DSP - Lab Manual Final

ST.
JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL
INDEX
CYCLE -I 1. 2. 3. 4. 5. 6. SUM OF THREE SINUSOIDAL SIGNALS FREQUENCY RESPONSE OF ANALOG FILTERS N-POINT FET FOR A GIVEN SEQUENCE POWER SPECTRA; DENSITY LINEAR CONVOLUTION FREQUENCY RESPONSE OF FIR FILTERS a. Rectangular Window b. Hanning window c. Hamming Window d. Kaiser Window 7. SAMPLING OF A SINUSOIDAL SIGNALS CYCLE II
1. 2. 3. 4. 5.
STUDY OF TMS320C6713 PROCESSOR LINEAR CONVOLUTION CIRCULAR CONVOLUTION FFT OF 1 DIMENSIONAL SIGNAL POWER SPECTRAL DENSITY OF A SEQUENCE
ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL
PROCEDURE: 1. Open the matlab software by double clicking the icon on desktop. 2. Open the new M-file by using file menu 3. Write the program in new file 4. Click on save and run the icon 5. Perform error check which displayed on command window 6. Plot the waveforms which displays on figure window. 7. Note down the values, which displays on the work space.
1. GENERATION OF SUM OF THREE SINUSOIDAL SIGNALS PROGRAM: fs=100; t=(1:100)/fs; s1=sin (2*pi*15*t); subplot (3,3,1); plot (t,s1); title('sin signal with 15HZ'); s2= sin (2*pi*20*t); subplot(3,3,2); plot(t,s2); title('sin signal with 20HZ'); s3=sin (2*pi*30*t); subplot(3,3,3); plot(t,s3); title('sin signal with 30HZ'); %sum of three signals s=s1+s2+s3; subplot(3,3,4); plot (t,s); title ('sum of three sin waves'); %design of elliptical BPF [b,a]=ellip(4,0.1,40,[10,20]*(2/fs)); [h,w]=freqz(b,a,512);
subplot(3,3,5); plot((w*fs)/(2*pi),abs(h)); title('elliptical bandpass filter'); grid; %filtering the sum signal z=filter(b,a,s); subplot(3,3,6); plot(z); title('filter output'); %fft of sum signal y=fft(s,512); w=((0:255)/256*(fs/2)); subplot (3,3,7); plot (w,abs(y(1:256))); title('fft of three sin waves'); grid; legend('before'); %fft of filter output k=fft (z,512); w=((0:255) /256*(fs/2)); subplot (3,3,8); plot (w,abs(k(1:256))); title('fft of filter output'); grid; legend ('after');
OUTPUT:
2. FREQUENCY RESPONSE OF ANALOG FILTERS

