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INTRODUCTION:

Wavelets are a mathematical development that may revolutionize the world of information
storage and retrieval according to many experts. They are a fairly simple mathematical tool
now being applied to the compression of datasuch as fingerprints, weather satellite
photographs, and medical x-raysthat were previously thought to be impossible to condense
without losing crucial details. In mathematics, a wavelet series is a representation of a square-
integrable (real- or complex-valued) function by a certain orthonormal series generated by
a wavelet. Nowadays, wavelet transformation is one of the most popular of the time-frequency-
transformations. The fundamental idea of wavelet transforms is that the transformation should
allow only changes in time extension, but not shape. This is affected by choosing suitable basis
functions that allow for this. Changes in the time extension are expected to conform to the
corresponding analysis frequency of the basis function. Based on the uncertainty principle of
signal processing,

1
t w
2

where t represents time and angular frequency ( = 2f, where f is temporal frequency).
The higher the required resolution in time, the lower the resolution in frequency has to be. The
larger the extension of the analysis windows is chosen, the larger is the value of t

When t is large,

1. Good Bad time resolution


2. frequency resolution
3. Low frequency, large scaling factor
When t is small,

1. Good time resolution


2. Bad frequency resolution
3. High frequency, small scaling factor
WAVELET TRANSFORM:

First of all, why do we need a transform, or what is a transform anyway?


Mathematical transformations are applied to signals to obtain further information from
that signal that is not readily available in the raw signal. In the following tutorial I will
assume a time-domain signal as a raw signal, and a signal that has been "transformed" by
any of the available mathematical transformations as a processed signal.

There are a number of transformations that can be applied, among which the Fourier
transforms are probably by far the most popular.

Most of the signals in practice are TIME-DOMAIN signals in their raw format. That is,
whatever that signal is measuring, is a function of time. In other words, when we plot the
signal one of the axes is time (independent variable), and the other (dependent variable) is
usually the amplitude. When we plot time-domain signals, we obtain a time-amplitude
representation of the signal. This representation is not always the best representation of the
signal for most signal processing related applications. In many cases, the most distinguished
information is hidden in the frequency content of the signal. The frequency SPECTRUM of
a signal is basically the frequency components (spectral components) of that signal. The
frequency spectrum of a signal shows what frequencies exist in the signal.
Intuitively, we all know that the frequency is something to do with the change in rate
of something. If something (a mathematical or physical variable would be the technically
correct term) changes rapidly, we say that it is of high frequency, where as if this variable
does not change rapidly, i.e., it changes smoothly, we say that it is of low frequency. If this
variable does not change at all, then we say it has zero frequency, or no frequency. For
example the publication frequency of a daily newspaper is higher than that of a monthly
magazine (it is published more frequently).

So how do we measure frequency, or how do we find the frequency content of a signal? The
answer is FOURIER TRANSFORM (FT). If the FT of a signal in time domain is taken, the
frequency-amplitude representation of that signal is obtained. In other words, we now have 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.

The frequency axis starts from zero, and goes up to infinity. For every frequency, we have an
amplitude value. For example, if we take the 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. No other signal, however, has a FT which is this simple. For
most practical purposes, signals contain more than one frequency component. The
following shows the FT of the 50 Hz signal:
Figure 1.4 The FT of the 50 Hz signal given in Figure 1.3

Note that two plots are given in Figure 1.4. The bottom one plots only the first half of the top
one. Due to reasons that are not crucial to know at this time, the frequency spectrum of a real
valued signal is always symmetric. The top plot illustrates this point. However, since the
symmetric part is exactly a mirror image of the first part, it provides no additional
information. And therefore, this symmetric second part is usually not shown. In most of the
following figures corresponding to FT, I will only show the first half of this symmetric
spectrum.

WHY DO WE NEED THE FREQUENCY INFORMATION?

