EXP.

NO: DATE:

SIMULATION OF DCT BASED SPEECH/AUDIO COMPRESSION METHOD

AIM:
To simulate and analyze an audio signal compression using discrete cosine transform.

EQUIPMENTS REQUIRED:
1. MATLAB 7.0.1 2. Personal Computer

THEORY:
Audio compression is removal of redundant or irrelevant information from the audio signal. Audio compression allows efficient storage and transmission of audio signal. Discrete cosine transform of an audio signal converts an audio block into its equivalent frequency coefficients. An audio sample is a sequence of real numbers X={x1,xN}. The dct of this audio sample is the sequence, DCT(X)=Y={y1,,yN} such that

where

w(k) =

The compression scheme The coefficients of the DCT are amplitudes of cosines that are within the original single. Small coefficients will result in cosines with small amplitudes, which we are less likely to hear. So instead of storing the original sample we could take the DCT of the sample, discard small coefficients, and keep that. We would store fewer numbers and so compress the audio data. Filling in the details When compressing with DCTs we typically compress small slices (windows) of the audio at once. This is partly so that seeking through the compressed stream is easier but mostly because we want the coefficients in our window to represent frequencies we hear (with large window the majority of the coefficients would represent frequencies well out of the human hearing range).

ALGORITHM:

Initialize the Matlab. Read the audio signal as input for compression. Initialize the compression matrices for compression factors 2,4 & 8. Compress the audio files by taking discrete cosine transform. Plot the original and compressed audio signals. Plot the expanded view of original and compressed audio signals. Plot the spectrogram of original and compressed audio signal

FLOW CHART:

START

Read the input signal

Initialize the compression matrices

Compress the audio signal using discrete cosine transform

Display the original and compressed audio signal

STOP

PROGRAM:
function[]=myDCT() [funky,f]=wavread('F:\SS\funky.wav'); windowsize=8192; sampleshalf=windowsize/2; samplesquarter=windowsize/4; sampleseighth= windowsize/8; funkycompressed2=[]; funkycompressed4=[]; funkycompressed8=[]; for i=1:windowsize:length(funky)-windowsize windowDCT=dct(funky(i:i+windowsize-1)); funkycompressed2(i:i+windowsize-1)=idct(windowDCT(1:sampleshalf),windowsize); funkycompressed4(i:i+windowsize-1)=idct(windowDCT(1:samplesquarter), windowsize); funkycompressed8(i:i+windowsize-1)=idct(windowDCT(1:sampleseighth), windowsize); end figure(1); h1=subplot(4,1,1); plot(funky) title('original waveform'); subplot(4,1,2); plot(funkycompressed2)
4

title('compression factor 2'),axis(axis(h1)); subplot(4,1,3); plot(funkycompressed4) title('compression factor 4'),axis(axis(h1)); subplot(4,1,4); plot(funkycompressed8) title('compression factor 8'),axis(axis(h1)); %expanded view of audio signal figure(2) h1=subplot(4,1,1);plot(funky(100000:120000)),title('portion of original waveform'); subplot(4,1,2) plot(funkycompressed2(100000:120000)),title('portion of compression factor2'); subplot(4,1,3) plot(funkycompressed4(100000:120000)),title('portion of compression factor4'); subplot(4,1,4) plot(funkycompressed8(100000:120000)),title('portion of compression factor8'); %spectogram of audio signals figure(3) subplot(4,1,1) specgram('funky'),title('original waveform'); subplot(4,1,2) specgram(funkycompressed2),title('compressionfactor2'); subplot(4,1,3) specgram(funkycompressed4),title('compressionfactor4'); subplot(4,1,4) specgram(funkycompressed8),title('compressionfactor8'); %saving to wave files wavwrite(funkycompressed2,'funky2') wavwrite(funkycompressed4,'funky4') wavwrite(funkycompressed8,'funky8') %playing files disp('original');
5

wavplay(funky,f); disp('compression factor2'); wavplay(funkycompressed2,f); disp('compression factor4'); wavplay(funkycompressed4,f); disp('compression factor8'); wavplay(funkycompressed8,f);

OUTPUT:
Portion of Original Wav eform 0.5 0 -0.5

0.5

1.5

2.5 x 10
4

Portion of Compression Factor 2 0.5 0 -0.5

0.5

1.5

2.5 x 10
4

Portion of Compression Factor 4 0.5 0 -0.5

0.5

1.5

2.5 x 10
4

Portion of Compression Factor 8 0.5 0 -0.5

0.5

1.5

2.5 x 10
4

O riginal W eform av 0.5 0 -0.5

0.5

1.5

2.5

3.5 x 10

4
5

C pression Factor 2 om 0.5 0 -0.5

0.5

1.5

2.5

3.5 x 10

4
5

C pression Factor 4 om 0.5 0 -0.5

0.5

1.5

2.5

3.5 x 10

4
5

C pression Factor 8 om 0.5 0 -0.5

0.5

1.5

2.5

3.5 x 10

4
5

Frequency

Original Waveform 1 0.5 0

10 Time

12

14

16

18 x 10
4

Frequency

Compression Factor 2 1 0.5 0

10 Time

12

14

16

18 x 10
4

Frequency

Compression Factor 4 1 0.5 0

10 Time

12

14

16

18 x 10
4

Frequency

Compression Factor 8 1 0.5 0

10 Time

12

14

16

18 x 10
4

EXP.NO: DATE:

SIMULATION OF DWT BASED SPEECH /AUDIO COMPRESSION METHOD

RESULT:
Thus the given audio signal has been compressed using discrete cosine transform. And its output was verified successfully.

AIM:
To compress and provide substantial improvements in audio quality at higher compression ratios using Wavelet Transform.

APPARATUS REQUIRED:
1. MATLAB 7.0.1 2. Personal Computer

THEORY: WAVELET COMPRESSION:

The wavelet transform has emerged as a cutting edge technology, Wavelet compression is a form of data compression well suited for image compression (sometimes also video
8

compression and audio compression). Notable implementations are MPEG, MP1, MP2 and MP3 for Audio signals. The goal of audio compression is to encode audio data to take up less storage space and less bandwidth for transmission .Wavelet compression can be either lossless or lossy. Using a wavelet transform, the wavelet compression methods are adequate for representing transients, such as percussion sounds in audio, or high-frequency components in two-dimensional images, for example an image of stars on a night sky. This means that the transient elements of a data signal can be represented by a smaller amount of information than would be the case if some other transform, such as the more widespread discrete cosine transform, had been used.

