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Until now we considered discrete sinusoidal transforms. This concept should be familiar to someone that has BSc in electrical engineering. However, there are numerous other transforms developed in the last 10 or 20 years that have found applications in numerous research areas. Digital image processing is attractive application for these transforms. Here, we will give and brief overview of advanced level transforms used in digital image processing.

Typical profile of the digital image (along single image line) is given below.

We can assume that image has almost constant values with small variations in several neighbor pixels and after that we have abrupt changes of luminance. For image filtering it is important that image energy is concentrated in the smallest transformation coefficients. Similar property is desirable for image compression.

When image energy is concentrated in relatively small number of non-zero transformation coefficients it means that we will be able to easily identify coefficients corrupted by noise and filter them out. In opposite, when we have large number of non-zero coefficients then they will have an average smaller amplitude and smaller amount of noise can disturb them. Similarly for compression of images it is important that we have small number of significant coefficients in order to be able to have high compression ratio neglecting insignificant coefficients.

When we are performing 2D sinusoidal wave transform on profile of digital image (see slide 3) we need large number of coefficients to accurately describe abrupt points since it is very difficult to represent abrupt changes with smooth functions. Then several discrete rectangular wave transforms are developed based on rectangular shaped periodic functions. We will introduce some of them on several forthcoming slides.

Hadamard transform

Here, we will introduce the Hadamard transform for N=2n samples. There are some alternative forms of Hadamard transform for other number of samples. Hadamard transform derivation begins with matrix:

1 1 2 = 1 1

Hadamard transform for N=2 is defined with transformation matrix:

1 H2 = 2 2

Hadamard transform

Hadamard transform of larger dimensions can be defined using recursively as: N / 2 N / 2 1 where HN = N N = N / 2 N / 2 N

1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1

8

OP 0 PP 7 3 PP 4 PP 1 PP 6 PP 2 Q5

Number on the end of each row is number of sign changes in the row (from 1 to 1 and vice versa).

Walsh ordering

Very often instead of the Hadamard transform we are using reordered form of the Hadamard transform (this reordering is called Walsh ordering and sometimes the corresponding transform is called the Walsh transform). Reordering is performed in such manner that on the top of the transform matrix is put the row of the Hadamard transform matrix with the smallest number of sign changes (from 1 to -1 and from -1 to 1) and below rows are ordered in increasing order of sign changes. Hadamard and Walsh transforms are equivalent to each other but the Walsh transform order has analogy with the sinusoidal transform with respect to increasing of frequencies represented with corresponding coefficients.

1 1 1 1 1 2 2 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 0 1 1 1 2 1 3 1 4 1 5 1 6 1 7

Your task (maybe it is not quite simple) is to determine the inverse Walsh transform. The simpler task is to graphically represent basis function of the Hadamard and Walsh transforms.

1 1 1 1 1 1 1 1 1 1 1 1 1 1

1 1

Several alternative rectangular transforms are reviewed in textbook. Here, we will due to its importance just mention the Haar expansion.

function on frequency 0 function on frequency 1 . . . frequency gradually increases

Columns of the Walsh matrix are quite similar to the sinusoidal function used in sinusoidal transforms (expansions).

Haar transform

Definition of the Haar transform is not quite simple. Consider integer k defined in domain 0kN-1. It can be written as: k=2p+q-1. Integers p and q are selected in such a manner that p is the largest integer that satisfy 2pk. Values of k represents rows of the Haar transform matrix (here the first row has index 0, next one has index 1 while the last one has the index N-1). The second index (corresponding to columns) is denoted with i and this index is also defined within the range 0iN-1. Now we can defined x as x=i/N.

Haar transform

There is direct relationship between x and column index i. Elements in rows of the Haar transform are denoted as hk(x) where k is row index described on the previous slide. These vectors are defined as: q 1/ 2 p/2 q 1 x< p 2 2 2p 1 p/2 q 1/ 2 q 1 hk ( x ) = x< p h0 ( x) = 2 p 2 2 N N elsewhere 0

In order to understand the Haar transform and role of (p,q) we give as an example matrix for N=8 and we give values of (p,q) next to each row.

1 Hr = 8 1 1 2 0 2 0 0 0 1 1 2 0 2 0 0 0 1 1 2 0 0 2 0 0 1 1 2 0 0 2 0 0 1 1 0 2 0 0 2 0 1 1 0 2 0 0 2 0 1 1 0 2 0 0 0 2 1 (0, 0) 1 (0,1) 0 (1,1) 2 (1, 2) 0 (2,1) 0 (2, 2) 0 (2,3) 2 (2, 4)

There is natural question how it looks like the inverse matrix in this case?

Lets try to understand roles of (p,q) in transform. Obviously, p corresponds to power of elements represented in matrix rows. This value is called the scale. Coefficient q define position on non-zero elements in the rows. Then it is called position or shift. Haar transform will be used later for introducing quite important concept of wavelets and then we will explain details about this transform.

x(0) x(1)

HT N=2

Haar transformer for N=2 is created in such manner than one branch is used to peek up two adjacent samples and to add them (with some multiple) while the second branch is used to subtract adjacent samples.

Assume that we have 4 samples and assume that we try to transform these 4 samples using HT for 2 samples. Then this transform will perform operation on the first pair of samples and after that on the second pair of samples. Results of operations applied to pairs of samples are submitted in upper and lower lines.

x(n), n=0,1,2,3

HT N=2

In order to get the HT for N=4 samples we can note that lower line already produces 2 samples of the output while we have to apply the HT for N=2 for upper line.

x(n), n=0,1,2,3

HT N=2

HT N=2

Haar transform for larger number of samples can be obtained in similar manner. For example for N=8: HT N=2

y(0) y(1) y(2), y(3) y(4), y(5), y(6), y(7)

x(n), n=0,...,7

HT N=2

HT N=2

It can be observed that lower lines of the Haar transform are not used for further processing. For self learning session students should try to realize the inverse Haar transformer. The Haar transform follows general logic of the image in transform domain. Namely, in the image more interesting things are on the lower frequencies where the high amount of energy is concentrated than on higher frequencies. Then it is more important to process samples on low than on high frequencies.

Basis images for N=8.

low frequencies with image energy

Optimal transform

The Haar transform will be used for introduction of very important class of transforms called wavelets. These transforms have slightly different logic behind with respect to the introduced transforms. They are appropriate for digital image processing since energy of images is hidden on low frequencies while details are on high frequencies. Before we proceed with wavelet transform description we want to address the question what the ideal (optimal) transform would be for us? Based on introduced facts we can conclude that the best transform would be the one concentrating the signal energy in the smallest number of transformation coefficients.

Optimal transform

Assume that we have transform with transformation matrix A. Then transformation can be written as: X=A x Introduce important concept of eigenvalues and eigenvectors of matrix A. Eigenvalues i are solution of equation:

| A I |= 0

Let solutions of this equation be: i, i=1,...,N. These solutions are called eigenvalues.

Eigenvalues

For a given i we should determine vector vi satisfying: Aqi=iqi It is easy to show that under given conditions (when i is solution of previously defined equation) there are infinite number of vectors satisfying this equation. We can limit number of solutions to 1 by normalization of vector qi in such a way that its amplitude (square root of sum of squares of vector elements) is equal to 1, ||qi||=1.

Matrix of eigenvectors

Matrix of eigenvectors can be formed by putting eigenvectors in rows of matrix. This matrix denoted with Q has numerous interesting properties. For example, it is orthogonal matrix: Q-1=QT System of equations Aqi=iqi can be written in matrix form as: AQ=Q where is diagonal matrix of eigenvalues.

Eigenvalues matrix

It holds: QTAQ= QQT=A Determine cross-correlation of transform X: CXX=E{XXT}=E{AxxTAT}=AE{xxT}AT

For self-exercise review basic matrix operations.

CXX=AcxxAT

Energy concentrated in transform can be connected with energy of matrix CXX and this energy has the smallest loss when CXX is a diagonal matrix.

Ideal transform

It is easy to conclude that ideal matrix CXX is produced when A is matrix of eigenvalues of cxx. When cxx is known we can determine ideal transform of A as a matrix of eigenvalues. It can be shown that when we use just M the largest transformation coefficients of vector obtained mean squared error is equal to:

MSE =

i = M +1

eigenvalues corresponding to truncated coefficients, i.e., sum of the smallest N-M eigenvalues

Ideal transform

It can be shown that this transform produces the smallest MSE when we take M transformation coefficients. This transform is called the Karhunen-Loeve (KL) and it represents ideal transform that is goal for compression and filtering applications. Why we are not using the KL? The answer is quite simple. The KL requires calculation of the auto-correlation matrix and eigenvalues. Both of these operations are quite demanding.

It would be great that the KL be applicable in digital images since most of images have structure that would produces extremely small number of significant eigenvalues while huge number corresponds to components of insignificant energy. Fortunately, the DCT approaches to the KL transform for numerous practical images!!!! This is the reason why one of the most common image compression standards is based on the DCT.

Eigenvalues - Examples

We will review steps in calculation of eigenvalues by hand and how it can be implemented in MATLAB. The following matrix is given:

1 2 1 R = 3 1 0 1 0 1 1 det(R ) = 3 1 2

In MATLAB kar_jed=poly(A)

1 0 = 3 + 2 + 6 + 6 0 1

Eigenvalues - Examples

There are no direct approach for solving polynomial equation of higher order but to use numerical means. Then it is quite usable MATLAB function eig(R) that produces eigenvalues. In our case eigenvalues are: 13.337 2,3-1.168 j0.658. Determine now one of the eigenvectors by hand. It is eigenvector corresponding to the first eigenvalue 1.

Eigenvalues - Examples

For this eigenvalue holds: 2 1 q11 2.337 3 0 q21 = 0 2.337 0 4.337 q31 1 q31=q11/4.337=0.231q11 q21=3/2.337q11=1.284q11 It is easy to prove that this expression satisfy the first equation (accuracy is limited by small error in rounding) that is not independent on the other two. -2.337q11+2q21-q31=0

Eigenvalues - Example

Then the corresponding eigenvector is [q11, 1.284q11,0.231q11]. This vector can be normalized to modulo 1:

2 2 q11 + (1.284q11 ) 2 + (0.231q11 ) 2 = 2.702q11 = 1

q11 = 0.608

Then the eigenvector is [0.608,0.781,0.140]. MATLAB function can perform this in quite simple manner: [Q,L]=eig([1 2 -1;3 1 0;1 0 -1])

matrix of eigenvectors diagonal matrix of eigenvalues

Eigenvalues - Example

The KL transform can be calculated for line of image as: clear Im=imread('cameraman.tif'); A=double(Im(180,:)); A=A(113:142); A=A-mean(A); [KL,L]=eig(A'*A) In numerous cases just one eigenvalue corresponds to the most of the image energy and from entire matrix in the KL transform it is enough to memorize just several transformation coefficients (eigenvalues). However, this should be paid with calculation complexity and with memorizing eigenvectors for this transform (KL is signal dependent).

Wavelet transform

Now, we can consider the Haar transform realized for N=8: HT N=2 HT N=2

y(0) y(1) y(2), y(3) y(4), y(5), y(6), y(7) We noted that in the lower branches there are no important components and we are not processing these branches further. Then the Haar transform matrix has large number of zero coefficients in transformation matrix.

x(n), n=0,...,7

HT N=2

The first decomposition stage can be illustrated in the following manner.

x(n) g(n) x(n)*g(n) 2 xg(2n) Circuit for decimation (downsampling). It takes even samples of input signal.

h(n)

x(n)*h(n)

xh(2n)

g (0) = 2 / 2 g (1) = 2 / 2

h(n) has impulse response and 0 for other n

h(0) = 2 / 2 h(1) = 2 / 2

Haar wavelet

In upper branch we have samples:

n=[0,N/2-1)

How input (original) signal can be reconstructed?

