure 2-5, then equations 24 obtain the following:

Figure 2-5

(Equation 2-5)

Since a, b, and r can be regarded as the three sides of a right-angled triangle, the following equation also
holds.

(Equation 2-6)

As seen above, the ball position is specifically determined by either using a and b in equations 2-5 or using r
and f, regardless of the number of rotations of the ball.
There are many physical phenomena that exhibit periodicity and are therefore classified as wave
phenomena. The sine and cosine function are used together with the tangent function as trigonometric
functions in the mathematical expressions that explain such phenomena.
Let's assume that the position of angle f is the starting point (in this case, f represents the initial phase).
Then, equation 2-4 can be expressed as shown below.

(Equation 2-7)

(Equation 2-8)

The expression representing the locus of ball motion may either be equation 2-7 or equation 2-8. Equation 27, the cosine-based function, is generally used, however, because a sine wave is considered to be a cosine
wave with a phase lag of p/2 and formulas representing physical phenomena are often expressed using
cosine-based functions for ease when solving.
The next section explains the related waveform expressions and the important common terms of FFT
waveform analysis.

2-3 Time Difference and Phase

If a person screams out towards a mountain, an echo will be heard a short time after. Let's consider how the
time-difference between the time of the scream and the time of echo return is expressed as the phase
difference.
If the period of the waveform is represented by T, the frequency by f, time difference by t and phase by f,
then:

(Equation 2-9)

For example, in the previous waveform of the "A" sound, if the frequency of "A" was 106 Hz and the timedifference to the echo return t=0.001 second, then the phase is calculated to have a lag of approximately
0.21p=38.1. If the time difference is the same, t= 0.001 second and the frequency of "A2" is assumed to be
212 Hz, then a phase lag of 76.3 occurs. This shows that if period T (frequency) is different, then the phase
will also be different, even if the time difference is the same.

2-4 Electrical Angle and Mechanical Angle

Besides amplitude, frequency, and phase, it is also important that the difference between the electrical angle
be understood.
Assume that an electromagnetic detector is prepared for a gear having 60 teeth and the gear rotates at 600
r/min. If the gear rotates one full revolution, a 60-cycle sine wave signal is output from the electromagnetic
detector. Here, the phase expressed by taking one period of the signal sine wave as 2p rad (360) is called an
"electrical angle" and the phase expressed by taking one rotation of the gear as 2p rad is called a
"mechanical angle."

Figure 2-6

As shown in Figure 2-6,

the frequency of the signal sine wave, f = 600r/min./60s60 gear teeth = 600(Hz), and
the frequency of rotation of the gear, f0 = 600r/min./60s = 10(Hz).
Although one period of the signal sine wave and that of the gear both have an angle of 2prad, their time
differs as shown below:

Period of the detector signal = Period of electrical angle expression = 2p = 1/600 second
Period of one turn of gear = Period of mechanical angle expression = 2p = 1/10 second

The electrical angle is based on one period of electrical signals, while the mechanical angle is based on one
period of one revolution of a body. When the phenomena of rotating bodies are analyzed, the frequency
analysis data must be considered in light of the relationship between the electrical and mechanical angles.
An FFT analyzer uses electrical angles.

2-5 Rpm Order and Harmonics

The rpm order is frequently used to analyze phenomena related to revolutions. If the gear rotation described
above is considered as a reference frequency, the sine-wave signal has a frequency of 60 times the reference
frequency of the gear. This is called the "sixtieth harmonic." In rotary machines, among other things, the
term "rotation" is used to note the rotational frequency (revolutions per second) of the gear. For example,
rotation at a frequency of 10Hz is referred to as the 1st-order rotation and rotation based on a 600-Hz sinewave signal is referred to as the 60th-order rotation.
The term, harmonics, refers to the frequencies of integer multiples of the fundamental frequency. For
example, if the frequency of fundamental wave "A1" of sound "A" is 106 Hz, then the frequency of its
second harmonic "A2" is 212Hz. These days, harmonic distortion analysis is carried out on the frequency of
power supplies. This involves analyzing the integer multiples of harmonic components or the amplitudes and
phases of these frequencies, when the fundamental power frequency is say 50 or 60 Hz.