METHOD OF COMPRESSION:
In order to determine what information in an audio signal is perceptually irrelevant, most lossy compression algorithms use transforms such as the (MDCT) to convert sampled waveforms into a transform domain. Once transformed, typically into the, component frequencies can be allocated bits according to how audible they are. Audibility of spectral components is determined by first calculating a, below which it is estimated that sounds will be
beyond the limits of human perception.

ALGORITHM:

Initialize the Matlab Read the input audio given for compression Add the multiplicative noise to the audio signal Transform the image using HAAR Transform Get the decomposition level from the user Compression ratio is calculated in percentage The image is compressed using the ratio Finalize by displaying the compressed audio
9

FLOW CHART:

START

Initialize by reading input Audio file

Add multiplicative noise to the Audio

Transform the Audio using Haar Transform

Get the decomposition level

10

Calculate the compression ratio

Display compression ratio and compressed Audio

STOP

PROGRAM:
clc; clear all; close all; [y, Fs, nbits, readinfo] = wavread('C:\Documents and Settings\WELCOME\Desktop\jjj.wav'); subplot(1,3,1); plot(y); n=input('enter the decomposition level'); [Lo_D,Hi_D,Lo_R,Hi_R]=wfilters('haar'); [c,s]=wavedec2(y,n,Lo_D,Hi_D); disp('the decomposition level is'); [THR,NKEEP]=wdcbm2(c,s,1.5,3*prod(s(1)));

11

[compressed_image,TREED,comp_ratio,PERFL2]=wpdencmp(THR,'s',n,'haar','threshold', 5,1); disp('compression ratio in percentage'); disp(comp_ratio); subplot(1,3,2); plot(compressed_image); re_im1=waverec2(c,s,'haar'); subplot(1,3,3); plot(re_im1);

OUTPUT:

12

Enter the decomposition level: 5 Compression ratio in percentage is: 20

RESULT:
Thus the given Audio signal has been compressed using the Wavelet Transform and its output was verified successfully.

13

EXP.NO: DATE:
AIM:

SIMULATION OF LPC BASED SPEECH /AUDIO COMPRESSION METHOD

To simulate and analyze an audio signal compression using linear predictive coding algorithm.

EQUIPMENTS REQUIRED:
1. MATLAB 7.0.1 2. Personal Computer

THEORY: LPC:
Linear predictive coding (LPC) is a tool used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model. It is one of the most powerful analysis methods for encoding good quality speech at a low bit rate and provides extremely accurate estimates of speech parameters. It expresses each sample of the signal as a linear combination of previous samples. Such an equation is called a linear predictor, which is called as Linear Predictive Coding. The coefficients of the difference equation characterize the formants, so the LPC system needs to estimate these coefficients. The estimate is done by minimizing the mean-square error between the predicted signal and the actual signal. It involves the computation of a matrix of coefficient values and the solution of a set of linear equations may be used to assure convergence to a unique solution with efficient computation.

ALGORITHM:
14

Initialize the Matlab. Read the audio signal as input for compression. Take transpose to the product of window signal and filtered input signal. Computation of a matrix of coefficient vaules used for convergence. Convergence of audio signal equations. may also be obtained by set of linear

Plot the durbin algorithm for linear prediction coefficients. Plot the predictor residual energy

15

FLOW CHART:

START

Initialize by reading input image

Transpose the product of window and the filtered input signal

Express each sample as a linear combination of previous

Estimation of coefficients by minimizing mean square error

16

Display original and compressed signal

STOP

PROGRAM:
clc; close all; clear all; [x,fs]=wavread('C:\Documents and Settings\WELCOME\desktop\123.wav');% read into the data % Preemphasis filter xx=double(x); y=filter([1 -0.9495],1,xx); N=160; y1=y(1:N); w1=hamming(1,N); y2=(y1.*w1)'; p=30;% predict the order of r=zeros(1,p+1); for k=1:p+1 sum=0; for m=1:N+1-k sum=sum+y2(m).*y2(m-1+k)'; end r(k)=sum;
17

end k=zeros(1,p); k(1)=r(2)/r(1); a=zeros(p,p); a(1,1)=k(1); e=zeros(1,p); e(1)=(1-k(1)^2)*r(1); for i=2:p c=zeros(1,i); sum=0; for j=1:i-1 sum=sum+(a(i-1,j).*r(i+1-j)); end c(i)=sum; k(i)=(r(i+1)-c(i))/e(i-1); if find(abs(k)>1) disp('default') else subplot(413);plot(abs(k));title('|k(i)|<=1') end a(i,i)=k(i); for j=1:i-1 a(i,j)=a(i-1,j)-k(i).*a(i-1,i-j); end e(i)=(1-k(i)^2)*e(i-1); subplot(414);plot(e); title('predictor residual energy E(i)'); end d=zeros(1,p); for t=1:p d(t)=a(p,t); end z=zeros(1,N); for i=1:p
18

z(i)=y2(i); end figure(1); subplot(411); plot(y2); title('Original Data'); subplot(412); plot(z); title('durbin algorithm for linear prediction coefficients');

OUTPUT:

Original Data 0.01 0 -0.01 1 0 -1 0.4 0.2 0 1.5 1 0.5 0 5 10 15 20 25 30 0 -3 x 10 5 10 15 20 predictor residual energy E(i) 25 30 0 20 40 60 80 |k(i)|<=1 100 120 140 160 0 20 40 60 80 100 120 durbin algorithm for linear prediction coefficients 140 160

19

EXP.NO: DATE:

SIMULATION OF SUBBAND BASED SPEECH/ AUDIO COMPRESSION METHOD

RESULT:
Thus the compression of given audio signal using linear predictive coding was simulated and its output has been plotted.

AIM:
To simulate audio and speech compression algorithm of subband coding using MATLAB.

APPARATUS REQUIRED:
1. MATLAB 7.0.1 2. Personal computer

THEORY:
A popular approach to decomposing the image into different frequency bands without the imposition of an arbitrary block structure is sub band coding. After the input has been decomposed into its constituents, we can use the coding technique best suited to each constituent to improve compression performance. Furthermore, each component of the source output may have different perceptual characteristics. Quantization error that is perceptually
20

objectionable in one component may be acceptable in a different component of the source output. Therefore, a coarser quantizer may be used for perceptually less important components. This is how the concept of sub-band coding comes into picture

ALGORITHM:

Initialize the Matlab. Read the audio file given for sub band coding. Plot the signals of speech and filter in time domain. Take Fourier transform for converting signals in time domain to frequency domain.

Plot the signals of speech and filter in frequency domain. Decimate the signals to get the four bands in synthesis. Compare the original band with synthesized band.

FLOW CHART:

START

Initialize by reading the audio files.

Convert the signals to frequency domain.