Let the first step in the procedure be upsampling defined as: xi (2n) = xi (n) xi (2n + 1) = 0 i = 0,1 In the next step we are putting both signals through the same filters as in the first stage (g(n) and h(n)). Finally, these signals should be summed. Output of g(n) filter in upper branch is signal with samples: [(x(0)+x(1))/2,(x(0)+x(1))/2], [(x(2)+x(3))/2,(x(2)+x(3))/2]... In lower branch we have: [(x(0)-x(1))/2,(-x(0)+x(1))/2], [(x(2)-x(3))/2,(-x(2)+x(3))/2]...

Decomposition (analysis) and reconstruction (synthesis) in the case of the wavelet transform with Haar wavelet can be given as.

x(n) g(n) 2 2 g(n) + h(n) 2 2 h(n) x(n)

In upper branch we are performing lowpass filtering with lowpass filter that average samples. Highpass filtering is performed in lower branch. This corresponds to the spectral characteristic of digital images: different amounts of energy on low and high frequencies. Since low frequency region has huge amount of energy we are performing further decomposition of this part.

elements on lower frequencies are called scale (analysis part or rough part) g(n) g(n) h(n) 2 h(n) 2 2 g(n) 2 h(n) 2 2

For exercise create system that performs synthesis of input signal from this decomposition. In practice after wavelet decomposition we perform filtering of coefficients corrupted by noise or compression of signal with removing from visual quality point of view unimportant coefficients. Then output signal is different from the input signal.

The Haar wavelet was the first wavelet transform. This wavelet (Haar) is good for modeling functions with abrupt changes in the considered interval. For slower variations this transform is not the best one. Can we construct the wavelet transform that is suitable for other slower variations in signal, i.e., constructed using alternative filtering functions. Model of general wavelet transform will be considered within this and the next lecture.

h0(n) 2 2 g0(n) + h1(n) 2 2 g1(n) x(n)

We are interested in conditions that should be satisfied with filters h0(n), h1(n), g0(n) and g1(n), in order that input signal be the same as the output one. It is obvious that filters h0(n) and h1(n) should contain all frequencies in order that we can reconstruct signal at the output.

The simplest method for filter bank analysis is using the Z-transform.

|H0()| low frequencies /2 |H1()| high frequencies

X ( z) =

n =

x ( n) z n

Check!

x ( n / 2) n = 0,2,4,... X u ( z) = X ( z 2 ) xu ( n ) = elsewhere 0

At the output of the wavelet system (after relatively simple mathematical derivations) we obtain:

it should be equal to 1

( z ) = 1 [G ( z ) H ( z ) + H ( z )G ( z )] X ( z ) X 0 0 1 1 2 1 + [G0 ( z ) H 0 ( z ) + H1 ( z )G1 ( z )] X ( z ) 2

it should be 0

Assuming that we know h0(n) and h1(n) equation for determination of the g0(n) and g1(n) in the Z-domain is:

H1 ( z ) G0 ( z ) 2 G ( z ) = det(H ( z )) H ( z ) 1 0 m

Denote:

H 0 ( z ) H 0 ( z ) Hm ( z) = H1 ( z ) H1 ( z )

Assume that det(Hm(z))=z-(2k+1) (delay only) and for =2 we obtain:

Under introduced assumptions it holds: P ( z ) = G0 ( z ) H 0 ( z ) = G1 ( z ) H1 ( z ) Further we obtain: G0 ( z ) H 0 ( z ) + G0 ( z ) H 0 ( z ) = 2 Calculating the inverse Z-transform it follows:

k =

g (k )h (n k ) + (1) g (k )h (n k ) = 2(k )

n

k =

This relationship represents the condition that should be satisfied by wavelet filters in time domain. In the next lecture we will conclude design of wavelet filters, some comments related to the 2D wavelets will be given and standard usage of wavelets explained.

For self-exercise

Write program for realization of the Hadamard and Walsh transforms. Determine inverse of the Walsh and Hadamard transforms. To which group they belong? Repeat the same procedure for the Haar transform. Assume that we have signal of dimensions 8x8 with non-zero coefficients x(1,2)=2, x(3,1)=1 and x(5,6)=1. Calculate the Hadamard, Haar and Walsh transforms of this signal. For NxM=8x8 determine basis images for (p,q)=(1,2), (p,q)=(3,3) and (p,q)=(6,7). Project for self-exercise. Application of the Hadamard transform in realization of the special purpose Hamming codes Reed-Muller codes. Write program for the Hadamard and other rectangular transforms of digital image. When image dimensions are not power of 2 realize transforms with zero-padded signal.

For self-exercise

Give hardware realization of the inverse Haar transformers. For filters used in the Haar transform (or Haar wavelet) determine spectral response. Since these filters have only 2 non-zero samples of the impulse response perform the zeropadding for obtaining clear results. Details of the eigenvalues decomposition and the KL transform in signal processing could be learnt in details from the book: For mini-project. Study singular value decomposition and cases when it can be used. Try to find proper references while wikipedia can be used as a starting point in search related to this topic.

M. R. Stoji, M. S. Stankovi, R. S. Stankovi: Diskretne transformacije u primjeni, Nauka, 1993 (strana 119).

For self-exercise

Write program that perform Wavelet Haar decomposition of the signal of the given length (power of 2) given number of times (for example 3 times) in such a manner that in each stage of decomposition it is performed on lowpass signal. Give graphical presentation of the signal decomposition using the wavelet transform. Wavelet decomposition is performed 3 times on the lowpass part. Give corresponding synthesis stage. Assume that in the lowpass branch of the wavelet we have ideal cut-off filter with cut-off frequency equal to half of the maximal frequency that corresponds to the sampling theorem. Determine filter for highpass branch and corresponding filters on the synthesis side. What is the main difference between this and the system with Haar wavelet?

For self-exercise

Prove formulas for Z-transform of the downsampling and upsampling circuit. Signal x(n) is transformed with downsampling and after that with upsampling circuit. Determine output of this system in time and Z-domain. Haar transform is determined. Try to determine methods for calculation of the Wavelet-Haar coefficients from the Haar transform. Haar transform and wavelet with Haar filter are defined with the same basic goal. Determine difference between these two transforms? Determine relationship between transform coefficients in the Hadamard and the Walsh transform? Mini-project. Consider several real images with their transforms (DFT, DCT, DHT and introduced rectangular ones). For each transform determine how many coefficients are kept in 50%, 70%, 80%, 90%, 95%, 98%, 99%, 99.5%, 99.9% of the signal power. Which of introduced transform is the best in each considered case?

For self-exercise

Repeat previous experiments when image is divided in blocks 2nx2n. Consider possibility to remove transformation coefficients that have small energy in function of n and used transforms. Can you make some consistent conclusion? Apply the wavelet-Haar decomposition on various images and try to repeat analysis from the previous two problems in this case form application of this transform in data compression. Mini-project. Study wavelets starting from the continuous time form. For this research start from the textbook.

Wavelet transform Filtering of digital images - Intro

We have proven that filters used in the wavelet transform should satisfy: G0 ( z ) H 0 ( z ) + H1 ( z )G1 ( z ) = 2

G0 ( z ) H 0 ( z ) + H1 ( z )G1 ( z ) = 0

P ( z ) = G0 ( z ) H 0 ( z ) = G1 ( z ) H1 ( z )

G0 ( z ) H 0 ( z ) + G0 ( z ) H 0 ( z ) = 2

G0 ( z ) H 0 ( z )G1 ( z ) + G0 ( z ) H 0 ( z )G1 ( z ) = 0

G0 ( z ) H 0 ( z ) + G0 ( z ) H 0 ( z ) = 2 H 0 ( z )G1 ( z ) + H 0 ( z )G1 ( z ) = 0 Inverse Z-transform of the first expression is:

k =

g 0 ( k )h0 ( n k ) + ( 1) n

k =

g ( k ) h ( n k ) = 2 ( n )

0 0

k =

k =

x(k ) y (k ) =

x, y

Inner product.

Similarly it can be proven that:

g1 ( k ), h1 (2n k ) = (n )

g0 (k ), h1 (2n k ) = 0

g1 (k ), h0 (2n k ) = 0

We can summarized these relationships as: the wavelet filters should satisfy the following relationship in the time domain: hi (2n k ), g j ( k ) = (i j )( n), i, j = {0,1}

When the following:

hi (2n k ), h j ( k ) = (i j )( n), i, j = {0,1}

holds we have orthogonal wavelets (the same type of wavelets used in analysis and synthesis stages). Next we will discuss the realization of the wavelet transforms for 2D signals and after that we will give some program realizations.

2D Wavelets

How to perform wavelet decomposition for 2D signals? In the same way as a generalization of all 1D transforms to 2D transforms we can perform operation separately along rows and columns of images. Therefore, we will perform wavelet transform along columns (or rows) followed by wavelet transforms along rows (or columns).

2D Wavelet transforms

h0(m) h0(n) x(n,m) 2 h1(m) 2 2

a(n,m)

h0(m) h1(n) applying filters to columns taking elements on even positions from each column 2 h1(m) applying filters to rows

2D Wavelet - Comments

a(n,m) is low-pass image (sometimes called coarse image). Notation a comes from the fact that it is product of analysis side of the wavelet transformers. In the next stages we can proceed with its analysis using wavelet transform. Images dV(n,m), dH(n,m) and dD(n,m) are called details in horizontal, vertical and diagonal directions respectively, These images are commonly not subject to further decomposition. Note that these images have dimensions N/2XN/2 and total dimension of data is not changed by this procedure. What is the advantage of wavelet transforms and how to realize it?

Wavelet realization

Very often instead of realization of the wavelet transform by using definition, numerous scientists use well-developed software tools. We will use the MATLAB Wavelet Toolbox but on the Internet it is possible to find alternative tools and MATLAB toolboxes. This is quite comprehensive and complicated toolbox and we will not consider all its possibilities but we will just concentrate on several features that can illustrate our theory in step-bystep manner. Note that there are single step functions in MATLAB toolbox able to perform very sophisticated operations. User should only learn how to use these functions and knowledge about wavelets is not required. This is popular for usage but it is source of numerous mistakes in practice.

Wavelet Toolbox

The most important recent contributions to the Wavelet Transform is given by Donoho and it is possible to find all MATLAB functions related to his research on his web-site. For a brevity we will present the wavelet transform in step-bystep manner without giving all details. Only on several places we will describe useful shortcuts that can be used in procedures. The first useful function has a format: [LD,HD,LS,HS] = wfilters('haar') Output of this function are wavelet function used in decomposition (analysis) stage LD and HD (lowpass and highpass filters) as well as those used in synthesis stage LS and HS. Argument of this function is used wavelet type. Here we use the Haar wavelet but there are numerous alternative predefined wavelet functions.

Wavelet filters

Instead of abrupt Haar wavelets we can use more smooth functions such as the Daubechies class of wavelet functions called dbn where n is integer that can be selected within range from n=1 to n=45 (db1 is in fact the Haar wavelet). Here, it will be used db23. Analysis stage can be performed using convolution of signal with corresponding wavelet functions: clear Test signal x=wnoise(1,10); [LD,HD,LS,HS]=wfilters('db23'); Filtering with corresponding filters xl=conv2(x,LD,'same'); (parameter same is introduced in xh=conv2(x,HD,'same'); order that obtained signals have

conv for 1D convolution has no argument same the same length as an original signal the first argument. Unrequired part of the signal is truncated).

xdl=xl(1:2:length(xl)); Decimation xdh=xh(1:2:length(xl)); xul=zeros(1,2*length(xdl)); xuh=zeros(1,2*length(xdl)); Upsampling xul(2:2:length(xul))=xdl; xuh(2:2:length(xul))=xdh; xr=conv2(xul,LS,'same')+conv2(xuh,HS,'same');

Filtering in the highpass and lowpass lines. Reconstructed signal

In the previous example we performed meaningless operation: signal is decomposed and after that reconstructed. Note that before the reconstruction stage lowpass part can be decomposed several times. Instead of showing the advantages of the wavelet transform on 1D we will do it on example of 2D signals. Here, we will use function dwt2 (there is similar function dwt for 1D signals) for decomposition that can perform it in more efficient manner than in the previous example where realization is performed in step-by-step manner.