Often times, the information that cannot be readily seen in the time-domain can be seen in the
frequency domain.
Let's give an example from biological signals. Suppose we are looking at an ECG signal
(Electrocardiography, graphical recording of heart's electrical activity). The typical shape of a
healthy ECG signal is well known to cardiologists. Any significant deviation from that shape
is usually considered to be a symptom of a pathological condition.
This pathological condition, however, may not always be quite obvious in the original time-
domain signal. Cardiologists usually use the time-domain ECG signals which are recorded on
strip-charts to analyze ECG signals. Recently, the new computerized ECG recorders/analyzers
also utilize the frequency information to decide whether a pathological condition exists.
A pathological condition can sometimes be diagnosed more easily when the frequency
content of the signal is analyzed.
This, of course, is only one simple example why frequency content might be useful.
Today Fourier transforms are used in many different areas including all branches of
engineering. Although FT is probably the most popular transform being used (especially in
electrical engineering), it is not the only one. There are many other transforms that are used
quite often by engineers and mathematicians.

For a better understanding of the need for the WT, let's look at the FT more closely. FT (as
well as WT) is a reversible transform, that is, it allows going backward and forward between
the raw and processed (transformed) signals. However, only either of them is available at
any given time. That is, no frequency information is available in the time-domain signal, and
no time information is available in the Fourier transformed signal. The natural question that
comes to mind is that is it necessary to have both the time and the frequency information at the
same time?
As we will see soon, the answer depends on the particular application and the nature of
the signal in hand. Recall that the 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. This information is not required when the
signal is so- called stationary.
Let's take a closer look at this stationary concept more closely, since it is of paramount
importance in signal analysis. Signals whose frequency content does not change in time are
called stationary signals. In other words, the frequency content of stationary signals does not
change in time. In this case, one does not need to know at what times frequency
components exist, since all frequency components exist at all times!!! .
For e.g.
x(t ) cos(2 10t ) cos(2 25t ) cos(2 50t ) cos(2 100t )
is a stationary signal, because it has frequencies of 10, 25, 50, and 100 Hz at any given
time instant. This signal is plotted below:

Figure 1.5
And the following is its FT:

Figure 1.6
The top plot in Figure 1.6 is the (half of the symmetric) frequency spectrum of the signal in
Figure 1.5. The bottom plot is the zoomed version of the top plot, showing only the range of
frequencies that are of interest to us.

Contrary to the signal in Figure 1.5, the following signal is not stationary. Figure 1.7
plots a signal whose frequency constantly changes in time. This signal is known as the
"chirp" signal. This is a non-stationary signal.

Figure 1.7
Figure 1.8

FT gives the spectral content of the signal, but it gives no information regarding where in time
those spectral components appear. Therefore, FT is not a suitable technique for non-
stationary signal, with one exception: FT can be used for non- stationary signals, if we
are only interested in what spectral components exist in the signal, but not interested where
these occur. However, if this information is needed, i.e., if we want to know, what spectral
component occur at what time (interval), then Fourier transform is not the right transform to
use.
Once again please note that, the FT gives what frequency components (spectral
components) exist in the signal. No more, no less.

When the time localization of the spectral components are needed, a transform giving
the TIME-FREQUENCY REPRESENTATION of the signal is needed.

THE ULTIMATE SOLUTION: THE WAVELET TRANSFORM

The wavelet transform is a transform of this type. It provides the time-frequency


