Name of Institution

ASET
M.Tech, 1st sem
Module 2

Time Frequency Analysis and Multi-resolution

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PART-I
Why wavelet Transform?

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 OVERVIEW: WHY WAVELET
TRANSFORM? Name of Institution

• Raw signal v/s Transformed signal

• Time domain signal v/s frequency domain signal
• How do we measure frequency? –by FT
• FT gives us a plot with one axis being the
frequency and the other being the amplitude. This plot
tells us how much of each frequency exists in our signal.
• FT of the electric current that we use in our houses, we
will have one spike at 50 Hz, and nothing elsewhere,
since that signal has only 50 Hz frequency component

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FT of AC mains Name of Institution

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 Why do we need the
frequency information? Name of Institution

• The information that cannot be readily seen in the time-

domain can be seen in the frequency domain.
• Transforms like Hilbert transform, STFT, Wigner
distributions, the Radon Transform, the wavelet
transform etc.
• Fourier and wavelet transform are reversible transforms.
• FT gives the frequency information of the signal, which
means that it tells us how much of each frequency exists
in the signal, but it does not tell us when in time these
frequency components exist

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 Fourier Transform Name of Institution

• Time information is not required if signal is stationary (all

frequency component exists at all the times)
• Example: x(t)=cos(2*pi*10*t)+cos(2*pi*25*t)
+cos(2*pi*50*t)+cos(2*pi*100*t)
• Plot of signal is:

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• Its FT is:

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• Plot of non stationary signal(chirp signal):

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Its FT is Name of Institution

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• It has four peak frequencies at 10,25,50

and 100 Hz.
• It is very similar to the FT of that stationary
signal.
• Avoid the ripples as lower amplitude is not
important.
• Hence, the FT of a stationary and non
stationary signal is same i.e FT shows
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• Only frequency components, it doesn’t tell

which frequency component exits at what
time. Hence time frequency resolution is
poor in FT.
• The ultimate solution to this problem is
“Wavelet Transform”

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• WT is capable of providing the time and

frequency information simultaneously,
hence giving a time-frequency
representation of the signal.
• It should be explained after STFT.

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 How WT works? Name of Institution

• we pass the time-domain signal from various HPF and

LPF, which filters out either HF or LF portions of the
signal. This procedure is repeated, every time some
portion of the signal corresponding to some frequencies
being removed from the signal.
• Let we have 1000Hz frequency signal. It is passed
through LPF and HPF. We got 0-500 Hz and 500-
1000Hz signal from LPF and HPF repectively.
• Then say LF signal 0-500Hz is again passed through
LPF and HPF now we got 0-250,250-500, and 500-
1000Hz signal.

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• This process is known as decomposition. It is repeated

untill we got the pre defined level.
• Now we have a bunch of signals, which
actually represent the same signal, but all corresponding
to different frequency bands.
• If we plot them on a 3-D graph, we will have time in one
axis, frequency in the second and amplitude in the third
axis.

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 Heisenberg uncertainty principle
Name of Institution

• We cannot know what spectral component exists at any

given time instant.
• The best we can do is to investigate what spectral
components exist at any given interval of time. This is a
problem of resolution, and it is the main reason why
researchers have switched to WT from STFT.
• STFT gives a fixed resolution at all times, whereas WT
gives a variable resolution.
• In WT Higher frequencies are better resolved in time,
and lower frequencies are better resolved in frequency.

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• In the above figure the top row shows high frequency

components which are best resolved in time and at lower
row we have less no. of components of low frequency
which are better resolved in frequency.