After reading image cameraman.tif we perform three steps of decomposition (wavelet transform is performed three times on lowpass part of image). clear x=imread('cameraman.tif'); [ca,ch,cv,cd] = dwt2(x,'db1','mode','sym'); [caa,cah,cav,cad]=dwt2(ca,'db1','mode','sym'); [caaa,caah,caav,caad]=dwt2(caa,'db1','mode','sym'); Slika=[caaa,caah;caav,caad]; Slika=[Slika,cah;cav,cad]; WavTransf=[Slika,ch;cv,cd];

Wavelet transform has the same dimensions like image but huge number of wavelet transform pixels is close to 0. It means that we can neglect many of them and that we can reconstruct image with high accuracy.

In image reconstruction we use 17% of the largest wavelet coefficients (according to absolute value). This is basis image and images representing edges, details and similar elements. Reconstruction is performed using idwt2 command.

Image reconstruction

Reconstructed Cameraman based on 17% of wavelet coefficients. Due to the described properties the wavelet transform has numerous application. Some of them will be given on next slides.

Here, we will present two the most important applications of the wavelet transform. The first one is compression. It is clear that for relatively small number of wavelet coefficients has significant energy. It means that only small number of coefficients has importance for visual image quality. Wavelets are note used alone for compression but they are combined with other compressing strategies (for example RLE, Huffman etc) but the wavelets are basis for some compression standards (for example JPEG 2000). Furthermore, the wavelet transform based compression is quite suitable since it can be applied in different manners for different channel state. For noisy channel with small capacity we are sending base image without details but for better conditions in the channels we are sending images with details.

Application in denoising

Assume that image is corrupted in channel or during acquisition process (for example medical images obtained using ultrasound devices could be very noisy). It can be proved that noise in various wavelet coefficients is (almost) mutually independent and it is uniformly distributed (for relatively large number of decomposition steps) on all wavelet coefficients. Then we can perform filtering independently for each coefficients. Two groups of techniques are developed using wavelets: hard thresholding (all coefficients below the threshold are set to zero and in reconstruction are used for reconstruction (threshold is determined based on estimated noise variance)); soft thresholding (small wavelet coefficients are attenuated proportionally more than large ones). Obtained wavelet-based results are the last word of science in denoising applications. It is quite difficult to be better than wavelets in this area in terms of quality of imaging and speed.

Wavelet toolbox

There are additional wavelet applications. Check wavelet toolbox and demo program wavedemo. There are single step functions in the toolbox for signal filtering. For example, check function wden with its parameters. Note that this function is used for the 1D denoising and also see denoising 2D images.

Drawbacks of wavelets

When signal subject of transform is dominantly on low frequencies with only details and edges of high frequencies the wavelet transform is close to ideal analysis, synthesis, filtering and compression tool. Digital images are commonly this signals of type and today wavelet transform is quite commonly used in the digital image processing field. However, when relatively large energy is concentrated on high frequencies or when we have abrupt changes of frequencies in the signal, wavelet transforms are not ideal tools.

These signals are not quite common in digital images but they exists. One of these signals are optical intereferograms commonly caused by mistakes in design of optical communication channels or due to diffraction in optical microscopy (optical or laser microscopes are cheaper than electronic counterparts but they suffer from several drawbacks including interferograms). Inteferogram: test example. Wavelet transform is not suitable for analysis and filtering of such signals.

Optical interferograms

TF (S/SF) Transforms

Time-frequency transforms are designed for analysis of 1D signals with high frequency content. The most important TFRs are:

Short-time Fourier transform (STFT); Wigner distribution (WD).

For multidimensional signals, multidimensional generalization of these transforms are used (sometimes called Space/Spatial-Frequency representations). We described several representations from this group in the textbook and here from the brevity reasons we will skip them.

Image filtering

The most important part of our course is related to image filtering and reconstruction. Under filtering we assume obtaining some image features (for example in communications signal carrier of information modulated on some frequency). Filtering in digital images is commonly assumed to be denoising operation. The goal of denoising process is obtaining signal as close as possible to original image. In addition we will consider in a brief manner image reconstruction for images corrupted by noise and distorted with some deterministic form of distortions.

Recall several information related to the image spectral characteristics. The 2D DFT of images is highly concentrated around origin (0,0) and it decreases rapidly with frequency increasing. Filters in the frequency domain can be divided in four groups:

Lowpass filters that remove frequencies away from the origin. Commonly, frequency response of these filters is symmetric around the origin; Highpass filters remove signal components around origin in frequency domain; Bandpass filters allows frequency in the band between lowest and the highest frequencies; Stopband filters remove frequency band.

Before we define these filters we will perform a simple experiment. Consider image Cameraman. Perform lowpass filtering with 17x17 frequency samples around the origin and perform highpass filtering for a region outside this zone. Results of this rough filtering scheme are given on the next slide.

Here, energy of lowpass image is larger approximately 18 times!!!

Lowpass filtered image is almost non-usable while highpass filtered counterpart is dark but recognizable (to be honest for better visualization we increased energy of highpass image).

Conclusion:

The largest amount of energy is concentrated on low frequencies, but it represents just image luminance and visually not so important part of image. Small energy on high frequency corresponds to visually very important image features such as edges and details.

This is the reason why filtering of digital images are quite complicated task and one of reasons for introducing the wavelet transform.

Filters in the frequency domain can be defined as

(here we give them in analog form but it can be discretized in straightforward manner):

( n, m ) = 1 f (2) 2

filtered image

X ( , ) H ( , ) d d

1 2 1 2 1

Filtering can be performed for denosing purposed, for removing distortions and/or for obtaining some important features from digital images.

Here, we will give the simplest filter types in frequency domain with value 1 in pass region and 0 in stop region. These filters are called cut-off filters. In addition, we will assume rectangular filter forms. This is just examples and more advanced filter forms (smooth) are commonly used in practice.

1 | 1 | 1 | 2 | 2 H ( 1 , 2 ) = elsewhere 0

Lowpass filter.

0 | 1 | 1 | 2 | 2 H ( 1 , 2 ) = elsewhere 1

High-frequency filter.

For exercise, determine stopband filter and filters with circle shape impulse responses. In addition determine filters with the shape of the Hanning window function in the pass part of the frequency response.

Up to now introduced filters are linear. Linear space invariant filters can be realized in both frequency and space domain. Realization in space domain by using convolution:

n0 m0

2D filter impulse response. Why we call this 2D filter impulse response and how it is connected with H(1,2)?

2D convolution.

Convolution or FFT

There are two alternatives for determination of the linear space invariant systems outputs: convolution or FFT in frequency domain. What to use?

The simpler technique.

At the first glance it is convolution in the space domain but in depth analysis can show that it is not the case in general. Let image x(n,m) has dimensions NxM and let impulse response h(n,m) has dimensions N1xM1. Convolution result has dimensions (N+N1-1)x(M+M1-1). Why?

Convolution or FFT

Under assumption that impulse response has smaller duration than image for filtering using convolution is required: (N+N1-1)x(M+M1-1)xM1xN1 additions and the same number of multiplications. Number of operation for

Number of pixels at the filter output. single pixels.

For the FFT-based realization it is required 3 FFTs (for filter response, input signal and inverse for output signal). These operations require: (N+N1-1)x(M+M1-1)log2(N+N1-1)x(M+M1-1)

All signals are properly zero-padded in order to avoid aliasing.

Convolution or FFT

In the FFT-based realization we need additional (N+N1-1)x(M+M1-1) multiplications X(1,2)H(1,2). In order that realization using convolution is simpler it is required to hold: (N+N1-1)x(M+M1-1)xM1xN1 < 3(N+N1-1)x(M+M1-1)log2(N+N1-1)x(M+M1-1)+ (N+N1-1)x(M+M1-1) After dividing with (N+N1-1)x(M+M1-1): M1xN1 <3log2(N+N1-1)x(M+M1-1)+1 Taking M1=N1 and N=M and N1=N we obtain.

Convolution or FFT

2N2<6log2(N+N-1)+1. Take for example N=100. The previous non-equality holds for: <0.065, i.e., for dimensions of the impulse response smaller than 7x7. To conclude that direct calculation of the convolution is better when impulse response of the filter is relatively small. Note that all derived formula are approximations but they reflect actual situations. In practice the FFT of the filter impulse response could be calculated in advance and stored in memory also we assumed that operations in direct evaluation of the convolution are performed for complex valued functions.

Large family of convolution filters can be applied both for denoisiong but also for producing some interesting effects in digital images. Since impulse response of these filters is commonly small they can be realized in faster manner using direct convolution and from that fact it is coming their name. Here, we will present some of the most important filters from this group.

Blur filters

The aim of Blur filters is to blur images. The most important blur filter is the Gaussian blur filter. Its primary aim is giving depth to images. Assume that we have image of the person in front of some background. This filter can be applied to background. Then due to given depth to the background it could look like that background is very far from the person. The impulse response of the Gaussian blur filter is:

h ( n, m ) = exp((m + n ) / ) exp((m 2 + n 2 ) / 2 )

2 2 2

n m ,( n ,m )D

Domain defined in such manner that it is symmetric around the origin (0,0).

Denumerator in the impulse response is introduced in order to keep the energy of blurred image (luminance) on the same level as in original image (sum of the impulse response is 1).

Original image.

The second important blur filter is motion blur. It is used to simulate motion of objects during image acquisition. Motion along horizontal direction can be simulated by averaging pixels in this direction. Impulse response of this filter can be defined as:

1/ N m h ( n, m ) = 0 n = 0, m [1, N m ] elsewhere

Width of the filter.

The simplest way to simulate the motion blur filtering along alternative directions is by rotating impulse response for a given angle and convolving image with rotated impulse response.

Laplace filter

+1 1 h (n, m) = +1 +1 1 1+ 4 +1 1 1+ 1+ 1 1+ 1+

Impulse response of the Laplace filter. Defined for [0,1]. For =0 the impulse response exhibits: 0 1 0 h1 (n, m) = 1 4 1 0 1 0

This filter is used in edge detection and for detection of abrupt changes in luminance.

Image sharpening

The most important filter for sharpening of digital images is the unsharpen mask. The impulse response of this filter for dimensions 3x3 is defined using the Laplace filter as:

0 0 0 h( n, m) = 0 1 0 h ( n, m) 0 0 0

We should be very careful in this operation since for large amount of sharpening it can appear the so-called grain noise.

Convolution filters

Convolution filters (we will learn some additional filters from this group) represents ground for numerous algorithms in image processing. At least 50% filters used in practice and in commercial tools for image processing are based on these filters. They are also important segment of highly specialized tools for image processing. In addition there are some other filters that will be taught next week that do not belong to this group.

For self-exercise

Study of wavelet toolbox including:

Single-step filters; Decomposition and decomposition tree; Compression algorithms etc. Write report on 5 pages.

Write missing commands required for creating images on slides 14-16. Study soft and hard thresholding techniques for wavelet transformation and write programs for filtering images in arbitrary number of decomposition levels using the wavelet transform. Find and study data about JPEG 2000 algorithm. The Gaussian noise with variance 2 is input in the wavelet filter. Determine threshold to minimize the amount of this noise at the filter output.

For self-exercise

Write report about time-frequency representations and their applications in image processing. Solve problems related to the time-frequency representations from the textbook. Determine the impulse noise for convolution filters given in spectral domain and spectral response for filters given in time domain. MATLAB realization of convolution filters. Study menu Filters in the Photoshop. Which filters from this menu can be realize using the convolution and try to write your version of these filters. Study options related to mask (selection) in the Photoshop. Describe what represents mask filtering. User wants to apply filter on part of image. How he can do it? Which problems arise on limits of this image part and how we can overcome them?

