SPARKS@ELECTROMANIA 2K9 
IMPLEMENTATION OF ANALOG TO INFORMATION COVERTER
1.Harishchandra dubey(E.C.E),2.Prateesh Raj(E.E)
1B.Tech ECE, Motilal Nehru National Institute of Technology,
E-mail: harish.dubey123@gmail.com
2
B.Tech EE, Motilal Nehru National Institute of Technology,
E-mail: xyz@abc.com
Abstract
In this paper, I have implemented anolog to
information (by information I mean something giving
the details input analog signal .In most cases ,we are
required it in the form of an image or another analog
signal) conversion based on theory of compressed
Sensing (also known as sampling).Sampling process
coverts the continuous time signals to sequences of
numbers (samples) that can be processed and stored in
digital devices (like buffer or registers).I have used
buffer for the storage of sampled signals. It is more
appropriate to do sampling of signals by considering
them as union of linear subspaces rather a single linear
space. .It is because union of sub spaces is a more
appropriate mathematical model for analog signals
than a single linear vector space.Instead .Recent
development in areas of compressive Sensing
(Sampling) and mathemtics including convex
programming,Uniform
principle and linear optimization theory have inspired
me do this project.Compressive Sensing allows us to
do sampling at a rate comparable to that of signal
frequency unlike the traditional Sensing (which was
based on Nyquist theorem).The stable sampling
condition is related to important concept of Restricted
Isometry Property(RIP)which helps in selection of
samples containing maximum information about input
signal.
Firstly,Fourier Sampling Algorithm is applied on
sampled signals. For storage of sampled signals we use
buffer(which is unity gain op-amp in our case).The
transformed signals are analysed by some appropriate
algorithms.(in our case it is based on convex
programming and uniform uncertainity principle)
.Upto this stage we have collected only those samples
that contain maximum information about signal. Now,
if required we can reconstruct the signal using
algorithm 890 in SPARCO : toolbox of MATLAB,else
required information can be taken from collected
samples. With the power of MATLAB (language of
mathematical and scientific calculations) ,we have thus
implemented a converter which not only eliminates the
problem posed by Nyquist sampling theorem but also
can serve multiple purposes for us.
1. Introduction
In real life-situations ,we mostly deal with analog
signals but the processing of signals is done on digital
systems due to a number of reasons.Also at the same
time after doing required manipulations or so-called
processing of digital signals we have to convert it back
into some information (like an image) or in another
analog signal so that it can be used by another device
or as some information .
In accordance with Nyquist Sampling theorem it is
very difficult to sample high bandwidth signal
especillay those in radio frequencies because of high
sampling rate required for efficient sampling.But ,it is
not the end ,today we are having the modern theory of
Compressive sensing/sampling which can be used to
eliminate the problems posed by Nyquist sampling
theorem and at the same time getting a satisfactory
level of accuracy in sparse recovery of signals.
Following the same way ,I have tried to implement it a
different way,instead of using all the sample we will be
using only a few which contain the maximum
information .The selection of these samples is based on
some mathematical theory involved.
2. PROBLEM STATEMENT
Many problems in radar and communication signal
processing involve radio frequency (RF) typically 30-
300 GHz signals of very high bandwidth. This presents
a serious challenge to systems that might attempt to
use a high-rate Analog-to-Digital Converter (ADC) to
sample these signals, as prescribed by the
Shannon/Nyquist Sampling Theorem(according to
which the highest frequency that can be accurately
represented is less than one half of the sampling rate ).
The dashed vertical lines are sample intervals, and the
blue dots are the crossing points – the actual samples
SPARKS@ELECTROMANIA 2K9 
taken by the conversion process. The sampling rate
here is below the Nyquist frequency, so when were
reconstruction the waveform we see the problem quite
readily.The power, stability, and low cost of digital
signal processing (DSP) have pushed the analog-to-
digital converter (ADC) increasingly close to the front-
end of many important sensing, imaging, and
communication systems. The power, stability, and low
cost of digital signal processing (DSP) have pushed the
analog-to-digital converter (ADC) increasingly close
to the front-end of many important sensing, imaging,
and
communication systems. The power, stability, and low