2-6 Representation of Waveforms <Sum of Waveform (composition)>

Consider the sum of the waves shown below (composition of waves).

(Equations 2-10)

The composite waveforms are as shown in (3) and (4) of Figure 2-7, each having the same frequency but
different amplitude and phase.
Let's consider x2 first. The equation of x2 can be rewritten as shown below using the trigonometric formula.

(Equation 2-11)

This means that, if a cosine wave and a sine wave having the same frequency and phase are composed, then
the resultant wave is equal to a cosine wave having root 2 times the original amplitude and a phase lag of
p/4. This can also be expanded as shown below.

(Equation 2-12)

If the "a" coefficient of the cosine wave is taken on the cosine axis and the "b" coefficient of the sine wave is
taken on the sine axis, then Figure 2-9 is obtained. As this figure is the same as the previous Figure 2-5,
equations 2-6 can be solved as follows.

Figure 2-8

(Equation 2-13)

That is, equations 2-14 hold true.

(Equation 2-14)

As seen above, a waveform with a single frequency (hereafter called a simple waveform) can be expressed
as an equation having the same frequency and coefficients of a and b as shown below.
Thus, if coefficients a and b are given, composite amplitude value r and phase difference can be determined.
For example, if the coefficients of the equation of the composite waveform,x1=
are a=2 and b=0, then the value of composite amplitude r=2 and phase difference f=0.

Components "A?", "A2", "A3",... of the sound "A" that was frequency-analyzed, are each simple waveforms
that can be expressed in the form
difference f is

. For your reference, please note that phase

2-7 Representation of Waveforms <Product of waveforms>

Now that we have explained how simple waveforms can be represented by the expression "
," this section will explain how the coefficients of the expression are actually
determined.
Let's consider the following products of waveforms.

(Equations 2-15)

Figure 2-9 shows waveforms of the following expressions when a1=2, a2=3, and b=2.
As shown in this figure, the product of periodical waveforms results in a waveform also having periodicity.
Let's consider the area obtained by integrating the waveform of x1 for one period (from 0 to 2p).
In waveform x1, the area on the positive side and the area on the negative side are equal, this the sum of
the areas is zero. Waveform X2 expresses the product of the fundamental waveform and the second
harmonic component. The area of waveform X2 is also zero. However, for the waveform of X3, there is no
negative side area, and the sum of the area.

That is,

(Equation 2-16)

This means that amplitude an can be determined from an arbitrary waveform of the form
," where n=1,2,3,...,by:
*

multiplying the waveform by

in order to determine the area for one period,

dividing the area obtained by T/2, and then

extracting the original

component.

Similarly, if the above arbitrary waveform is multiplied by

, the area of the product
waveform determined and then divided by T/2, then amplitude bn can be determined.

This report contains a conceptual explanation of FFT and the fundamentals of sine and cosine waves
necessary for understanding Fourier transform and Fourier series. Within this explanation, appear many
mathematical expressions. Did you understand them? We have tried to give these mathematical expressions
meaning by showing them with real figures. If you have any questions, please feel free to discuss them with
us.

In the next report, we will explain the remaining Fourier series items and describe the expression of Fourier
series by way of a complex exponential function. In addition, we will give a practical example of
calculating FFT.