FILTERS

Signal Denoising

Probably the most important application of digital filters is denoising (removing of additive noise from signals). The denoising problem can be described in the following manner (here given for 1D signals):

Signal of interest is f(n). It is corrupted by white additive noise (n), x(n)=f(n)+(n). Our goal is to get filtered signal s(n) as close as possible to original signal f(n) based on noisy observations x(n).

Signal denoising

As a similarity measure between f(n) and s(n) we will use mean squared error: N is total signal duration

1 N MSE = E{[ s (n) f (n)] } = [ s (n) f (n)]2 N n=1 Under certain assumptions such as to know the probability density function we can construct filters that minimize mean squared error. They are called ML filters (Maximum likelihood).

2

ML filter

Minimization problem for defining ML filters is:

s( n ) = arg min

n+ N

In this case 2N+1 is width of symmetric local neighborhood around instant n for which we want to determine filter output. General rule: wider window (larger N) better removing of noise but with signal disturbances. argmin means: value of that minimizes. F() is called loss function (term is coming from economics) and for ML filter it is equal to F()=-logp() where p() is probability density function of noise.

k =n N

F ( x(k ) )

Gaussian noise (one of the most important noise models) has probability density function: 2 noise 2 2 1 p () = e / 2 variance 2 Loss function for this noise is:

additive and multiplicative constant does not influence results of filtering

F () = log 2+ | |2 / 2 2

MA filter

The output of the filter for the Gaussian input noise follows from:

J () =

n+ N k =n N l = m M

m+ M

[ x(k , l ) ]2

that minimizes this expression can be obtained by calculating the first derivative of J() with respect to : J () n+ N m+ M = 2[ x(k , l ) ] = 0 k =n N l =m M

n+ N k =n N l =m M

m+ M

x(k , l ) =

n+ N

k = n N l = m M

m+ M

MA filter

n+ N m+ M 1 s ( n, m ) = = N l =M x(k , l ) = (2 N + 1)(2 M + 1) k =n m N M 1 = N l = M x(n + k , m + l ) (2 N + 1)(2 M + 1) k =

It can be noted that this filter called MA filter (moving average) is convolutional filter that can be realized using convolution of image with impulse response: What is this?

clear a=imread('Baboon.jpg'); imshow(a) a=rgb2gray(a); b3=uint8(conv2(double(a),ones(3)/9,'same')); b5=uint8(conv2(double(a),ones(5)/25,'same')); b7=uint8(conv2(double(a),ones(7)/49,'same'));

image converted to double format convolution with impulse response Performs rounding and returns data in image format. same causes that result of convolution has the same dimension as the first argument of the command. What dimension would be otherwise?

Determine the frequency response of MA filters. Are they lowpass or highpass filters?

Original image with image filtered with 7x7 MA filter. Obviously, this very wide window blurs and disturbs images, etc. Then we want to perform filtering with the MA filter with relatively narrow window in order to avoid distorting of important information in images (details and edges). Exercise: Add Gaussian noise to digital images with certain variance (you can use MATLAB functions imnoise and randn) and calculate mean squared error between original image, noisy image and image filtered with MA filter with various widths of the window.

Drawbacks of MA filter

We already noted that MA filter has simple, semi-intuitive form, but that it can disturb edges and details of images. This filter can be realized in simple manner. However, image is not subject only to Gaussian noise but also some other noises that can have rare but very strong amplitudes are quite common in this area. This type of noise is called impulsive noise. MA filter is very sensitive to this noise and single pixel with impulse noise can be spread using the MA over neighbor pixels with decreased amplitude.

1 1 1 2 1 1 2 26 1 2 4 4 2 1 1 1 2 4 4 1 1 1 2 1 3 3 2 1 3 1.67 1.44 1.55

Assume that this is an image and that we want to filter out it with 3x3 MA filter. Red color denotes impulse that corrupted image. It has significantly different value of other adjacent pixels. Filter out inner 3x3 pixels. Several neighbor pixels has luminance significantly increased due to averaging impulsive (corrupted) pixel. MA filter in fact enlarges zone of impulse influence.

4.11 4

Comparison of pdfs

p()

Probability density function of Gaussian noise is smooth bellshaped curve having fast decay to zero meaning that there is high probability that underlying process would have extremely large values. Probability density function of Laplace noise is typical representative of impulse noise class since it has long tail with small but finite probability of taking very large values.

Probability density function of Laplace noise is proportional to exp(-||) with corresponding loss function F()=||.

MEDIAN FILTER

Median filter is ML filter for signal corrupted with Laplace noise. Derivation begins with:

J () =

n+ N k =n N l = m M

m+ M

| x(k , l ) |

Sign function +1 for positive values of argument, -1 for negative and 0 for zero.

MEDIAN FILTER

Sum of signs is equal to zero when number of positive signs is the same as the number of negative signs. This can be understood in an alternative manner. Sort x(k,l) for k[n-N,n+N] and l[m-M,m+M] into nondecreasing order. Denote sorted values as x(i) where: x(1)x(2)...x(i-1)x(i) ... x(2N+1)(2M+1) Median is value in the middle of sorted sequence x((2N+1)(2M+1)+1)/2. This value is those for which exactly half of samples produces x(k,l)- less than zero (has sign function equal to -1) while half are greater than 0 (has sign +1).

Consider sequence with even number of samples 1 3 11 4 2 5 Sorted sequence is: 1 2 3 4 5 11 We can take as a median any value between two middle samples in the sequence (in this case between 3 and 4) since corresponding minimizing function will produce minimal result. Commonly as the median for even number of samples sequence mean between two middle samples is adopted (in this case 3.5).

1 1 1 2 1 1 2 26 1 2 1 1 1 2 1 1 1 2 1 1 1 1 1 1 2 1 1 1 1 3 3 2 1 3

Assume that we have image and we want to filter it with median filter of dimension 3x3. Red represents impulse that is significantly of different value than neighboring pixels. Perform filtering of inner 3x3 square. Obviously, impulse influence is removed from the pixels of filtered image.

For filtering of this pixel we have a sequence with 5x1, 3x2 and 1x3.

Median filter is nonlinear! It means that both convolution and FFT algorithm cannot be used for its realization but we need to employ some sorting algorithms for its evaluation (for example quick sort or insertion sort algorithms). This implies that median filter realization is significantly demanding than the moving average filter realization. Then it is interesting to find spectral characteristics of this filter. Is it lowpass or highpass filter?

Since the median filter is not the LSIS then we can only to estimate spectral response of the median filter. It is performed here in the following manner. We considered image corrupted by noise and we perform median filtering. The 2D DFTs of the input and output images are calculated and its ratio is considered as the spectral response of the median filter. For improving precision we performed these evaluation 10 times and obtained results are averaged.

Logarithm of the approximate spectral characteristics of the median filter is depicted. Red and yellow represent large values while green and blue represent small values. Conclusion. Median is lowpass filter (as the MA filter) but it preserves edges in image. For self exercise reconstruct this experiment.

Median filter is excellent for removing impulse noise while it is just slightly worse than the MA filter for Gaussian noise environment.

In order to illustrate this property consider asymptotic formula for mean square error in image filtering for optimization technique with the loss function F() and probability density function is p() where loss function could be different from optimal one F()=-logp():

1 MSE = E{| x(n, m) f (n, m) | } 2 (2 N + 1)(2 M + 1) Noisy image F "() p ()d Non-noisy (original) image

2

( F '()) 2 p ()d

Previous relationship is just rough approximation. Its derivation is rather complicate and it is derived under assumption that original image in local neighborhood is constant that is not the true in realistic applications.

For exercise perform the following experiments: check the mean squared error for Gaussian noise and median and MA filters; then repeat this analysis for Laplace noise and finally for Cauchy noise. Cauchy noise is model of the impulse noise with probability density function: p()=a/(2+a2). Conclusions derived from this analysis? Suggested conclusions: median is slightly worse than the MA filter for Gaussian noise, slightly better than the Laplace noise, and much better than the Cauchy noise. Median filter performs relatively good for impulse noise.

Very good properties of the median filters is the fact that it preserves edges and abrupts while the MA filter smooth them.

Perform the following experiment. Consider signal with followed with K consecutive zeros. Apply 1D filtering with MA and median filter of 2N+1 and consider results.

The drawback of the median filter is complexity and the fact that very small details in images are treated as impulses and eliminated. In order to keep small details size of the median filter is small commonly not larger than 9x9 and very often just 3x3.

Median filter realized as the function y=med_filt(x,p) MATLAB function: x is image mM=p(1); mM=round((mM-1)/2); of interest, p is vector of two parameters that represent nN=p(2); nN=round((nN-1)/2); dimensions of neighborhood. if(mM<0) mM=0; end if(nN<0) nN=0; end For p=[9 9] we have mM=4 if(mM==0&nN==0) and nN=4, i.e., we are y=x; performing filtering in the region else [m-mM,m+mM]x[n-nN,n+nN]. [M,N]=size(x); Negative values of mM and nN are avoided. x=double(x); In this case not perform fitlering. y=x; Preparation for filtering, conversion of image in double format. y is output image (here we perform its initialization).

for m=1:M For all image pixels. for n=1:N z=x(max(1,m-mM):min(M,m+mM),max(1,n-nN):min(N,n+nN)); y(m,n)=median(z(:)); end Region of image from m-mM to m+mM end and from n-nN to n+nN but having in mind z(:) converts matrix in end that image cannot be outside of domain vector, median

determines median of sequence. [1,M]x[1,N]. This is not the fastest way but it is suitable for our course purposes.

y=uint8(y);

Filter comparison

Image corrupted by Gaussian noise and filtered with moving average filter and median filter. It is difficult to notice significant difference in image quality. Image corrupted by Laplace noise and filtered with median filter and moving average filter. Median filter produces significantly better quality.

Test noises

Noise for testing filters can be crated in various manners. For example Gaussian noise can be obtained as s*randn(N,M) where NxM is image size while s is the standard deviation of noise. Image that is corrupted by noise should be returned to the image format before visualization: truncated (or rounded) noninteger parts and given within limits between 0 and 255 for image presented with 8 bits per pixel. Alternative function is: B=imnoise(A,tip,parametri); A is original non-noisy image, tip is type of noise for example salt & papper for typical impulse noise, while parametri are noise parameters (in the salt & papper it is percentage of noise).

Generation of noise

In the textbook, it can be found details about Laplacian noise generation based on the uniform noise generator (available in MATLAB with function rand). In the recent MATLAB versions there are numerous realized noise environment. For example poissrnd realizes multiplicative Poisson random noise with mean value and standard deviation equal to the luminance of non-noisy image. Ultrasound images are subject to this kind of noise. Search for other random noise generators can be performed by using the MATLAB help system and option search entering phrase Random numbers generator. For testing different algorithms and techniques it is required to generate different noises and sometimes MATLAB functions are not available or they are not generated on machines where technique should be implemented (for example medical devices) so we have to realize them from the scratch.

The most widely used technique for numerical quality assessment of filtering technique (and other signal processing algorithms) is the SNR (Signal to Noise Ratio). Mean squared value SNR is defined as:

SNR = 10log10

Given in the dB.

M N M N

Original image.

Filtered image. Mean squared filtering error (Mean Squared Error). It is also used as quality measure.

Quality measure

The most popular technique for image filtering is the pseudo signal-noise ratio (PSNR) defined as:

Maximal signal value (maximal luminance).

PSNR = 10log10

2 fmax

1 M N [ f ( n, m ) f ( n, m )]2 MN m =1 n =1 Excluding some non-realistic situations (for example dark image with single bright pixel) PSNR is very good quality measure for filtering and other image processing algorithms.