Decimate the signals to get the bands in synthesis

Compare the original band with synthesized band

21

Display the output waveforms

STOP

PROGRAM:
clc; close all; clear all; num=36000; [x,fs,nbits] = wavread(' sub1.wav',num); x=x(:,1)'; lnx=length(x); L = 2; len = 25; wc = 1/L; %cut-off frequency is pi/2. freq=-pi:2*pi/(lnx-1):pi;% the frequency vector lp = fir1(len-1, wc,'low'); hp = fir1(len-1, wc,'high'); yl=conv(x,lp); yh=conv(x,hp); %Time domain plots of signal and filters
22

figure(1); subplot(311); plot(x);axis([0 lnx min(x) max(x)]);ylabel('speech'); Title('Speech and filters in time domain'); subplot(312); stem(lp);axis([0 length(lp) (min(lp)+0.1) (max(lp)+0.1)]); ylabel('lp'); subplot(313); stem(hp);axis([0 length(hp) min(hp)+0.1 max(hp)+0.1]); ylabel('hp'); pause %plotting filter response of filters and the two speech bands(lower and upper) in freq domian figure(2); X=fftshift(fft(x,lnx)); Lp=fftshift(fft(lp,lnx)); Hp=fftshift(fft(hp,lnx)); YL=fftshift(fft(yl,lnx)); Yh=fftshift(fft(yh,lnx)); subplot(3,2,1); plot(freq/pi, abs(X)); ylabel('|X|'); axis([0 pi/pi min(abs(X)) max(abs(X))]); title('Freq domain representation of speech and the two bands'); subplot(3,2,3); plot(freq/pi, abs(Lp),'g'); ylabel('|Lp|'); axis([0 pi/pi min(abs(Lp)) max(abs(Lp))]); subplot(3,2,4); plot(freq/pi, abs(Hp), 'g'); ylabel('|Hp|'); axis([0 pi/pi min(abs(Hp)) max(abs(Hp))]); subplot(3,2,5); plot(freq/pi, abs(YL), 'y');
23

ylabel('|YL|'); axis([0 pi/pi min(abs(YL)) max(abs(YL))]); legend('Low bandafter filtering'); subplot(3,2,6); plot(freq/pi, abs(Yh), 'y'); ylabel('|Yh|'); axis([0 pi/pi min(abs(Yh)) max(abs(Yh))]); legend('High band after filtering'); pause ydl =yl(1:2:length(yl)); ydh=yh(1:2:length(yh)); s0=conv(ydl,lp); s1=conv(ydl,hp); s2=conv(ydh,lp); s3=conv(ydh,hp); % now finally decimating to get the four bands b0 =s0(1:2:length(s0)); b1=s1(1:2:length(s1)); b2 =s2(1:2:length(s2)); b3=s3(1:2:length(s3)); %freq plots of decimated signals(four bands) figure(3); title('Four bands in freq domain'); subplot(4,1,1); plot(freq/pi,abs(fftshift(fft(b0,lnx)))); ylabel('|B0|'); axis([0 pi/pi min(abs(fft(b0))) max(abs(fft(b0)))]); title('Four bands in freq domain'); subplot(4,1,2); plot(freq/pi,abs(fftshift(fft(b1,lnx)))); ylabel('|B1|'); axis([0 pi/pi min(abs(fft(b0))) max(abs(fft(b1)))]); subplot(4,1,3); plot(freq/pi,abs(fftshift(fft(b2,lnx))));
24

ylabel('|B2|'); axis([0 pi/pi min(abs(fft(b2))) max(abs(fft(b2)))]); subplot(4,1,4); plot(freq/pi,abs(fftshift(fft(b3,lnx)))); ylabel('|B3|'); axis([0 pi/pi min(abs(fft(b3))) max(abs(fft(b3)))]); pause; % now synthesizing L=2; N1=length(b0); Ss0=zeros(1,L*N1); Ss1=zeros(1,L*N1); Ss2=zeros(1,L*N1); Ss3=zeros(1,L*N1); Ss0(L:L:end)=b0; Ss1(L:L:end)=b1; Ss2(L:L:end)=b2; Ss3(L:L:end)=b3; %Passing through reconstruction filters % making a low pass filter with cutoff at 1/L and gain L reconst_fil=L*fir1(len-1,1/L); % finding the freq response of the filter sb0=conv(reconst_fil,Ss0); sb1=conv(reconst_fil,Ss1); sb2=conv(reconst_fil,Ss2); sb3=conv(reconst_fil,Ss3); Slow=sb0-sb1; Shigh=sb2-sb3; subl=zeros(1,length(Slow)*2); subh=zeros(1,length(Shigh)*2); subl(L:L:end)=Slow; subh(L:L:end)=Shigh; subll=conv(reconst_fil,subl); subhh=conv(reconst_fil,subh);
25

sub=subll-subhh; %Freq plots of final two bands and their merging into a single band figure(4); subplot(3,1,1); plot(freq/pi,abs(fftshift(fft(subll,lnx)))); ylabel('|low band|'); axis([0 pi/pi min(abs(fft(subll))) max(abs(fft(subll)))]); title('Final two bands in synthesis'); subplot(3,1,2); plot(freq/pi,abs(fftshift(fft(subhh,lnx)))); ylabel('|High band|'); axis([0 pi/pi min(abs(fft(subhh))) max(abs(fft(subhh)))]); subplot(3,1,3); plot(freq/pi,abs(fftshift(fft(sub,lnx)))); ylabel('|Band|'); axis([0 pi/pi min(abs(fft(sub))) max(abs(fft(sub)))]); pause %Comparison figure(5); subplot(2,1,1); plot(freq/pi, abs(X)); ylabel('|X|'); axis([0 pi/pi min(abs(X)) max(abs(X))]); title('Comparison'); legend('original band'); subplot(2,1,2); plot(freq/pi,abs(fftshift(fft(sub,lnx))),'r'); ylabel('|Band|'); axis([0 pi/pi min(abs(fft(sub))) max(abs(fft(sub)))]); legend('Synthesized Band');

26

OUTPUT:
FIGURE 1:

FIGURE2:

27

FIGURE 3:

FIGURE 4:

28

FIGURE 5:

29

EXP.NO: DATE:

SIMULATION OF EZW IMAGE COMPRESSION ALGORITHM

RESULT:
Thus audio and speech compression algorithm of subband coding using MATLAB was simulated.

AIM:
To compress an image using simulate EZW image coding algorithm.