representation. (There are other transforms which give this information too, such as
short time Fourier transform, Wigner distributions, etc.)
Often times a particular spectral component occurring at any instant can be of particular
interest. In these cases it may be very beneficial to know the time intervals these particular
spectral components occur. For example, in EEGs, the latency of an event-related
potential is of particular interest (Event-related potential is the response of the brain to a
specific stimulus like flash-light, the latency of this response is the amount of time elapsed
between the onset of the stimulus and the response).
Wavelet transform is capable of providing the time and frequency information simultaneously,
hence giving a time-frequency representation of the signal.
The WT was developed as an alternative to the short time Fourier Transform (STFT) to
overcome some resolution related problems of the STFT.
To make a real long story short, we pass the time-domain signal from various high pass filters
and low pass filters, which filters out either high frequency or low frequency portions of the
signal. This procedure is repeated, every time some portion of the signal corresponding to
some frequencies being removed from the signal.
Here is how this works: Suppose we have a signal which has frequencies up to 1000 Hz. In
the first stage we split up the signal in to two parts by passing the signal from a high pass and
a low pass filter (filters should satisfy some certain conditions, so-called admissibility
condition) which results in two different versions of the same signal: portion of the signal
corresponding to 0-500 Hz (low pass portion), and 500-1000 Hz (high pass portion).
Then, we take either portion (usually low pass portion) or both, and do the same thing
again. This operation is called decomposition.
Assuming that we have taken the low pass portion, we now have 3 sets of data, each
corresponding to the same signal at frequencies 0-250 Hz, 250-500 Hz, 500-1000 Hz.
Then we take the low pass portion again and pass it through low and high pass filters; we now
have 4 sets of signals corresponding to 0-125 Hz, 125-250 Hz, 250-500 Hz, and 500-1000
Hz. We continue like this until we have decomposed the signal to a pre-defined certain
level. Then we have a bunch of signals, which actually represent the same signal, but all
corresponding to different frequency bands. We know which signal corresponds to which
frequency band, and if we put all of them together and plot them on a 3-D graph, we will have
time in one axis, frequency in the second and amplitude in the third axis. This will show
us which frequencies exist at which time (there is an issue, called "uncertainty principle",
which states that, we cannot exactly know what frequency exists at what time instance, but
we can only know what frequency bands exist at what time intervals.
The frequency and time information of a signal at some certain point in the time-frequency
plane cannot be known. In other words: 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 as follows:
Higher frequencies are better resolved in time, and lower frequencies are better resolved in
frequency. This means that, a certain high frequency component can be located better in time
(with less relative error) than a low frequency component. On the contrary, a low frequency
component can be located better in frequency compared to high frequency component.
Take a look at the following grid:

F
r
e
q

^
|******************************************* continuous
|* * * * * * * * * * * * * * * wavelet transform
|* * * * * * *
|* * * *
|* *
--------------------------------------------> Time
Interpret the above grid as follows: The top row shows that at higher frequencies we have
more samples corresponding to smaller intervals of time. In other words, higher frequencies
can be resolved better in time. The bottom row however, corresponds to low frequencies, and
there are less number of points to characterize the signal, therefore, low frequencies are not
resolved well in time.

^
fr
e
q
u
e
n
c
y
|
|
|
|
************************************************
*******
|
|
|
| * * * * * * * * * * * * * * * * * * *
discrete time
| wavelet
transform
|* * * * * * *
* * *
|
|* * *
* *
|* *
*
|----------------------------------------------------------> time

In discrete time case, the time resolution of the signal works the same as above, but now,
the frequency information has different resolutions at every stage too.

Below, are some examples of continuous wavelet transforms:


Let's take a sinusoidal signal, which has two different frequency components at two different
times:

Note the low frequency portion first, and then the high frequency.
Figure 1.10

The continuous wavelet transform of the above signal:

Figure 1.11
Consequently, the little peak in the plot corresponds to the high frequency components in
the signal, and the large peak corresponds to low frequency components (which appear
before the high frequency components in time) in the signal.

You might be puzzled from the frequency resolution shown in the plot, since it shows good
frequency resolution at high frequencies. Note however that, it is the good scale resolution
that looks good at high frequencies (low scales), and good scale resolution means poor
frequency resolution and vice versa.

MULTIRESOLUTION ANALYSIS &THE CONTINUOUS WAVELET:


Although the time and frequency resolution problems are results of a physical phenomenon
(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). MRA, as implied by its name, analyzes the signal at different frequencies with
different resolutions. Every spectral component is not resolved equally as was the case in the
STFT.
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. This
approach makes sense especially when the signal at hand has high frequency components for
short durations and low frequency components for long durations. Fortunately, the signals
that are encountered in practical applications are often of this type. For example, the
following shows a signal of this type. It has a relatively low frequency component throughout
the entire signal and relatively high frequency components for a short duration somewhere
around the middle.

