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Hi, and welcome to module 4.3 of Digital
Signal Processing.

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In this module we will look at the DFT

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in practice by showing some analysis
example by showing

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you how to label the DFT axis to relate

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the DFT to physical quantities like real
world frequency.

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We will talk about DFT synthesis, how we

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move from the frequency domain to the time
domain.

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And finally we will introduce

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a close relative of DFT called the
discreet Fourier series, which

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is just a different flavor of DFT applied
to periodic signals.

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You remember that we tried to wet your
appetite with respect to Fourier

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analysis in the beginning of module 4.1,
by showing you a mystery signal.

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And by showing you that by changing the
basis in which the signal is represented,

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we all of a sudden could see some
structure in the data.

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So now we can go back to that example

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and analyze it applying what we know about
DFT.

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So we take the DFT of the mystery signal,

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we plot the real and imaginary part of the
DFT,

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and we see that the structure that appears
is

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a couple of spikes at symmetric positions
in the spectrum.

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Now, we already know that this

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indicates the presence of a strong
sinusoidal component

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and more importantly, a sinusoidal
component has a

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frequency which is a multiple of the basic

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frequency for the space that the signal
lives in.

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Now, if you look in detail at what happens
in the spectrum, we see

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that the peaks are for k equal to 64 and k
equal to 960.

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We also see that the

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peaks appear only in the real part of the
spectrum and we remember that

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when this happens the underlying sinusoid,
the

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sinusoid represented by those peaks, is a
cosine.

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So from this simple visual inspection, we
can already

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write our signal as such, it will be a

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cosine component and we will have to
determine both

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the frequency and the initial phase of
this cosine.

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And there will be another part that we can
probably call

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a noise component, in the sense that it
doesn't have any structure.

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Since the imaginary part of the Fourier
transform doesn't exhibit any particular

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peak, we can assume that the phase of the
cosine is zero.

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And as far as the frequency is concerned,
we

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know that the peak occurs at k equals 64.

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We are in a space of 1,024 points, and
therefore omega will be equal to

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2 pi over 1,024, the basic frequency for
the space, times 64.

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If it blocked the two components
separately at this point, we see

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that we have indeed a cosine that
oscillates 64 times in 1,024 points.

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And we have an additional noise component

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that doesn't really seem to have any
structure.

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Let's now look at another signal, one that

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we showed in the introduction to this
class.

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And that is a time series that maps the
solar activity since the 1700s.

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Astronomers have noticed that solar
activity could be

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related to what they call a sunspot
number.

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The details are not really important, but
fundamentally

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a measure of how many solar spots are

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at a given point in time on the face of
the sun.

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So we have a data set that goes from 1749
to 2003,

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that is equivalent to 2,900 months, solar
spots are computed every month.

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And as we plot this data we see that there
is an oscillatory behavior.

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We're interested in finding out if

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there's any fundamental periodicity in
solar activity.

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So we can take the Fourier transform.

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The sunspot time series is a real signal.

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Now, if we are interested just in the
magnitudes of the DFT

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coefficients, you remember we need only
show the first half of them.

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Even so, if you look at the first 1,500
coefficients you see that

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after the 100th coefficient or so, their

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magnitude becomes too small to be
relevant.

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So in this plot, for clarity, we just

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show the first 100 DFT coefficients in
magnitude.

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And indeed we can see that there is not
just one peak but a series of peaks.

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Now, what is the fundamental period?

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What is the most dominant mode of this
time series?

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The main peak happens at k equal 22.

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There are 22 cycles over the entire data
set.

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So the data set is 2900 months, so the

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frequency of the solar spot activity is
2900 divided

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by 22 which is approximately 11 years.

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So solar activity seems to have an
inherent period of 11 years.

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We can perform the same analysis with a
data set that records the daily

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temperature over a total of 2920 days.
If we take the DFT of this time

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series, we see there is actually a very
pronounced periodicity in the spectrum,

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there is basically just one peak.

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And if we plot the normalized DFT
coefficient, so

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if we divide the DFT coefficients by the
length

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of the temperature vector, we can actually
extract some

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information about the temperature values
in this time series.