APPARATUS REQUIRED:
1. MATLAB 7.0.1
2. Personal Computer

THEORY: EZW Algorithm:

Embedded Zerotrees of Wavelet Transforms is a lossy compression algorithm . At low bit rates i.e. high compression ratios most of the coefficients produced by a such as the will be

30

zero, or very close to zero. This occurs because "real world" images tend to contain mostly low frequency information By considering the transformed coefficients as a with the lowest frequency coefficients at the root node and with the children of each tree node being the spatially related coefficients in the next higher frequency subband, there is a high probability that one or more subtrees will consist entirely of coefficients which are zero or nearly zero, such subtrees are called zerotrees. Due to this, we use the terms node and coefficient interchangeably, and when we refer to the children of a coefficient, we mean the child coefficients of the node in the tree where that coefficient is located. We use children to refer to directly connected nodes lower in the tree and descendants to refer to all nodes which are below a particular node in the tree, even if not directly connected. In zerotree based image compression scheme such as EZW and, the intent is to use the statistical properties of the trees in order to efficiently code the locations of the significant coefficients. Since most of the coefficients will be zero or close to zero, the spatial locations of the significant coefficients make up a large portion of the total size of a typical compressed image. A coefficient is considered significant if its magnitude is above a particular threshold. By starting with a threshold which is close to the maximum coefficient magnitudes and iteratively decreasing the threshold, it is possible to create a compressed representation of an image which progressively adds finer detail. Due to the structure of the trees, it is very likely that if a coefficient in a particular frequency band is insignificant, then all its descendants will also be insignificant.

ALGORITHM:
Initialize the Matlab Read the input image given for compression Specify the maximum number of steps for the compression algorithm. Compress the image using EZW algorithm. Uncompress the compressed image. Finalize by plotting the compressed and uncompressed image
31

FLOW CHART:

START

Initialize by reading input image

32

Specify the number of input levels for compression

Compress the image using EZW algorithm

Uncompress the compressed image.

Display original image and compressed image

STOP

PROGRAM:
clc; clear all; close all; X = imread('wpeppers.jpg'); image(X)
33

axis square colormap(pink(255)) title('Original Image: peppers') meth = 'gbl_mmc_h'; % Method name option = 'c'; % 'c' stands for compression [CR,BPP] = wcompress(option,X,'peppers.wtc',meth,'BPP',0.5); option = 'u'; % 'u' stands for uncompression Xc = wcompress(option,'peppers.wtc'); colormap(pink(255)) figure(1) subplot(1,2,1); image(X); axis square; title('Original Image') subplot(1,2,2); image(Xc); axis square; title('Compressed Image') xlabel({['Compression Ratio: ' num2str(CR,'%1.2f %%')], ... ['BPP: ' num2str(BPP,'%3.2f')]}) meth = 'ezw'; % Method name wname = 'haar'; % Wavelet name nbloop = 6; % Number of loops 'wname','haar'); Xc = wcompress('u','peppers.wtc'); colormap(pink(255)) figure(2) subplot(1,2,1); image(X); axis square; title('Original Image') subplot(1,2,2); image(Xc); axis square; title('Compressed Image - 6 steps') xlabel({['Compression Ratio: ' num2str(CR,'%1.2f %%')], ... ['BPP: ' num2str(BPP,'%3.2f')]})
34

[CR,BPP] = wcompress('c',X,'peppers.wtc',meth,'maxloop', nbloop, ...

[CR,BPP] = wcompress('c',X,'peppers.wtc',meth,'maxloop',9,'wname','haar'); Xc = wcompress('u','peppers.wtc'); colormap(pink(255)) figure(3) subplot(1,2,1); image(Xc); axis square; title('Compressed Image - 9 steps') xlabel({['Compression Ratio: ' num2str(CR,'%1.2f %%')],... ['BPP: ' num2str(BPP,'%3.2f')]}) [CR,BPP] = wcompress('c',X,'peppers.wtc',meth,'maxloop',12,'wname','haar'); Xc = wcompress('u','peppers.wtc'); subplot(1,2,2); image(Xc); axis square; title('Compressed Image - 12 steps') xlabel({['Compression Ratio: ' num2str(CR,'%1.2f %%')], ... ['BPP: ' num2str(BPP,'%3.2f')]}) [CR,BPP] = wcompress('c',X,'peppers.wtc','ezw','maxloop',12, ... 'wname','bior4.4'); Xc = wcompress('u','peppers.wtc'); colormap(pink(255)) figure(4) subplot(1,2,1); image(Xc); axis square; title('Compressed Image - 12 steps - bior4.4') xlabel({['Compression Ratio: ' num2str(CR,'%1.2f %%')], ... ['BPP: ' num2str(BPP,'%3.2f')]}) [CR,BPP] = wcompress('c',X,'peppers.wtc','ezw','maxloop',11, ... 'wname','bior4.4'); Xc = wcompress('u','peppers.wtc'); subplot(1,2,2); image(Xc); axis square; title('Compressed Image - 11 steps - bior4.4')
35

xlabel({['Compression Ratio: ' num2str(CR,'%1.2f %%')], ... ['BPP: ' num2str(BPP,'%3.2f')]}) [CR,BPP] = wcompress('c',X,'peppers.wtc','spiht','maxloop',12, ... 'wname','bior4.4'); Xc = wcompress('u','peppers.wtc'); colormap(pink(255))

OUTPUT:

36

O inl I ae r a mg ig

Cmr s e I ae-6s p o pes dmg tes

10 0 20 0 30 0 40 0 50 0 10 0 20 0 30 0 40 0 50 0

10 0 20 0 30 0 40 0 50 0 10 0 20 0 30 0 Cmr s io Rtio 0 6% o pes n a : .0 B P0 2 P : .0 40 0 50 0

O in l I a e r a mg ig

Cmr s e I a e o pe s d mg

10 0 20 0 30 0 40 0 50 0 10 0 20 0 30 0 40 0 50 0

10 0 20 0 30 0 40 0 50 0 10 0 20 0 30 0 40 0 Cmr s io Rtio 1 7% o pe s n a : .5 B P0 8 P : .3 50 0

37

C m re s dIm g - 9s p o p s e ae te s

C m re s dIm g - 1 s p o p s e a e 2 te s

10 0 20 0 30 0 40 0 50 0 10 0 20 0 30 0 40 0 C m re io R tio 0 1% o p ss n a : .8 B P0 9 P : .1 50 0

10 0 20 0 30 0 40 0 50 0 10 0 20 0 30 0 40 0 C m re s nR tio 7 7% o p s io a : .4 BP1 9 P : .7 50 0

Cmr s e I a e- 1 s p - b r .4 o pe s d mg 2 te s io4

Cmr s e I a e- 1 s p - b r .4 o pe s d mg 1 te s io4

10 0 20 0 30 0 40 0 50 0 10 0 20 0 30 0 40 0 Cmr s io Rtio 4 2% o pe s n a : .9 B P1 8 P : .1 50 0

10 0 20 0 30 0 40 0 50 0 10 0 20 0 30 0 40 0 Cmr s io Rtio 2 1% o pe s n a : .5 B P0 0 P : .6 50 0

38

EXP.NO: DATE:

SIMULATION OF SPIHT IMAGE COMPRESSION ALGORITHM

RESULT:
Thus the given image has been compressed using the EZW algorithm.