FFT Analyzer II
3. Fourier Series and Fourier Transforms
As mentioned in the previous report, waveforms that periodically vary in the time domain can be expressed
using a Fourier series. In this chapter we will actually try to express these periodical waveforms by using the
Fourier series and Fourier transform.
3-1 Fourier Series
Let us begin by considering the sound "A" (pronounced as "aah"), as introduced in Figure 1-1, Chapter 1, of
the previous report.
As we noted, the sound "A" could be decomposed into three elements, "A1", "A2" and "A3". Moreover, the
waveform of "A1" could be represented as cosine waves having a phase difference in the following manner:

Similarly, "A2" and "A3" are also expressed as follows:

Since the sound x(t) of "A" is composed of the waves of "A1", "A2", "A3" ..., the following relation holds:

By rearranging f1, f2, f3, ..., a1, a2, a3 ..., and b1, b2, b3, ... , respectively, as fn, an, bn, (n = 1, 2, 3, ...) here, we
have:

(Equation 3-1)

If we compare Equation 3-1 for the sound "A" with Definition Formula 3-2 for a Fourier series, shown
below, we notice the two are strikingly alike. If the basic frequency is f0, components f1, f2, f3, --- of the
sound "A" are f1 = f0, f2 = 2f0, f3 = 3f0, and correspond to the 1st harmonic, 2nd harmonic, 3rd harmonic, --(frequencies in integer multiples of f0). Since the example analysis of the sound "A" in Figure 1-2 of the
previous report adopted T = 160 ms (f0 = 6.25 Hz), these harmonics correspond to f1 = 106.25 Hz = 17f0
(17th harmonic), f2 = 34f0, and f3 = 51f0, in the figure. The harmonics that do not correspond to f1, f2, f3, --can be considered other signal elements different from "A", such as background noise.
As can be gathered from the above, the concept we have discussed since Section 2-2 of the previous report is
nothing other than the Fourier series.
So, let us take a look at the definition formula of this Fourier series.
If the period of a waveform periodically varying in the time domain is represented by T, we have:
Fundamental frequency
f0 = 1/T
Fundamental angular frequency
0 = 2 f0
Therefore, the waveform can be expressed in the form of a Fourier series as shown below:

(Equation 3-2)

Here, by making use of the multiplications of cosine and sine given in Section 2-6 of the previous report,
and the appropriate area values, the components of Equation 3-2 can be found:

(Equation 3-3)

a0 is a DC component, and an and bn are the amplitudes of cosine waves and sine waves whose angular
frequency is n 0. These last two are called Fourier coefficients and are collectively referred to as a Fourier
coefficient pair.
Now, let us review the explanation thus far.

Equation 3-2 can be interpreted as follows:

n 0t(n=1, 2, 3, ...) are 1x, 2x, 3x, ... harmonics of basic angular frequency
components of n 0t are expressed by:

, where the waveforms of the

The corresponding Fourier coefficients an and bn can be found by Equation 3-3. Furthermore, based on the
relationship between an and bn shown in Figure 3-1 below, Equation 3-3 can be represented as follows:

Figure 3-1

(Equation 3-4)

Where rn and

are the amplitude and phase of the n-th harmonic, respectively.

An FFT analyzer stores the calculated Fourier coefficients an and bn in memory, then uses them to calculate
the amplitude rn and phase n of frequency fn. Based on this calculation, the analyzer produces a spectral
display showing the relationship between frequency fn and amplitude rn. Likewise, it gives a phase spectrum
showing the relationship between frequency fn and phase fn. The original time waveform can also be restored
from an and bn using Equation 3-2. We should note that a spectrum is just a quantity of magnitude (i.e., does
not have the phase information < >) and, therefore, spectral data alone does not enable the original
waveform to be restored. Among its functions, the FFT analyzer offers a technique to seek time waveforms
from Fourier transform data (i.e., an inverse Fourier transform). To execute this function, it is necessary to
store previously a Fourier spectrum with phase information (consisting of a real part and an imaginary part).

In addition, if original waveforms are stored, the analyzer is also able to perform an FFT from this data
again.
* The terms "real part" and "imaginary part" will be explained in the following section.

