Adsp Lab Manual-1

Department of Electronics & Communication Engineering
ADVANCED DIGITAL SIGNAL PROCESSING LAB
NARAYANA
ENGINEERING COLLEGE
GUDUR

MANUAL
DEPARTMENT OF
ELECTRONICS AND COMMUNICATION
ENGINEERING
1
JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY ANANTAPUR

M.Tech I year I Semester (DSCE) L T
P C
0 0 3
2
(17D38108) ADVANCED DIGITAL SIGNAL PROCESSING LAB
Note:
A. Minimum of 10 Experiments have to be conducted
B. All Simulations are be carried out using MATLAB/DSP Processors/Labview
Software & DSP Kits
1. Study of various addressing modes of DSP using simple programming examples

2. Generation of waveforms using recursive/filter methods
3. Sampling of input signal and display
4. Implementation of Linear and Circular Convolution for sinusoidal signals
5. Framing & windowing of speech signal.
6. Finding voice & unvoiced detection for each frame of the speech signal.
7. IIR Filter Implementation using probe points
8. Implementation of FIR filters on DSP processor
9. Loop back using DSK kit
10. Real-time signal enhancement using Adaptive Filter.
11. Representation of different Q-formats using GEL function
12. Verification of Finite word length effects (Overflow, Coefficient Quantization,
Scaling and Saturation mode in DSP processors)
13. Image enhancement using spatial & frequency domain
14. Implementation of Image segmentation techniques
15. Extraction of frames from Video signal
2
NARAYANA ENGINEERING COLLEGE::GUDUR

LIST OF EXPERIMENTS TO BE CONDUCTED
1. Study of various addressing modes of DSP using simple programming examples
2. Implementation of Linear and Circular Convolution for sinusoidal signals
3. Framing & windowing of speech signal
4. Finding voice & unvoiced detection for each frame of the speech signal
5. IIR Filter Implementation using probe points
6. Implementation of FIR filters on DSP processor
7. Representation of different Q-formats using GEL function
8. Image enhancement using spatial & frequency domain
9. Implementation of Image segmentation techniques
10. Extraction of frames from Video signal
3
EXP NO:1 STUDY OF VARIOUS ADDRESSING MODES OF DSP DATE:

USING SIMPLE PROGRAMMING EXAMPLES
Aim:
To Study the various addressing mode of TMS320C6745 DSP processor.
Addressing Modes:
The addressing modes on the DSP are linear, circular using BK0, and circular
using BK1. The addressing mode is specified by the addressing mode register (AMR).
All registers can perform linear addressing. Only eight registers can perform circular
addressing: A4-A7 are used by the .D1 unit, and B4-B7 are used by the .D2 unit. No
other units can perform circular addressing. LDB(U)/LDH(U)/LDW, STB/STH/STW,
LDNDW, LDNW, STNDW, STNW, LDDW, STDW,
ADDAB/ADDAH/ADDAW/ADDAD, and SUBAB/SUBAH/SUBAW instructions all
use AMR to determine what type of address calculations are performed for these
registers. There is no SUBAD instruction.
Addressing Mode Register (AMR):