AIM:
To compress an image using simulate SPIHT image coding algorithm.

APPARATUS REQUIRED:
1. MATLAB 7.0.1 2. Personal Computer

THEORY: SPIHT ALGORITHM:

One of the most efficient algorithms in the area of image compression is the Set Partitioning in Hierarchical Trees (SPIHT). It uses a sub-band coder, to produce a pyramid structure where an image is decomposed sequentially by applying power complementary low pass and high pass filters and then decimating the resulting images. These are one-dimensional filters that are applied in cascade (row then column) to an image whereby creating four-way decomposition: LL (low-pass then another low pass), LH (low pass then high pass), HL (high and low pass) and finally HH (high pass then another high pass). The resulting LL version is again four-way decomposed. This process is repeated until the top of the pyramid is reached. This pyramid structure is commonly known as spatial orientation tree.

ALGORITHM:
Initialize the Matlab Read the input image given for compression
39

Specify the maximum number of steps for the compression algorithm. Compress the image using SPIHT algorithm. Uncompress the compressed image. Finalize by plotting the compressed and uncompressed image.

FLOW CHART:

START

Initialize by reading input image

Specify the number of input levels for compression

Compress the image using SPIHT algorithm

Uncompress the compressed image.

Display original image and compressed image

STOP

40

PROGRAM:
clc; clear all; close all; x=imread(wpeppers.jpg); [cr,bpp]=wcompress(c,x,wpeppers.wtc,spiht,maxloop,12); xc=wcompress (u,wpeppers.wtc); Colormap (pink(255)); Subplot(1,2,1); Image(x); Axis square; Title(original image); Subplot(1,2,2); Image(xc); Axis square; Title(compressed image-12-steps-bior 4.4); xlabel({[compression ratio:num2str(cr,%1.2f%%)] [bpp:num2str(bpp,%3.2f)]}); delete(wpeppers.wtc);

41

OUTPUT:

Original Image

Compressed Image - 12 steps - bior4.4

100 200 300 400 500 100 200 300 400 500

100 200 300 400 500 100 200 300 400 500 Compression Ratio: 1.65 % BPP: 0.40

42

EXP.NO: S- PARAMETER ESTIMATION OF COUPLER DATE:

RESULT:
Thus the given image has been compressed using the SPIHT algorithm and its output was verified successfully.

AIM:
To design and analyze the characteristics of a coupler using ADS.

APPARATUS REQUIRED:
1. ADS 2. Personal Computer

THEORY:
A very commonly used basic element in microwave system is the directional coupler. Its basic function is to sample the forward and reverse travelling waves through a transmission line or a waveguide. The common use of this element is to measure the power level of a transmitted or received signal. The model of a directional coupler is shown in Figure 1.

43

As seen in the figure, the coupler is a four-ports device. The forward travelling wave goes into port 1 and exit from port 2. A small fraction of it goes out through port 4. In a perfect coupler, no signal appears in port 4. Since the coupler is a lossless passive element, the sum of the signals power at ports 1 and 2 equals to the input signal power. The reverse travelling wave goes into port 2 and out of port 1. A small fraction of it goes out through port 3. In a perfect coupler, no signal appears in port 4. The directional coupler S-parameters matrix is:

Where k is the coupling factor (a linear value). One popular realization technique of the directional coupler is the coupled lines directional coupler; two quarter wavelength line are placed close to each other. The wave travelling through one line is coupled to the other line. Such a coupler is shown in Figure 2.

44

Since there is no ideal coupler available, some of the forward travelling wave is coupled into port 3. This mean that we may think that there is a reverse travelling wave when there isnt. This is very critical in application where the directional coupler is used to measure the return loss of the device. By calculating 20 log(S31/S41) we can find the return loss of the device connected to port 2. If out coupler has no perfect directivity then out measurement is not accurate. There are few simple parameters to describe the functionality of a coupler: Insertion Loss: 20 log(S21) or 10 log(1 k2). Return Loss: 20 log(S11). Coupling: 20 log(S31) or 20 log(k). Directivity: 20 log(S31) 20 log(S41).

CIRCUIT DIAGRAM:

45

OUTPUT:

46

RESULT:
Thus the coupler was designed and analyzed using ADS and its output was verified successfully.

47

EXP.NO: DESIGN OF OSCILLATOR DATE:

AIM:
To Design the oscillator using ADS.

APPARATUS REQUIRED:
1. ADS SOFTWARE
2. Personal Computer.

THEORY: MICROWAVE OSCILLATOR:

An electronic oscillator is an electronic circuit that produces a repetitive electronic signal, often a sine wave or a square wave. They are widely used in many electronic devices. Common examples of signals generated by oscillators include signals broadcast by radio and television transmitters, clock signals that regulate computers and quartz clocks, and the sounds produced by electronic beepers and video games. Oscillators are often characterized by the frequency of their output signal: an audio oscillator produces frequencies in the audio range, about 16 Hz to 20 kHz. An RF oscillator produces signals in the radio frequency (RF) range of about 100 kHz to 100 GHz. A lowfrequency oscillator (LFO) is an electronic oscillator that generates a frequency below 20 Hz. This term is typically used in the field of audio synthesizers, to distinguish it from an audio frequency oscillator. Oscillators designed to produce a high-power AC output from a DC supply are usually called inverters. There are two main types of electronic oscillator: the harmonic oscillator and the relaxation oscillator.

48

PROCEDURE:
1. Initialize the Advanced Design System software 2. Create a new project and save it. 3. Select the simulation format. 4. Select an application type as oscillator. 5. Select the required sample design. 6. Specify the simulation template for oscillator. 7. Simulate the project.

CIRCUIT DIAGRAM:

49

OUTPUT:

RESULT:
50

EXP.NO: DESIGN OF FILTER DATE:

Thus the Oscillator has been designed using ADS software and its output was verified successfully .

AIM:
To stimulate the performance of a Filter using ADS

APPARATUS REQUIRED:
1. Personal Computer 2. ADS software

THEORY:
A filter is a two port network used to control the frequency response at a certain point in a system by providing transmission within the passband of the filter and attenuation in the stop band of the filter. The basic filter types are low-pass, high-pass, bandpass and band-reject (notch) filters.