cost of digital signal processing (DSP) have pushed the
analog-to-digital converter (ADC) increasingly close
to the front-end of many important sensing, imaging,
and communication systems.
The power, stability, and low cost of digital signal
processing (DSP) have pushed the analog-to-digital
converter (ADC)increasingly close to the front-end of
many important sensing, imaging, and communication
Systems.Unfortunately, many systems, especially those
operating in the radio frequency (RF) bands, severely
stress current ADC technologies.For example, some
important radar and communications applications
would be best served by an ADC sampling over 5
GSample/s and resolution of over 20 bits, a
combination that greatly exceeds current capabilities.
It could be decades before ADCs based on current
technology will be fast and precise enough for these
applications.And even after better ADCs become
available, the deluge of data will swamp back-end DSP
algorithms. For example, sampling a 1GHz band using
2 GSample/s at 16 bits-persample generates data at a
rate of 4GB/s, enough to fill a modern hard disk in
roughly one minute. In a typical application, only a
tiny fraction of this information is actually relevant;
the wideband signals in many RF applications often
have a large bandwidth but a small “information rate”.
Fortunately, recent developments in mathematics and
signal processing have uncovered a promising
approach to the ADC bottleneck that enables sensing
at a rate comparable to the signal’s information rate. A
new field, known as Compressive Sensing (CS)
establishes mathematically that a relatively small
number of non-adaptive, linear measurements can
harvest all of the information necessary to faithfully
reconstruct sparse or compressible signals.
Traditional sampling theories consider the problem of
reconstructing an unknown signal x from a series of
samples. A prevalent assumption which often
guarantees recovery from the given measurements is
that x lies in a known subspace. Recently, there has
been growing interest in nonlinear but structured signal
models, in which x lies in a union of subspaces. In this
project I develop a general framework for efficient
recovery of such signals from a given set of samples.
More specifically, we treat the case in which x lies in
a sum of k subspaces, chosen from a larger set of m
possibilities. The samples are modelled as inner
products with an arbitrary set of sampling functions.
To derive an efficient recovery algorithm, we show
that our problem can be formulated as that of
recovering a block-sparse vector whose non-zero
elements appear in fixed blocks. Our main result is an
equivalence condition under which the proposed
Convex Algorithm along with uniform uncertainity
principle is guaranteed to recover the original signal.
This result relies on the notion of block restricted
isometry property (RIP), which is a generalization of
the standard RIP used extensively in the context of
compressed sensing. Based on RIP we also prove
stability of our approach in the presence of noise and
have a large bandwidth but a small “information rate”.
Fortunately, recent developments in mathematics and
signal processing have uncovered a promising
approach to the ADC bottleneck that enables sensing at
a rate comparable to the signal’s information rate. A
new field, known as Compressive Sensing (CS)
establishes mathematically that a relatively small
number of non-adaptive, linear measurements can
harvest all of the information necessary to faithfully
reconstruct sparse or compressible signals.
Traditional sampling theories consider the problem of
reconstructing an unknown signal x from a series of
samples. A prevalent assumption which often
guarantees recovery from the given measurements is
that x lies in a known subspace. Recently, there has
been growing interest in nonlinear but structured signal
models, in which x lies in a union of subspaces. In this
project I develop a general framework for efficient
recovery of such signals from a given set of samples.
More specifically, we treat the case in which x lies in a
sum of k subspaces, chosen from a larger set of m
possibilities. The samples are modelled as inner
products with an arbitrary set of sampling functions.
To derive an efficient recovery algorithm, we show
that our problem can be formulated as that of
recovering a block-sparse vector whose non-zero
elements appear in fixed blocks. Our main result is an
equivalence condition under which the proposed
Convex Algorithm along with uniform uncertainity
principle is guaranteed to recover the original signal.
This result relies on the notion of block restricted
isometry property (RIP), which is a generalization of
the standard RIP used extensively in the context of
SPARKS@ELECTROMANIA 2K9 