For each of the eight registers (A4-A7, B4-B7) that can perform linear or
circular addressing, the addressing mode register (AMR) specifies the addressing
mode. A 2-bit field for each register selects the address modification mode: linear (the
default) or circular mode. With circular addressing, the field also specifies which BK
(block size) field to use for a circular buffer. The mode select & block size fields are
choose by writing values to AMR. The register details about AMR is available
at TMS320C674x DSP CPU and Instruction Set Reference Guide.
Linear Addressing Mode
LD and ST Instructions
For load and store instructions, linear mode simply shifts the offsetR/cst operand to
the left by 3, 2, 1, or 0 for doubleword, word, halfword, or byte access, respectively;
and then performs an add or a subtract to baseR (depending on the operation
specified). The LDNDW and STNDW instructions also support nonscaled offsets. In
nonscaled mode, the offsetR/cst is not shifted before adding or subtracting from the
baseR.
For the preincrement, predecrement, positive offset, and negative offset address
generation options, the result of the calculation is the address to be accessed in
memory. For postincrement or postdecrement addressing, the value of baseR before
the addition or subtraction is the address to be accessed from memory.
ADDA and SUBA Instructions
4
For integer addition and subtraction instructions, linear mode simply shifts the src1/cst
operand to the left by 3, 2, 1, or 0 for doubleword, word, halfword, or byte data sizes,
respectively, and then performs the add or subtract specified.
Circular Addressing Mode
The BK0 and BK1 fields in AMR specify the block sizes for circular addressing
LD and ST Instructions
As with linear address arithmetic, offsetR/cst is shifted left by 3, 2, 1, or 0 according

to the data size, and is then added to or subtracted from baseR to produce the final
address. Circular addressing modifies this slightly by only allowing bits N through 0
of the result to be updated, leaving bits 31 through N + 1 unchanged after address
arithmetic.
The resulting address is bounded to 2(N + 1) range, regardless of the size of the
offsetR/cst.
The circular buffer size in AMR is not scaled; for example, a block-size of 8 is 8
bytes, not 8 times the data size (byte, halfword, word). So, to perform circular
addressing on an array of 8 words, a size of 32 should be specified, or N =
4. Example shows an LDW performed with register A4 in circular mode and BK0 = 4,
so the buffer size is 32 bytes, 16 halfwords, or 8 words. The value in AMR for this
example is 0004 0001h
ADDA and SUBA Instructions
As with linear address arithmetic, offsetR/cst is shifted left by 3, 2, 1, or 0 according

to the data size, and is then added to or subtracted from baseR to produce the final
address. Circular addressing modifies this slightly by only allowing bits N through 0
of the result to be updated, leaving bits 31 through N + 1 unchanged after address
arithmetic. The resulting address is bounded to 2(N + 1) range, regardless of the size
of the offsetR/cst.
The circular buffer size in AMR is not scaled; for example, a block size of 8 is 8 bytes,
not 8 times the data size (byte, halfword, word). So, to perform circular addressing on
an array of 8 words, a size of 32 should be specified, or N = 4.
Example shows an ADDAH performed with register A4 in circular mode and BK0 =

4, so the buffer size is 32 bytes, 16 halfwords, or 8 words. The value in AMR for this
example is 0004 0001h.
5
Result
Thus, the various addressing mode of DSP processor TMS320C6745 was studied.
IMPLEMENTATION Of LINEAR AND

EXP NO:2 CIRCULAR CONVOLUTION USING MATLAB DATE:
AIM: To verify Linear Convolution using MATLAB.
6
EQUIPMENT REQUIRED:
Hardware required Software Required