CIRCUIT DIAGRAM:

51

Digital Filter Impulse Response

DF DF DefaultNumericStart= 0 DefaultNumericStop= 100

123

Numeric

ImpulseFloat I1 Level= 1.0 Period= 0 Delay= 0

SW_DFILT_FIR X1

NumericSink N1 Plot= Rectangular Start= DefaultNumericStart Stop= DefaultNumericStop ControlSimulation= YES

PROCEDURE:

Step1: Open the ADS software.

Step2: Create a new project from the file menu.

Step3: Open the schematic window of ADS.

Step4: From the components library select the appropriate necessary fo the required model.

Step5: Click on the necessary components and place them on the schematic windows of ADS.

Step6: Design the filters using discrete elements

Step7: Determine the SWR of each filter using hand analysis

52

Step8: Compare experimental results to theory and simulation

Step9: Comment on your result

OUTPUT:

0 . 4 0 . 3

N1

0 . 2 0 . 1 0 . 0 -. 0 1 0 5 0 10 0

I dx ne

53

EXP.NO: DESIGN OF MIXER USING ADS DATE:

RESULT:
Thus the Filter was simulated using ADS and its output was verified successfully.

AIM:
To stimulate the performance of a Mixer using ADS

APPARATUS REQUIRED:
1. 2. Personal Computer ADS software

THEORY:
Mixers are three port active or passive devices, are designed to yield both a sum and a difference frequency at a single output port when two distinct input frequencies are inserted into the other two ports. This process, called frequency conversion (or heterodyning), is found in most communications gear, and is used so that we may increase or decrease a signals frequency. One of the two input frequencies will normally be a CW wave, produced within the radio by a local oscillator (LO), while the other input will be the RF signal received from the antenna. If we would like to produce an output frequency within the mixer circuit that is lower than the input RF signal, then this is called down conversion; if we would like to produce an
54

output signal that is at a higher frequency than the input signal, it is referred to as up conversion. Indeed, most AM, SSB, and digital transmitters require mixers to convert up to a higher frequency for transmission into space, while superheterodyne receivers require a mixer to convert a received signal to a much lower frequency. This lower received frequency available at the mixers output port is called the intermediate frequency (IF). Receivers use this lower-frequency IF signal because it is much easier to efficiently amplify and filter with all the IF stages tuned and optimized for a single, low band of frequencies, which increases the receivers gain and selectivity. Again, the frequency conversion process within the nonlinear mixer stage produces the intermediate frequency by the RF input signal heterodyning, or beating, with the receivers own internal LO. This heterodyning mixer circuit will consist of either a diode, BJT, or FET that is overdriven, or biased to run within the nonlinear area of its operation. However, the beating of the mixers RF and LO input signals yields not only the RF, the LO, and the sum and difference frequencies of these two primary signals, but also many spurious frequencies at the mixers output port. Most of these undesired frequencies will be filtered out within the receivers IF stages, resulting in the new desired signal frequency, consisting of the converted carrier and any sidebands, now at the difference frequency. This new, lower difference frequency will then be amplified and further filtered as it passes through the fixed-tuned IF strip. There are three basic classifications for both active and passive mixers: Unbalanced mixers have an IF output consisting of fS,fLO, fS fLO, fS + fLO, and other spurious outputs. They will also exhibit little isolation between each of the mixers three ports, resulting in undesired signal interactions and feedthroughs to another port. Singlebalanced mixers will at least strongly attenuate either the original input signal or the LO (but not both), while sending less of the above mixing products on to its output than the unbalanced type. A double-balanced mixer, or DBM for short, supplies superior IF-RF-LO inter-port isolation, while outputting only the sum and difference frequencies of the input signal and the local oscillator, while attenuating both the LO and RF signals, and significantly attenuating three quarters of the possible mixer spurs at the output of the IF port. This makes the job of filtering and selecting a frequency plan a much easier task.

55

CIRCUIT DIAGRAM:

56

PROCEDURE:

Step1: Open the ADS software.

Step2: Create a new project from the file menu.

Step3: Open the schematic window of ADS.

Step4: From the components library select the appropriate necessary of the required model.

Step5: Click on the necessary components and place them on the schematic windows of ADS.

Step6: Design the Mixer using discrete elements

Step7: Determine the SWR of each filter using hand analysis

Step8: Compare experimental results to theory and simulation

57

Step9: Comment on your result

OUTPUT:

58

RESULT:
Thus the Mixer was simulated using ADS and its output was verified successfully.

59

EXP.NO: DESIGN OF AMPLIFIER DATE:

AIM:
To design and analyze the characteristics of Amplifier using ADS.

APPARATUS REQUIRED:
1. ADS software 2. Personal Computer

THEORY:
An amplifier is an electronic device that increases the voltage, current, or power of a signal. Amplifiers are used in wireless communications and broadcasting, and in audio equipment of all kinds. They can be categorized as either weak-signal amplifiers or power amplifiers. Weak-signal amplifiers are used primarily in wireless receivers. They are also employed in acoustic pickups, audio tape players, and compact disc players. A weak-signal amplifier is designed to deal with exceedingly small input signals, in some cases measuring only a few nano volts (units of 10-9 volt). Such amplifiers must generate minimal internal noise while increasing the signal voltage by a large factor. The most effective device for this application is the field-effect transistor. The specification that denotes the effectiveness of a weak-signal amplifier is sensitivity, defined as the number of micro volts (units of 10-6 volt) of signal input that produce a certain ratio of signal output to noise output (usually 10 to 1). Power amplifiers are used in wireless transmitters, broadcast transmitters, and hi-fi audio equipment. The most frequently-used device for power amplification is the bipolar transistor. However, vacuum tubes, once considered obsolete, are becoming increasingly popular, especially among musicians. Many professional musicians believe that the vacuum tube (known as a "valve" in England) provides superior fidelity.

60

CIRCUIT DIAGRAM:

PROCEDURE:
1. Initialize the Advanced Design System software 2. Create a new project and save it. 3. Select simulation format. 4. Select an application type as amplifier. 5. Select the required sample design. 6. Specify the simulation template for Amplifier. 7. Simulate the project.

61

OUTPUT:

62

EXP.NO: SIMULATION OF GPS DATE:

RESULT:
Thus the Amplifier was simulated using ADS and its output was verified successfully.

