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Digital Signal Processing

Digital Signal Processing - Lecture 9 Soma Biswas teaches digital signal processing at IISc Bangalore. The class meets on Tuesdays and Thursdays from 5:15 to 6:40 pm in room EE 218. The document discusses representing periodic sequences using the discrete Fourier series. It explains that a periodic sequence with period N can be represented as a sum of N complex exponentials with frequencies that are integer multiples of the fundamental frequency. It also describes properties of the discrete Fourier series such as linearity, shift property, duality, and periodic convolution.
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0% found this document useful (0 votes)
29 views34 pages

Digital Signal Processing

Digital Signal Processing - Lecture 9 Soma Biswas teaches digital signal processing at IISc Bangalore. The class meets on Tuesdays and Thursdays from 5:15 to 6:40 pm in room EE 218. The document discusses representing periodic sequences using the discrete Fourier series. It explains that a periodic sequence with period N can be represented as a sum of N complex exponentials with frequencies that are integer multiples of the fundamental frequency. It also describes properties of the discrete Fourier series such as linearity, shift property, duality, and periodic convolution.
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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Digital Signal Processing

Soma Biswas
Department of Electrical Engineering
IISc, Bangalore
Office: 215 A (EE)
E-mail: [email protected]

Class Hours: Tuesday and Thursday 5:15 to 6.40 pm

Class Room: EE 218

Representation of periodic sequences:


The Discrete Fourier Series
Consider a sequence [] that is periodic with period , so that =
[ + ] for any integer value of and .
We will use tilde (~) to denote periodic sequences whenever it is important
to clearly distinguish between periodic and aperiodic sequence.
As with continuous-time periodic signals, such a sequence can be
represented by a Fourier series corresponding to a sum of harmonically
related complex exponential sequences,
i.e., complex exponentials with frequencies that are integer multiples of the
fundamental frequency (/) associated with the periodic sequence [].
These periodic complex exponentials are of the form

Where is an integer, and the Fourier series representation has the form

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The Fourier series representation of a continuous-time periodic signal


generally requires infinitely many harmonically related complex
exponentials.
Whereas the Fourier series of any discrete-time signal with period
requires only harmonically related complex exponentials.
Because of the inherent periodicity of the complex exponentials with
period .

is an integer. Consequently, the set of periodic complex exponentials


, , , defines all the distinct periodic complex exponentials
with frequencies that are integer multiples of (/).
Thus, the Fourier series representation of a periodic sequence []
contains only of these complex exponentials, and hence, it has a form

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The Fourier series coefficients [] are obtained by:

Note that the sequence [] is periodic with period N: i.e.,


= , = [ + ], and more generally,

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The Fourier series coefficients [] can be interpreted to be a sequence of


finite length, for = , , , ( ), and zero otherwise, or as a periodic
sequence defined for all .
Both of these interpretation are acceptable. Advantage of interpreting the
Fourier series coefficients as a periodic sequence is that there is a duality
between the time and frequency domains for the Fourier series
representation of periodic sequences.
For convenience in notation, we use = (/).
With this notation, the discrete Fourier series (DFS) analysis-synthesis pair
is expressed as follows:

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Example

Representation of periodic impulse train in terms of sum of complex exponentials.


Complex exponentials with same magnitude and phase adds to unity at integer multiples of
and zero for all other integer.

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Example

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Any periodic signal can be representation as


a sum of complex exponential sequences.

Digital Signal Processing - Lecture 9

Properties of DFS

Linearity:

Shift:

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If the shift is greater than or equal to the period, we have to take the modulo or
remainder when is divided by .