2a) Program for the design of butter worth analog LPF:
rp=input ('enter passband ripple'); rs=input ('enter stopband ripple'); wp=input ('enter passband frequency'); ws=input ('enter stopband frequency'); fs=input ('enter sampling frequency'); w1=2*(wp/fs); w2=2*(ws/fs); [n,wn]=buttord(w1,w2,rp,rs,'s'); [z,p,k]=butter (n,wn); [b,a]=zp2tf (z,p,k); [b,a]= butter(n,wn,'s'); w=0:0.01:pi; [n,om]=freqs(b,a,w); m=20*log10(abs(n)); an=angle(n); subplot(2,1,1); plot(om/pi,m); ylabel ('grid in db....'); xlabel ('(a) normalised freq...'); subplot (2,1,2); plot (om/pi,an); xlabel ('(b) normalised freq...'); ylabel ('phase in radians....');
INPUT: enter passband ripple enter stopband ripple 0.15 30
enter passband frequency 500 enter stopband frequency 1000 enter sampling frequency 2000 OUTPUT:
2b) Program for the design of butter worth analogHPF:
rp=input ('enter passband ripple'); rs=input ('enter stopband ripple'); wp=input ('enter passband frequency'); ws=input ('enter stopband frequency'); fs=input ('enter sampling frequency'); w1=2*(wp/fs); w2=2*(ws/fs); [n,wn]=buttord(w1,w2,rp,rs,'s'); [b,a]= butter(n,wn,'high','s'); w=0:0.01:pi; [n,om]=freqs(b,a,w); m=20*log10(abs(n)); an=angle(n); subplot(2,1,1); plot(om/pi,m); ylabel ('grid in db....'); xlabel ('(a) normalised freq...'); subplot (2,1,2); plot (om/pi,an); xlabel ('(b) normalised freq...'); ylabel ('phase in radians....');
INPUT: enter passband ripple0.15 enter stopband ripple20 enter passband frequency500 enter stopband frequency1000 enter sampling frequency2000 OUTPUT:
ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL 2c) Program for the design of butter bandpass filter
rp=input ('enter passband ripple'); rs=input ('enter stopband ripple'); wp=input ('enter passband frequency'); ws=input ('enter stopband frequency'); fs=input ('enter sampling frequency'); w1=2*wp/fs; w2=2*ws/fs; [n,wn]=buttord(w1,w2,rp,rs,'s'); wn=[w1,w2]; [b,a]= butter(n,wn,'bandpass','s'); w=0:0.01:pi; [b,om]=freqs(b,a,w); m=20*log10(abs(b)); an=angle(b); subplot(2,1,1); plot(om/pi,m); ylabel ('gain in db....'); xlabel ('(a) normalised freq...'); subplot (2,1,2); plot (om/pi,an); xlabel ('(b) normalised freq...'); ylabel ('phase in radians....');
INPUT: enter passband ripple0.15 enter stopband ripple20 enter passband frequency500 enter stopband frequency1000 enter sampling frequency10000 OUTPUT:
ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL 2d) Program for the design of butter bandstop filter
rp=input ('enter passband ripple'); rs=input ('enter stopband ripple'); wp=input ('enter passband frequency'); ws=input ('enter stopband frequency'); fs=input ('enter sampling frequency'); w1=2*wp/fs; w2=2*ws/fs; [n,wn]=buttord(w1,w2,rp,rs,'s'); wn=[w1,w2]; [b,a]= butter(n,wn,'stop','s'); w=0:0.01:pi; [b,om]=freqs(b,a,w); m=20*log10(abs(b)); an=angle(b); subplot(2,1,1); plot(om/pi,m); ylabel ('gain in db....'); xlabel ('(a) normalised freq...'); subplot (2,1,2); plot (om/pi,an); xlabel ('(b) normalised freq...'); ylabel ('phase in radians....');
INPUT: enter passband ripple0.1 enter stopband ripple20 enter passband frequency500 enter stopband frequency1000
OUTPUT:
3. N - POINT FFT OF GIVEN SEQUENCE PROGRAM: fs=100; %INPUT SEQUENCE xn=input('enter the input sequence'); %ORDER OF FFT N=input('enter the no of points in the FFT'); subplot(3,1,1); stem(xn); title('input sequence'); L=length(xn); if(N<L) disp('N must be greater than or equal to L'); end; %FFT COMPUTATION xk=fft(xn,N); subplot(3,1,2); stem(xk); title('output sequence'); xn1=fft(xk); subplot(3,1,3); stem(xn1); title('inverse FFT of a given input sequence');
INPUT: enter the input sequence[1 1 1 1] enter the no of points in the FFT4 OUTPUT:
4. POWER SPECTRAL DENSITY PROGRAM: fs=100; t=(1:100)/fs; s= sin(2*pi*15*t); subplot(2,1,1); plot(t,s); title('Sine Wave'); xlabel('-------> ylabel('-------> grid; pxx=psd(s); subplot(2,1,2); plot(pxx); grid; xlabel('-------> ylabel('-------> Frequency(Hz)'); Magnitudue'); time(sec)'); Amp(volts)');
OUTPUT:
5. LINEAR CONVOLUTION PROGRAM: x=input('enter the first sequence'); h=input('enter the second sequence'); % COMPUTING CONVERSION y=conv(x,h); subplot(3,1,1); stem(x); title('First Sequence'); xlabel('Amp ------>'); ylabel('(a)n------->'); subplot(3,1,2); stem(h); title('Second Sequence'); ylabel('Amp-------->'); xlabel('(b)--------->'); subplot(3,1,3); stem (y); title('output sequence'); ylabel('amp...'); xlabel('(c)n...'); disp (' the resultant signal is');
INPUT: enter the first sequence[1 3 1 2] enter the second sequence[1 2 1 2]
OUTPUT:
6. PROGRAM FOR WINDOWING TECHNIQUES