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PART-II
THE FOURIER TRANSFORM
AND
THE SHORT TERM FOURIER
TRANSFORM

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FT and IFT Name of Institution

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• x(t)=cos(2*pi*5*t)+cos(2*pi*10*t)+cos(2*pi*20*t)+cos(2*pi*50*t)
is:

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FT of signal Name of Institution

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Non stationary signal
Name of Institution

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FT Name of Institution

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 STFT Name of Institution

• What if the part that we can consider to be stationary is

very small?
• This approach of researchers ended up with a revised
version of the Fourier transform, so-called : The
STFT.
• In STFT, the signal is divided into small enough
• segments, where these segments (portions) of the signal
can be assumed to be stationary
• window function "w" is chosen. The width of this
window must be equal to the segment of the signal
where its stationarity is valid.
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 STFT Name of Institution

• Where x(t) is the signal itself, w(t) is the window

function, and * is the
complex conjugate.
• As you can see from the equation, the STFT of the
signal is nothing but the FT of the signal multiplied by a
window function.
• The following figure will help you for its better
understanding:
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These will correspond to three different FTs at three

different times. Therefore, we will obtain a true time-
frequency representation
(TFR) of the signal 27
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Let's take a non-stationary signal, such as the following one

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 IT’s STFT Name of Institution

The interval 0 to 250 ms is a simple sinusoid of 300 Hz, and the other 250 ms
intervals are sinusoids of 200 Hz, 100 Hz, and 50 Hz, respectively.
Apparently, this is a non-stationary signal. Now, let's look at its STFT:

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• note that the graph is symmetric with respect to midline

of the frequency axis.
• FT of a real signal is always symmetric, since STFT is
nothing but a windowed version of the FT, it should
come as no surprise that STFT is also symmetric in
frequency.
• note that there are four peaks corresponding to four
different frequency components.
• Also note that, unlike FT, these four peaks are located
at different time intervals along the time axis .
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• we have a true time-frequency representation of the

signal.
• We not only know what frequency components are
present in the signal, but we also know where they are
located in time.
• since STFT gives the TFR of the signal, why do we need
the wavelet transform
• Reason lies in Heisenberg Uncertainty Principle.
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• One cannot know what spectral components exist at

what instances of times. What one can know are the
time intervals in which certain band of frequencies exist,
which is a resolution problem.
• The problem with the STFT has something to do with the
width of the window function that is used.
• This width of the window function is known as the
support of the window. If the window function is narrow,
than it is known as compactly supported.
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• Window used in the FT is its kernel, the exp{jwt}

function, which lasts at all times from minus infinity to
plus infinity.
• Now, in STFT, our window is of finite length, thus it
covers only a portion of the signal, which causes the
frequency resolution to get poorer.
• What I mean by getting poorer is that, we no longer
know the exact frequency components that exist in the
signal, but we only know a band of frequencies that
exist.
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• why don't we make the length of the window in the STFT

infinite, just like as it is in the FT, to get perfect frequency
resolution?
• Ans: you loose all the time information, you basically end
up with the FT instead of STFT.
• Narrow window ===>good time resolution, poor
frequency resolution.
• Wide window ===>good frequency resolution, poor time
resolution.
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• Say we have four windows of different length, and we will

use these to compute the STFT, and see what happens:
• The window function we use is simply a Gaussian
function in the form:
w(t)=exp(-a*(t^2)/2)
• ‘a’ is the length of the Windows and t Is time.

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• The above example given was computed

with the second value, a=0.001 .
• STFT of the same signal given above
computed with the other windows
• We expect the STFT to have a very good
time resolution,but relatively poor
frequency resolution:

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STFT Name of Institution

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• four peaks are well separated from each

other in time.
• Frequency domain, every peak covers a
range of frequencies, instead of a single
frequency value.
• Now let‘s make the window wider, and
look at the third window.

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• Note that the peaks are not well separated

from each other in time, unlike the
previous case,
• however, in frequency domain the
resolution is much better.
• Now let's further increase the width of the
window, and see what happens

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• Well, this should be of no surprise to

anyone now, since we would expect a
terrible (and I mean absolutely terrible)
time resolution.

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 Problem in STFT Name of Institution

• These examples should have illustrated the implicit

problem of resolution of the STFT.
• Anyone who would like to use STFT is faced with this
problem of resolution. What kind of a window to use?
• Narrow windows give good time resolution, but poor
frequency resolution.
• Wide windows give good frequency resolution, but poor
time resolution; furthermore, wide windows may violate
the condition of stationarity.
• The best solution is to use Wavelet transform.