Figure 2.1
THE CONTINUOUS WAVELET TRANSFORM

The continuous wavelet transform was developed as an alternative approach to the short time
Fourier transforms to overcome the resolution problem. The wavelet analysis is done in a
similar way to the STFT analysis, in the sense that the signal is multiplied with a function,
similar to the window function in the STFT, and the transform is computed separately for
different segments of the time-domain signal. However, 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 continuous wavelet transform is defined as follows:
CWTx ( , s) x ( , s)
1 * t
x(t ) s dt (3.1)
s
As seen in the above equation, the transformed signal is a function of two variables, and s,
the translation and scale parameters, respectively. is the transforming function, and it is
called the mother wavelet. The term mother wavelet gets its name due to two important
properties of the wavelet analysis as explained below:
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. 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 related to the location of the window, as the window is shifted
through the signal. This term, obviously, corresponds to time information in the transform
domain. However, we do not have a frequency parameter, as we had before for the STFT.
Instead, we have scale parameter which is defined as frequency. The term frequency is
reserved for the STFT. Scale is described in more detail in the next section.

THE SCALE

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 corresponding to
various scales are given as examples in the following figure.
Figure 2.2

Fortunately in practical applications, 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, as a mathematical operation, either dilates or compresses a signal. Larger scales
correspond to dilated (or stretched out) signals and small scales correspond to
compressed signals. All of the signals given in the figure are derived from the same cosine
signal, i.e., they are dilated or compressed versions of the same function. In the above figure,
s=0.05 is the smallest scale, and s=1 is the largest scale.
In terms of mathematical functions, 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.
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 THE CWT

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''. However,
depending on the signal, a complete transform is usually not necessary. For all practical
purposes, the signals are bandlimited, and therefore, computation of the transform for a
limited interval of scales is usually adequate.
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.
The wavelet is placed at the beginning of the signal at the point which corresponds to
time=0. The wavelet function at scale1is 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. In other words, it is
the value that corresponds to the point =0, s=1 in the time-scale plane.
The wavelet at scale s=1 is then shifted towards the right by amount to the location t= ,
and the above equation is computed to get the transform value at t= , 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 time-scale plane for the scale s=1 is now completed.
Then, s is increased by a small value. Note that, this is a continuous transform, and
therefore, both and s must be incremented continuously. However, if this transform
needs to be computed by a computer, then both parameters are increased by a sufficiently
small step size. This corresponds to sampling the time-scale plane.
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

Figure 2.3
In Figure 2.3, the signal and the wavelet function are shown for four different values of . The
signal is a truncated version of the signal shown in Figure 3.1. The scale value is 1,
corresponding to the lowest scale, or highest frequency. 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. At every location, it is multiplied by the signal. Obviously, the product is
nonzero only where the signal falls in the region of support of the wavelet, and it is zero
elsewhere. 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).

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 Figure 2.3 has spectral components comparable to the
window's width at s=1 around t=100 ms.
The continuous wavelet transform of the signal in Figure 2.3 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.

Figure 2.4
Figure 2.5

Figures 2.4 and 2.5 illustrate the same process for the scales s=5 and s=20, respectively.
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.

As a result, for every scale and for every time (interval), one point of the time-scale plane is
computed. The computations at one scale construct the rows of the time-scale plane, and the
computations at different scales construct the columns of the time-scale plane.

Now, let's take a look at an example, and see how the wavelet transform really looks like.
Consider the non-stationary signal in Figure 2.6.
Figure 2.6

Figure 2.7 is the continuous wavelet transform (CWT) of this signal.


Figure 2.7

Figure 2.8

THE WAVELET THEORY: A MATHEMATICAL APPROACH

This section describes the main idea of wavelet analysis theory, which can also be considered
to be the underlying concept of most of the signal analysis techniques. The FT defined by
Fourier use 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). The synthesis, or the
reconstruction, corresponds to computing the linear combination equation.