PC MATLAB
DSK6713 DSP Starter kit
USB Cable OS: Windows XP
Power Adapter
THEORY:
Convolution is a formal mathematical operation, just as multiplication, addition, and
integration. Addition takes two numbers and produces a third number, while convolution
takes two signals and produces a third signal. Convolution is used in the mathematics of
many fields, such as probability and statistics. In linear systems, convolution is used to
describe the relationship between three signals of interest: the input signal, the impulse
response, and the output signal. In this equation, x1(k), x2(n-k) and y(n) represent the
input to and output from the system at time n.
Here we could see that one of the inputs is shifted in time by a value every time it is
multiplied with the other input signal. Linear Convolution is quite often used as a method
of implementing filters of various types.
Linear Convolution Using MATLAB:-
Program:
clc;
clear all;
close all;
disp('linear convolution program');
x=input('enter i/p x(n):');
m=length(x);
h=input('enter i/p h(n):');
n=length(h);
x=[x,zeros(1,n)];
subplot(2,2,1), stem(x);
title('i/p sequence x(n)is:');
xlabel('---->n');
ylabel('---->x(n)');grid;
h=[h,zeros(1,m)];
subplot(2,2,2), stem(h);
title('i/p sequence h(n)is:');
xlabel('---->n');
ylabel('---->h(n)');grid;
disp('convolution of x(n) & h(n) is y(n):');
y=zeros(1,m+n-1);
for i=1:m+n-1
y(i)=0;
for j=1:m+n-1
if(j<i+1)
y(i)=y(i)+x(j)*h(i-j+1);
7
end
end
end
y
subplot(2,2,[3,4]),stem(y);
title('convolution of x(n) & h(n) is :');
xlabel('---->n');
ylabel('---->y(n)');grid;
CIRCULAR CONVOLUTION:
AIM: To verify Circular Convolution using MATLAB.
EQUIPMENT REQUIRED:
PC MATLAB
Power Adapter
THEORY
Circular convolution is another way of finding the convolution sum of two input signals.
It resembles the linear convolution, except that the sample values of one of the input
signals is folded and right shifted before the convolution sum is found. Also note that
circular convolution could also be found by taking the DFT of the two input signals and
finding the product of the two frequency domain signals. The Inverse DFT of the product
would give the output of the signal in the time domain which is the circular convolution
output. The two input signals could have been of varying sample lengths. But we take the
DFT of higher point, which ever signals levels to. For example, If one of the signal is of
length 256 and the other spans 51 samples, then we could only take 256 point DFT. So
the output of IDFT would be containing 256 samples instead of 306 samples, which
follows N1+N2 – 1 where N1 & N2 are the lengths 256 and 51 respectively of the two
inputs. Thus the output which should have been 306 samples long is fitted into 256
samples. The256 points end up being a distorted version of the correct signal. This
process is called circular convolution.
Circular Convolution using MATLAB:-

Program:
clc;
clear all;
close all;
disp('circular convolution program');
x=input('enter i/p x(n):');
m=length(x);
h=input('enter i/p sequence h(n)');
n=length(h);
subplot(2,2,1),
stem(x);
title('i/p sequence x(n)is:');
xlabel('---->n');
ylabel('---->x(n)');
8
grid;
subplot(2,2,2),
stem(h);
title('i/p sequence h(n)is:');
xlabel('---->n');
ylabel('---->h(n)');
grid;
disp('circular convolution of x(n) & h(n) is y(n):');
if(m-n~=0)
if(m>n)
h=[h,zeros(1,m-n)];
n=m;
end
x=[x,zeros(1,n-m)];
m=n;
end
y=zeros(1,n);
y(1)=0;
a(1)=h(1);
for j=2:n
a(j)=h(n-j+2);
end
%circular convolution
for i=1:n
y(1)=y(1)+x(i)*a(i);
end
for k=2:n
y(k)=0;
% circular shift
for j=2:n
x2(j)=a(j-1);
end
x2(1)=a(n);
for i=1:n
if(i<n+1)
a(i)=x2(i);
y(k)=y(k)+x(i)*a(i);
end
end
end
y
subplot(2,2,[3,4]),stem(y);
title('convolution of x(n) & h(n) is:');
xlabel('---->n');
ylabel('---->y((n)');grid;
EXP NO:3
Framing & windowing of speech signal. DATE:
9
AIM: To verify Framing & windowing of speech signal using MATLAB.
EQUIPMENT REQUIRED:
PC MATLAB
Power Adapter
Framing & windowing of speech signal using MATLAB:-
Program:
[signal,fs]=wavread('Noor.wav');
%Shift percentage is 40%to cancel the background effect
%and Convert the signal to row vector
s=signal(.4*length(signal):end)';
Number_of_frames=240; %Number of frames
window_length=round(fs/Number_of_frames);%size of the frame
Frame_No=zeros(Number_of_frames);%The frame number
Frame_No(1)=1;
for i=1:Number_of_frames-1
Frame_No(i+1)=Frame_No(i)+window_length;%Construct an array for the start
number of each frame
end
j=input('Enter the frame number: ');
N=Frame_No(j);%specify which frame
plot((N:N+window_length-1),s(N:N+window_length-1));%plot the frame
title(['Frame Number ',num2str(j)]);xlabel('Time');ylabel('Amplitude');
figure,
plot(s);title('input signal');xlabel('Time');ylabel('Amplitude');
Finding voice & unvoiced detection for each