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END
of
PART II

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 PART-III Name of Institution

MULTIRESOLUTION ANALYSIS
&
THE CONTINUOUS WAVELET
TRANSFORM

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 MULTIRESOLUTION ANALYSIS
Name of Institution

• Time and frequency resolution problems are results of

the Heisenberg uncertainty principle and exist regardless
of the transform used,
• It is possible to analyze any signal by using an
alternative approach called the multiresolution
analysis (MRA)
• It analyzes the signal at different frequencies with
different resolutions
• Every spectral component is not resolved equally as was
the case in the STFT.

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• MRA is designed to give good time resolution

and poor frequency resolution at high
frequencies and good frequency resolution and
poor time resolution at low frequencies.
• signal at hand has high frequency components
for short durations and low frequency
components for long durations.
 For eg Name of Institution

It has a relatively low frequency component

throughout the entire signal and relatively high frequency
components for a short duration
 THE CONTINUOUS WAVELET
Name of Institution

TRANSFORM
• It is an alternative approach to the STFT to overcome
the resolution problem.
• There are two main differences between the STFT and
the CWT:
1. The Fourier transforms of the windowed signals are not
taken, and therefore single peak will be seen
corresponding to a sinusoid, i.e., negative frequencies
are not computed.
2. The width of the window is changed as the transform is
computed for every single spectral component, which is
probably the most significant characteristic of the
wavelet transform.
 The CWT Name of Institution

• the transformed signal is a function of two variables,

tau and s , the translation and scale parameters,
• psi(t) is the transforming function, and it is called the
mother wavelet
• The term mother wavelet gets its name due to two
important properties
1. The term wavelet means a small wave . The
smallness refers to the condition that this (window)
function is of finite length ( compactly supported).
The wave refers to the condition that this function is
oscillatory .
 Name of Institution

• The term mother implies that the functions with

different region of support that are used in the
transformation process are derived from one main
function, or the mother wavelet.
• In other words, the mother wavelet is a prototype for
generating the other window functions.
• The term translation is used in the same sense as it
was used in the STFT; it is related to the location of the
window, as the window is shifted through the signal
• we have scale parameter which is defined as
1/frequency.
 The Scale Name of Institution

• The parameter scale in the wavelet analysis is

similar to the scale used in maps.
• As in the case of maps, high scales correspond
to a non-detailed global view (of the signal), and
low scales correspond to a detailed view.
• Similarly, in terms of frequency, low frequencies
(high scales) correspond to a global information
of a signal (that usually spans the entire signal),
whereas high frequencies (low scales)
correspond to a detailed information of a hidden
pattern in the signal (that usually lasts a
relatively short time)
 Cosine signals Name of Institution

corresponding to various scales

 Name of Institution

• low scales (high frequencies) do not last for the

entire duration of the signal, unlike those shown
in the figure, but they usually appear from time
to time as short bursts, or spikes.
• High scales (low frequencies) usually last for the
entire duration of the signal.
• Scaling either dilates or compresses a signal.
• Larger scales correspond to
• dilated (or stretched out) signals and small
scales correspond to compressed signals
 Name of Institution

• In the above figure, s=0.05 is the smallest

scale, and s=1 is the largest scale.
• If f(t) is a given function f(st) corresponds
to a contracted (compressed) version of
f(t) if s > 1 and to an expanded (dilated)
version of f(t) if s < 1.
 Name of Institution

• However, in the definition of the wavelet

transform, the scaling term is used in the
denominator, and therefore, the opposite
of the above statements holds, i.e., scales
s > 1 dilates the signals whereas scales s
<1 , compresses the signal
 COMPUTATION OF Name of Institution

THE CWT
• Let x(t) is the signal to be analyzed
• mother wavelet is chosen to serve as a prototype for all
windows in the process
• All the windows that are used are the dilated (or
compressed) and shifted versions of the mother wavelet.
• The Morlet wavelet and the Mexican hat function used
for the wavelet analysis
• Once the mother wavelet is chosen the computation
starts with s=1 and the continuous wavelet transform is
computed for all values of s , smaller and larger than
``1''.
 Name of Institution