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So remember, for instance, the DFT
coefficient for k equal to

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0 is the known normalized average of all
the data points

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because x of 0 is simply the sum of all
the points in

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the time series for n that goes from 0 to
big N minus 1.

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So if you plot the normalized
coefficients,

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the coefficient in 0 will be the average

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temperature for the time series, which in

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this case happened to be 12.3 Celsius
degrees.

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And then we remark that there is a peak in
the DFT for k equal to 8.

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So if we sum up what we learned from the
DFT about the temperature signal, we

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know the average value of the temperature
is

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the 0th DFT coefficient normalized, so
12.3 degrees Celsius.

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The main peak is at k equal to 8, for a
value of 6.4 degrees.

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What that means is there are eight cycles

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of the temperature signal over the entire
duration of

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our data set, which is 2,920 days.

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So the frequency is 2,900 divided by 8,
which happens to be 365 days.

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So indeed the temperature has a yearly
periodicity, as we all know very well.

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Now, the value of the DFT main peak is 6.4
degrees Celsius.

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Now, if we have a sinusoid of the form A
cosine of omega

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n, and we take the DFT of this signal, we
will have a peak in magnitude

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at some, for some index k, that has a
value of A

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over 2, remember the definition of the DFT
of a cosine function.

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So from this we can say that the
temperature excursion around

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the average is twice the value of the DFT
main peak.

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And, so we can say,

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that the yearly temperature at the point
of measurement for the

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year was an average of 12.3 degrees plus
or minus 12.8.

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So this is a second example in which, in

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order to find the real world frequency of
a certain

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DFT component, we take the total length of
the

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signal and we divide by the location of
the peak.

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This is actually a general method to label
the frequency

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axis of a DFT plot and it will be very
useful in the future

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when we start analyzing generic signals
that

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have been sampled in a variety of
contexts.

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So if you remember, in module 2.2, we
informally introduced

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the notion of a clock for a digital signal
processing system.

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And this is really equivalent to
associating a certain

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time interval, Ts, measured in actual
physical seconds between successive

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samples in the signal.

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So, for instance for the solar spot
signal, we had a Ts of

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one month, whereas in the temperature
signal, the Ts was equal to one day.

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We will see that for audio signal Ts
becomes very small

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because we will need to take at least
8,000 samples per second.

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Now if we have a value for Ts, which is
determined

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by the experimental setup, we can reason
like so, the fastest positive

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frequency in a digital signal is omega
equal to pi.

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So a sinosoid at that frequency needs two
samples

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to do a full revolution, remember the unit
circle.

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Something that moves at a speed of pi will
be here at instance n, here at n

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plus one, here at n plus two, so two
samples to complete the full revolution.

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Now, the clock Ts can

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also be expressed as 1 over Fs, where Fs
is the frequency of the system.

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This is a standard relationship between
period and frequency for any system.

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So, if the real world period for the
fastest sinusoid in a digital system

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is 2 times Ts measured in seconds, the
real world frequency for the fastest

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sinusoid is Fs over 2.
So the maximum frequency,

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once we've fixed the period between
samples, is

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Fs over 2, or equivalently, 1 over 2 Ts.

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So let's take a concrete example.

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Suppose I give you an audio file that
records a train whistle

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and I'm asking you to find out which notes
make up the whistle

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[SOUND].

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And of course your idea is to take the DFT
of this file and find out the peaks

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and the Fourier transform, which will
probably correspond to

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the fundamental frequencies of the notes
that are being played.

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So in order to do so you need some extra
information and namely you

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need to know the sampling frequency of

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this file, namely the time between
successive samples.

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So the sampling frequency is Fs

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equal to 8000 hertz, which means that Ts
is 1 over 8000 seconds,

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this is the time between samples, and also
the file contains 32768 samples.

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We will see it later why we often choose
signal lengths that are a power of two.

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So if you take the DFT of this file and
plot

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its magnitude you see that there are three
peaks that probably

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corresponds to the three notes being
played.

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Now in order to label the axis remember
what we said, the

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highest frequency for a digital system

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corresponds to half the inherent sampling
rate.

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And on the DFT vector, this point will
correspond to the midpoint in

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the vector, namely the point for k is
equal to N over 2.