EXP NO:4 DATE:
frame of the speech signal
AIM:
10
The main aim of this experiment is to determine whether the speech waveform is voiced
or unvoiced, in other words speech or silence.
Theory:
The input speech waveform is segmented with the help of a window either a rectangular
of hamming window. Then the auto correlation function of the speech segment is
calculated by using formula
m= N−1−k
Rn (k) = ∑ ¿ ¿x(n+m)ώ(m)][x(n+m+k)ώ(k+m)]
m =0
The property of the short time auto correlation to reveal periodicity in a signal is used
here. The auto correlation of the voiced spech segment retains the periodicity. On the
other hand , the autocorrelation of the unvoiced speech signal looks more like noise. In
general, the auto correlation is considered as a robust indicator of periodicity. Thus we
will decide whether a speech wave form is voiced or unvoiced.
Algorithm of voiced – unvoiced speech detection:
Step-1: record the wave file that is to be analyzed. (wav record).
Step-2: take some samples from the entire array set into a variable.
Step-3: find the autocorrelation of the segmented section. (xcorr)
Step-4: take another segment from the same wave file at a different time instant.
Step-5: perform the autocorrelation for the second segment just as step-3.
Step-6: plot the autocorrelation of these two segments.
Step-7: the speech segment with higher magnitude is the voice speech.
MATLAB CODE DETECTION OF VOICED AND UNVOICED SPEECH

SIGNAL:
Clc;
clear all;
close all;
% voiced and unvoiced speech as input
% Fs=8000;
%x= wavrecord(4*Fs,Fs,’double’);
[x Fs ]= wavread(‘SP06.wav’);
Ss1=x(2920:2980) ;
[ac4, lags4]=xcorr(ss1) ;
Ss4 =x (2920:2980) ;
[ac4, lags4]=xcorr(ss4) ;
Subplot(2,2,1) ;
Plot(ss1) ;
legend(‘voiced speech’)
Subplot(2,2,2) ;
Plot (lags1,ac1);
11
Xlim([lags1(1) lags1(end)];
Legend(‘ autocorrelation of voiced speech’)
Grid on;
Subplot (2,2,3);
Plot(ss4);
Legend(‘unvoiced speech’)
Subplot(2,2,4);
Plot(lags4, ac4);
Xlim ([lags1(1) lags1(end)]) ;
Legend(‘ autocorrelation of unvoiced speech’)
Grid on ;
CONCLUSION:
Notice from the output waveforms that the autocorrelation of the voiced speech
segment retains the periodicity .on the other hand ,the autocorrelation of the unvoiced
speech segment looks like more noise.
EXP NO:5 IIR FILTER IMPLEMENTATION USING PROBE DATE:

POINTS
12
AIM: To verify IIR Filter Implementation Using Probe Points using MATLAB
EQUIPMENT REQUIRED:
PC MATLAB
Power Adapter
IIR Filter Implementation Using Probe Points using MATLAB:-
Program
clear all
load y2;
x = y2;
x = x/max(abs(x));
load SOS
N = length(x);

b1(1:3) = SOS(1,1:3); %B(z) coefficients
a1(1:3) = SOS(1,4:6); %A(z) coefficients
h_a1 = impz(1, a1, 100);
h_a2 = impz(1, a2, 100);
h_a3 = impz(1, a3, 100);
%compute scaling factors for each section. This computation is done
%offline prior to coding your filter on the DSP.