3-2 Representing the Fourier Series by Complex Exponential Function

An FFT analyzer will display the expressions "Real Part" and "Imaginary Part" when indicating Fourier
spectra and transfer functions among the diverse types of functions it computes and processes. These
expressions appear when a Fourier series is represented by a complex plane (Gauss plane) and are used in a
technique for functional representation. Each part has an important meaning. In this section we will attempt
to transform a Fourier series into an indication of complex exponential function by using Euler's formula
and will, in this way, explain the concepts of "Real Part" and "Imaginary Part."
The explanation below may be a bit hard to understand, but let's start by gaining an understanding of Figure
3-3 below. Here, we will not go into details of the complex number and exponent; those who wish to know
more about them should refer to the relevant specialized literature.
Euler's formula is:

(Equation 3-5)

In Euler's formula indicated above, , e and j are understood as follows:

: Pi, 3.141592....
: Base of Napierian logarithm, e = 2.71828.... et is "et", whether differentiated or integrated. d/dt (e t ) =

e
et
j : Value where j2 = -1.

As you are well aware, numbers consist of real numbers and imaginary numbers. Those without a j value are
called real numbers, and those with a j value are termed imaginary numbers.
The complex number Z = 1+1j on a complex plane (Gauss plane) is expressed as shown in Figure 3-2(a).
Meanwhile, X = ejn t represents a circle having radius = 1 on a complex plane as indicated in Figure 3-2(b),
which circle rotates counterclockwise (i.e., in a positive direction) at an angular speed of n per second. This
means that the two equations of cosine and sine that represented the locus of a ball in Equations 2-7 and 2-8
(given in the previous report) are represented here by one single equation. Similarly, Y = e-jn t - indicates a
clockwise (negative) rotation. X and Y relate to each other symmetrically with respect to the real-number
axis (Re-axis), Y being called a conjugate complex number of X and X, a conjugate complex number of Y.
Conjugate complex numbers are indicated with an "*", e.g., Z*.

Figure 3-2

We rearrange Equation 3-5 above by substituting it into Fourier-series Equations 3-2 and 3-3, as follows:

(Equation 3-6)

Multiplying both sides by j yields the following:

Noting (an-jbn) and (an+jbn) here, we express X0, Xn, Xn*, respectively, as follows:
(Equation 3-7)
(Equation 3-7a)
(Equation 3-7b)
(Equation 3-7c)

Considering the case where n = 0 is adopted in Equation 3-7b above, the right side of Equation 3-7b matches
that of Equation 3-7a, because e0 = 1. Therefore, Equation 3-7 can be rewritten as follows, assuming that n
takes n = 0 to :

(Equation 3-8)

Moreover, if we assume that n in Equation 3-7c takes n = -1 to - , it matches the right side (n = 1 to ) of
Equation 3-7b. Therefore, Xn* can be handled as one that exists when n of Xn is n = -1 to - . Accordingly, to
recap the above, Equations 3-7, 7a, 7b and 7c can be represented in a very clear form as follows:
(Equation 3-9a)

(Equation 3-9b)

Equation 3-9a above is termed the Fourier series of a complex exponential indication and Equation 3-9b, the
Fourier expansion of a complex exponential indication.
Because ejn 0t and e-jn 0t in Euler's formula consist of a cosine and sine, this formulation replaces the
periodicity of cosine and sine we have seen to this point, and has the nature of an exponent at the same time.
If we replace ejn 0t and e-jn 0t with sine and cosine, respectively, for example, and replace Equation 3-9b with
cosine, it corresponds to the equation for an in Equation 3-3. Furthermore, if we consider that the cosine
term and sine term of a Fourier coefficient are the real part and imaginary part of a complex number
indication, in light of the relation shown below, then it will be easier to understand the relationship between
the Fourier series and its complex number indication. (To see this, compare Figure 3-1 and Figure 3-3 with
each other.)