Quality measure

For PSNR>60dB difference between two images are very difficult for observations even in the case of direct comparison. For PSNR>40dB (and mostly for PSNR>35dB) difference can be observed by comparing original and filtered image. For 15dB<PSNR<35dB image can be recognized but it is of lower quality (close to lower bound of very bad while on the upper bound relatively good). For PSNR<15dB (or PSNR<20dB) image is not usable. These bounds are one of reasons for wide usage of this measure in practice.

Numerical: One relatively common technique for quality measure is maximal absolute value of error; Currently very popular measure is ISNR (Improvement in SNR): 1 M N [ x(n, m) f (n, m)]2 MN m=1 n=1 ISNR = 10log10 There are 1 M N [ f (n, m) f (n, m)]2 numerous other MN m=1 n=1 numerical

measures!

As all other measures where the logarithm is used this is also given in decibels dB. It represents ratio between MSE for input signal and for output signal, i.e., improvement achieved with our filter. For ISNR=0dB filtering does not improve image quality.

Non-numerical. All numerical measures have some drawbacks. For example, the PSNR is good for numerous realistic images but for dark images with small number of light spots it is not good. Then we are using measures based on human observations that are giving non-numerical marks for example very good. These measures are also quantified since we are giving images to larger number of users and they are giving marks, we quantify marks for example in the range between 1 and 5 and average these marks and for example average mark is 3.62. Maybe it sounds strange but these measurement are not rare in practice and the main task is to find fair user to take part in assessment.

There are numerous modifications of the median filter. We can in addition to rectangular neighborhood consider other forms: lines, plus shape, diamond shape, circle etc. Try to modify our median function that can work for these or arbitrary form neighborhood. Simplification of the median filter realization was important issue until recent advance in the computer technology. Simplifications were in direction of reduced memory of calculation burden. Here we will consider two simplified median filter form:

Separable median filter (reduces calculation complexity) Recursive median filter (reduces memory complexity).

In the separable median filter we perform median filtering of columns (or rows) and after that perform median filtering of obtained results. Here we give only part of realizations with nested for cycles: function y=sep_med_filt(x,p) %%%the same commands as in the median realization

for m=1:M for n=1:N Filtering along t(m,n)=median(x(m,max(1,n-nN):min(N,n+nN)); rows. end end for m=1:M for n=1:N y(m,n)=median(t(max(1,m-mM):min(M,m+mM),n); Filtering along end columns. end y=uint8(y);

In the median filter we have sorting sequence of dimension (2K+1)x(2K+1). The fastest sorting algorithm (for example quick sort) under some conditions requires Nlog2N comparisons, i.e., in our case it is 2(2K+1)2log2(2K+1). Separable median filter requires 2K+1 sorting along rows (or columns) and one additional sorting of obtained results and complexity is approximately: (2K+2)(2K+1)log2(2K+1) i.e., complexity is reduced for (2K+1)/(K+1) times. Obtained results are just approximations and they are not quite accurate for small number of samples but they give some idea about complexity. Results of the separable median filter is of similar quality as for the median filter.

The other form of the simplified median filter is recursive median filter. It was applied in systems with small memory space. It works as follows. Assume that we want to filter out pixel (n,m). Apply median filter and obtained result we put in the original image on the same position. For filtering of (n+1,m) we are using already filtered pixel (n,m)!!! For recursive filtering we are avoiding output image since all results are memorized directly to the original (noisy) image.

%the same as in the median Eliminated step (see [M,N]=size(x); comment sign %). x=double(x); %%%y=x;!!!!! for m=1:M for n=1:N z=x(max(1,m-mM):min(M,m+mM),max(1,n-nN):min(N,n+nN)); x(m,n)=median(z(:)); end end Filtering result recorded in

original image!

Recursive median reduces memory requirements. Presently memory is not expensive and excluding specific machines we have enough memory for median filtering. Recursive median is able to reduce impulse noise influence in better manner than the median filter (filtering is performed on already filtered samples). However, recursive median (for relatively large filter size) is able to blur image in direction of filtering. For engineering application it is very bad property but for some artistic (for example Photoshop) tools it is desired.

ML filter is introduced for noise that corrupts images. ML filter for Gaussian noise is the MA, while the median is ML filter for Laplace noise. When we do not know type of noise we can perform median filtering since it produces relatively accurate results for wide range of impulse noise environments. Sometimes we have the Gaussian noise almost all the time but sometimes impulses appear. This noise type is called the mixed noise. What to do in this case?

L - filters

We can apply the median filters since they produce relative accurate results. Better results can be obtained using the ML filter for particular noise type. However, the ML filters for mixed noise case cannot be written in closed form and their realization requires iterative procedures. Instead of the ML filter design we are using approximation called L-filters that combines properties of the MA and median filters.

L-filters - definition

L-filter is abbreviation of the linear combination of order statistics. Assume that we have image and that we consider pixel (n,m) with symmetric local neighborhood of size (2N+1)x(2M+1). In the case of the median filter we sorted pixels from the neighborhood into non-decreasing sequence: x(1) x(2) x(3) ... x[(2N+1)(2M+1)+1]/2 ... x[(2N+1)(2M+1)-1] x(2N+1)(2M+1) Values x(i) satisfy x(i)x(i+1). Median is equal to: x[(2N+1)(2M+1)+1]/2.

L-filters - Definition

L-filter can be defined as:

y ( n, m ) =

(2 N +1)(2 M +1)

i =1

ai x((in),m )

L-filter coefficients.

Coefficients ai commonly satisfy the following properties: (2 N +1)(2 M +1) ai = a(2 N +1)(2 M +1)+1i ai = 1 Un-biasness condition that take

i =1

Condition that keeps energy (luminance) of the output signal to the same energy as in the input signal.

values from on the same distance from the median with the same weights.

Two introduced filters: MA and median are special cases of the L-filter class:

MA filter follows for ai=1/(2N+1)(2M+1) for each i. Since we assume that number of pixels in neighborhood is odd we can obtain median for a[(2N+1)(2M+1)+1]/2=1 and ai=0 for i[(2N+1)(2M+1)+1]/2. Two special cases of the L-filters class are the MAX filter giving the maximal luminance in the neighborhood: a(2N+1)(2M+1)=1 and ai=0 and i(2N+1)(2M+1) and MIN filter for: a1=1 and ai=0 for i1.

The most important form of the L-filter is the trimmed mean that takes average of several pixels from the around the median. Its coefficients ai can be calculated as: K=(2N+1)(2M+1)

1 i [( K + 1) / 2 K ,( K + 1) / 2 + K ] ai = 2K + 1 0 elsewhere

[0,0.5]

For =0 the -trimmed mean is equal to the median while for =0.5 it is equal to the MA filter. Values of practical importance are values inside of these two extreme values (0,0.5) since they can offer possibility to remove impulse noise and reduce influence of the Gaussian noise. An alternative form of the L-filter is case when we calculate median and after that average pixels that are on distance smaller than from adopted parameter .

For self-exercise

Create functions for MA and median filters for non-rectangular neighborhood. Create function that realize the -trimmed mean with as an input argument.

Hint. Realization is close to median. We perform sorting (function sort in MATLAB), and after that perform averaging of central values in the sorted sequence (depending on ).

Realize the L-filter described in lower part of the slide 45. Realize the myriad filter for local neighborhood of given size. Myriad filter is the ML filter for Cauchy noise that has probability density function proportional to s/(2+K2) where K is the socalled linearization parameter.

Hint. This filter output cannot be represented in the closed form and we need to develop some iterative algorithm. Initial iteration can be output of the MA or the median filter.

For self-exercise

Real image is corrupted by the Gaussian noise. Can the mean squared error of the output in the case of the MA filter be larger than in the case of the median filter? Why? Repeat the same analysis for the images corrupted by the Laplacian noise and filtering with these two filters. Can the ML for this noise (median) produce worse results than the MA filter and why? Create separable and recursive median filter. Compare the separable median, median, -trimmed mean and MA filters for filtering of the salt&papper noise, Gaussian noise and Laplacian noise.

Hint. Experiment can be performed as follows: consider several images, consider different noise levels, for each level repeat simulations for various noise realizations (Monte-Carlo simulation), consider different filters parameters (size, , etc), various noise parameters and try to make some conclusions.

For self-exercise

Determine spectral characteristics of the median filter. Determine the spectral characteristics of the MA filters (MA filters in frequency domain). Does the L-filter defined in the lower part of the slide 45 follows the common requirements of the L-filters presented on slide 42. Consider a noise with uniform probability density function on interval from to . Define the ML filter for this noise. Can we define the filter from the L-filter class to filter out this noise? Based on which two special filter types from the L-filter class we can filter-out this noise? Noise has with probability (1-p) the Gaussian pdf with mean 0 and variance 2 (it is sometimes denoted as N(0,2) where N comes from term normal pdf) and with probability p (p is relatively small commonly below 10%) Gaussian pdf with mean 0 and variance K22 where K is greater than 1 and commonly it is 3 or 5. Determine corresponding ML filter for this noise. Can it be given in closed form? Is, for this noise type, better the MA or median filter?

For self-exercise

Realize the ML filter for mixed Gaussian noise (probably you will need some iterative procedure). Compare it with the median and the MA filter. Are obtained results in line with expectations and why? Median filter takes as its output arbitrary pixel from the neighborhood. We want to keep original pixel if it is not corrupted by the impulse noise. There are modification of the median filter called weighted median where in sequence used for sorting, central pixel is repeated several times in order to increase probability that it is output of the filter. Realize this filter and compare results obtained with various number of repetitions? Rayleigh noise is equal to the square root of the sum of squares of two Gaussian noises. It is common in practice. Can median, MA and other introduced filter forms be effectively used for filtering of this noise? What is the ML filter for this noise? How to modify the L-filter or median filter in order to have acceptable results for this noise environment?

For self-exercise

Create functions for calculation of PSNR, SNR, and other quality measures (based on original and noisy or filtered image. Create function for the ISNR evaluation. Can result obtained with the ISNR be negative and what is conclusion that we can drawn based on this result?

FILTERS AND OTHER STORIES

ML filters are used when we know the noise type. Can we improve filtering ability when we know something more about noise process. Yes, we can. Consider the following case. Let an image x(n,m) be corrupted by white noise (n,m): y(n,m)=x(n,m)+(n,m) Lets try to find coefficients of the linear (convolution) filter h(n,m): s(n,m)=y(n,m)*n*mh(n,m)

Wiener filter

Assume that the filtering goal is minimization of the mean squared error:

1 MSE = NM

N M

N M

n =1 m=1

Easy! We should calculate partial derivatives of the MSE along h(k,l) and solve system of equations for these derivatives equal to zero. I will skip several steps in this procedure (try it yourselves).

Wiener filter

The obtained set of equations could be written as:

h(n k , m l ) R

k l

yy

(k , l ) = Rxy (n, m)

Ryy (k , l ) = E[ y (n, m) y (n k , m l )]

Rxy (k , l ) = E[ x(n, m) y (n k , m l )]

Convolution is obtained on the left-hand side of this equation. Then we can apply the 2D DFT on both sides of equation.

Wiener filter

The calculation of 2D DFT gives:

Cross-spectral power. Spectral power.

Wiener filter

The considered filter can be written as:

This filter is called the | X (1 , 2 ) |2 H (1 , 2 ) = Wiener filter. 2 2 | X (1 , 2 ) | + | N(1 , 2 ) | | Y (1 , 2 ) |2 | N(1 , 2 ) |2 H (1 , 2 ) = | Y (1 , 2 ) |2

| N (1 , 2 ) |2 = 1 | Y (1 , 2 ) |2

Unknown Known

For determination of the filter response function in spectral domain we need to have information about noise spectral power, i.e., to know parameters of noise.