compressed sensing. Based on RIP we also prove

stability of our approach in the presence of noise and Here denotes the standard
modeling errors. Adapting our results to this context
leads to new MMV recovery methods as well as
equivalence conditions under which the entire set can
be determined efficiently. inner product on Ή.For example, if Ή =L 2 is
the space of real finite energy signals then
3.PROPOSED SOLUTION
3.1. OUTLINE OF SOLUTION  
Our“Analog-To-Informationconverter” (AIC)is
inspired by the recent theory of Compressive Sensing
(CS), which states that a discrete signal having a
sparse representation in some domain can be
recovered from a small number of linear projections of
that signal. We generalize the CS theory to continuous- …………………..(2)
time sparse signals, explain our proposed AIC system
in the CS context, and discuss practical issues
regarding implementation.
 
Analog signals are sampled considering them as union
of linear sub spaces rather than a single space.In most
of the practical applications it is found union of sub
spaces is a better model for the signal of intertest that
using Fast Fourier Transform and then stored in a
buffer .Using Convex Programming and Uniform …………..(3)
Uncertainity Principle we take the sparsed sampled Non-linear sampling is also there but our focus will be
signals which collect maximum information.Finally on the Linear one only.When Ή= IRN ,the unknown x
using 890 algorithm in SPARCO toolbox of matlab = x as well as the sampling functions si = si are vectors
we reconstruct the signal as information in some in IRN. Therefore, the samples can be written
desired format. conveniently in matrix form as y = ST x, where S is the
matrix with columns functions si .In the more general
3.2..”UNION OF FINITE DIMENSIONAL case in which Ή =L2 or any other abstract Hilbert
LINEAR SUB SPACES MODEL” FOR space, we can use the set transformation notation in
SIGNALS OF INTEREST order to conveniently represent the samples. A set
transformation S : IRN Ή corresponding to sampling
A.Subspace Sampling vectors
Traditional sampling theory deals with the problem of
{ si € Ή, 1 ≤ i ≤ n} is defined by
recovering an unknown signal ϰ ∈ Ή from a set of n
samples yi=fi(ϰ) where fi(ϰ) is some function of ϰ. The
n
signal ϰ can be a function of time ϰ = x(t), or can
Sc = Σ si
represent a finite-length vector ϰ = x. The most
i=1
common type of sampling is linear sampling in which
……………………(4)
yion here. for all c € IRN. From the definition of the adjoint,
if c = s*x, then c(i) = <si,x>.
Note that when Ή= IR N , S = S and S* = ST .Using
this notation, we can always express the samples as
…………… (1)
for a set of function y= S* x
…………………….(5)