Basis Vectors
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. There may be more than one
basis for a vector space. However, all of them have the same number of vectors, and this
number is known as the dimension of the vector space. For example in two-dimensional
space, the basis will have two vectors.
k bk (3.2)
k
Equation 3.2 shows how any vector v can be written as a linear combination of the basis
vectors bk and the corresponding coefficients k .
This concept, given in terms of vectors, can easily be generalized to functions, by replacing
the basis vectors bk with basis functions k (t ) , and the vector with a function f (t ) .
Equation 3.2 then becomes

f (t ) kk (t ) (3.2a)
k
The complex exponential (sines and cosines) functions are the basis functions for the FT.
Furthermore, they are orthogonal functions, which provide some desirable properties for
reconstruction.
Let f(t) and g(t) be two functions in L2 (a, b) . L2 (a, b) denotes the set of square integrable
functions in the interval (a, b) . The inner product of two functions is defined by Equation 3.3:
b
f (t ), g (t ) f (t ).g (t ) dt (3.3)
a

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 , s (t ) :

CWTx ( , s) x , s x(t ). *, s (t )dt (3.4)


where,
1 t
,s (3.5)
s s
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. The calculated CWT coefficients refer to the closeness of the
signal to the wavelet at the current scale.

Inner Products, Orthogonality, and Orthonormality


Two vectors v, w are said to be orthogonal if their inner product equals zero:
, nn 0 (3.6)

Similarly, two functions f(t) and g(t) are said to be orthogonal to each other if their inner
product is zero:
f (t ), g (t ) a f (t ).g*(t )dt 0
b
(3.7)

A set of vectors {v1 , v2 ,...vn } is said to be orthonormal, if they are pair wise orthogonal to each
other, and all have length 1. This can be expressed as:
vm , vn mn
Similarly, a set of functions { k (t ) }, k=1,2,3,..., is said to be orthonormal if
*
a k (t )l (t ) dt 0, k l (orthogonality condition)
b
(3.8)
a k (t ) dx 1
b 2
(3.9)
b
*
k (t )l (t )dt kl (3.10)
a

where, kl is the Kronecker delta function, defined as:


1 if k l
kl (3.12)
0 if k l

The Mexican Hat wavelet is defined as the second derivative of the Gaussian function:
2
t
1
(t ) e 2 2
(3.15)
2
which is
t 2 t 2
2
1
(t ) e 2
. 1 (3.16)
2 3
2

The Morlet wavelet is defined as


t 2
(t ) e . e
iat 2
(3.16a)
where a is a modulation parameter, and is the scaling parameter that affects the width of
the window.

EXAMPLES

The following signal shown in Figure 3.11 belongs to a normal person.

Figure 3.11
and the following is its CWT.

Figure 3.12
and the Figure 3.13 plots the same transform from a different angle for better
visualization.
Figure 3.14
Figure 3.14 plots an event related potential of a patient diagnosed with Alzheimer's disease

Figure 3.14
Figure 3.15

and Figure 3.15 illustrates its CWT.


Here is another view from a different angle
Figure 3.16

WHY IS THE DISCRETE WAVELET TRANSFORM NEEDED

Although the discrete continuous wavelet transform enables the computation of the
continuous wavelet transform by computers, it is not a true discrete transform. As a matter
of fact, 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, on the other hand, requires a significant amount of computation time and
resources. The discrete wavelet transform (DWT), on the other hand, 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. The basic
concepts of the DWT will be introduced in this section along with its properties and the
algorithms used to compute it. As in the previous chapters, examples are provided to aid in the
interpretation of the DWT.
THE DISCRETE WAVELET TRANSFORM (DWT)

The foundations of the DWT go back to 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. Later in 1989,
Vetterli and Le Gall made some improvements to the subband coding scheme, removing the
existing redundancy in the pyramidal coding scheme. Subband coding is explained below.