On the first glance the Wiener filter seems non useful since we have to know noise and signal in advance. However when we know signal in advance we will not apply the filtering algorithm Fortunately, the Wiener filter can work accurately when we have relatively good estimation of noise parameters. This can be available in numerous applications. For example, during TV signal transmission (how it looks like noise during on TV) we can remove one frame. Then on the receiving side of the channel noise parameters can be estimated and obtained noise spectral power can be used for filtering in adjacent frames. This procedure should be repeated from time to time.

We know that the Gaussian noise tends to increase the spectral power of corrupted image for amount that is proportional to variance. For variance estimation we can take some percentage of the smallest 2D DFT samples. Then mean or median value of this samples can be used as estimation of the variance. This value can be used as an estimated spectral power of noise.

Very often we are using the spectral threshold (denoted with T) and the Wiener filter is designed based on the threshold as:

1 | Y (1 , 2 ) | > T H (1 , 2 ) = 0 | Y (1 , 2 ) |2 T

2

This simplified form produces very often quite accurate results that are sometimes better than those produced with some sophisticated methods.

Wiener filters are implemented in various equipment since we can know in advance type of disturbance that is common for that machines. For example, it is known that in ultrasound imaging we are dealing with Poisson noise and we can know parameters of this noise in advance. The Wiener filters are popular since they are linear and they can be efficiently realized in real-time.

What is assumed under image reconstruction? Rough definition: Reconstruction is obtaining estimate of original image that is subject to various distortions. In practice, under distortion, we assumed different events and techniques that can be applied to digital images. For example: Artistic image is damaged. We want to reconstruct it. The first step is digitalization. Using the software tools we can try to reconstruct original shape of the image. This is the task for image restorers in process of real image reconstruction. Similar procedure can be applied to images, icons, but also to pottery and statues. There are software tools that can reconstruct broken paintings or pottery similarly like puzzles.

We will not so deep in this area since we will consider just this model. y(n,m)=x(n,m)*n*md(n,m)+(n,m) Noise

Damaged image. Signal of interest that we want to Distortion modeled reconstruct. as linear space relation exhibits: invariant process.

Now we can apply Wiener filtering assuming that N(1, 2) is known.

Inverse Filtering

Wiener filter in this case is:

| N(1 , 2 ) |2 H w (1 , 2 ) = 1 | Y (1 , 2 ) |2 H w (1 , 2 ) X (1 , 2 ) = Y (1 , 2 ) D (1 , 2 )

denotes approximately Inverse filter.

This filter is estimator of X(1,2)D (1,2). When we know D(1,2) then X(1,2) (2D FT of original image) can be calculated as:

In order to illustrate the problem that can appear in the case of the inverse filtering, consider an example of the motion blur in x-axis direction with length K. 2D DFT of the motion blur calculated over the entire image dimensions NxM is: N 1 M 1 2mk2 2nk1 D(k1 , k2 ) = d (n, m) exp j j = N M n =0 m = 0 2Kk2 1 exp j K 1 1 M 2mk2 1 = exp j = = K m =0 M K 2k2 1 exp j M

Kk2 sin 1 M j ( K 1) k2 / M D(k1 , k2 ) = e K k2 sin M The problem is the fact that the 2D DFT has zeros for each integer k2 that can be written as k2=Mr/K where r is integer. Then we have problem of division with zero that can damage significantly reconstructed image. Transfer function of the inverse filter can be modified as:

H (k1 , k2 ) = 1/ D (k1 , k2 ) H (k1 , k2 ) = 0

for for

| D ( k1 , k2 ) |> | D ( k1 , k2 ) |

The inverse filtering is often performed in combination with the Wiener filtering. Commonly, these two filters are two stages in image reconstruction and order how they are applied is not important. It is important that the inverse filter does not amplify noise at high frequencies. From this reason the inverse filter transfer function in frequency domain can be limited to 1 for highfrequency region:

Adopted parameter.

In addition, there are iterative form of the inverse filters commonly applied in space domain. They are using the obtained image as initialization and they are approaching to the inverse filter results in iterative manner. One of this techniques is given in the textbook (from brevity reason we will avoid it here). For application of the inverse filters we need to know distortion! Fortunately, producers of sensors commonly know distortion introduced by their sensors. Then they can perform inverse filtering using software tools that is commonly cheaper than by using more expensive sensors. In addition, in other applications it is very often possible to estimate parameters of disturbance and to design proper inverse filter.

In numerous application instead of entire frequency domain it is used for inverse filtering only region up to the first zeros from origin in corresponding directions. Inverse filtering is a form of convolution equation. There are specific mathematic discipline related to solving such equations. Then there is a room that some of results from this area can be used in design of inverse filters.

Linear (convolution) filtering including MA, Wiener, inverse filters can be applied for color images separately on each channel. In the case of nonlinear filters (median, L-filter, etc) can also be adopted similar strategy. Then these filters are called marginal. For example in the marginal median we assume that median filter is applied to each channel of image separately. Better results can be achieved with more complex vector median filters.

Definition of median

We considered instant n and local neighborhood [n-K,n+K]. Then output of the median filter is calculated as a value that minimizes:

J () =

n+ K k =n K

| x(k ) |

This can be assumed as a value from the set {x(k)|k[n-K,n+K]} with minimal sum of distances to all other points from the set.

Vector median

Luminance of color-pixel can be written as a vector x(n,m)=(r(n,m),g(n,m),b(n,m)). Output of the vector median for local neighborhood (here we give 2D local neighborhood) is from the set {x(k,l)|(k,l)[n-K,n+K]x[m-L,m+L]} having the smallest sum of distances to all other points from the set

J () =

n+ K

k =n K l = m L

d (x(k , l ), )

m+ L

Distance in 3D space

For distance function defined as:

d ( x1 ( n, k ), x 2 ( n, k )) = ( r1 ( n, k ) r2 ( n, k )) 2 + ( g1 ( n, k ) g 2 ( n, k )) 2 + (b1 ( n, k ) b2 ( n, k )) 2

output is MA filter for channels of image. For Euclidian distance:

d ( x1 ( n, k ), x 2 ( n, k )) = ( r1 ( n, k ) r2 ( n, k )) 2 + ( g1 ( n, k ) g 2 ( n, k )) 2 + (b1 ( n, k ) b2 ( n, k )) 2

we obtain vector median filter output.

For vector median calculation we need to determine distance between NxM pixels. It means that vector median filtering is more demanding than the marginal median filter evaluation. Vector median produces significantly better color that is equal to luminance of some of local neighbor pixels but not to artificial combination of luminance of three possible various pixels. Then the marginal median can produce some non-realistic colors that can be easily observed in color image. Thus, the vector median filter represents colors in image in more realistic manner than the marginal median filter.

Here, we describe pseudocode of the vector median realization (there is a tool developed for this purpose):

In nested loops we should process all image pixels Determine local neighborhood for each pixel For each pixel from the considered neighborhood determine sum of distances between its colors and colors of all pixels in the local neighborhood. Pixel producing the smallest distance is used as an output of the vector median filter.

For self learning realize the vector median filter, consider its numerical complexity and compare it with the marginal median filter.

Vector L-filters

The Euclidian distance is just one possible method for evaluation color differences. Then, alternative vector median forms can be developed. How to realize the vector L-filters for color images? It is relatively simple technique. It should be sorted pixels from the local neighborhood according to sum of distances between colors. The output of the L-filter is average produced by using several pixels having the smallest sums. For self-exercise realize this filter. There are numerous variants of the adaptive vector filters.

Tool

We created Tool for non-linear image filtering (for grayscale and color images) that realizes introduced and also numerous other filters. Students can download version 1.02 of this tool. Major revision is under way and it should produce more working comfort and increased computation efficiency.

Pseudocolors

Determination of pseudocolors is procedure where grayscale image (image without color information) is replaced with color version. The first idea is to try to get color image of old photos but commonly under pseudocoloring is assumed alternative operation. This operation is commonly performed when we want to get mode realistic image in color than in grayscale, i.e., image based on grayscale where some details can be more easily observed than in grayscale images (here, property of human vision that it is more sensitive to colors than on luminance is used).

Pseudocolor applications

Pseudocolors are used in video surveillance systems (X-ray systems at airports) where security clerks are trying to find explosives, drugs and other goods subject to smuggling. The most important set of pseudocolor applications is in medicine. For example, small blood vessels are searched in tissue that produces similar luminance, ultrasound imaging of kidney, color ultrasound systems, are just part of applications in medicine.

Pseudocolor methods

Here, we will explain three techniques for obtaining pseudocolors. The first technique is the simplest. RGB color is given to any shade of grayscale. This is similar to the color map described earlier but only here the grayscale is assumed to be represented with corresponding color map. Companies that produce equipment study different color maps and try to find those that produce the best results. In fact buying this product you are buying the colormap. Similar technique is situation where we are adopting several levels of luminance and values between two levels we represent with one color while between other levels with alternative color.

More sophisticate (and harder for implementation) is pseudocolor technique in spectral domain. Assume that we have grayscale image with small blood vessels and background tissue of the similar luminance. Small vessels are on the higher frequency than surrounding tissue. The basic idea is to represent details on the higher frequencies with blue and background tissue on lower frequencies with red. How to do it?

Spectral domain

Assume the original image f(n,m), than red, green, and blue channels can be obtained as: CR(n,m)=f(n,m)*n*mhL(n,m) CG(n,m)=f(n,m)*n*mhB(n,m) CB(n,m)=f(n,m)*n*mhH(n,m) Where hH(n,m), hB(n,m) and hL(n,m) are impulse responses for highpass, bandpass and lowpass filters. These filters in spectral domain can be defined as:

0 D(1 , 2 ) DH H H (1 , 2 ) = 1 D (1 , 2 ) > DH

1 D (1 , 2 ) DL H L (1 , 2 ) = 0 D (1 , 2 ) > DL

1 DL < D ( 1 , 2 ) DH H B ( 1 , 2 ) = elsewhere 0

2 2 where: D(1 , 2 ) = 1 + 2

What we are getting buying equipment with pseudocolors in spectral domain? Companies sell to us parameters DL, DH, modifications related to function D(1,2) as well as different filters than here described cut-off filter. Finally, the most important are weights associated with corresponding channels. Research related to the pseudocolors is assembled together with sensors and other devices and sold as a machine.

Example

Pseudocolors in security systems is very important issue. Here image of the suitcase obtained using X-rays is given with its pseudo version.

Red rectangular of the right image represent the TNT explosive. This material is very difficult for examination from the grayscale. Since this pseudocolor image is dominated with red and blue colors we can learn that for imaging is used some technique in spectral domain.

In medicine (and sometimes in other fields) pseudocolors are determined in one specific manner. Two images of the same object are formed. We are depicting with colors differences between these images. For example in medical scanners initial images are created while the other set of images is obtained after some substance is injected in patient body. Difference between two images is created. Positions without differences can suggest that these blood vessels are not in proper function. In similar manner it can be performed monitoring of the kidney. Physicists considers kidney and it makes several frames in different time instants. Subtracted image is superposed in different color to original image. Differences in various parts of kidney suggest that it performs correctly since it corresponds to production of urine while positions without differences means that it does not work properly.

Halftoning - Dithering

Halftoning and dithering are strategies for printing images in continuous scale to printers that have single color. Here, drawback of the human eye that it recognizes several dots on small distance as a color is used. Then when we want to print black we put more dots while when we want to print gray we put less dots while when color close to white is printed very small number of dots is used.

Halftoning

Halftoning is simpler strategy for obtaining discrete tones from continuous ones in order to produce image suitable for printing. Behind this strategy is quite simple idea. One pixel of original image represents x black-white spots on printer (standard for text in Europe is 8x8 or 10x10 while in Japan Assume that luminance can be within limits of [Amin,Amax].

and some other Asian countries it is 12x12 in order to allow reproduction of kanji Chinese characters).