where S is a set transformation for arbitrary Ή and an

appropriate matrix when Ή= IRN.Our goal is to
SPARKS@ELECTROMANIA 2K9 
recover x from the samples y ∈IRN. If the vectors si do
not span the entire space Ή , then there are many
possible signals x consistent with y. More specifically,
if we define by S the sampling space spanned by the
vectors si, then clearly S* v= 0 for any v
€S*.Therefore, if S*. is not the trivial space then adding
such a vector v to any solution xof (5) will result in
the same samples y. However, by exploiting prior
knowledge on x in many cases uniqueness can be
guaranteed. A prior very often assumed is that lies in
a given subspace A of Ή . If A and S have the same
finite dimension, and S⊥and A intersect only at the 0
vector, then x can be perfectly recovered from the
samples y.
B. Union of Subspaces
When subspace information is available, perfect
reconstruction can often be guaranteed.Furthermore,
recover can be implemented by a simple linear
transformation of the given samples (5). However,
there are many practical scenarios in which we are
given prior information about x that is not necessarily
in the form of a subspace.Here we focus our attention
on the setting where x lies in a union of subspaces
n k
u=U v i ……i=1……(6)
where each v i is a subspace. Thus, x belongs to one
of the v i , but we do not know a priori to which one.
Note that the set u is no longer a subspace. Indeed, if
v i is, for example, a one-dimensional space spanned
by the vector vi, then U contains vectors of the form
αvi, for some i but does not include their linear
combinations. Our goal is to recover a vector ϰ lying
in a union of subspaces, from a given set of samples.
In principle, if we knew which subspace ϰ belonged
to, then reconstruction can be obtained using standard
sampling results. However, here the problem is more
involved because conceptually we first need to identify
the correct subspace and only then can we recover the
signal within the space. Previous work on sampling
over a union focused on invertibility and stability
results in some generaliastions which are useful for us.
To achieve this goal,we limit our attention to a
subclass of (6) for which stable recovery algorithms
can be developed and analyzed. Specifically, we treat
the case in which each v i has the additional structure
v i= AJ
|j| = k
…… …………………………
(7)
where { A J , 1 ≤ j ≤ m} are a given set of disjoint
subspaces, and |j| = k denotes a sum over k indices.
Thus, each subspace vi corresponds to a different
choice of
subspaces AJ that comprise the sum. We assume
throughout the paper that m and the dimensions di =
dim(AJ ) of the subspaces A J are finite. Given n
samples
y= S* x
……………..(8)
and the knowledge that x lies in exactly one of the
subspaces v i ,we would like to recover the unknown
signal x .In this setting, there are
possible subspaces comprising the
union.An alternative interpretation of our model is as
follows. Given an observation vector y, we seek a
signal xfor which y= S* x and in addition x can be
written as
ϰ = Σ xi
i=1
…………………………………. (9)
where each ϰi lies in AJ for some index j.A special
case is the standard CS problem in which x = x is a
vector of length N, that has a sparse representation in
a given basis defined by an invertible matrix W. Thus,
x = Wc where c is a sparse vector that has at most k
nonzero elements. This fits our framework by
choosing A i as nthe space.
spanned by the ith column of W. In this setting m = N,
and there are
subspaces comprising the union.
A.Problem Formulation and Main Results
Given k and the subspaces A i, ,we would like to
address the following questions:
1) What are the conditions on the sampling vectors si,
1 ≤ i ≤ n in order to guarantee that the sampling is
invertible and stable?
2) How can we recover the unique x (regardless of
computational complexity)?
SPARKS@ELECTROMANIA 2K9 
3) How can we recover the unique x in an efficient and
stable manner? However, no concrete methods were
proposed in order to recover x. Here we provide
efficient convex algorithms that recover x in a stable
way for arbitrary k under appropriate conditions on the
sampling functions si another spaces A i. My results
are based on an equivalence between the union of
subspaces problem assuming (7) and that of recovering
block-sparse vectors. This allows us to recover x from
the given samples by first treating the problem of
recovering a block k-sparse vector c from a given set
of measurements. This relationship is established in
the next section. In the reminder of the paper we
therefore focus on the block k-sparse model and
develop our results in that context. In particular, we
introduce a block RIP condition that ensures
uniqueness and stability of our sampling problem. We
then suggest an efficient convex optimization problem
which approximates an unknown block-sparse vector
c. Based on block RIP we prove that c can be
recovered exactly in a stable way using the proposed
optimization program. Furthermore, in the presence of
noise and modeling errors, this algorithm can
approximate the best block-k sparse solution.
D. UNIQUENESS AND STABILITY
In this section we study the uniqueness and stability of
our sampling method. These properties are intimately
related to the RIP, which we generalize here to the
block-sparse setting.The first questi we address is that
of uniqueness, namely conditions under which a block-
sparse vector c is uniquely determined by the
measurement vector y = Dc.
Proposition 1: There is a unique block-k sparse vector
c consistent with the measurements y = Dc if and only
if Dc ǂ 0 for every c ǂ 0 that is block 2k-sparse.
Proof: The proof follows from [22, Proposition 4]. We
next address the issue of stability. A sampling operator
is stable for a set T if and only if there exists constants
α> 0, β < ∞ such that
α II x 1 −x 2 II 2 Ή <= IIS* x 1 -- S* x 2 II2 <= βIIx 1 −x 2 II
2
Ή most M nonzero elements. If we require D to satisfy
…………… (18)
for every x 1, x 2 in T . The ratio κ = β/α provides a
measure for stability of the sampling operator. The
operator is maximally stable when κ = 1. In our
settings S is replaced by D, and the set T contains
block-k sparse vectors. The following proposition
follows immediately from (18) by noting that given
two block-k sparse vectors c1,c2 their difference c1 −c2
is block-2k sparse.
Proposition 2: The measurement matrix D is stable for
every block k-sparse vector c if and only if there exists
c1 > 0 and c2 < ∞ such that
c1 II vII 22 <= II Dv II 22 <= II vII 22 ………(19)
for every v that is block 2k-sparse.It is easy to see that
if D satisfies (19) then Dc ǂ 0 for all block 2k-sparse
vectors c. Therefore, this condition implies both
invertibility and stability .
A.Block RIP
Property (19) is related to the RIP used in several
previous works in CS [9], [13], [14]. A matrix D of
size n × N is said to have the RIP if there exists a
constant δk€ [0, 1) such that for every k-sparse c €IRN,
(1 − δk) II c II2 2 <= II Dv II 22 <= (1 + δk) II c II2 2
…………. (20)
Extending this property to block-sparse vectors leads
to the following definition:
Definition 2: Let D : IRN →IRN be a given matrix.
Then D has the block RIP over I = {d1, . . . , dm} with
parameter δk|I if for every c ∈ IRN that is block k-
sparse over I we have that
(1 − δk/I) II c II2 2 <= II Dv II 22 <= (1 +
δk/I) II c II2 2
…………………..
(21)
By abuse of notation, we use δk for the block-RIP
constant δk|I when it is clear from the context that we
refer to blocks. Block-RIP is a special case of the A
-restricted isometry defined in [23]. From Proposition
1 it follows that if D satisfies the RIP (21) with δ2k <
1, then there is a unique block-sparse vector c
consistent with (16). Note that a block k-sparse vector
over I is M-sparse in the conventional sense where M
is the sum of the k largest values in I, since it has at
RIP for all M-sparse vectors, then (21) must hold for
all 2M-sparse vectors c. Since we only require the RIP
for block sparse signals, (21) only has to be satisfied
for a certain subset of 2M sparse signals, namely those
that have block sparsity. As a result, the block-RIP
constant δk|I is typically smaller than δM (where M
depends on k; for blocks with equalsize d, M = kd).To
emphasize the advantage of block RIP over standard
RIP, consider the following matrix, separated into three
blocks of two columns each:
SPARKS@ELECTROMANIA 2K9 