6(a)PROGRAM FOR RECTANGULAR WINDOW
rp= input ('enter the passband ripple'); rs= input ('enter the stopband ripple'); fp= input ('enter the passband frequency'); fs= input ('ente the stopband frequency'); f= input ('enter the sampling frequency'); wp=2*fp/f; ws=2*fs/f; num=-20*log10(sqrt (rp*rs))-13; den=14.6*(fs-fp)/f; n= ceil(num/den); n1=n+1; if (rem(n,2)~=0); n1=n; n=n-1; end; y=boxcar(n1); % LOWPASS FILTER b=fir1(n,wp,y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,1); plot(om/pi,m); title('LOW PASS FILTER')
ylabel('gains in dB....'); xlabel('normalised frequency...'); %HIGHPASS FILTER b=fir1(n,wp,'HIGH',y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,2); plot(om/pi,m); title('HIGH PASS FILTER'); ylabel('gains in dB....'); xlabel('normalised frequency...'); %BANDPASS FILTER wn= [wp ,ws]; b=fir1(n,wn,y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,3); plot(om/pi,m); title('BAND PASS FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); %BANDSTOP FILTER b=fir1(n,wn,'STOP',y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h));
subplot(2,2,4); plot(om/pi,m); title('BAND STOP FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...');
INPUT: enter the passband ripple0.03 enter the stopband ripple0.01 enter the passband frequency1400 ente the stopband frequency2000 enter the sampling frequency8000 OUTPUT:
ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL 6(b)PROGRAM FOR KAISER WINDOW
rp= input ('enter the passband ripple'); rs= input ('enter the stopband ripple'); fp= input ('enter the passband frequency'); fs= input ('ente the stopband frequency'); f= input ('enter the sampling frequency'); beta=input ('enter the beta value'); wp=2*fp/f; ws=2*fs/f; num=-20*log10(sqrt (rp*rs))-13; den=14.6*(fs-fp)/f; n= ceil(num/den); n1=n+1; if (rem(n,2)~=0); n1=n; n=n-1; end; y=kaiser(n1,beta); % LOWPASS FILTER b=fir1(n,wp,y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,1); plot(om/pi,m); title('LOW PASS FILTER')
ylabel('gains in dB....'); xlabel('normalised frequency...'); %HIGHPASS FILTER b=fir1(n,wp,'HIGH',y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,2); plot(om/pi,m); title('HIGH PASS FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); %BANDPASS FILTER wn= [wp ws]; b=fir1(n,wn,y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,3); plot(om/pi,m); title('BAND PASS FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); %BANDSTOP FILTER b=fir1(n,wn,'STOP',y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h));
subplot(2,2,4); plot(om/pi,m); title('BAND STOP FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); INPUT: enter the passband ripple0.02 enter the stopband ripple0.01 enter the passband frequency1000 ente the stopband frequency1500 enter the sampling frequency10000 enter the beta value5.8 OUTPUT:
ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL 6(c) PROGRAM FOR HAMMING WINDOW
rp= input ('enter the passband ripple'); rs= input ('enter the stopband ripple'); fp= input ('enter the passband frequency'); fs= input ('ente the stopband frequency'); f= input ('enter the sampling frequency'); wp=2*fp/f; ws=2*fs/f; num=-20*log10(sqrt (rp*rs))-13; den=14.6*(fs-fp)/f; n= ceil(num/den); n1=n+1; if (rem(n,2)~=0); n1=n; n=n-1; end; y=hamming(n1); % LOWPASS FILTER b=fir1(n,wp,y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,1); plot(om/pi,m); title('LOW PASS FILTER') ylabel('gains in dB....');
xlabel('normalised frequency...'); %HIGHPASS FILTER b=fir1(n,wp,'HIGH',y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,2); plot(om/pi,m); title('HIGH PASS FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); %BANDPASS FILTER wn= [wp ws]; b=fir1(n,wn,y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,3); plot(om/pi,m); title('BAND PASS FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); %BANDSTOP FILTER b=fir1(n,wn,'STOP',y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,4);
plot(om/pi,m); title('BAND STOP FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); INPUT: enter the passband ripple0.02 enter the stopband ripple0.01 enter the passband frequency1200 ente the stopband frequency1700 enter the sampling frequency9000 OUTPUT:
ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL 6(d) PROGRAM FOR HANNING WINDOW
rp= input ('enter the passband ripple'); rs= input ('enter the stopband ripple'); fp= input ('enter the passband frequency'); fs= input ('ente the stopband frequency'); f= input ('enter the sampling frequency'); wp=2*fp/f; ws=2*fs/f; num=-20*log10(sqrt (rp*rs))-13; den=14.6*(fs-fp)/f; n= ceil(num/den); n1=n+1; if (rem(n,2)~=0); n1=n; n=n-1; end; y=hanning(n1); % LOWPASS FILTER b=fir1(n,wp,y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,1); plot(om/pi,m); title('LOW PASS FILTER') ylabel('gains in dB....');
xlabel('normalised frequency...'); %HIGHPASS FILTER b=fir1(n,wp,'HIGH',y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,2); plot(om/pi,m); title('HIGH PASS FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); %BANDPASS FILTER wn= [wp ws]; b=fir1(n,wn,y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,3); plot(om/pi,m); title('BAND PASS FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); %BANDSTOP FILTER b=fir1(n,wn,'STOP',y); [h,om]=freqz(b,1,256); m=20*log10 (abs(h)); subplot(2,2,4);
plot(om/pi,m); title('BAND STOP FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...'); INPUT: enter the passband ripple0.03 enter the stopband ripple0.01 enter the passband frequency1400 ente the stopband frequency2000 enter the sampling frequency8000 OUTPUT:
7. SAMPLING OF A SINUSOIDAL SIGNALS