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Properties of DFS

Duality:

Periodic convolution:

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Fourier Transform of periodic signals


Uniform convergence of Fourier transform of a sequence requires the
sequence to be absolutely summable, and mean-square convergence
requires that the sequence be square summable.
Periodic sequences satisfy neither condition, because they do not
approach zero as approaches .
However, sequences that can be expressed as a sum of complex
exponentials can be considered to have a Fourier transform representation.
It is useful to incorporate the discrete Fourier series representation of
periodic signals within the framework of the Fourier transform.
This can be done by interpreting the Fourier transform of a periodic signal
to be an impulse trans in the frequency domain within the impulse values
proportional to the DFS coefficients for the sequence.
Specifically, if [] is periodic with period and the corresponding
discrete Fourier series coefficients are [] , then the Fourier transform of
[] is defined to be the impulse train

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Inverse Fourier transform

Fourier transform, DTFT

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Although the Fourier transform of periodic sequence does not converge in


the normal sense, the introduction of impulses permits us t include
periodic sequences formally within the framework of Fourier transform
analysis.
This approach can be used to obtain a Fourier transform representation of
other nonsummable sequences.

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In other words, the periodic sequence [] of DFS coefficients has an


interpretation as equally spaced samples of the Fourier transform of the
finite-length sequence obtained by extracting one period of [].

This corresponds to sampling the Fourier transform at equally spaced


frequencies between = and = with frequency spacing of /.

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Discrete Fourier Transform (DFT)

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Example

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Properties of DFT

Linearity:

Clearly if 1 [] has length 1 and 2 [] has length 2 , then the maximum length of
3 [] will be 3 = max[1 , 2 ].
Thus both the DFTs must be computed with the same length > 3 .
Zero padding is done to match length.

Circular shift:

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Duality:

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Properties of DFT

Symmetry:

Circular convolution:

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Circular convolution of finite length


Sequence with single delayed impulse

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Linear convolution using DFT


Efficient algorithms are available for computing the discrete Fourier
transform of finite-duration sequence. They are known as fast
Fourier Transform (FFT) algorithms.
Because of these algorithms are available, it is computationally
efficient to implement a convolution of two sequences by the
following procedure:
Compute N-point DFTs () and () of the two sequences
[] and [], respectively.
Compute the product = .
Compute the sequence = []
as the inverse DFT of .
This will result in circular convolution, however, in most cases we
want to implement linear convolution of two sequences, i.e., in case
of LTI systems.
To obtain linear convolution, we must ensure that the circular
convolution has the effect of linear convolution.

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Linear convolution of two finite-length sequences

Therefore, ( + 1) is the
maximum length of the sequence
3 [] resulting from the linear
convolution of a sequence of length
with a sequence of length .

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Time aliasing in the circular


convolution of two finite length
sequences can be avoided if +
1, Also it is clear that if = =
, all of the sequence values of the
circular convolution may be different
from those of the linear convolution.

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Implementing LTI system using DFT


Since LTY systems can be implemented by convolution, circular
convolution can be used to implement these systems.
Lets us first consider an L-point input sequence [] and a P-point
impulse response []. The linear convolution of these two
sequences, which will be denoted by [], has finite duration length
( + ).
If a circular convolution is done with at least ( + ) points, it
will be identical to linear convolution.
The circular convolution can be achieved by multiplying the DFTs
of [] and []. Both [] and [] must be augmented with
sequence values of zero amplitude. This process is often referred to
as zero-padding.
The output of a FIR system whose input also has finite length can
be computed with DFT.
In many applications, such as filtering of speech waveform, the
input signal is of indefinite duration. While, theoretically, we might
be able to store the entire waveform and then implement the
procedure using DFT for a large number of points, however, such
DFT is generally impractical to compute.
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Another drawback is that in this method, no filtered output samples


can be computed until all the input samples have been collected.
Generally we would like to avoid such delay in processing.
The solution to both problems is block convolution, in which the
signal to be filtered is segmented into sections of length .
Each section can then be convolved with the finite-length impulse
response and the filtered sections fitted together in an appropriate
way.
The linear filtering of each block can then be implemented using the
DFT.

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We assume = 0 < 0 and


the length of [] is much greater
than .
The sequence [] can be
represented as a sum of shifted finitelength segments of length ; i.e.,

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Overlap-save method

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