AIM:
To Simulate the GPS system using MATLAB

APPARATUS REQUIRED:
1. MATLAB 7.0.1 2. Personal Computer

THEORY:
Trilateration is a method of determining the relative position of objects using the geometry of triangles in a similar fashion as triangulation. Unlike triangulation, which uses angle measurements to calculate the subjects location, triangulation uses the known locations of two or more reference points, and the measured distance between the subject and each reference point. To accurately and uniquely determine the relative location of a point on a 2D plane using trilateration alone, generally at least 3 reference points are needed. Standing at B, you want to know your location relative to the reference points P1, P2 and P3 on a 2D plane. Measuring r1 narrows your position down to a circle. Next, measuring r2 narrows it down to two points, A and B. A third measurement, r3, gives your coordinates at B. A fourth measurement could also be made to reduce error.

63

A mathematical derivation for the solution of a three-dimensional trilateration problem can be found by taking the formulae for three spheres and setting them equa;l to each other. To do this, we must apply three constraints to the centers of these spheres; all three must be on the z=0 plane, one must be on the origin, and one other must be on the x-axis. Starting with three spheres,

and

We subtract the second from the first and solve for x:

Substituting this back into the formula for the first sphare produces the formula for a circle, the solution to the intersection of the first two spheres:

Setting this formula equal to the formula for the third sphere finds:

Now that we have the x-and y- coordinates of the solution point, we can simply rearrange the formula for the first sphere to find the z- coordinate:

Now we have the solution to all three points x, y and z. because z is expressed as a square root, it is possible for there to be zero, one or two solutions to the problem.
64

ALGORITHM:
Initialize the Matlab Get the co-ordinate values for x,y and z Determine the absolute location of the points Locate the area of intersections of three spheres Display the latitude and longitude values

PROGRAM:
clc; clear all; close all; x=input('the x-coordinate value='); if(x>100) disp('i/p exceeds the axis value'); return end y=input('the y-coordinate value='); if(y>100) disp('i/p exceeds the axis value'); return end z=input('the z-coordinate value='); if(z>100) disp('i/p exceeds the axis value'); return end a=6378137; b=6356752.31425; f=(a-b)/b;
65

display(f); doubletemp=0; doubletemp1=0; temp=((a*a)-(b*b))/(b*b); e1=sqrt(temp); disp('the value of e1 is'); display(e1); temp=2*f-(f*f); e=sqrt(temp); disp('the value of e is'); disp(e); temp=(x*x)+(y*y); p=sqrt(temp); disp('the value of p is'); display(p); theta=atan(z*a/(p*b)); display(theta); temp=z+(e1*e1*b*sin(theta)*sin(theta)*sin(theta)); disp('the value of temp is'); display(temp); temp1=p-(e*e*a*cos(theta)*cos(theta)*cos(theta)); disp('the value of temp1 is'); display(temp1); fi=atan(temp/temp1); disp('the value of fi is'); display(fi); lam=atan2(y,x); disp('the value of lam is'); display(lam); temp=1-(e*e*sin(fi)*sin(fi)); temp1=sqrt(temp); n=a/temp1; h=(p/cos(fi))-n; disp('The value of Altitiude (h) is');
66

display(h); OUTPUT: the x-coordinate value=5 the y-coordinate value=4 the z-coordinate value=3 f =0.0034 the value of e1 is e1 =0.0821 the value of e is 0.0820 the value of p is p =6.4031 theta =0.4394 the value of temp is temp =3.3018e+003 the value of temp1 is temp1 = -3.1747e+004 the value of fi is fi = -0.1036 the value of lam is lam = 0.6747 The value of Altitiude (h) is h =-6.3784e+006
67

EXP.NO: DATE:

PERFORMANCE EVALUATION OF SIMULATION OF CDMA SYSTEM

RESULT:
Thus the GPS was simulated using MATLAB and its output was verified successfully.

AIM:
To design and analyze the performance evaluation of simulation of CDMA system.

APPARATUS REQUIRED:
1. MATLAB Version 7.0.1 2. Personal computer

THEORY:
DMA is a spread spectrum multiple access technique. A spread spectrum technique spreads the bandwidth of the data uniformly for the same transmitted power. A spreading code is a pseudo-random code that has a narrow Ambiguity function, unlike other narrow pulse codes. In CDMA a locally generated code runs at a much higher rate than the data to be transmitted. Data for transmission is combined via bitwise XOR (exclusive OR) with the faster code. Code division multiple access (CDMA) is a channel access method used by various radio communication technologies. It should not be confused with the mobile phone standards called cdmaOne, CDMA2000 (the 3G evolution of cdma One) and WCDMA (the 3G standard used by GSM carriers), which are often referred to as simply CDMA, and use CDMA as an underlying channel access method.One of the concepts in data communication is the idea of allowing several transmitters to send information simultaneously over a single communication channel. This allows several users to share a band of frequencies (see bandwidth). This concept is called multiple access.
68

CDMA employs spread-spectrum technology and a special coding scheme (where each transmitter is assigned a code) to allow multiple users to be multiplexed over the same physical channel. By contrast, time division multiple access (TDMA) divides access by time, while frequency-division multiple access (FDMA) divides it by frequency. CDMA is a form of spread-spectrum signalling, since the modulated coded signal has a much higher data bandwidth than the data being communicated.An analogy to the problem of multiple access is a room (channel) in which people wish to talk to each other simultaneously. To avoid confusion, people could take turns speaking (time division), speak at different pitches (frequency division), or speak in different languages (code division). CDMA is analogous to the last example where people speaking the same language can understand each other, but other languages are perceived as noise and rejected. Similarly, in radio CDMA, each group of users is given a shared code. Many codes occupy the same channel, but only users associated with a particular code can communication.

BLOCK DIAGRAM OF CDMA SYSTEM:

MULTI USER:

69

TRANSMITTER SIDE SYSTEM:

RECEIVER SIDE SYSTEM:

70

SINGLE USER:

71

PROCEDURE:
1. Convert input bits to bipolar bits.. 1 to 1 and 0 to -1 for user1 and user2

72

2. Take 100 samples per bit for both user1 and user2 and then plot base band signal which is in bipolar NRZ format. 3. Then BPSK modulate the signal. Take care that sampling rate of sinusoidal carrier matches the sampling rate per bit. Here it is 100 samples per carrier and then plot the BPSK signal 4. Multiply the BPSK modulated signal with the PN code. Here again the care should be taken to match the sampling rate. i.e. no. of chip per bit* no of samples per chip = no of samples per bit of BPSK modulated signal. 5. Same procedure is carried out for user2 bits. 6. The channel is AWGN channel with SNR 5 dbs. In channel the signal from user1 is added to signal from user2 and white Gaussian noise is added. 7. At receiver end , first received signal is multiples with PN then BPSK demodulated by multiplying with the carrier(coherent demod) 8. Then the samples over 1 bit interval are summed. And if the sum is greater than 0 than the received bit is 1 else rx bit is 0. Summation is used in place of integration because it is a discrete time system 9. Same procedure is repeated for user2.