• In this study, some finite interval of values for s

were used.
• For convenience, the procedure will be started
from scale s=1 and will continue for the
increasing values of s , i.e., the analysis will start
from high frequencies and proceed towards low
frequencies
• This first value of s will correspond to the most
compressed wavelet. As the value of s is
increased, the wavelet will dilate
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• The wavelet is placed at the beginning of the signal at

the point which corresponds to time=0.
• The wavelet function at scale ``1'' is multiplied by the
signal and then integrated over all times
• The result of the integration is then multiplied by the
constant number 1/sqrt{s}
• This multiplication is for energy
• normalization purposes so that the transformed signal
will have the same energy at every scale
• The final
• result is the value of the transformation, i.e., the value of
the continuous wavelet transform at time zero and scale
s=1
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• In other words, it is the value that corresponds to the

point tau =0 , s=1 in the time-scale plane
• The wavelet at scale s=1 is then shifted towards the right
by tau amount to the location t=tau , and the above
equation is computed to get the transform value at
t=tau , s=1 in the time-frequency plane
• This procedure is repeated until the wavelet reaches the
end of the signal. One row of points on the timescale
plane for the scale s=1 is now completed
• Then, s is increased by a small value.
• this is a continuous transform, and therefore, both tau
and s must be incremented continuously
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• The above procedure is repeated for every value

of s
• Every computation for a given value of s fills the
corresponding single row of the time-scale plane
• When the process is completed for all desired
values of s, the CWT of the signal has been
calculated.
• The figures below illustrate the entire process
step by step
Name of Institution
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• Note how compact it is (the blue window). It

should be as narrow as the highest frequency
component that exists in the signal.
• Four distinct locations of the wavelet function
are shown in the figure at to=2 , to=40, to=90,
and to=140 .
• By shifting the wavelet in time, the signal is
localized in time, and by changing the value of
s,the signal is localized in scale (frequency).
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• If the signal has a spectral component that

corresponds to the current value of s (which is 1
in this case), the product of the wavelet with the
signal at the location where this spectral
component exists gives a relatively large value.
• If the spectral component that corresponds to
the current value of s is not present in the signal,
the product value will be relatively small, or zero
• The signal in has spectral components
comparable to the window's width at s=1 around
t=100 ms.
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• The continuous wavelet transform of the

signal in will yield large values for low
scales around time 100 ms, and small
values elsewhere
• For high scales, on the other hand, the
continuous wavelet transform will give
large values for almost the entire duration
of the signal, since low frequencies exist at
all times.
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• Note how the window width changes with

increasing scale (decreasing frequency).
• As the window width increases, the
transform starts picking up the lower
frequency components.
Non stationary signal
Name of Institution
CWT Name of Institution
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• Note that the axes are translation and scale, not time
and frequency.
• translation is strictly related to time, since it indicates
where the mother wavelet is located
• Scale S is actually inverse of frequency.
• In other words, whatever we said about the properties of
the wavelet transform regarding the frequency
resolution, inverse of it will appear on the figures
showing the WT of the time-domain signal.
 Name of Institution

• that smaller scales correspond to higher

frequencies i.e., frequency decreases as
scale increases,
• therefore, that portion of the graph with
scales around zero, actually correspond to
highest frequencies in the analysis, and
that with high scales correspond to lowest
frequencies
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• Remember that the signal had 30 Hz (highest

frequency) components first, and this appears at
the lowest scale at a translations of 0 to 30.
• Then comes the 20 Hz component, second
highest frequency, and so on the 5 Hz
component appears at the end of the translation
axis (as expected), and at higher scales (lower
frequencies) again as expected.
Name of Institution
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• lower scales (higher frequencies) have

better scale resolution which correspond
to poorer frequency resolution .
 THE WAVELET THEORY: Name of Institution
A MATHEMATICAL APPROACH
• The FT basis functions to analyze and reconstruct a
function.
• Every vector in a vector space can be written as a
linear combination of the basis vectors in that vector
space , i.e., by multiplying the vectors by some
constant numbers, and then by taking the summation of
the products.
• The analysis of the signal involves the estimation of
these constant numbers (transform coefficients, or
Fourier coefficients, wavelet coefficients,etc).
 Basis Vectors Name of Institution