  uniquely recovered by solving the optimization

problem

Min

…………………. (23)
To show that (23) will indeed recover the true value of
c, suppose that there exists a c′ such that Dc′ = y and II
………………………(22) c′II 0,I <= II c′II 0,I <= k. Since both c, c′ are consistent
with the measurements,
where B is a diagonal matrix that results in unit-norm
columns of D, i.e., B = diag (1, 15, 1, 1, 1, 12)−1/2 . In 0 = D(c − c′) = Dd, ……………
this example m = 3 and I = {d1 = 2,d2 = 2,d3 = 2}. ………(24)
Suppose that c is block-1 sparse, which corresponds to
at most two non-zero values. Brute-force calculations
show that the smallest value of δ2 satisfying the where II d II 0,I <= 2k so that d is a block 2k-sparse
standard RIP (20) is δ2 = 0.866. On the other hand, the vector. Since D satisfies(21) with δ2k < 1, we must
block-RIP (21) corresponding to the case in which the have that d = 0 or c = c′.
two non-zero elements are restricted to occur in one
block is satisfied with δ1|I = 0.289. Increasing the 3.3.The Fast Fourier TransformAlgorithm
number of non-zero elements to k = 4, we can verify
that the standard RIP (20) does not hold for any δ4 € [0, This is how the DFT may be computed efficiently.
1). Indeed, in this example there exist two 4-sparse 1D Case
vectors that result in the same measurements. In
contrast, δ2|I = 0.966 satisfies the lower bound in (21)
when restricting the 4 non-zero values to two blocks.
Consequently, the measurements y = Dc uniquely
specify a single block-sparse c. In the next section, we
has to be evaluated for N values of u, which if done in
will see that the ability to recover c in a
computationally efficient way depends on the constant the obvious way clearly takes multiplications.It is
δ2k|I in the block RIP (21). The smaller the value possible to calculate the DFT more efficiently than
ofδ2k|I , the fewer samples are needed in order to this, using the fast Fourier transform or FFT
guarantee stable recovery. Both standard and block algorithm, which reduces the number of operations to
RIP constants δk,δk|I are by definition increasing with
k. Therefore, it was suggested in [12] to normalize
each of the columns of D to 1, so as to start with δ1 = 0. .
In the same spirit, we recommend choosing the bases We shall assume for simplicity that N is a power of 2,
for A I such that D = S* A has unit-norm columns, . If we define to be the root of unity
corresponding to δ1|I = 0. given by