Now, the k-th harmonic in this condition is represented as follows:

Figure 3-3

(Equation 3-10)

Xk* is a conjugate complex number of Xk. If the above is represented by complex coordinates with the real
part (Re) plotted on the x-axis and the imaginary part (Im) on the y-axis, Xk* and Xk appear symmetrical to
each other with respect to the Re-axis, as shown in Figure 3-3.
Since Figure 3-3 can be considered in the same way as was Figure 2-9 in Section 2-6 of the previous report,
the following Equation 3-11 can be established:

(Equation 3-11)

Xk here is called a Fourier spectrum, 1/2ak the real part (Re) of the Fourier spectrum, and 1/2bk the
imaginary part of the Fourier spectrum (Im). Moreover, |Xk|2 is termed a power spectrum. Incidentally, the
Fourier spectrum in an FFT analyzer indicates ak and bk, and the power spectrum shows 4|Xk|2. This is
because it is convenient to represent these terms with the rn, an, and bn used in Figure 3-1.
3-3 Fourier Transform
Since Equation 3-9a is a Fourier series, it represents a continuous, periodic time-dependent waveform.
Substituting Equation 3-9b into this equation yields Equation 3-12 below. (For simplicity of calculation, the
0 to T section of the waveform was replaced with -2/T to 2/T.)

(Equation 3-12)

Let us consider here extending period T to - to + , so that waveforms without periodicity can also be
handled. If, in Equation 3-12, we adopt for df a very small frequency ultimately reached as a result of
enlarging T in 1/T, we have:

(Equation 3-13)

Substituting X(f) for the part in {} in the equation above, we obtain the following equations:
(Equation 3-14)

(Equation 3-15)

Equation 3-14 represents a Fourier transform and Equation 3-15, an inverse Fourier transform. Although 1/T
disappears in Equation 3-14, the Fourier transform, it can be regarded to have moved to the inverse
transform side of Equation 3-15. In this way, it can be understood that the pair of Fourier transform and
inverse transform corresponds to the pair of Fourier expansion and series handled in Equations 3-9 above.
However, be careful not to confuse Equations 3-14 and 3-15 with each other, as the two are very much alike.
X(f) represents the frequency domain and x(t), the time domain
While the spectrum was shown in the form of discontinuous harmonics in the Fourier series, it appears as a
spectrum of continuous frequency in the Fourier transform, where the periodicity is expanded infinitely,
causing the basic frequency f0(=1/T) to take on a very small value.
Now, let us consider the Fourier transform of x(t)=1.
We cut a certain time portion T out of an infinitely continuing value of 1 and we regard the rest as zero (0).
Let us try to make a Fourier transform on this assumption.
Figure 3-4 illustrates two cases, T = 1 and T = 5. The conjecture is shown in Figure 3-4a, but it differs
substantially from Figure 3-4b. However, if T is made larger, the waveform becomes narrower in width and
the amplitude value also increases, creating a waveform closer to that in Figure 3-4a. This difference arises
because we simply assumed the invisible portion as 0, and a larger T shows the characteristic of the whole
waveform more clearly as a matter of course. As can be gathered, if we select a longer time T for which a
waveform is cut out (called "time resolution") in a Fourier transform, the frequency resolution (= 1/T) will
become higher, allowing us to view a more detailed spectrum. Conversely, if the time for which a waveform
is cut out is set to be short (i.e., time base resolution is high), the frequency resolution will turn rough and
indicate a spectrum only on a broad basis. In this way, the Fourier transform is possessed with a natural
uncertainty, as it is called, between time resolution and frequency resolution. The FFT analyzer allows this
time length T to be modified as a data length (number of samples), according to the intended application.