%compute scaling factors
G1 = sum(abs(h_a1));

%scale input
x_scaled = x/G1;
%scale b coefficients
b1 = b1*G1/G2;
b2 = b2*G2/G3;
b3 = b3*G3;
%initialize ripple buffers
%SOS1
x01 = 0;
x11 = 0;
x21 = 0;
%SOS2
13
x02 = 0;
x12 = 0;
x22 = 0;
%SOS3
x03 = 0;
x13 = 0;
x23 = 0;
for n = 1:N,
%SOS1 computations
x01 = x_scaled(n) - a1(2)*x11 - a1(3)*x21;
y1 = b1(1)*x01 + b1(2)*x11 + b1(3)*x21;
x21 = x11; %shift ripple buffer
x11 = x01;
x1(n) = x01;
%SOS2 computations
x02 = y1 - a2(2)*x12 - a2(3)*x22;
y2 = b2(1)*x02 + b2(2)*x12 + b2(3)*x22;
x12 = x02;
x2(n) = x02;
%SOS3 computations
x03 = y2 - a3(2)*x13 - a3(3)*x23;
y(n) = b3(1)*x03 + b3(2)*x13 + b3(3)*x23;
x13 = x03;
x3(n) = x03;
end
%this is just to compare with the Matlab 'sosfilt' function
y1 = sosfilt(SOS,x);
norm(y1' - y)
IMPLEMENTATION OF FIR FILTERS ON DSP

EXP NO:6 PROCESSOR DATE:
AIM: To verify Implementation Of FIR Filters On DSP Processor using MATLAB
EQUIPMENT REQUIRED:
14
PC MATLAB
Power Adapter
Implementation Of FIR Filters On DSP Processor using MATLAB:-
Program
Fs=8e3; %Specify Sampling Frequency

Ts=1/Fs; %Sampling period.
Ns=512; %Number of time samples to be plotted.
t=[0:Ts:Ts*(Ns-1)]; %Make time array that contains Ns elements
%t = [0, Ts, 2Ts, 3Ts,..., (Ns-1)Ts]
f1=500;
f2=1800;
f3=2000;
f4=3200;
x1=sin(2*pi*f1*t); %create sampled sinusoids at different frequencies
x2=sin(2*pi*f2*t);
x3=sin(2*pi*f3*t);
x4=sin(2*pi*f4*t);
x=x1+x2+x3+x4; %Calculate samples for a 4-tone input signal
grid on;
N=16; %FIR1 requires filter order (N) to be EVEN
%when gain = 1 at Fs/2.
W=[0.4 0.6]; %Specify Bandstop filter with stop band between
%0.4*(Fs/2) and 0.6*(Fs/2)
B=fir1(N,W,'DC-1');%Design FIR Filter using default (Hamming window.
B %Leaving off semi-colon causes contents of
%B (the FIR coefficients) to be displayed.
A=1; %FIR filters have no poles, only zeros.
freqz(B,A); %Plot frequency response - both amp and phase response.
pause; %User must hit any key on PC keyboard to go on.
figure; %Create a new figure window, so previous one isn't lost.
subplot(2,1,1); %Two subplots will go on this figure window.
Npts=200;
plot(t(1:Npts),x(1:Npts)) %Plot first Npts of this 4-tone input signal
title('Time Plots of Input and Output');
xlabel('time (s)');
ylabel('Input Sig');
%Now apply this filter to our 4-tone test sequence
y = filter(B,A,x);
subplot(2,1,2); %Now go to bottom subplot.
plot(t(1:Npts),y(1:Npts)); %Plot first Npts of filtered signal.
xlabel('time (s)');
Filtered Sig');
pause;
15
figure; %Create a new figure window, so previous one isn't lost.
subplot(2,1,1);
xfftmag=(abs(fft(x,Ns))); %Compute spectrum of input signal.
xfftmagh=xfftmag(1:length(xfftmag)/2);
%Plot only the first half of FFT, since second half is mirror imag
%the first half represents the useful range of frequencies from
%0 to Fs/2, the Nyquist sampling limit.
f=[1:1:length(xfftmagh)]*Fs/Ns; %Make freq array that varies from
%0 Hz to Fs/2 Hz.
plot(f,xfftmagh); %Plot frequency spectrum of input signal
title('Input and Output Spectra');
xlabel('freq (Hz)');
ylabel('Input Spectrum');
subplot(2,1,2);
yfftmag=(abs(fft(y,Ns)));
yfftmagh=yfftmag(1:length(yfftmag)/2);
%Plot only the first half of FFT, since second half is mirror image
%the first half represents the useful range of frequencies from
%0 to Fs/2, the Nyquist sampling limit.
plot(f,yfftmagh); %Plot frequency spectrum of input signal
xlabel('freq (Hz)');
EXP NO:7 Representation of different Q-formats using DATE:

GEL function
AIM: To verify Representation of different Q-formats using GEL function using

MATLAB
EQUIPMENT REQUIRED:
PC MATLAB
16
Power Adapter
Representation of different Q-formats using GEL function using

MATLAB:-
Program
function y = dec2q(x,a,b,format)
% function y = dec2q(x,a,b,format)
% Input is a vector of decimal numbers and the output is the equivalent numbers in Qa.b
format
% in either binary or hex format where "a' is number of bits to the left of the
% binary point not including the sign bit, and 'b' is the number of bits to the
% right of the binary point.
% Input numbers outside the range are clipped
% An example of a Q0.15 (also called Q15) number is 1.100 0000 0000 0000 = -1 + 0.5
= -0.5.
% x (vector) input in decimal
% a (scalar) number of bits to the left of the binary point not including the sign bit
% (optional, default a = 0)
% b (scalar) number of bits to the right of the binary point
% (optional, dafault b = 15)
% format (string) format of the output 'bin' or 'hex' (optional, default 'hex')
% y (string matrix) output in Qa.b format (each output number on a row of the matrix)
% written by Joe Hoffbeck, 2/5/17
% default a = 0
if nargin < 2,
a = 0;
end
if isempty(a),
a = 0;
end
% default b = 15
if nargin < 3,
b = 15;
end
if isempty(a),
b = 15;
end
% default format = 'hex'

if nargin < 4,
format = 'hex';
end
format = lower(format); % convert to lower case (to accept 'HEX' or 'BIN')
17
if a < 0,
error('dec2q: input a must be greater than or equal to 0')
end
if b < 0,
error('dec2q: input b must be greater than or equal to 0')
end
if ~strcmp(format,'hex') & ~strcmp(format,'bin'),

error('dec2q: format must be either ''hex'' or ''bin''')
end
N = a + b + 1; % number of bits in output number

y = [];
for i = 1:length(x),
% clip input to range of Qa.b
if x(i) < -2â,
x(k) = -2â;
disp('dec2q: input clipped (too small)')
end
if x(i) > 2â - 2^-b,
x(i) = 2â - 2^-b;
disp('dec2q: input clipped (too big)')
end
% scale input to an integer and convert to two's complement format

if x(i) >= 0,
d = dec2bin(x(i)*2^b,N); % if positive, convert to binary
else
d = dec2bin(2^N - abs(x(i)*2^b),N); % if negative, take two's complement
and convert to binary
end
% convert to hex (if needed)

if strcmp(format,'hex'),
if round(N/4) ~= N/4,
error('dec2q: hex output not supported for these values of a and b because the
number of bits is not a multiple of 4')
end
d = dec2hex(bin2dec(d),N/4);
end
y(i,:) = d;
end
y=char(y);
18
EXP NO:8 IMAGE ENHANCEMENT USING SPATIAL AND DATE:

FREQUENCY DOMAIN
AIM: To verify Image Enhancement Using Spatial And Frequency Domain using
MATLAB
EQUIPMENT REQUIRED:
PC MATLAB
19
Power Adapter
Image Enhancement Using Spatial And Frequency Domain using

MATLAB:-
Program
footBall=imread('football.jpg');
%Convert to grayscale
footBall=rgb2gray(footBall);
imshow(footBall)
%Determine good padding for Fourier transform

PQ = paddedsize(size(footBall));
%Create a Gaussian Lowpass filter 5% the width of the Fourier transform

D0 = 0.05*PQ(1);
H = lpfilter('gaussian', PQ(1), PQ(2), D0);
% Calculate the discrete Fourier transform of the image

F=fft2(double(footBall),size(H,1),size(H,2));
% Apply the highpass filter to the Fourier spectrum of the image

LPFS_football = H.*F;
% convert the result to the spacial domain.

LPF_football=real(ifft2(LPFS_football));
% Crop the image to undo padding

LPF_football=LPF_football(1:size(footBall,1), 1:size(footBall,2));
%Display the blurred image

figure, imshow(LPF_football, [])
% Display the Fourier Spectrum

% Move the origin of the transform to the center of the frequency rectangle.
Fc=fftshift(F);
Fcf=fftshift(LPFS_football);
% use abs to compute the magnitude and use log to brighten display
S1=log(1+abs(Fc));
S2=log(1+abs(Fcf));
figure, imshow(S1,[])
figure, imshow(S2,[])
20
EXP NO:9 Implementation of Image segmentation techniques DATE:
AIM: To verify Implementation of Image segmentation techniques using MATLAB
EQUIPMENT REQUIRED:
PC MATLAB
Power Adapter
Implementation of Image segmentation techniques using MATLAB:-
Program
21
% This is a program for extracting objects from an image. Written for vehicle number
plate segmentation and extraction
% Authors : Jeny Rajan, Chandrashekar P S
% U can use attached test image for testing
% input - give the image file name as input. eg :- car3.jpg
clc;
clear all;
k=input('Enter the file name','s'); % input image; color image
im=imread(k);
im1=rgb2gray(im);
im1=medfilt2(im1,[3 3]); %Median filtering the image to remove noise%
BW = edge(im1,'sobel'); %finding edges
[imx,imy]=size(BW);
msk=[0 0 0 0 0;
0 1 1 1 0;
0 1 1 1 0;
0 1 1 1 0;
0 0 0 0 0;];
B=conv2(double(BW),double(msk)); %Smoothing image to reduce the number of
connected components
L = bwlabel(B,8);% Calculating connected components
mx=max(max(L))
% There will be mx connected components.Here U can give a value between 1 and mx
for L or in a loop you can extract all connected components
% If you are using the attached car image, by giving 17,18,19,22,27,28 to L you can
extract the number plate completely.
[r,c] = find(L==17);
rc = [r c];
[sx sy]=size(rc);
n1=zeros(imx,imy);
for i=1:sx
x1=rc(i,1);
y1=rc(i,2);
n1(x1,y1)=255;
end % Storing the extracted image in an array
figure,imshow(im);
figure,imshow(im1);
figure,imshow(B);
figure,imshow(n1,[]);
22
EXP NO:10 Extraction of frames from video signal DATE:
AIM: To verify Extraction of frames from video signal using MATLAB
EQUIPMENT REQUIRED:
PC MATLAB
Power Adapter
Extraction of frames from video signal using MATLAB:-
Program
%How to get frames from a video
mov=aviread('fun.avi');
23
Im=frame2im(mov(1,58));
figure,imshow(Im);
The above process can be done using 'mmreader' function also.
MATLAB CODE:
Vobj=mmreader('fun.avi');
Im=read(Vobj,58);
imshow(Im);
Lmov=aviread('smiley.avi');
MIm=frame2im(Lmov(1,1));
figure,imshow(MIm);
%Preallocating the structure array

frame=struct('cdata',1,'colormap',cell([1 91]));
I added the modified frames to the original video.