B. Recovery Method ,
We have seen that if D satisfies the RIP (21) with δ2k and set M=N/2, we have
< 1, then there is a unique block-sparse vector c
consistent with (16). The question is how to find c in
practice. Below we present an algorithm that will in
principle find the unique c from the samples y.
Unfortunately, though, it has exponential complexity. This can be split apart into two separate sums of
In the next section we show that under a stronger alternate terms from the original sum,
condition on δ2k we can recover c in a stable and
efficient manner.Our first claim is that c can be
SPARKS@ELECTROMANIA 2K9 

Thus, we can compute an N-point DFT by dividing it

into two parts:

The first half of F(u) for

Now, since the square of a root of unity is an can be found from

Eqn. 28,
root of unity, we have that
• The second half for
can be found simply be reusing the same
and hence terms differently as shown by Eqn. 30.
• This is obviously a divide and conquer
method.
To show how many operations this requires, let T(n) be
the time taken to perform a transform of size
, measured by the number of multiplications
performed. The above analysis shows that
10
If we call the two sums demarcated above

the first term on the right hand side coming from the
and two transforms of half the original size, and the second
term coming from the multiplications of

respectively,then we have
by

.
Induction can be used to prove that

Note that each of

and A similar argument can also be applied to the number

of additions required, to show that the algorithm as a
whole takes time.
for
.
is in itself a discrete Fourier transform over N/2=M Also Note that the same algorithm can be used with a
points. little modification to perform the inverse DFT too.
How does this help us? Going back to the definitions of the DFT and its
Well inverse,

and we can also write

10
SPARKS@ELECTROMANIA 2K9 
and
If we take the complex conjugate of the second
equation, we have that
This now looks (apart from a factor of 1/N) like a
forward DFT, rather than an inverse DFT. Thus to
compute an inverse DFT,
• take the conjugate of the Fourier space data,
• put conjugate through a forward DFT
algorithm,
• take the conjugate of the result, at the same
time multiplying each value by N.
2D Case :
The same fast Fourier transform algorithm can be used
-- applying the separability property of the 2D
transform.
Rewrite the 2D DFT as
The right hand sum is basically just a one-dimensional
DFT if x is held constant. The left hand sum is then
another one-dimensional DFT performedwith the
numbers that come out of the first set of sums.
So we can compute a two-dimensional DFT by
• performing a one-dimensional DFT for each
value of x, i.e. for each column of f(x,y), then
• performing a one-dimensional DFT in the
opposite direction (for each row) on the
resulting values.
This requires a total of 2 N one dimensional
transforms, so the overall process takes time
3.4. STORAGE OF SAMPLED SIGNALS
For storage of sampled signals I have used buffer as
described below.I have designed a non-inverting
amplifier with a gain of exactly 1.The gain for a non-
inverting amplifier is given by the formula :
So, if we make R2 zero, and R1 infinity, we'll have an
amp with a gain of exactly 1. How can we do this? The
circuit is surprisingly simple. Here, R2 is a plain wire,
which has effectively zero resistance. We can think of
R1 as an infinite resistor -- we don't have any
connection to ground at all.This arrangement is called
an Op-Amp Follower, or Buffer. The buffer has an
output that exactly mirrors the input (assuming it's
within range of the voltage rails), so it looks kind of
useless at first.
However, the buffer is an extremely useful circuit,
since it helps to store the signal for sometime . The
input impedance of the op-amp buffer is very high:
close to infinity. And the output impedance is very
low: just a few ohms.
3.5.SPARSE RECOVERY BY 890 ALGORITHM
USING SPARCO TOOLBOX OF MATLAB
Now in this section ,I am giving a brief account of
SPARCO with emphasis on applicatiopn part .Sparco
is organized as a fexible framework providing test
problems for sparse signal recon-struction as well as a
library of operators and tools. The problem suite
currently contains 25 problems and 28 operators.The
latest version of Sparco and releted stuffs like
installation guides ,prerequisites code forc sparase
SPARKS@ELECTROMANIA 2K9 