73

OUTPUT:

74

RESULT:
Thus the performance evaluation of simulation of CDMA was simulated using MATLAB and its output was verified successfully.
75

EXP.NO: DESIGN AND TESTING OF A MICROSTRIP COUPLER DATE:

AIM:
To design and analyze the characteristics of a microstrip coupler using MATLAB.

APPARATUS REQUIRED:
3. MATLAB 7.0.1 4. Personal Computer

THEORY:
Mini-Circuits microstrip couplers are reactive devices featuring very low insertion loss. Most models have 3 ports, and are manufactured with an internal 50-ohm termination. In the case, power coupled from any power incident to the output port (the reflected power) is absorbed and not available to the user. However, all 4-port (bi-microstrip) models have both the incident and reflected coupled power available. Examples are the ZFDC-20-1H and the BDsuffix models. The basic function of a microstrip coupler is to operate on an input so that two input signals are available. However, when the input is applied to the opposite port of an internally terminated coupler, only one output signal is produced.

MICROSTRIP COUPLER CHARACTRISTICS

1. The output signals are unequal in amplitude. The larger signal is at the main-line output port. The smaller signal is at the coupled port. 2. The main-line insertion loss depends upon the signal level at the coupled port, as determined by design. 3. There is high isolation between the coupled port and the output of the main-line.

76

Key characteristics of a microstrip coupler include coupling coefficient, coupling flatness, main-line loss and directivity, defined in the next page. Mini-Circuits full line of microstrip couplers, spanning 5 KHz to 2GHz, provide excellent performance. They feature:

1. flat coupling over a broad bandwidth 2. low main-line loss, as low as 0.1 dB 3. directivity as high as 55 dB and
4. A wide range of coupling values, from 6dB to 30dB.

MICROSTRIP COUPLER APPLICATIONS

The high performance characteristics of these units enable the following signal processing functions to be accomplished: 1. Measure incident and reflected power to determine VSWR 2. Signal sampling 3. Signal injection

PROGRAM:
clc; clear all; close all; for(f=10^8:10^8:10^11) C=47*10^-12; L=1542*10^-9*((sqrt(f)^-1)); Rs=4.8*sqrt(f)*10^-6; Re=33.9*10^12*(f^-1); impedance1=abs((i*2*pi*f*C)^-1); impedance=abs(i*2*pi*f*L+Rs+Re*((1+i*2*pi*f*C*Re)^-1));
77

axis on; grid on; hold on; axis auto; xmin=10^5; xmax=10^11; ymin=0; ymax=3.5; axis([xmin,xmax,ymin,ymax]); xlabel('frequency'); ylabel('impedance'); plot(f,impedance,'red'); plot(f,impedance1,'blue'); end

78

OUTPUT:
3.5

2.5

impedance

1.5

0.5

5 frequency

9 x 10

10
10

RESULT:
Thus the microstrip coupler was simulated using MATLAB and its output was verified successfully.

79

EXP.NO: DATE:

SIMULATION OF MICROSTRIP ANTENNA USING MATLAB

AIM:
To simulate the micro strip antenna using MATLAB.

EQUIPMENTS REQUIRED:
1. Personal computer.

2. MATLAB 7.0 version.

THEORY:
In telecommunication, there are several types of micro strip antennas (also known as printed antennas) the most common of which is the micro strip patch antenna or patch antenna. A patch antenna is a narrowband, wide-beam antenna fabricated by etching the antenna element pattern in metal trace bonded to an insulating dielectric substrate, such as a printed circuit board, with a continuous metal layer bonded to the opposite side of the substrate which forms a ground plane. Common micro strip antenna shapes are square, rectangular, circular and elliptical, but any continuous shape is possible. Some patch antennas do not use a dielectric substrate and instead made of a metal patch mounted above a ground plane using dielectric spacers; the resulting structure is less rugged but has a wider bandwidth. Micro strip antennas are relatively inexpensive to manufacture and design because of the simple 2-dimensional physical geometry. They are usually employed at UHF and higher frequencies because the size of the antenna is directly tied to the wavelength at the resonant frequency. A single patch antenna provides a maximum directive gain of around 6-9 dB. It is relatively easy to print an array of patches on a single (large) substrate using lithographic techniques. The directivity of patch antennas is approximately 5-7 dB

80

ALGORITHM:

Start the program. Calculate the width, effective dielectric constant, length, effective length of microstrip as follows; w= ((sqrt (2/er+1))*c)/(2*fr) preff =((er+1)/2)+(((er-1)/2)*(1+12*1/wbyh)) len=(c/(2*fr*sqrt(preff)))-(2*incleng) eff=len+(2*incleng)

Where er=dielectric constant value fr=resonant frequency

Stop the program.

81

FLOWCHART

START

ENTER THE DIELECTRIC CONSTANT,RESONANT FREQUENCY,HEIGHT OF MICROSTRIP ANTENNA

FIND THE WIDTH OF MICROSTRIP ANTENNA

CALCULATE EFFECTIVE DIELECTRIC CONSTANT,LENGTH,EFFECTIVE LENGTH OF MICROSTRIP ANTENNA

STOP

CODING:
82

clc; close all; clear all; er=input('the dielectric constant value'); fr=input('the resonant frequency value in Ghz'); h=input('the height of microstrip antenna in cm'); c=30; w=((sqrt(2/er+1))*c)/(2*fr); disp('width of microstrip in cm'); display(w); wbyh=w/h; preff=((er+1)/2)+((er-1)/2)*(1+12*1/wbyh); disp('effective dielectric constant of microstrip'); disp(preff); a=((preff+0.3)/(preff-0.258)); b=((wbyh+0.264)/(wbyh+0.813)); incleng=0.4128*h*a*b; disp('increase in length of microstrip in cm'); disp(incleng); len=((c/(2*fr+sqrt(preff)))-(2*incleng)); disp('length of microstrip in cm'); disp(len); eff=len+(2*incleng); disp('effective length of microstrip in cm'); disp(eff);

OUTPUT:
83

the dielectric constant value 1 the resonant frequency value in Ghz 50 the height of micro strip antenna in cm 5 width of micro strip in cm w = 0.5196 effective dielectric constant of micro strip 1 increase in length of micro strip in cm 1.4510 length of micro strip in cm -2.6050 effective length of micro strip in cm 0.2970

RESULT:
Thus the micro strip antenna using MATLAB has been simulated.

84