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So here indeed we have the midpoint of the
DFT vector and

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therefore, this will correspond to a
frequency of four kilohertz.

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We also have three peaks to take place for
different values of k.

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To find the real frequency there, we just
apply a

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linear mapping from zero to four
kilohertz, and so the

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frequency of any intermediate point will
be 4 kilohertz divided

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by the total number of points multiplied
by the index k.

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And with this rule, we find that the peaks

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take place at 494 hertz, 559 hertz, 739
hertz.

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And if we look up what these frequencies
correspond to in the standard

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western scale, we find that the whistle is
a simple B minor chord.

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Okay, so far we have looked at DFT as an
analysis tool, let's now spend

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some time thinking about the synthesis
properties

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of the DFT, how we build the signal

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starting from DFT coefficients.

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And to better understand how the DFT

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reconstruction formula works, it's useful
to look at

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the standard sinusoidal generator that
works at one

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of the standard frequencies in the Fourier
basis.

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So here you have a complex exponential
with a frequency which is a

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multiple of 2 pi over big N and with an
initial phase, phi k.

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The way this generator works is

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by successively generating points on the
unit circle, starting at the phase

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phi k and proceeding in increments of 2 pi
over big N.

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We can draw this concisely with this
notation where here inside you

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have the index of the Fourier frequency
that the generator will produce.

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And the input parameters for the
generators are

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of course the initial phase here, and
again

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factor A of k, that will lead to A of k
times to

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e to the j 2 pi over big N, kn plus phi k.

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So the DFT reconstruction formula can be
drawn

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graphically like a machine composed of big
N sinusoidal

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generators, each one of which will be
initialized

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with a gain factor and with a phase
factor.

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All their outputs will be summed together
and

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this will give us back the original
signal.

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How do we initialize each one of these
blocks?

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Well the amplitude will be the amplitude

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of the corresponding Fourier analysis
coefficient normalized by

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big N, and the phase will be

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the phase of the corresponding Fourier
analysis coefficient.

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00:13:14,960 --> 00:13:20,030
So, lets look at an example.
Lets take a simple signal, a seven point

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long signal that looks like a triangle,
and we perform a Fourier analysis

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on this vector and we get seven Fourier
coefficients

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00:13:30,770 --> 00:13:35,360
that we list here in terms of amplitude
and phase.

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Now we initialize the seven generators
with the As

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and the phis that we found in the analysis
part.

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00:13:40,380 --> 00:13:43,160
And then we turn the crank seven times and
we

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will sum the outputs together to obtain
the original signal.

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00:13:47,130 --> 00:13:51,460
So here is how it works.
In the first line,

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we will show the output of each generator
in turn, starting

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for k equal to 0 up to k equal to 6.

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And here on the bottom line, we will show
the cumulative

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sum of the outputs as k proceeds from 0 to
6.

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00:14:06,830 --> 00:14:10,100
So the first generator is the one for k
equal to 0,

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so the frequency is 0, and the output will
be simply a constant.

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00:14:14,410 --> 00:14:15,630
As a matter of fact, the value of this
contes,

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00:14:15,630 --> 00:14:18,750
constant will be equal to A of 0.

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00:14:18,750 --> 00:14:21,840
And so the cumulative sum now has only one
term,

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so the output at this point will look like
a constant.

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00:14:25,370 --> 00:14:28,110
For k equal to 1, we have now a sinusoid

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that will have a certain initial phase and
a certain amplitude.

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00:14:32,200 --> 00:14:35,510
And we can see that as we add this to the
previous constant, here in

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grey you see the previous sum, and here
the updated sum for k equal to 1,

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the real part of the signal starts to look
a little bit like a triangle.

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00:14:44,740 --> 00:14:49,050
We also have a significant non-negligible
imaginary part which will have to

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disappear in the end because remember, we
started from a real signal.

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00:14:53,190 --> 00:14:57,170
So as we proceed for k equal to 2, we see
that the coefficient

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00:14:57,170 --> 00:15:01,380
amplitude is very small and so this will
be just small adjustments in the sum.