Halftoning

Region of possible luminance can be divided in x+1 in the following manner: i-th region is [Amin+(i-1)A, Amin+iA) where A=(Amax-Amin)/(x+1) and i[1,x+1]. The first region (i=1) of luminance represents the darkest region of the image and we give to it x black dots. Any further region has smaller number of dots with respect to previous one. For example i-th region has xi+1 black dots. We can illustrate this on the example with zone of 2x2 pixels and for luminance within [0,255].

Halftoning - Example

4 black Luminance [0,51) Luminance [51,102) 3 black 2 black Luminance [102,153) Luminance [153,204) 1 black Luminance [204,255] without black

Halftoning is simple procedure but it has a significant drawback that limits its applications. Note that technical details of this procedure can be represented in slightly different form.

Halftoning example

Assume that image in larger zone has uniform luminance and that this represent case with two black pixels.

Here is given enlarged image of this pattern. Humans cannot distinguish black and white dots on small distance but we are able to recognize periodic patterns even when they are quite small. The biggest problems are lines and this image could look like:

Dithering

Dithering is more complicated methodology for obtaining binary image from grayscale that includes some additional (stochastic) elements. Halftoning can be assumed as truncation of colors algorithm. In dithering we want to reduce and distribute errors that is produced by process of obtaining binary from continuous scale image (or close to continuous scale). Here we are using dithering matrix that is commonly defined using recursive relationships.

Dithering matrix

Dithering matrix has dimensions nxn and it can be obtained using matrix of dimensions (n/2)x(n/2):

2 4 D n / 2 + D00U n / 2 Dn = n / 2 4 D + D120U n / 2 2 4 D n / 2 + D01U n / 2 2 n/2 n/2 4 D + D11U

matrix of smaller dimensions matrix of dimensions (n/2)x(n/2) with all ones (ones in MATLAB) Element on position (i,j) in dithered matrix of dimension 2x2. Here index begins with 0.

LM0 2OP = N 3 1Q

Initial matrix.

Dithering matrix

Dithering matrix 4x4 is (check it):

LM 0 8 2 10OP 4 6 MM12 11 14 9 PP D = 3 1 MN15 7 13 5 PQ The most common dithering matrix has dimension 8x8.

4

Determine it for self-exercise. Obtained dithering matrix is periodically repeated in both directions until entire image is covered.

Since maximal value from the dithering matrix is smaller than maximal luminance of image commonly dithering matrix is scaled (in fact multiplied) with some constant before other operation. Commonly this constant is Amax/Dmax where Amax is some (large) value of the image luminance and Dmax is maximal value written in the dithering matrix. Binary image can be determined as:

1 g (k , l ) = 0

f (k , l ) > T (k , l ) elsewhere

where threshold is: T (k , l ) = ( Amax / Dmax ) D n (k mod n, l mod n)

original image

We performed experiment with small number of pixels of original image in order to results of experiment be more obvious. Careful look at halftoned image can even tell to us info about shape of binary font.

The most popular dithering scheme today is the FloydSteinbergova technique. It uses one additional probabilistic level since errors in quantization are distributed left and bellow considered pixel. This technique is designed in such a manner that it can be applied on pixel-by-pixel base without creating large dithering matrix. Halftoning (dithering) in color is additionally complicated since it should be performed on larger number of channels and we should keep in mind possible interaction between various halftoned channels. Then there are special techniques for angle adjustment between various channels in order to get optimal halftoned image.

For self-exercise

Create the Wiener filter for image corrupted by the Gaussian noise. Filter should estimate standard deviation of the noise using approximate relationship:

= median{| x(n) x(n 1) |, n [2, N ]}/ 0.6745 as a median of absolute difference between neighbor pixels. Note. This is 1D relationship. It should be adjusted to the 2D case. Realize inverse filter. Do you obtain expected results and if not try to determine reasons. You can find on Internet or from the lecturer more details related to inverse filters and their practical application. Consider methodology for determination of the motion blur parameters for motion in arbitrary direction. Try to include this information in inverse filter for motion blur.

For self-exercise

Inverse filtering and Gaussian blur. How to estimate parameters of the blur. Consider methods for realization and try to determine problems in the realization. Try to find additional references (textbooks, Internet and lecturer) where you can find more details how to overcome problems in realization of this filter. Realize marginal median, vector median, vector L-filter for color images. Consider realization of the toolbox for non-linear filtering especially realization of the vector filters for colored images. Try to find methods for reducing complexity of these algorithms. Realize technique for pseudocoloring described within lecture. Describe problems you have found in realization. Realize both techniques for halftoning (binary font and dithering).

For self-exercise

Find more data about the Floyd-Steinbergovom algorithm for dithering and try to realize it. Try to find more details related to color dithering and realize some of these techniques. Miniproject. One very important problem in practice is so called inverse dithering. It is technique that produce continuous scale image from binary one. This in fact filtering of binary image that produces non-binary output. Propose your technique for solving this problem and try to find something in available publications.

EDGE DETECTION

Importance of Edges

From the really beginning of the course we heard that there is simple rule in the images: low frequencies have significant amount of image energy but they are relatively unimportant from the point of view of visual image quality from both human and machine vision; high frequency region has small energy but it is very useful part from the point of human and machine vision. This high frequency region can be very prone to the noise influence. The most important part of the high frequency region are edges: object and image are recognizable just based on images; they can be recorded in the binary format and they can be efficiently processed; based on detected images we can easily adjust local neighborhood for filtering algorithms, etc.

There is no precise definition but we can tell that edges are regions of the images with abrupt luminance variations. How did we detect similar abrupt changes in functions during our previous courses? One of the simplest techniques to determine fast variation is based on first derivative. Can we apply this strategy for edge deterction? Yes, we can but with slight modifications.

First derivative

The first derivative is defined as: df ( x) f ( x + x) f ( x) f '( x) = = lim x0 dx x For the 2D functions derivative is determined as: f ( x, y ) f ( x, y ) f ( x, y ) = [ f x ( x, y ), f y ( x, y )] = , x y

Abrupt variation of luminance can be detected based on magnitude (amplitude) of derivative that can be calculated as:

Partial derivatives along x and y

e( x, y ) =

f x2 ( x, y ) + f y2 ( x, y )

Alternatively edge detector e(x,y) can be calculated as: e( x, y ) =| f x ( x, y ) | + | f y ( x, y ) | Local edge direction can be calculated as: f y ( x, y ) ( x, y ) = arctan f x ( x, y ) Obviously, e(x,y) will be compared with some threshold in deciding if it is edge or not while (x,y) will give us some additional information related to the edge.

The basic idea related to the edge detection is comparison of e(x,y) with some threshold that can be positiondependent T(x,y). We are making binary decision: edge or not edge. Therefore, the image representing edges is binary image: (x,y) belongs to edge 1 e ( x , y ) T ( x , y ) iv( x, y ) = 0 e ( x, y ) < T ( x, y )

(x,y) does not belong to edge

The first noticeable problem is fact that we have no continuous image but discrete and that we need to use difference instead of derivatives.

As a derivative approximations we can consider the following differences:

f x (n, m) = f (n + 1, m) f (n, m)

Instead of (x,y) we consider here pixels (m,n).

The second significant problem is the fact that differences amplify noise.

Assume that we have image without image corrupted by Gaussian white noise with variance 2 and that noise is independent on position of pixels (non-correlated in neighbor pixels). Then, variance of difference along x-coordinate is:

=0 increased variance

= E{ 2 ( n + 1, m )} 2 E{( n + 1, m )( n, m )} + E{ 2 ( n, m )} = 2 2

Basic idea for overcoming noise related problems is averaging certain number of pixels along edge direction. Thus, edge detection along x-direction for pixel (n,m) is not using only luminance in (n,m+1) and (n,m) but also luminance for pixels (n+1,m+1), (n+1,m) and (n-1,m+1), (n-1,m). There is large class of edge detectors of this type including very popular Sobelov and Prewittov detectors. Before we proceed with these detectors we will describe the Roberts detector. It uses 4 neighbor pixels for calculation of the detector response.

Roberts detector

This very simple detector can be evaluated as:

e ( n , m ) = [ f ( n, m )

f (n + 1, m + 1)]2 + [ f (n + 1, m)

f (n, m + 1)]2

4 pixels are used for edge detector evaluation but without information about edge direction. In design of this detector it was used knowledge about human visual system and its ability to detect edges. This operation is assumed to be performed by eyes in process of edge detection but in practice results achieved with this detector are poor.

Mask

The most common group of edge detectors is based on mask. Mask is 3x3 matrix. Here edge detector response can be evaluated using convolution with this matrice. Two matrices are designed for edge detection along x- and yaxis. Rules for matrix selection.

Matrix used for detection along x-axis should be the same as transpose or rotated for /2 version of matrix used for detection along y axis. In this manner the same importance is given to both orthogonal directions. Here we will consider detection along x-axis with matrix:

a11 a 21 a31

Mask design

Central pixel is pixel of interest. We want to subtract pixels on position a12 and a32. However, we are using neighbor pixels to reduce noise influence. Then right and left pixels should be taken with the same strength and from these facts follows that a11=a31, a21=a23 and a31=a33:

a11 a12 a11 a a22 a21 21 a31 a32 a31 Pixels in bottom row are subtracted from pixels in top row and these coefficients should have opposite signs:

a11 a 21 a11

Mask design

For uniform image we want that edge detector produces result equal to 0. Then we need that sum of mask coefficients be 0:

Commonly edge detector used for detection along one direction should produce result equal to 0 for edge in normal direction. Then we set central position in mask equal to 0:

a11 a 21 a11

a11 0 a11

a12 0 a12

a11 0 a11

Mask design

Mask can be divided with a11 since it is unimportant multiplicative constant:

K 1 0 0 1 K

1 K 1

1 0 K 0 1 0

K=1Prewitt matrix K=2Sobel matrix The most important edge detector based on mask.

MATLAB Function

Matlab function for edge detection is edge. It is quite simple for using: E=edge(G,type,parameters); where G is a grayscale, type is used detector type (all described so far and some other roberts, prewitt and sobel) while parameters are parameters of considered detector. Function output E is binary image. Commonly the third parameter (it can be avoided and MATLAB would take some default value) is threshold i.e, within domain [0,1] (it can be also empty matrix []), while the fourth parameter for sobel and prewitt detector is direction of edges: horizontal, vertical or `both.

The threshold determination is rather difficult in some systematic manner. Namely, for dark region of image, detector threshold should be small since response function is small while in bright regions this threshold could be larger. Even more difficult problem could be the fact that there are lines that are in other directions not just in x and yaxis directions and that both detector responses along x and y axes could be small. Finally, it can happen that several consecutive pixels have large detector response since we can have gradual variation of luminance instead of abrupt variation.

It is not quite easy to find in serious image processing books details related to the threshold in the edge detection since it is not quite simple to present in consistent way. Due to the completeness reasons we will here present how it is usually performed in the case of detectors from the class of the Sobel and Prewitt detectors. Detector response function can be adopted as:

2 2 e ( n , m ) = ex ( n , m ) + e y ( n , m )

T= 4 e(n, m) MN m n

When we want to detect edge in the x-axis direction it is required that |ex(n,m)|>|ey(n,m)| and when we want to detect edge along y-axis it is required that: |ey(n,m)|>|ex(n,m)|. In addition, it is required that local maximum be larger than adjacent samples of detector response in corresponding directions: |ex(n,m)|>|ex(n-1,m)| and |ex(n,m)|>|ex(n+1,m)| for detection of edge in x-axis direction or |ey(n,m)|>|ey(n,m-1)| and |ey(n,m)|>|ey(n,m+1)| for edge detection in direction of y-axis.