7(a) DOWNSAMPLING
N=50; n=0:1:N-1; X=sin(2*pi*n/20)+sin(2*pi*n/15); M=2; X1=X(1:M:N); n1=1:1:N/M; subplot(2,1,1); stem(n,X); xlabel('n'); ylabel('X'); title('Input Sequence'); subplot(2,1,2); stem(n1-1,X1); xlabel('n'); ylabel('X1'); title('Downsampling Sequence');
OUTPUT:
ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL 7(b) UPSAMPLING
N=10; n=0:1:N-1; X=sin(2*pi*n/10)+sin(2*pi*n/5); L=3; X1=[zeros(1,L*N)]; n1=1:1:L*N; j=1:L:L*N; X1(j)=X; subplot(2,1,1); stem(n,X); xlabel('n'); ylabel('X'); title('Input Sequence'); subplot(2,1,2); stem(n1,X1); xlabel('n'); ylabel('X1'); title('Upsampling Sequence');
OUTPUT:
CYCLE-II
1. INTRODUCTION TO DSP PROCESSORS

A signal can be defined as a function that conveys information, generally about the state or behavior of a physical system. There are two basic types of signals viz Analog (continuous time signals which are defined along a continuum of times) and Digital (discrete-time). Remarkably, under reasonable constraints, a continuous time signal can be adequately represented by samples, obtaining discrete time signals. Thus digital signal processing is an ideal choice for anyone who needs the performance advantage of digital manipulation along with todays analog reality. Hence a processor which is designed to perform the special operations(digital manipulations) on the digital signal within very less time can be called as a Digital signal processor. The difference between a DSP processor, conventional microprocessor and a microcontroller are listed below. Microprocessor or General Purpose Processor such as Intel xx86 or Motorola 680xx family Contains - only CPU -No RAM -No ROM -No I/O ports -No Timer Microcontroller such as 8051 family Contains - CPU - RAM - ROM -I/O ports - Timer & - Interrupt circuitry Some Micro Controllers also contain A/D, D/A and Flash Memory DSP Processors such as Texas instruments and Analog Devices Contains - CPU - RAM -ROM - I/O ports - Timer
ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL Optimized for fast arithmetic - Extended precision - Dual operand fetch - Zero overhead loop - Circular buffering The basic features of a DSP Processor are Feature Fast-Multiply accumulate Multiple access memory architecture Specialized addressing modes Specialized program control On-chip peripherals and I/O interfaces Use Most DSP algorithms, including filtering, transforms, etc. are multiplication- intensive Many data-intensive DSP operations require reading a program instruction and multiple data items during each instruction cycle for best performance Efficient handling of data arrays and first-in, first-out buffers in memory Efficient control of loops for many iterative DSP algorithms. Fast interrupt handling for frequent I/O operations. On-chip peripherals like A/D converters allow for small low cost system designs. Similarly I/O interfaces tailored for common peripherals allow clean interfaces to off-chip I/O devices.
Key Features
The C6713 DSK is a low-cost standalone development platform that enables users to evaluate and develop applications for the TI C67xx DSP family. The DSK also serves as a hardware reference design for the TMS320C6713 DSP. Schematics, logic equations and application notes are available to ease hardware development and reduce time to market.
The DSK comes with a full compliment of on-board devices that suit a wide variety of application environments. Key features include: A Texas Instruments TMS320C6713 DSP operating at 225 MHz. An AIC23 stereo codec 8 Mbytes of synchronous DRAM 512 Kbytes of non-volatile Flash memory (256 Kbytes usable in default configuration) 4 user accessible LEDs and DIP switches Software board configuration through registers implemented in CPLD Configurable boot options Standard expansion connectors for daughter card use JTAG emulation through on-board JTAG emulator with USB host interface or external emulator Single voltage power supply (+5V)
2.CHITECTURE OF 6713 DSP PROCESSOR