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00:15:01,380 --> 00:15:04,220
It's very hard to see the difference
between this step and the previous step.

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00:15:05,270 --> 00:15:08,670
K equal to 3 is very similar, small
adjustments.

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00:15:08,670 --> 00:15:14,150
For k equal to 4, we're going in the,
counter-clockwise part of the spectrum,

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00:15:14,150 --> 00:15:19,310
so more adjustments still, that will start
to bring down the imaginary part to 0.

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00:15:19,310 --> 00:15:21,080
Here you see another small adjustment.

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00:15:21,080 --> 00:15:24,700
And finally k equal to 6, the last
coefficient will completely kill the

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00:15:24,700 --> 00:15:29,230
imaginary part and give us back exactly
the signal we were starting from.

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00:15:29,230 --> 00:15:30,670
So why do we think

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00:15:30,670 --> 00:15:34,340
about the DFT reconstruction formula as a
machine?

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00:15:34,340 --> 00:15:36,730
Because in reality these machines were
used in

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00:15:36,730 --> 00:15:39,330
the past for instance to predict the
tides.

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00:15:39,330 --> 00:15:43,960
The tide is a periodic phenomenon and once
you find the modes you can actually use

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00:15:43,960 --> 00:15:46,710
the experimental data to predict the
evolution of

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00:15:46,710 --> 00:15:50,160
the tide as a superposition of sinusoidal
components.

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00:15:50,160 --> 00:15:52,570
So this machine, that was originally
invented by

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00:15:52,570 --> 00:15:56,860
Lord Kelvin, is nothing but a mechanical
implementation

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00:15:56,860 --> 00:15:59,540
of the DFT reconstruction machine that we
just saw before.

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00:16:00,940 --> 00:16:03,100
So here's a question, what happens if you
turn

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00:16:03,100 --> 00:16:06,170
the crank of the machine more than big N
times?

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00:16:06,170 --> 00:16:10,562
Well it turns out that the output will
become periodic, in other words,

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00:16:10,562 --> 00:16:13,174
x of small n plus big N will be equal to x
of n.

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00:16:13,174 --> 00:16:16,055
This is actually quite apparent from the
structure

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00:16:16,055 --> 00:16:19,370
of the synthesis and analysis formulas of
the DFT.

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00:16:19,370 --> 00:16:21,510
Let's start from the synthesis formula.

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00:16:21,510 --> 00:16:22,160
So in

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00:16:22,160 --> 00:16:26,890
theory, the output index small n should go
from 0 to big N minus 1.

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00:16:26,890 --> 00:16:31,050
But you can see that it only appears here,
inside this complex exponential.

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00:16:31,050 --> 00:16:36,930
Now this guy is 2 pi periodic, so if you
push n over big N minus 1, this will

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00:16:36,930 --> 00:16:44,340
simply cycle over once again as if n was
looping over the 0 to n minus 1 range.

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00:16:44,340 --> 00:16:47,260
So in the end, you can actually safely
take n

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00:16:47,260 --> 00:16:51,990
to be from the set of integers and the
output

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00:16:51,990 --> 00:16:54,390
will be an N periodic signal in the time
domain.

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00:16:55,410 --> 00:16:57,970
Actually the same holds for the analysis
formula.

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00:16:57,970 --> 00:17:00,980
You can let the index k roam over the
entire

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00:17:00,980 --> 00:17:03,930
set of integers and since it appears only
in this

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00:17:03,930 --> 00:17:07,580
complex exponential here, again, it will
be as if k

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00:17:07,580 --> 00:17:11,860
was looping over the 0 to n minus 1 range.

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00:17:11,860 --> 00:17:12,590
So what happens

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00:17:12,590 --> 00:17:15,720
really, is that you can consider the
sequence of DFT

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00:17:15,720 --> 00:17:19,550
coefficients as an N periodic signal in
the frequency domain.

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00:17:21,200 --> 00:17:26,820
Now, when we talk explicitly about the
periodicity of the Fourier

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00:17:26,820 --> 00:17:32,570
transform, we prefer to use the term
discrete Fourier series or DFS for short.

280
00:17:32,570 --> 00:17:36,720
So the DFS is just the DFT with the
periodicity explicit.