THE SUBBAND CODING AND THE MULTIRESOLUTION ANALYSIS

The main idea is the same as it is in the CWT. A time-scale representation of a digital signal is
obtained using digital filtering techniques. Recall that the CWT is a correlation
between a wavelet at different scales and the signal with the scale (or the frequency)
being used as a measure of similarity. 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.
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.
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.
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:

x n h n x k .h n k

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.
The unit of frequency is of particular importance at this time. 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 Nyquists rate (which is twice
the maximum frequency that exists in the signal); that is, the Nyquists rate corresponds
to p rad/s in the discrete frequency domain. Therefore using Hz is not appropriate for
discrete signals. However, Hz is used whenever it is needed to clarify a discussion, since it
is very common to think of frequency in terms of Hz. It should always be remembered that
the unit of frequency for discrete time signals is radians.
After passing the signal through a half band lowpass filter, half of the samples can be
eliminated according to the Nyquists rule, since the signal now has a highest frequency of
p/2 radians instead of p radians. Simply discarding every other sample will sub sample the
signal by two, and the signal will then have half the number of points. 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. 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 numbers of samples are redundant. This doubles the scale.
This procedure can mathematically be expressed as

y[n] h[k ].x[2n k ]
k
Having said that, we now look how the DWT is actually computed: The DWT analyzes
the signal at different frequency bands with different resolutions by decomposing 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 decomposition of the signal into different frequency bands
is simply obtained by successive highpass and lowpass filtering of the time domain signal.
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 Nyquists
rule, since the signal now has a highest frequency of p/2 radians instead of p. The signal
can therefore be subsampled by 2, simply by discarding every other sample. This
constitutes one level of decomposition and can mathematically be expressed as follows:

yhigh [k ] x[n].g[2k n]
n

ylow[k ] x[n].h[2k n]
n

Where yhigh [k ] and ylow[k ] are the outputs of the highpass and lowpass filters, respectively,
after subsampling by 2.
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, can be repeated for further decomposition. At every level,
the filtering and subsampling will result in half the number of samples (and hence half the
time resolution) and half the frequency band spanned (and hence doubles the frequency
resolution). Figure 4.1 illustrates this procedure, where x[n] is the original signal to be
decomposed, and h[n] and g[n] are lowpass and highpass filters, respectively. The bandwidth
of the signal at every level is marked on the figure as "f".
Figure 4.1. The Sub-band Coding Algorithm
ORTHONORMAL WAVELET:
A function F L2 () is said to be band-limited if the support of f is contained in a finite
interval. Our purpose is to study orthonormal wavelets that are band-limited.
A function L2 () is called an orthonormal wavelet if it can be used to define a Hilbert
basis, that is a complete orthonormal system, for the Hilbert space L2 () of square
integrable functions.
The Hilbert basis is constructed as the family of functions { jk : j , k } by means
of dyadic translations and dilations of ,
j
jk ( x) 2 2 (2 j x k ) for integers j, k .

If under the standard inner product on L2 () ,



f , g f ( x) g ( x)dx

This family is orthonormal, it is an orthonormal system:



jk , lm jk lm ( x) dx jl km , where jl is the Kronecker delta.
Completeness is satisfied if every function h L2 () may be expanded in the basis as

h( x) c jk jk ( x)
j , k

with convergence of the series understood to be convergence in norm. Such a representation of


a function f is known as a wavelet series. This implies that an orthonormal wavelet is self-dual.
The integral wavelet transform is the integral transform defined as

1 x b
W f (a, b) f ( x)dx
| a | a
The wavelet coefficients c jk are then given by
c jk W f 2 j , k 2 j
Here, a 2 j is called the binary dilation or dyadic dilation, and b k 2 j is
the binary or dyadic position.