This chapter provides an overview of the architectural structure of the TMS320C67xx DSP, which comprises the central processing unit (CPU), memory, and on-chip peripherals. The C67xE DSPs use an advanced modified Harvard architecture that maximizes processing power with eight buses. Separate program and data spaces allow simultaneous access to program instructions and data, providing a high degree of parallelism. For example, three reads and one write can be performed in a single cycle.Instructions with parallel store and application-specific instructions fully utilize this architecture. In addition, data can be transferred between data and program spaces. Such Parallelism supports a powerful set of arithmetic, logic, and bit-mani pulation operations that can all be performed in a single machine cycle. Also, the C67xx DSP includes the control mechanisms to manage interrupts, repeated operations, and function calling.
Fig 2 1 BLOCK DIAGRAM OF TMS 320VC 6713
Bus Structure
The C67xx DSP architecture is built around eight major 16-bit buses (four program/data buses and four address buses): _ The program bus (PB) carries the instruction code and immediate operands from program memory. _ Three data buses (CB, DB, and EB) interconnect to various elements, such as the CPU, data address generation logic, program address generation logic, on-chip peripherals, and data memory. _ The CB and DB carry the operands that are read from data memory. _ The EB carries the data to be written to memory. _ Four address buses (PAB, CAB, DAB, and EAB) carry the addresses needed for instruction execution. The C67xx DSP can generate up to two data-memory addresses per cycle using the two auxiliary register arithmetic units (ARAU0 and ARAU1). The PB can carry data operands stored in program space (for instance, a coefficient table) to the multiplier and adder for multiply/accumulate operations or to a destination in data space for data move instructions (MVPD and READA). This capability, in conjunction with the feature of dual-operand read, supports the execution of single-cycle, 3-operand instructions such as the FIRS instruction. The C67xx DSP also has an on-chip idirectional bus for accessing on-chip peripherals. This bus is connected to DB and EB through the bus exchanger in the CPU interface. Accesses that use this bus can require two or more cycles for reads and writes, depending on the peripherals structure.
Central Processing Unit (CPU)

The CPU is common to all C67xE devices. The C67x CPU contains: _ 40-bit arithmetic logic unit (ALU) _ Two 40-bit accumulators _ Barrel shifter _ 17 17-bit multiplier _ 40-bit adder _ Compare, select, and store unit (CSSU) _ Data address generation unit _ Program address generation unit
Arithmetic Logic Unit (ALU)