%Modify the video in the middle
frame(1,1:57)=mov(1,1:57);
frame(1,58:87)=Lmov(1,1:30);
frame(1,88:91)=mov(1,64:67);
implay(frame,5);
%Part of the video

scz=figure;
set(scz, 'position', [300 100 size(Im,1)+130 size(Im,2)-100]);
FRange=[1 27:33];
%Here 1 is for the number of times the video has to be repeated
%27:33 indicates the frames from 27 to 33.
movie(scz,mov,FRange,5);
%To save the desired part alone
obj=avifile('VidPart.avi','FPS',5);
obj=addframe(obj,mov(27:33));
obj=close(obj);
24

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Department of Electronics & Communication Engineering

ADVANCED DIGITAL SIGNAL PROCESSING LAB

ADVANCED DIGITAL SIGNAL PROCESSING LAB

JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY ANANTAPUR

1. Study of various addressing modes of DSP using simple programming examples

NARAYANA ENGINEERING COLLEGE::GUDUR

1. Study of various addressing modes of DSP using simple programming examples

2. Implementation of Linear and Circular Convolution for sinusoidal signals

3. Framing & windowing of speech signal

5. IIR Filter Implementation using probe points

6. Implementation of FIR filters on DSP processor

7. Representation of different Q-formats using GEL function

8. Image enhancement using spatial & frequency domain

9. Implementation of Image segmentation techniques

10. Extraction of frames from Video signal

EXP NO:1 STUDY OF VARIOUS ADDRESSING MODES OF DSP DATE:

Addressing Mode Register (AMR):

ADDA and SUBA Instructions

Circular Addressing Mode

As with linear address arithmetic, offsetR/cst is shifted left by 3, 2, 1, or 0 according

ADDA and SUBA Instructions

As with linear address arithmetic, offsetR/cst is shifted left by 3, 2, 1, or 0 according

Example shows an ADDAH performed with register A4 in circular mode and BK0 =

IMPLEMENTATION Of LINEAR AND

AIM: To verify Linear Convolution using MATLAB.

Hardware required Software Required

Linear Convolution Using MATLAB:-

AIM: To verify Circular Convolution using MATLAB.

Circular Convolution using MATLAB:-

AIM: To verify Framing & windowing of speech signal using MATLAB.

Framing & windowing of speech signal using MATLAB:-

Finding voice & unvoiced detection for each

MATLAB CODE DETECTION OF VOICED AND UNVOICED SPEECH

EXP NO:5 IIR FILTER IMPLEMENTATION USING PROBE DATE:

IIR Filter Implementation Using Probe Points using MATLAB:-

IMPLEMENTATION OF FIR FILTERS ON DSP

AIM: To verify Implementation Of FIR Filters On DSP Processor using MATLAB

Implementation Of FIR Filters On DSP Processor using MATLAB:-

Fs=8e3; %Specify Sampling Frequency

EXP NO:7 Representation of different Q-formats using DATE:

AIM: To verify Representation of different Q-formats using GEL function using

Representation of different Q-formats using GEL function using

% written by Joe Hoffbeck, 2/5/17

% default format = 'hex'

if ~strcmp(format,'hex') & ~strcmp(format,'bin'),

N = a + b + 1; % number of bits in output number

% scale input to an integer and convert to two's complement format

% convert to hex (if needed)

EXP NO:8 IMAGE ENHANCEMENT USING SPATIAL AND DATE:

Image Enhancement Using Spatial And Frequency Domain using

%Determine good padding for Fourier transform

%Create a Gaussian Lowpass filter 5% the width of the Fourier transform

% Calculate the discrete Fourier transform of the image

% Apply the highpass filter to the Fourier spectrum of the image

% convert the result to the spacial domain.

% Crop the image to undo padding

%Display the blurred image

% Display the Fourier Spectrum