DAUBECHIES WAVELETS:
The Daubechies wavelets, based on the work of Ingrid Daubechies, are a family of orthogonal
wavelets defining a discrete wavelet transform and characterized by a maximal number of
vanishing moments for some given support. With each wavelet type of this class, there is a
scaling function (called the father wavelet) which generates an orthogonal multiresolution
analysis.
CONSTRUCTION:

Both the scaling sequence (Low-Pass Filter) and the wavelet sequence (Band-Pass Filter) will
here be normalized to have sum equal 2 and sum of squares equal 2. In some applications, they
are normalised to have sum 2 , so that both sequences and all shifts of them by an even
number of coefficients are orthonormal to each other.
Using the general representation for a scaling sequence of an orthogonal discrete wavelet
transform with approximation order A,
a( Z ) 21 A (1 Z ) A p(Z ) ,
with N=2A, p having real coefficients, p(1)=1 and degree(p)=A-1, one can write the
Orthogonality condition as
a( Z )a(Z 1 ) a( Z )a( Z 1 ) 4 ,
or equally as (2 X ) A P( X ) X A P(2 X ) 2 A (*)

with the Laurent-polynomial X : 1/ 2(2 Z Z 1 ) generating all symmetric sequences


and X ( Z ) 2 X ( Z ) .
Further, P(X) stands for the symmetric Laurent-polynomial
P( X ( Z )) p(Z ) p(Z 1 ) .
Since X (eiw ) 1 cos( w) and p(eiw ) p(e iw ) | p(eiw ) |2 , P takes nonnegative values on the
segment [0,2].
Equation (*) has one minimal solution for each A, which can be obtained by division in the
ring of truncated power series in X,
A1 A k 1 k k
PA ( X ) 2 X .
k 0
A 1
Obviously, this has positive values on (0,2)
The homogeneous equation for (*) is antisymmetric about X=1 and has thus the general
solution X A ( X 1) R(( X 1) 2 ) , with R some polynomial with real coefficients. That the sum

P( X ) P ( X ) X A ( X 1) R(( X 1)2 )
A
shall be nonnegative on the interval [0,2] translates into a set of linear restrictions on the
coefficients of R. The values of P on the interval [0,2] are bounded by some quantity 4 A r ,
maximizing r results in a linear program with infinitely many inequality conditions.

To solve P( X ( Z )) p( Z ) p( Z 1) for p one uses a technique called spectral factorization


resp. Fejr-Riesz-algorithm. The polynomial P(X) splits into linear
factors P( X ) ( X 1 )( X N ) , N=A+1+2deg(R) . Each linear factor represents a
1 1
Laurent-polynomial ( X ( Z ) ) Z 1 Z 1 that can be factored into two linear
2 2
factors. One can assign either one of the two linear factors to p(Z), thus one obtains 2N possible
solutions. For extreme phase one chooses the one that has all complex roots of p(Z) inside or
on the unit circle and is thus real.
For Daubechies wavelet transform, a pair of linear filters is being used. This pair of filters
should have a property which is called as quadrature mirror filter. Solving the coefficient of the
linear filter ci 's using the quadrature mirror filter property results in the below solution for the
coefficient values for filter of order 4.
1 3 3 3 3 3 1 3
c0 c1 c2 c3 c4
4 2 4 2 4 2 4 2

BALIAN LOW THEOREM:


The BalianLow theorem in Fourier analysis is named for Roger Balian and Francis E. Low.
The theorem states that there is no well-localized window function (or Gabor atom) g either
in time or frequency for an exact Gabor frame (Riesz Basis).
Suppose g is a square-integrable function on the real line, and consider the so-called Gabor
system
g m,n ( x) e2 imbx g ( x na),
for integers m and n, and a,b>0 are fixed lattice parameters satisfying ab=1
where g L2 () is a fixed window function.
The BalianLow theorem states that if
{g m,n : m, n }
is an orthonormal basis for the Hilbert space L2 (), then either

x | g ( x) | dx or | g ( ) | d .
2 2 2 2

In other words, if X and P are the position and momentum operators, respectively and
g L2 () generates a basis of coherent states, then X g and Pg both cannot be square-
integrable.
This is a no-go theorem for localizing phase space with a windowed Fourier transform. It states
that such functions must have infinitely long tails in either position space or momentum space.
The window function g maximizes the uncertainty principle in some sense. This result was
originally stated by Balian and Low independently. It is one of the best examples of the fact
that stability and good-time frequency localization cannot be achieved.

____________________________ .END. _________________

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