The C67x DSP performs 2s-complement arithmetic with a 40-bit arithmetic logic unit (ALU) and two 40-bit accumulators (accumulators A and B). The ALU can also perform Boolean operations. The ALU uses these inputs: _ 16-bit immediate value _ 16-bit word from data memory _ 16-bit value in the temporary register, T _ Two 16-bit words from data memory _ 32-bit word from data memory _ 40-bit word from either accumulator The ALU can also function as two 16-bit ALUs and perform two 16-bit operations simultaneously.
Fig 2 2 ALU UNIT
Accumulators
Accumulators A and B store the output from the ALU or the multiplier/adder block. They can also provide a second input to the ALU; accumulator A can be an input to the multiplier/adder. Each accumulator is divided into three parts: _ Guard bits (bits 3932) _ High-order word (bits 3116) _ Low-order word (bits 150) Instructions are provided for storing the guard bits, for storing the high- and the low order accumulator words in data memory, and for transferring 32-bit accumulator words in or out of data memory. Also, either of the accumulators can be used as temporary storage for the other.
Barrel Shifter
The C67x DSP barrel shifter has a 40-bit input connected to the accumulators or to data memory (using CB or DB), and a 40-bit output connected to the ALU or to data memory (using EB). The barrel shifter can produce a left shift of 0 to 31 bits and a right shift of 0 to 16 bits on the input data. The shift requirements are defined in the shift count field of the instruction, the shift count field (ASM) of status register ST1, or in temporary register T (when it is designated as a shift count register).The barrel shifter and the exponent encoder normalize the values in an accumulator in a single cycle. The LSBs of the output are filled with 0s, and the MSBs can be either zero filled or sign extended, depending on the state of the sign-extension mode bit (SXM) in ST1. Additional shift capabilities enable the processor to perform numerical scaling, bit extraction, extended arithmetic, and overflow prevention operations.
Multiplier/Adder Unit
The multiplier/adder unit performs 17 _ 17-bit 2s-complement multiplication with a 40- bit addition in a single instruction cycle. The multiplier/adder block consists of several elements: a multiplier, an adder, signed/unsigned input control logic, fractional control logic, a zero detector, a rounder (2s complement), overflow/saturation logic, and a 16-bit temporary storage register (T). The multiplier has two inputs: one input is selected from T, a data-memory operand, or accumulator A; the other is selected from program memory, data memory, accumulator A, or an immediate value. The fast, on-chip multiplier allows the C54x DSP to perform operations efficiently such as convolution, correlation, and filtering. In addition, the multiplier and ALU together execute multiply/accumulate (MAC) computations and ALU operations in parallel in a single instruction cycle. This function is used in determining the Euclidian distance and in implementing symmetrical and LMS filters, which are required for complex DSP algorithms. See section 4.5, Multiplier/Adder Unit, on page 4-19, for more details about the multiplier/adder unit.
Fig 2 3 MULTIPLIER/ADDER UNIT
These are the some of the important parts of the processor and you are instructed to go through the detailed architecture once which helps you in developing the optimized code for the required application.
PROCEDURE: 1. Open Code Composer Studio, make sure the DSP kit is turned on. 2. Start a anew project using Project-new pull down menu, save it in a separate directory (c:\ti\myprojects)with name iconv.pjt. 3. Add the source files conv.asm to the project using Projectadd files to project pull down menu 4. Add the linker command file hello.cmd (Path:c;\ti\tutorial\dsk6713\hello1\hello.cmd) 5. Add the run time support library file rts6700.lib (Path:c;\ti\c600\cgtools\lib\rts6700.lib) 6. Compile the program using the Project-compile pull down menu or by checking the shortcut icon on the left side of program window. 7. Build the program using the Project-Build pull down menu or by clicking the shortcut icon on the left side of program window. 8. Load the program(icnov.out) in program memory of DSP chip using the Fileload program pull down menu 9. To View output graphically Select viewgraphtime and frewuency.