Figure 3-4

3-4 Discrete Fourier Transform

An FFT analyzer handles a time waveform x(t) as discontinuous data (discrete data) acquired at sampling
interval h.
If the sampling value sequence of an N-number of finite data points sampled at sampling interval h is
represented by x(n) (n = 0, ... N-1), the Fourier transform likewise holds true for that data. Since multiplying
a time waveform x(n) by a given waveform e-j2pft and integrating one period of it provides the amplitude of
the waveform, we apply this discrete data to the formula by using Equation 3-9b as a guide:

From the above, the Fourier transformation formula of the k-th harmonic is expressed as follows:

(Equation 3-16)

Equation 3-16 is called a discrete Fourier transform (DFT). Moreover, the relation of Equation 3-11 also
holds for this equation in the same way as for Equation 3-7b. Similarly, the inverse discrete Fourier
transform (IDFT) is expressed by:

(Equation 3-17)

Please refer to Figure 3-5 (click on it), where we have given a conceptual representation of Equation 3-16
for your reference.
3-5 Sampling Theorem
So, to what extent can we calculate a harmonic k?
To know an original waveform, it is necessary to perform sampling at a frequency at least twice that of the
waveform. This is called the sampling theorem.
Now, let us consider the case where an N-number of samples are taken for a waveform with period T. In this
case, the sampling interval h and its frequency fs are expressed by:

(Equation 3-18)

Therefore, the frequency fm that can be analyzed using the sampling theorem is:
(Equation 3-19)

Namely, the analysis can be conducted up to the N/2-th harmonic.

If we sample a waveform with band limited from 0 to fm at fs and perform a DFT calculation, we obtain a
spectrum of frequency - to + Figure 3-6 illustrates a typical spectrum of frequencies from -fm to 4fm.

Figure 3-6

As can be seen in the figure above, the spectrum obtained has the same value repeated, just like a paper
folded at integer multiples of fm. A similar result to Figure 3-6 will also be obtained if we sample a
waveform with band limited from fm to 2 fm at the same frequency fs and then perform the DFT calculation.
This calculation will, however, provide us with the components of 0 to fm which do not exist in actuality.
This is called an "aliasing" phenomenon (or "folding frequency") and a frequency fm in this condition is
termed a Nyquist frequency. A simple look at just the 0 to fm range of the spectrum obtained will not enable
you to tell whether the original waveform is from 0 to fm or from fm to 2 fm. For this reason, an FFT analyzer
passes signals through a low-pass filter (dubbed an "aliasing filter") that previously limits them to fm and
thus defines only signals below fm in order to meet the sampling theorem. Moreover, since frequencies up to
fm, plus a, need to be considered in order to obtain frequencies that can fully be attenuated by the said lowpass filter, an FFT analyzer is designed to indicate spectra up to 1/1.28 of fm only (Figure 3-7).

Suppose the frequency range of FFT is set to 1 kHz, for example, the
FFT analyzer will sample at frequency fs = 2.56 kHz, which is 2.56
times the frequency range.

Figure 3-7

Please note that Figure 3-8 shows the sampling theorem scrutinized from the standpoint of time waveforms,
and Figures 3-9a and 3-9b (please click on them) exhibit the waveforms displayed by an FFT analyzer as the
aliasing filter is turned on and off.

 : Sampling point
-- : Waveform reproduced at sampling points
If the original signal has a frequency smaller than 1/2 (=fm) of
sampling frequency fs, it can be reproduced as shown in (a), but if
the frequency is larger, the original signal cannot be reproduced and
instead, frequency garbling like the waveform indicated by the
dotted line in (b) will take place. When the original signal has a
frequency that matches the sampling frequency fm, a straight
waveform will show up like the dotted line in (c).
Figure 3-8

This discussion has assumed quite a complex and mathematical tone. We hope it has been clear to you up to
this point.
At this time, we would like to review the Fourier series, Fourier transform, DFT, and sampling from the
viewpoint of an FFT analyzer, with some imagery concepts mixed in. Part of this explanation will include
quotations from the book "Introduction to Digital Signal Processing," written by Ken'ichi Kido.
Let's start by referring to Figure 3-10. Waveform "A" is a time waveform that is repeated infinitely with a
band limited to less than a Nyquist frequency fm. "B" is the frequency spectrum obtained from "A", which is
distributed over fm from its origin, and the part outside that range is zero (0). Then, assuming that spectrum
X(f) of "B" is arranged infinitely on the frequency axis with period 2fx, the cases indicated below are likely:

Figure 3-10

When fx = fm, spectrum waveform of "C".