• subscripts are shown by the underscore character _ ,

and superscripts are shown by the ^ character.
• Also note that all letters or letter names written in
bold type face represent vectors, Some important
points are also written in bold face
• A basis of a vector space V is a set of linearly
independent vectors, such that any vector v in V can
be written as a linear combination of these basis vectors.
• For example in two-dimensional space, the basis will
have two vectors.
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•shows how any vector v can be written as a linear

combination of the basis vectors b_k and
the corresponding coefficients nu^k.
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• This concept, given in terms of vectors,

can easily be generalized to functions, by
replacing the basis vectors b_k with basis
functions phi_k(t), and the vector v with
a function f(t). Then above equation
becomes:

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• The complex exponential (sines and

cosines) functions are the basis functions
for the FT
• Let f(t) and g(t) be two functions in L^2
[a,b]. Their inner product is:

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• According to the above definition of the

inner product, the CWT can be thought of
as the inner product of the test signal with
the basis functions psi_(tau ,s)(t):

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• where,

•This definition of the CWT shows that the wavelet analysis is a measure of
similarity between the basis functions (wavelets) and the signal itself.
• Here the similarity is in the sense of similar frequency content.

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 Inner Products, Orthogonality, Name of Institution
and Orthonormality

• Two vectors v , w are said to be

orthogonal if their inner product equals
zero

Similarly, two functions f(t) and g(t) are said to be orthogonal to each
other if their inner product is zero

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• A set of vectors {v_1, v_2, ....,v_n} is said

to be orthonormal , if they are pairwise
orthogonal to each other, and all have
length ``1''. This can be expressed as:

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• Similarly, a set of functions {phi_k(t)},

k=1,2,3,..., is said to be orthonormal if

and

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• or equivalently

where, delta_{kl} is the Kronecker delta function, defined as

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• As stated above, there may be more than one set of

basis functions (or vectors).
• Among them, the orthonormal basis functions (or
vectors) are of particular importance because of the nice
properties they provide in finding these analysis
coefficients.
• The orthonormal bases allow computation of these
coefficients in a very simple and straightforward way
using the orthonormality property.

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• For orthonormal bases, the coefficients, mu_k , can be

calculated as

and the function f(t) can then be reconstructed as

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• Orthonormal bases may not be available for every type

of application where a generalized version,biorthogonal
bases can be used.
• The term ``biorthogonal'' refers to two different bases
which are orthogonal to each other, but each do not form
an orthogonal set
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• The Mexican Hat wavelet is defined as the

second derivative of the Gaussian function

Which is
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• The Morlet wavelet is defined as

where a is a modulation parameter, and sigma is the scaling parameter

that affects the width of the window
EXAMPLES Name of Institution
 Name of Institution

• CWT was computed at 350 translation and 60 scale

locations
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 THE WAVELET Name of Institution

SYNTHESIS
• The CWT is a reversible transform
provided that the below condition is
satisfied even though the basis functions
are in general may not be orthonormal

……1
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• The reconstruction is possible by using the

following reconstruction formula

•where C_psi is a constant that depends on the wavelet used

•The success of the reconstruction depends on this constant called,
the admissibility constant
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• Equation 1 implies that psi^hat(0) = 0,

which is

As stated above, is not a very restrictive requirement since many wavelet

functions can be found whose integral is zero. For Equation to be satisfied,
the wavelet must be oscillatory.
 Discretization of the Continuous Name of Institution
Wavelet Transform: The Wavelet Series

• It is apparent that neither the FT, nor the STFT, nor the
CWT can be practically computed by using analytical
equations, integrals, etc
• It is therefore necessary to discretize the transforms
• As in the FT and STFT, the most intuitive way of doing
this is simply sampling the time-frequency (scale) plane
• Again intuitively, sampling the plane with a uniform
sampling rate sounds like the most natural choice.
• However, in the case of WT, the scale change can be
used to reduce the sampling rate
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• At higher scales (lower frequencies), the sampling rate