MRIB toolbox ,test problems packaged with the GPSR y = D(x,1); % gives y := Dx
solver can be obtained from x = D(y,2); % gives x := Dty
www.cs.ubc.ca/labs/scl/sparco.Also an open source A full list of the basic operators available in the Sparco
pacakage Rice Wavelet Toolbox can be of great help.A library is given in Tables 3 and 4:
brief description of various sparco operators is as
follows : Matlab classes can be used to overload operations
commonly used for matrices so that the objects in that
At the core of the Sparco architect architecture is a class behave exactly like explicit matrices. Although
large library of linear operators. Where possible, this mechanism is not used for the implementation of
specialized code is used for fast evaluation of matrix- the Sparco operators, operator overloading can provide
vector multiplications. Once an operator has been a very convenient interface for the user. To facilitate
created D = opDCT(128) this feature, Sparco provides the function classOp:
matrix-vector products with the created operator can
be accessed as follows:
Matlab function Description:

opBinary binary (0/1) ensemble

opBlockDiag compound operator with operators on the diagonal
opBlur two-dimensional blurring operator
opColumnRestrict restriction on matrix columns
opConvolve1d one-dimensional convolution operator
opCurvelet2d two-dimensional curvelet operator
opDCT one-dimensional discrete cosine transform
opDiag scaling operator
opDictionary compound operator with operators abutted
opDirac identity operator
opFFT one-dimensional FFT
opFFT2d two-dimensional FFT
opFFT2C centralized two-dimensional FFT
opFoG subsequent application of a set of operators
opGaussian Gaussian ensemble
opHaar one-dimensional Haar wavelet transform
opHaar2d two-dimensional Haar wavelet transform
opHeaviside Heaviside matrix operator
opKron Kronecker product of two operators
opMask vector entry selection mask
opMatrix wrapper for matrices
opPadding pad and unpad operators equally around each side
opReal discard imaginary components
opRestriction vector entry restriction
opSign sign-ensemble operator
opWavelet wavelet operator
opWindowedOp overcomplete windowed operator

opWindowedOp overcomplete windowed operator

Table 3: The operators in the Sparco library

C = classOp(op); % Create matrix object C from op

C = classOp(op,'nCprod'); % Additionally, create a global counter variable nCprod

the main matrix-vector operations are de_ned. In its second form, the classOp function
SPARKS@ELECTROMANIA 2K9 