Direction of edge

Obviously, determination of the detection threshold is quite complex issue. Even threshold setup can be different for various detectors. There is a problem of estimating direction of edges. For the mask based detectors (Sobel, Prewitt etc) we have detector responses for x and y coordinates. There is possibility to estimate edge direction based on ex(n,m) and ey(n,m) as

e y ( x, y ) ( x, y ) = arctan ex ( x , y )

Kirch detectors

Kirch designed additional masks for detection of edges in other directions. For example detector that is sensitive to edges in directions of x, y-axes and axes that are giving angles of /4 and 3/4 with respect to main axes can be designed using 4 matrices:

1 1 1 0 0 0 1 1 1

1 0 1 1 0 1 1 0 1

0 1 1 1 0 1 1 1 0

1 1 0 1 0 1 0 1 1

Direction of edge is determined using the largest detector response produced with these matrices.

Kirch detectors

Kirch designed detector with eight matrices corresponding to eight various directions. 5 5 5 3 5 5 3 3 5 3 3 3 3 0 3 3 0 5 3 0 5 3 0 5 3 3 3 3 3 3 3 3 5 3 5 5

3 3 3 3 3 3 5 3 3 5 5 3 3 0 3 5 0 3 5 0 3 5 0 3 5 5 5 5 5 3 5 3 3 3 3 3

Laplace detector

The logic behind the Laplace detector is quite different than the mask-based detectors. The Laplace differential operator is defined as:

1 2 f (n, m) f (n, m) [ f (n + 1, m) + f ( n 1, m) + f ( n, m + 1) + f (n, m 1)] 4

Edge (abrupt change in luminance) can be detected as zero in the Laplace operator (position with zero-crossing).

If red is equal to 100 and green 50 Laplace detector response for pixel X is: 100-350/4>0 while for o: 50250/4<0 suggesting that zero-crossing (edge) is between these two pixels.

o X

The main drawback of the Laplace detector is its sensitivity to noise influence. Namely, when image is uniform and corrupted with just small noise the Laplace detector would detect large number of edges as positions of zerocrossing. There are numerous (more or less acceptable techniques) for removing noise in edge detection process. One of them is to calculate local variance of pixels in local neighborhood:

1 ( n, m ) = (2 M + 1) 2

2 n+ M k =n M l =n M

n+ M

f ( k , l ) f ( n, m )

local neighborhood

In the previous relation holds:

1 f ( n, m ) = (2 M + 1) 2

n+ M k =n M l =m M

m+ M

f (k , l )

The Laplace detector introduces threshold of variance. For small variance we assume that edge is not detected (image is almost constant) while for large variance and zero-crossing we assume that edge is detected. There is alternative that region where edges can be detected be calculated based on difference:

w( n, m) = max A{ f ( n, m)} min A { f ( n, m)}

Maximal and minimal luminance in local neighborhood

Laplace-Gauss detector

A technique for reducing noise influence in the Laplace detector is filtering of image or detector response with Gaussian filter (Gaussian blur). Both operations are linear and they can change order without affecting results. Obtained detector is in literature called LoG (Laplacian of Gaussian). Matrix in the Gaussian is given in the shape:

h(n, m) = exp((n 2 + m 2 ) / 2 2 )

LoG detector

The most common matrix for LoG detector is:

0 0 0 1 1 2 0 1 0 0 0 0 16 2 1 2 1 0 1 0 0 1 0 2 1

However, for various values of follows different matrix for LoG but given matrix is the most common in practice.

The Canny detector is a special form of the LoG detector assumed to be one of the best (but not the simplest one).

Canny detector

Three different filters are employed in the Canny detector realization. The Gaussian with impulse response:

h(n, m) = exp((n 2 + m 2 ) / 2 2 ) /(2 2 )

adopted constant

Filters with impulse response equal to partial derivatives of the Gaussian filter impulse response:

hx (n, m) = h(n, m) / n = n exp((n 2 + m 2 ) / 2 2 ) /( 2 ) hy (n, m) = h(n, m) / m = m exp((n 2 + m 2 ) / 2 2 ) /( 2 )

g (n, m) = f (n, m) *n *m h(n, m)

This operation is used to reduce the noise influence.

Canny detector

After this operation output of the Gaussian filter is filtered with filters having responses equal to derivatives of the Gaussian function:

ex (n, m) = g (n, m) *n *m hx (n, m)

2 2 e ( n, m ) = e x ( n, m ) + e y ( n , m )

Commonly instead of e(n,m) we present binary image obtained by comparing it with a threshold.

There are several steps in obtaining edges in the Canny detector using e(n,m):

Determine percentage of image that could belong to image. This percentage corresponds to the largest values of e(n,m). Eliminate all points that are not local maximum along the corresponding direction. In the Canny detector check it is performed also for diagonal directions. Candidates for edges are pixels that are larger than the selected threshold (threshold 1), that among the largest detector responses, and that are local maximum. Sometimes we perform determination of the edges by additional comparison with second threshold (larger than threshold 1) and final response is obtained as some combination of edges obtained with these two thresholds (larger number of pixels is recognized as edges with respect to smaller threshold and larger than in the case of higher threshold).

HUHUHUH!!!

Isolated points can be detected by using mask function of the shape: 1 1 1

1 8 1 1 1 1

When edge is not abrupt change in luminance but single line passing over a background in other color we can use matrices of the form:

1 1 1 2 2 2 1 1 1 1 1 2 1 2 1 2 1 1 1 2 1 1 2 1 1 2 1 2 1 1 1 2 1 1 1 2

The simplest technique is to calculate edges for separate channels and to connect them using logical operations: Detectors for various A=imread('flowers.tif'); channels calculated with er=edge(A(:,:,1)); default parameters of the MATLAB edge function. eg=edge(A(:,:,2)); eb=edge(A(:,:,3)); Logical operation OR applied on e=er|eg|eb; edges detected in channels. imshow(e)

This is very simple and produces effective results but recently some more sophisticated techniques for color edge detection are proposed. Interested students can take this techniques as a topic for self-excercise.

We reviewed almost all important edge detectors within our lecture. Recently with a lot success several novel techniques where size of mask is determined in adaptive manner is used. Theoretically almost exact values of the image derivatives are determined in such a manner. This is significant advantage with respect to usage of differences. We have seen that threshold determination is complicated and commonly heuristic (ad-hoc) technique are applied for this purpose.

Hough transform

Images are in some application transformed to binary form using edge detectors and after they are used for recognition of geometrical objects primitives. The simplest primitive object is straight line. It can be represented using linear function y=ax+b (or m=an+b). The main goal is to determine if pixels belong to line and to determine parameters of line (a,b). The procedure is not quite simple. Consider (x1,y1) that belongs to edge but we do not know if this edge is straigh line and if it is we do not know its parameters.

We adopt domain of possible values a and b as Cartisian product AxB where aiA and bjB are possible values from these sets. Denote ai=amin+ia, bj=bmin+jb where amin and bmin are minimal values from corresponding sets. Let amax are bmax maximal values from sets and let number of elements in sets is NA and NB respectively. Then: a=(amax-amin)/(NA-1) and b=(bmax-bmin)/(NB-1). Number of elements in sets should not be too large and too small. Large number increases burden and sometimes makes detection more difficult while small number produces not precise results.

Create matrix P(i,j)=P(ai,bj) where each element from the set has value P(i,j). Initialize all P(i,j) i=1,..., Na and j=1,...,Nb to zero. For each image pixel (xk,yk) with edge detector giving 1 (detected edge) e(xk,yk)=1 determine all straight lines to which this pixel can belong br=yk-aqxk and increment values of P corresponding to these lines: P(q,r)=P(q,r)+1.

Procedure is repeated for each pixel from edges and after that we detect maximum of P and this value corresponds to line with given parameters. After that we remove this maximum and region around the maximum from matrix P and we repeat procedure for detecting next maximum. This procedure is repeated until we detect all lines or when there are no dominant maxima in the matrix P. Even very simple the Hough transform is quite effective line detector. The main problem in the Hough transform is related to lines parallel to x-axis such as x+b=0 where coefficient a approaches toward infinity.

This problem can be overcome by the following parameterization: r=ycos+xsin

y

Rr0

/2>-/2

Limit of domain can be determined by using image dimensions.

r x

In this manner both parameters used for line parameterization have limited domain.

clear Realization is illustrative and without % elements that can produce more efficient N=256; algorithm or more robust result to noise B=zeros(N); influence. %Linije Rectangle (four straight lines). for k=60:150,B(k,k)=1;end for k=20:110,B(k,k+80)=1;end for k=60:-1:20,B(k,120-k)=1;end for k=150:-1:110,B(k,300-k)=1;end M=256; Matrix P, initialized and space of parameters. P=zeros(M); tacnost=0.2; R=linspace(-M,M,M); Additional parameter used in the th=linspace(-pi/2,pi/2,M); algorithm.

for y=1:N for x=1:N if(B(x,y)==1) If pixel belongs to edge... for r=1:M for p=1:M if(abs(R(r)-x*cos(th(p))y*sin(th(p)))<tacnost), P(r,p)=P(r,p)+1;end end ... Increase for 1 elements from P end that represents lines that pass end through the pixel. end end figure(1),imshow(B) Graphical representation. figure(2),pcolor(th,R,P),shading interp

Results

This is very interesting, semi-intuitive, technique that can be subject to numerous improvements (accuracy improvement, robustness to noise influence, determination of short lines) and even today it is one of the most popular algorithms for detection of straight lines. Comparative analysis has shown that it is better than other algorithms almost on all criterion. It can be connected with the Radon transform. It can be used in modified form for other shapes that can be parameterized using 2 parameters or when they can be reduced to case with 2 parameters in some acceptable manner.

The circle can be represented using coordinates of center and radius as: (x-a)2+(y-b)2=r2 Then for parameterization in the Hough algorithm is required three parameters (a,b,r). All pixels belonging to the circuit can be given as: x=a+rcos y=b+rsin Strategies for simplification are developed in order to reduce 3D search (for 3 parameters) to less demanding realization.

Circle parameters

In the case of the circle parameterization we can use edge direction (angle). Namely, we can employ some edge detection algorithm that produces edge e(x,y) and edge direction (angle) (x,y). In the case of the circle the edge direction is normal to radius and search for the circle center can be reduced to 1D search along the line normal with respect to the edge direction. Unfortunately, edge direction can be determined up to limited accuracy and we cannot employ this idea in straightforward manner. Instead we will use segment of the circle in the search procedure.

detected edge direction

For self-exercise

Realize all introduced edge detectors: Roberts, Sobel, Prewitt, Laplace, Gauss-Laplace, Canny detector. Realize described methods for mapping edge detector to binary image. Study methodology for determination of the threshold for various edge detectors (especially for Canny detector) in the MATLAB function edge. For mini-project: Consider edge detectors for color images using the Canny and Kumani operators. In this case, detectors are not evaluated separately along channels but based on channels combinations. For mini-project: Application of the edge detector in realization of adaptive filters.

For self-exercise

For mini-project: Application of color edge detectors in creation of the adaptive color filters. For mini-project: Realize detector of isolated point and detector of contrast lines with methods for threshold selection. For mini-project: Realization of the Kirch detector and possibility to determine edge direction more precisely than as multiple of 45 degrees. For mini-project: Advanced adaptive techniques for detection of edges using precise estimation of derivatives. Create faster version of the algorithm for the Hough transform. Hint: For detected edge point we can go along detected direction but not for all (x,y).

For self-exercise

Establish relationship between the Hough and Radon transform. For mini-project: SLIDE algorithm for edge detection. For mini-project: Improvements of the Hough transform. For mini-project: Compare the improved Hough transform and the SLIDE algorithm. Hint: References related to the SLIDE algorithm can be found from the lecturer. Realizujte algoritam za prepoznavanje krugova i krunih lukova. Za miniprojekat: Algoritam za prepoznavanje elipsi. Kako se pomou Houghove transformacije moe prepoznati poetak i kraj linije? Za miniprojekat: Realizacija algoritama za praenje ivica. Pogledati skriptu.

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