can be decreased, according to Nyquist's rule
• In other words, if the time-scale plane needs to be
sampled with a sampling rate of N_1 at scale s_1 , the
same plane can be sampled with a sampling rate of
N_2 , at scale s_2 , where, s_1 < s_2 (corresponding to
frequencies f1>f2 ) and N_2 < N_1 . The actual
relationship between N_1 and N_2 is
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•In other words, at lower frequencies the sampling rate can be

decreased which will save a considerable amount of computation
time
•It should be noted at this time, however, that the discretization can
be done in any way without any restriction as far as the analysis of
the signal is concerned
•If synthesis is not required, even the Nyquist criteria does not
need to be satisfied. The restrictions on the discretization and the
sampling rate become important if, and only if, the signal
reconstruction is desired.
•Nyquist's sampling rate is the minimum sampling rate that allows
the original continuous time signal to be reconstructed from its
discrete samples
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• As mentioned earlier, the wavelet psi(tau,s)

satisfying Equation 1, allows reconstruction of
the signal
• However, this is true for the continuous
transform.
• The question is: can we still reconstruct the
signal if we discretize the time and scale
parameters?
• The answer is ``yes'', under certain conditions
 Name of Institution

• The scale parameter s is discretized first

on a logarithmic grid
• The time parameter is then discretized
with respect to the scale parameter ,
i.e., a different sampling rate is used for
every scale
• In other words, the sampling is done on
the dyadic sampling grid shown below
Name of Institution
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• The CWT assigns a value to the continuum of points on

this plane.
• Therefore, there are an infinite number of CWT
coefficients
• Among that infinite number of points, only a finite
number are taken, using a logarithmic rule.
• The base of the logarithm depends on the user. The
most common value is 2 because of its convenience. If 2
is chosen, only the scales 2, 4, 8, 16, 32, 64,...etc. are
computed. If the value was 3, the scales 3, 9, 27, 81,
243,...etc. would have been computed.
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• The time axis is then discretized according to the

discretization of the scale axis.
• Since the discrete scale changes by factors of 2 , the
sampling rate is reduced for the time axis by a factor of 2
at every scale.
• Note that at the lowest scale (s=2), only 32 points of the
time axis are sampled (for the particular case given
in above Figure ).
• At the next scale value, s=4, the sampling rate of time
axis is reduced by a factor of 2 since the scale is increased
by a factor of 2, and therefore, only 16 samples are taken.
 Name of Institution

• At the next step, s=8 and 8 samples are taken in time,

and so on.
• Although it is called the time-scale plane, it is more
accurate to call it the translation-scale plane
• Similar to the relationship between continuous Fourier
transform, Fourier series and the discrete Fourier
transform, there is a continuous wavelet transform, a
semi-discrete wavelet transform (also known as wavelet
series) and a discrete wavelet transform.

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• Expressing the above discretization

procedure in mathematical terms, the
scale discretization is s = s_0^j ,and
translation discretization is tau =
k.s_0^j.tau_0 where s_0>1 and tau_0>0
• Note, how the translation discretization is
dependent on scale discretization with s_0

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The continuous wavelet function

by inserting s = s_0^j , and tau = k.s_0^j.tau_0 .

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If {psi_(j,k)} constitutes an orthonormal

basis, the wavelet series transform
becomes
…..1

or

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• A wavelet series requires that {psi_(j,k)}

are either orthonormal, biorthogonal, or
frame. If {psi_(j,k)} are not orthonormal,
Equation 1 becomes:

where hat{ psi_{j,k}^*(t)} , is either the dual biorthogonal basis or

dual frame

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• If {psi_(j,k) } are orthonormal or

biorthogonal, the transform will be non-
redundant,
• where as if they form a frame, the
transform will be redundant.
• On the other hand, it is much easier to find
frames than it is to find orthonormal or
biorthogonal bases.