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00:17:36,720 --> 00:17:38,270
The DFS maps

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00:17:38,270 --> 00:17:42,810
an N-periodic signal onto an N-periodic
sequence of Fourier coefficients.

283
00:17:42,810 --> 00:17:45,270
And the inverse DFS maps an N-periodic

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00:17:45,270 --> 00:17:49,130
sequence of Fourier coefficients onto an
N-periodic signal.

285
00:17:49,130 --> 00:17:53,360
But mathematically the two are the same,
it's just a conceptual difference.

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00:17:53,360 --> 00:17:58,100
However, DFS helps understand, for
instance, why we said that the

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00:17:58,100 --> 00:18:03,420
natural shift for finite length sequences
is actually a periodic shift.

288
00:18:03,420 --> 00:18:06,710
Let's consider an N-periodic sequence x
tilda of n.

289
00:18:06,710 --> 00:18:09,540
For periodic sequences, shifts are
well-defined.

290
00:18:09,540 --> 00:18:12,550
We can take x tilda of n minus big M

291
00:18:12,550 --> 00:18:16,000
and it's just a normal shift of an
infinite two-sided sequence.

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00:18:17,040 --> 00:18:22,600
If we take the DFS of this periodic
sequence with a shift, we can work out

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00:18:22,600 --> 00:18:28,690
very easily that the DFT coefficient for
index k is equal to the DFT coefficient

294
00:18:28,690 --> 00:18:31,790
for index k of the original sequence
without the shift.

295
00:18:31,790 --> 00:18:38,860
So, x of k is simply the DFS of tilde x of
n

296
00:18:40,960 --> 00:18:46,720
with a phase shift term, e to the minus j,
2 pi over n, big M times k.

297
00:18:46,720 --> 00:18:48,600
Similarly, it's very easy to verify that
the

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00:18:48,600 --> 00:18:51,670
inverse discrete Fourier series of a
vector of

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00:18:51,670 --> 00:18:56,030
DFS coefficients multiplied by a phase
delay factor,

300
00:18:56,030 --> 00:19:00,860
is simply the periodic sequence delayed by
M.

301
00:19:00,860 --> 00:19:02,980
So this term, e to the minus j, 2 pi

302
00:19:02,980 --> 00:19:07,420
over n, Mk is a delay factor in the
frequency domain.

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00:19:07,420 --> 00:19:09,130
So what happens for finite length signals?

304
00:19:10,160 --> 00:19:14,820
We saw that in this case, a shift is not a
well-defined operation and we need to

305
00:19:14,820 --> 00:19:18,070
embed the signal into an infinite length
signal, either

306
00:19:18,070 --> 00:19:21,260
by building a finite support or a periodic
signal.

307
00:19:22,280 --> 00:19:28,110
Now we can always build x tilda of n as x
of n modules big N.

308
00:19:28,110 --> 00:19:32,580
Now mathematically, the DFS of this
periodic signal is equal to

309
00:19:32,580 --> 00:19:37,530
the DFT of the original signal.
So the inverse DFT of the DFT

310
00:19:37,530 --> 00:19:42,700
of the signal times this delay factor,
that we showed in the previous slide,

311
00:19:42,700 --> 00:19:48,360
is numerically equal to the inverse DFS of
the same quantity.

312
00:19:48,360 --> 00:19:54,490
But we know that the inverse DFS of this
will be a delayed periodic sequence.

313
00:19:54,490 --> 00:19:57,610
And if we go back to the definition of how
we build this periodic

314
00:19:57,610 --> 00:20:02,990
sequence this will be simply x of n minus
big N modulus big N.

315
00:20:02,990 --> 00:20:07,680
What we have shown here is that if we
think of the Fourier representation of a

316
00:20:07,680 --> 00:20:13,550
finite length signal, circular shifts are
the natural extension of the shift

317
00:20:13,550 --> 00:20:16,550
operator to finite length signals because
the

318
00:20:16,550 --> 00:20:18,830
underlying Fourier representation is the
same as

319
00:20:18,830 --> 00:20:23,280
that what would use for a periodic symbol
built on the finite length signal.