H = opDictionary(A1,A2,...); % H := [A1 j A2 j _ _ _ j
These calls take an operator op and return an object An]
from the operator class for which H = opTranspose(A); % H := AT
the main matrix-vector operations are de_ned. In its H = opBlockDiag(A1,A2,...); % H := diag(A1;A2; : : :)
second form, the classOp function
accepts an optional string argument and creates a H = opKron(A1,A2); % H := A1 A2
global variable that keeps track of the
number of multiplications with C and CT . The A sixth meta-operator, opWindowedOp, is a mixture
variable can be accessed from Matlab's between opDictionary and opBlockDiagv in which
base workspace. The following example illustrates the blocks can partially overlap rather than fully
use of classOp: (opDictionary), or not at all (opBlockDiag). A further
two di_erences are that only a single operator is
repeated and that each operator is implicitly preceded
F = opFFT(128); by a diagonal window operator.
G = classOp(F);
g1 = F(y,2); % gives g1 := FTy
g2 = G'*y; % gives g2 := GTy _ Fty Ensemble operators and general matrices:The three
ensemble
Operator type Matlab function operators (see Table 4) can be instantiated by simply
specifying their dimensions and a mode that
determines the normalization of the ensembles. Unlike
Ensembles opBinary, opSign, theother operators in the collection, the ensemble
opGaussian operators can be instantiated as explicit matrices
Selection opMask, (requiring O(m_ n) storage), or as implicit operators.
opColumnRestrict, opRestriction When instantiated as implicit operators, the random
Matrix opDiag, opDirac, number seeds are saved and rows and columns are
opMatrix, opToMatrix generated on the y during multiplication, requiring
Fast operators opCurvelet, only O(n) storage for the normalization coeffcients.
opConvolve1d, opConvolve2d, opDCT,
opFFT, opFFT2d, Selection operators:
opFFT2C, opHaar, opHaar2d,
opHeaviside, Two selection operators are provided: opMask and
opWavelet opRestriction. In forward mode, the restriction
Compound operators opBlockDiag, operator selects certain entries from the given vector
opDictionary, opFoG, opKron, and returns a correspondingly shortened vector. In
opWindowedOp contrast, the mask operator evaluates the dot-product
Nonlinear opReal, with a binary vector thus zeroing out the entries
opPadding instead of discarding them, and returns a vector of the
same length as the input vector.

Table 4: Operators grouped by type:
Meta operators:
Several tools are available for conveniently assembling
more complex operators from the basic operators. The
_ve meta-operators opFoG, opDictionary,
opTranspose, opBlockDiag, and opKron take one or
more of the basis operators as inputs, and assemble
them into a single operator:
H = opFoG(A1,A2,...); % H := A1 _ A2 _ : : : _ An
Fast operators:
Sparco also provides support for
operators with a special structure for which fast
algorithms are available. Such operators in the library
include Fourier, discrete cosine, wavelet, two-
dimensional curvelet, and one-dimensional
convolution of a signal with a kernel.For example, the
following code generates a partial 12 Fourier
measurement operator (F), a masked version with 30%
of the rows randomly zeroed out (M), and a
dictionary consisting of an FFT and a scaled Dirac
basis (B):
m = 128;
SPARKS@ELECTROMANIA 2K9 

D = opDiag(m,0.1); F = opFFT(m); % D is a diagonal

operator, F is an FFT
M = opFoG(opMask(rand(m,1) < 0.7),F); % M is a
masked version of F
B = opDictionary(F,D); % B = [F D]

Utilities:
For general matrices there are three operators:
opDirac, opDiag, and opMatrix. The Dirac operator
coincides with the identity matrix of desired size.
Diagonal matrices can be generated using opDiag
which takes either a size and scalar, or a vector
containing the
5.RESULTS AND INFERENCES
diagonal entries. General matrix operators can be
created using opMatrix with a (sparse) matrix as an
Though the experimentation was not so successful
argument.
completely due to some inevitable and unavoidable
The opToMatrix utility function takes an implicit
circumstances , the result is still satisfactory.Since
linear operator and forms and returns an explicit
always there is a
matrix. Figure 1 shows the results of using this utility
function on the operator M and B :
scope of improvement , design of such a system in
Mexplicit = opToMatrix(M); imagesc(Mexplicit);
practical form is still a challenge. The
Bexplicit = opToMatrix(B); imagesc(Bexplicit);
top view of DSP kit used is shown below :
Using the appropriate algorithms and tools I have done
experimentation which is discussed in next section.

4. EXPERIMENTASTION AND OBSERVATION :

4.1. Aim: to reconstruct an image given input image is

provided .

4.6.Output image is as follows:

SPARKS@ELECTROMANIA 2K9 
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SPARKS@ELECTROMANIA 2K9 

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