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END
OF
PART- III

113
 PART IV Name of Institution

MULTIRESOLUTION ANALYSIS:

THE DISCRETE

WAVELET TRANSFORM

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 Why is the Discrete Name of Institution
Wavelet Transform Needed?
• The wavelet series is simply a sampled version of the
CWT, and the information it provides is highly redundant
as far as the reconstruction of the signal is concerned.
• This redundancy requires a significant amount of
computation time and resources.
• The DWT provides sufficient information both for
analysis and synthesis of the original signal, with a
significant reduction in the computation time.
• The DWT is considerably easier to implement when
compared to the CWT

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 THE DISCRETE WAVELET Name of Institution

TRANSFORM (DWT)
• In 1976 when Croiser, Esteban, and Galand devised a
technique to decompose discrete time signals.
• Crochiere, Weber, and Flanagan did a similar work on
coding of speech signals in the same year. They named
their analysis scheme as subband coding.
• In 1983, Burt defined a technique very similar to
subband coding and named it pyramidal coding which
is also known as multiresolution analysis

116
 The Subband Coding and Name of Institution
The Multiresolution Analysis
• The continuous wavelet transform was computed by
changing the scale of the analysis window, shifting the
window in time, multiplying by the signal, and integrating
over all times
• In the discrete case, filters of different cutoff frequencies
are used to analyze the signal at different scales.
• The signal is passed through a series of high pass filters
to analyze the high frequencies, and it is passed through
a series of low pass filters to analyze the low
frequencies.

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 Downsampling Name of Institution

• The resolution of the signal, which is a measure of the

amount of detail information in the signal, is changed by
the filtering operations, and the scale is changed by
upsampling and downsampling (subsampling)operations
• Subsampling a signal corresponds to reducing the
sampling rate, or removing some of the samples of the
signal
• For example, subsampling by two refers to dropping
every other sample of the signal.
• Subsampling by a factor n reduces the number of
samples in the signal n times

118
 Upsampling Name of Institution

• Upsampling a signal corresponds to increasing the

sampling rate of a signal by adding new samples to the
signal.
• For example, upsampling by two refers to adding a new
sample, usually a zero or an interpolated value, between
every two samples of the signal.
• Upsampling a signal by a factor of n increases the
number of samples in the signal by a factor of n.
 Name of Institution

• DWT coefficients are usually sampled from the CWT on

a dyadic grid, i.e., s_0 = 2 and t _0 = 1, yielding s=2j and
t =k*2j
• The procedure starts with passing this signal (sequence)
through a half band digital lowpass filter with impulse
response h[n].
• Filtering a signal corresponds to the mathematical
operation of convolution of the signal with the impulse
response of the filter. The convolution operation in
discrete time is defined as follows:
 Name of Institution

• A half band lowpass filter removes all

frequencies that are above half of the
highest frequency in the signal.
• For example, if a signal has a maximum of
1000 Hz component, then half band
lowpass filtering removes all the
frequencies above 500 Hz.
 Name of Institution

• In discrete signals, frequency is expressed

in terms of radians.
• Accordingly, the sampling frequency of the
signal is equal to 2p radians in terms of
radial frequency. Therefore, the highest
frequency component that exists in a
signal will be p radians, if the signal is
sampled at Nyquist’s rate
 Name of Institution

• After passing the signal through a half band

lowpass filter signal now has a highest
frequency of p/2 radians instead of p radians
• Simply discarding every other sample will
subsample the signal by two, and the signal will
then have half the number of points.
• Simply discarding every other sample will
subsample the signal by two, and the signal will
then have half the number of points
 Name of Institution

• The scale of the signal is now doubled. Note that

the lowpass filtering removes the high frequency
information, but leaves the scale unchanged
• Only the subsampling process changes the
scale
• Resolution, on the other hand, is related to the
amount of information in the signal, and
therefore, it is affected by the filtering operations
• Half band lowpass filtering removes half of the
frequencies, which can be interpreted as losing
half of the information. Therefore, the resolution
is halved after the filtering operation.
 Name of Institution

• Note, however, the subsampling operation after

filtering does not affect the resolution, since
removing half of the spectral components from
the signal makes half the number of samples
redundant anyway.
• Half the samples can be discarded without any
loss of information.
• In summary, the lowpass filtering halves the
resolution, but leaves the scale unchanged. The
signal is then subsampled by 2 since half of the
number of samples are redundant. This doubles
the scale.
 Name of Institution