When fx < fm, spectrum waveform of "D".
When fx > fm, spectrum waveform of "E".
Here, the spectrum waveform of "E" represents the state in which the different spectra are overlapped on
each other, causing aliasing to take place. The spectrum waveforms of "C" and "D" are identical to spectrum
X(f) of x(t) in the range from -fm to +fm. Changing the angle of view, now let us consider sample train xn,
where x(t) is sampled at 2fx. Spectrum xn in this condition is:

(Equation 3-20)

This represents "C", "D" and "E" with an equation. In other words, if we sample a continuous waveform x(t)
with band limited to fx at sampling frequency 2fx, seek spectrum X(f) from the data of its infinite sample
train {xn}, and look at its frequency range from -fx to +fx, then we find that it is a frequency spectrum X(f) of
x(t) in its own right ("C", "D").
Since application of inverse Fourier transform to X(f) yields x(t), the sampling value sequence at every 1/
(2fx) has sufficient information to indicate a continuous waveform x(t). If the original waveform has a
frequency that is higher than the sampling frequency, overlapped spectra (such as those shown in "E") will
appear because of the aliasing effect, and the resultant spectrum will be different from X(f), making it
impossible, therefore, to reproduce x(t).
The discussion up to this point has centered on the Fourier transform.
Next, suppose the waveform of "A" is repeated infinitely with period T like "F", such that the frequency
spectrum resulting from the Fourier expansion of that waveform will present itself as a linear spectrum
arranged on the frequency axis at interval 1/T. Since this waveform has its frequency component limited to
less than fx, the spectrum will appear like "G", and the envelope of this linear spectrum will match the
continuous spectrum of "B". If we assume here that the spectrum of "G" is identical to a spectrum "H" that is
repeated infinitely with period 2fx, like "C" and "D", the time waveform obtained from applying an inverse
Fourier transform to "H" will appear as a sampling value sequence that is repeated infinitely with period T,
as shown in "I". That is, both the time waveform and the spectrum appear as a sampling value sequence
repeated periodically and infinitely. These relationships are represented by DFT and IDFT of Equations 3-16
and 3-17. Since the following parts of Equations 3-16 and 3-17 are periodic functions that take the same
values repetitively with a certain period, like the sine and cosine waveforms, it will be understood that DFT
and IDFT represent "H" and "I" of the figure:

Furthermore, it will also be clear that they give sufficient information to the user, provided that the spectra of
0 to fx can be obtained correctly. Given below for reference purposes is an equation to restore a continuous
waveform from a spectrum.
Sample train {xn} is expressed by:

(Equation 3-21)

And, the Fourier coefficient in the term of -n of its spectrum X(f) corresponds to xn. Moreover, a continuous
original waveform x(t) can be obtained from sampling value sequence {xn} by the following equation (see
Figure 3-11):

(Equation 3-22)

Figure 3-11

Let's wrap up this explanation of Fourier series and Fourier transforms for now by giving a brief roundup of
information on DFTs.
<Summary>
Suppose signal x(t) which is infinitely repeated with period T has its band limited to below a certain
frequency, and that a discrete spectrum (linear spectrum) X(f) can be found performing DFT at every 1/T
=1/Nh, based on a discrete sample train {xn} acquired from signals of period T in an N-number at constant
interval h conforming to the sampling theorem. Likewise, a discrete sample train {xn} can be reproduced by
performing an IDFT.
An FFT analyzer shows a sample train of 0 to N-1 samples as a time waveform in the time domain and a
frequency spectrum (linear spectrum) from 0 to fm/1.28 in the frequency domain.