1. LINEAR CONVOLUTION
PROGRAM: #include<stdio.h> int x[15],h[15],y[15]; main() { int i,j,m,n; printf("\n enter value for m"); scanf("%d",&m); printf("\n enter value for n"); scanf("%d",&n); printf("Enter values for i/p x(n):\n"); for(i=0;i<m;i++) scanf("%d",&x[i]); printf("Enter Values for i/p h(n) \n"); for(i=0;i<n; i++) scanf("%d",&h[i]); // padding of zeros for(i=m;i<=m+n-1;i++) x[i]=0; for(i=n;i<=m+n-1;i++) h[i]=0; /* convolution operation */ for(i=0;i<m+n-1;i++)
{ y[i]=0; for(j=0;j<=i;j++) { y[i]=y[i]+(x[j]*h[i-j]); }} //displaying the o/p for(i=0;i<m+n-1;i++) printf("\n The Value of output y[%d]=%d",i,y[i]); } Result: enter value for m4 enter value for n4 Enter values for i/p 1234 Enter Values for n 1234 The Value of output y[0]=1 The Value of output y[1]=4 The Value of output y[2]=10 The Value of output y[3]=20 The Value of output y[4]=25 The Value of output y[5]=24 The Value of output y[6]=16
OUTPUT WAVE FORM:
2. CIRCULAR CONVOLUTION PROGRAM: #include<stdio.h> int m,n,x[30],h[30],y[30],i,j, k,x2[30],a[30]; void main() { printf(" Enter the length of the first sequence\n"); scanf("%d",&m); printf(" Enter the length of the second sequence\n"); scanf("%d",&n); printf(" Enter the first sequence\n"); for(i=0;i<m;i++) scanf("%d",&x[i]); printf(" Enter the second sequence\n"); for(j=0;j<n;j++) scanf("%d",&h[j]); if(m-n!=0) /*If length of both sequences are not equal*/ { if(m>n) /* Pad the smaller sequence with zero*/ { for(i=n;i<m;i++) h[i]=0; n=m; } for(i=m;i<n;i++)
x[i]=0; m=n; } y[0]=0; a[0]=h[0]; for(j=1;j<n;j++) /*folding h(n) to h(-n)*/ a[j]=h[n-j]; /*Circular convolution*/ for(i=0;i<n;i++) y[0]+=x[i]*a[i]; for(k=1;k<n;k++) { y[k]=0; /*circular shift*/ for(j=1;j<n;j++) x2[j]=a[j-1]; x2[0]=a[n-1]; for(i=0;i<n;i++) { a[i]=x2[i]; y[k]+=x[i]*x2[i]; } } /*displaying the result*/ printf(" The circular convolution is\n");
for(i=0;i<n;i++) printf("%d \t",y[i]); } OUTPUT:Enter the length of the first sequence 4 Enter the length of the second sequence 3 Enter the first sequence 1234 Enter the second sequence 123 The circular convolution is 18 16 10 16
3.FAST FOURIER TRANSFORM PROGRAM: #include<math.h> #define PTS 64 #define PI 3.14159265358979// # of points for FFT typedef struct{float real,imag;} COMPLEX; void FFT(COMPLEX*Y, int n); float iobuffer[PTS]; float x1[PTS]; short i; short buffercount=0; int upper_leg,lower_leg; //iobuffer short flag=0; //full COMPLEX w[PTS]; COMPLEX samples[PTS];
main() {
for (i=0;i<PTS;i++) {
// set up twiddle constants in w
w[i].real=cos(2*PI*i/(PTS*2.0) ); w[i].imag=-sin(2*PI*i/(PTS*2.0)
//Re component of twiddle constants ); //Img component of twiddle constants
for (i=0;i<PTS;i++) {
// swap buffers
iobuffer[i]=sin(2*PI*10*i/64.0);/*10--> freq, 64---> sampling freq*/ samples[i].real=0.0; samples[i].imag=0.0; } for (i=0;i<PTS;i++) { samples[i].real=iobuffer[i]; // buffer with new data } for (i=0;i<PTS;i++) { samples[i].imag=0.0; } FFT(samples, PTS); for (i=0;i<PTS;i++) { x1[i]=sqrt(samples[i].real*samples[i].real+samples[i].imag*samples[i].imag); } } //#define PTS 64 //typedef struct {float real,imag;} COMPLEX; //extern COMPLEX W[PTS]; // CALL FUNCTION FFT.c // compute magnitude // imaginary components=0 // swap buffers // swap buffers
void FFT(COMPLEX *y,int N) { COMPLEX temp1,temp2; int i,j,k; int upper_leg,lower_leg; int leg_diff; int num_stages=0; int index,step; i=1; do { num_stages +=1; i=i*2; }while(i!=N); leg_diff=N/2; for(i=0;i<num_stages;i++); { index=0; for(j=0;j<leg_diff;j++) { for(upper_leg=j;upper_leg<N;upper_leg+=(2*leg_diff)) { lower_leg=upper_leg + leg_diff; temp1.real=(y[upper_leg]).real+(y[lower_leg]).real; temp1.imag=(y[upper_leg]).imag+(y[lower_leg]).imag;
temp2.real=(y[upper_leg]).real-(y[lower_leg]).real; temp2.imag=(y[upper_leg]).imag+(y[lower_leg]).imag; (y[lower_leg]).real=temp2.real*(w[index]).real-temp2.imag*(w[index]).imag; (y[lower_leg]).imag=temp2.real*(w[index]).imag+temp2.imag*(w[index]).real; (y[upper_leg]).real = temp1.real; (y[upper_leg]).imag = temp1.imag; } index += step; } leg_diff = leg_diff/2; step *=2; } j=0; for (i=1; i<(N-1); i++) { k=N/2; while (k<=j) { j=j-k; k=k/2; } j=j+k; if (i<j) { temp1.real = (y[j]). real; // bit reversal for resequencing data