• This procedure can mathematically be

expressed as
 DWT computation Name of Institution

• The DWT decompose the signal into a coarse

approximation and detail information
• DWT employs two sets of functions, called scaling
functions and wavelet functions, which are associated
with low pass and highpass filters, respectively
• The original signal x[n] is first passed through a halfband
highpass filter g[n] and a lowpass filter h[n].
• After the filtering, half of the samples can be eliminated
according to the Nyquist’s rule, since the signal now has
a highest frequency of p /2 radians instead of p
 Name of Institution

• This constitutes one level of

decomposition and can mathematically be
expressed as follows:

where yhigh[k] and ylow[k] are the outputs of the highpass and lowpass
filters, respectively, after subsampling by 2

128
 Subband coding Name of Institution

• This decomposition halves the time resolution

since only half the number of samples now
characterizes the entire signal.
• However, this operation doubles the frequency
resolution, since the frequency band of the
signal now spans only half the previous
frequency band, effectively reducing the
uncertainty in the frequency by half
• The above procedure, which is also known as
the subband coding
Name of Institution
 Name of Institution

• This process continues until two samples are

left. For this specific example there would be 8
levels of decomposition, each having half the
number of samples of the previous level
• The DWT of the original signal is then obtained
by concatenating all coefficients starting from the
last level of decomposition
• The DWT will then have the same number of
coefficients as the original signal.
 Name of Institution

• The difference of this transform from the Fourier

transform is that the time localization of these
frequencies will not be lost.
• If the main information of the signal lies in the high
frequencies, as happens most often, the time localization
of these frequencies will be more precise, since they are
characterized by more number of samples.
• If the main information lies only at very low frequencies,
the time localization will not be very precise, since few
samples are used to express signal at these frequencies.
• This procedure in effect offers a good time resolution at
high frequencies, and good frequency resolution at low
frequencies. Most practical signals encountered are of
this type.
Name of Institution
 Name of Institution

• Figure shows a typical 512-sample signal that is normalized to unit

amplitude.
• The horizontal axis is the number of samples, whereas the vertical
axis is the normalized amplitude.
• Figure b shows the 8 level DWT of the signal Fig a.
• The last 256 samples in this signal correspond to the highest
frequency band in the signal, the previous 128 samples correspond
to the second highest frequency band and so on.
• It should be noted that only the first 64 samples, which correspond
to lower frequencies of the analysis, carry relevant information and
the rest of this signal has virtually no information.
• Therefore, all but the first 64 samples can be discarded without any
loss of information. This is how DWT provides a very effective data
reduction scheme
 Name of Institution

• One important property of the DWT is the

relationship between the impulse
responses of the HPF &LPF. The HPF &
LPF are not independent of each other,
and they are related by
 Name of Institution

• where g[n] is the HPF, h[n] is the LPF, and L is the filter
length (in number of points).
• Two filters are odd index alternated reversed versions of
each other. LPF to HPF conversion is provided by the (-
1)n term.
• Filters satisfying this condition are commonly used in
signal processing, and they are known as the
Quadrature Mirror Filters (QMF).
• The two filtering and subsampling operations can be
expressed by
 Name of Institution

• The reconstruction in this case is very

easy since halfband filters form
orthonormal bases
• The above procedure is followed in
reverse order for the reconstruction
• The signals at every level are upsampled
by two, passed through the synthesis
filters g’[n], and h’[n] (highpass and
lowpass, respectively), and then added
 Name of Institution

• The interesting point here is that the

analysis and synthesis filters are identical
to each other, except for a time reversal.
• Therefore, the reconstruction formula
becomes (for each layer)
 Name of Institution

• However, if the filters are not ideal halfband,

then perfect reconstruction cannot be achieved
• Although it is not possible to realize ideal filters,
under certain conditions it is possible to find
filters that provide perfect reconstruction.
• The most famous ones are the ones developed
by Ingrid Daubechies, and they are known as
Daubechies’ wavelets
 Name of Institution

• Note that due to successive subsampling

by 2, the signal length must be a power of
2, or at least a multiple of power of 2, in
order this scheme to be efficient.
• The length of the signal determines the
number of levels that the signal can be
decomposed to.
• For example, if the signal length is 1024,
ten levels of decomposition are possible.
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END
OF
PART-IV