temp1.imag = (y[j]). imag; (y[j]).real = (y[i]).real; (y[j]).imag = (y[i]).imag; (y[i]).real = temp1.real; (y[i]).imag = temp1.imag; } } return; }
RESULT: Input
OUTPUT:
5.POWER SPECTRAL DENSITY OF A SEQUENCE PROGRAM: #include <math.h> #define pts 128 #define pi 3.14159265358979 typedef struct { float real,imag;}complex; void fft (complex *y, int n); float iobuffer[pts]; float x1[pts],x[pts]; int n,k,i; //short i; short buffercount=0; short flag =0; float y[128]; complex w[pts]; complex samples [pts]; main() { float sum=0.0; int n,k,i; for (i=0;i<pts;i++) { w[i].real=cos(2*pi*i/(pts*2.0)); w[i].imag= -sin (2*pi*i/(pts*2.0)); }
for (i=0;i<pts;i++) x[i] = sin (2*pi*5* i / pts); samples[i].real=0.0; samples[i].imag=0.0; for(n=0;n<pts;n++) { sum=0.0; for(k=0;k<pts-n;k++) { sum=sum+(x[k] * x[n+k]); } iobuffer[n]=sum; } for (i=0;i<pts;i++) { samples [i].real = iobuffer[i]; } for (i=0;i<pts;i++) { samples[i].real = iobuffer[i]; } for (i=0;i<pts;i++) samples[i].imag=0.0; fft (samples, pts); for (i=0;i<pts;i++)
{ x1[i] = sqrt (samples[i].real*samples[i].real+samples[i].imag*samples[i].imag); } } //fft.c: void fft(complex *y, int N) { complex temp1,temp2; int i,j,k; int upper_leg,lower_leg; int leg_diff; int num_stages =0; int index,step; i=1; do { num_stages +=1; i=i*2; } while(i!=N); leg_diff=N/2; step = (pts *2)/N; for (i=0; i<num_stages;i++) { index=0;
for(j=0; j<leg_diff;j++) { for (upper_leg=j; upper_leg<n;upper_leg = (2*leg_diff)) { lower_leg=upper_leg+leg_diff; temp1.real=(y[upper_leg]).real + (y[lower_leg]).real; temp1.imag=(y[upper_leg]).imag + (y[lower_leg]).imag; temp2.real=(y[upper_leg]).real - (y[lower_leg]).real; temp2.imag=(y[upper_leg]).imag - (y[lower_leg]).imag; (y[lower_leg]).real = temp2.real*(w[index]).real - temp2.imag * (w[index]).imag; (y[lower_leg]).imag = temp2.real*(w[index]).imag + temp2.imag * (w[index]).real; (y[upper_leg]).real = temp1.real; (y[upper_leg]).imag = temp1.imag; } index+=step; leg_diff =leg_diff/2; step *=2; } } j=0; for (i=1;i<(n-1);i++) { k=N/2; while (k<=j)
{ j=j-k; k=k/2; } j=j+k; if (i<j) { temp1.real = (y[j]).real; temp1.imag = (y[j]).imag; (y[j]).real = (y[i]).real; (y[j]).imag = (y[i]).imag; (y[i]).real = temp1.real; (y[i]).imag = temp1.imag; } } return; }
INPUT:
OUTPUT:

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JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

2. FREQUENCY RESPONSE OF ANALOG FILTERS

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

INPUT: enter passband ripple enter stopband ripple 0.15 30

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

2b) Program for the design of butter worth analogHPF:

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

INPUT: enter the first sequence[1 3 1 2] enter the second sequence[1 2 1 2]

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

6. PROGRAM FOR WINDOWING TECHNIQUES

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

subplot(2,2,4); plot(om/pi,m); title('BAND STOP FILTER') ylabel('gains in dB....'); xlabel('normalised frequency...');

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

7. SAMPLING OF A SINUSOIDAL SIGNALS

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

1. INTRODUCTION TO DSP PROCESSORS

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

2.CHITECTURE OF 6713 DSP PROCESSOR

Fig 2 1 BLOCK DIAGRAM OF TMS 320VC 6713

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

Central Processing Unit (CPU)

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

Arithmetic Logic Unit (ALU)

Fig 2 2 ALU UNIT

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

Fig 2 3 MULTIPLIER/ADDER UNIT

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

OUTPUT WAVE FORM:

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

ST.JOHNS COLLEGE OF ENGINEERING & TECHNOLOGY DSP LAB MANUAL

// set up twiddle constants in w

//Re component of twiddle constants ); //Img component of twiddle constants