Manuel S.

Enverga University Foundation College of Engineering and Technical Department

Research on Introduction of Digital Signal Processing and Other Related Topics Digital Signal Processing

Submitted By: Jefferson S. Berbano Submitted To: Engr. Sherwin Lagrama Date: June 29, 2011

SIGNALS AND SYSTEM

INTRODUCTION A SIGNAL is defined as any physical quantity that changes with time, distance, s peed, position, pressure, temperature or some other quantity. A SIGNAL is physic al quantity that consists of many sinusoidal of different amplitudes and frequen cies. Ex x(t) = 10t X(t) = 5x2+20xy+30y A System is a physical device that performs an operations or processing on a sig nal. Ex Filter or Amplifier. CLASSIFICATION OF SIGNAL PROCESSING 1) ASP (Analog signal Processing) : If the input signal given to the system is a nalog then system does analog signal processing. Ex Resistor, capacitor or Indu ctor, OP-AMP etc. Analog Input Analog Output

2) DSP (Digital signal Processing) : If the input signal given to the system is digital then system does digital signal processing. Ex Digital Computer, Digita l Logic Circuits etc. The devices called as ADC (analog to digital Converter) co nverts Analog signal into digital and DAC (Digital to Analog Converter) does vi ce-versa. Analog Analog signal

signal Most of the signals generated are analog in nature. Hence these signals are conv erted to digital form by the analog to digital converter. Thus AD Converter gene rates an array of samples and gives it to the digital signal processor. This ar ray of samples or sequence of samples is the digital equivalent of input analog signal. The DSP performs signal processing operations like filtering, multiplica tion, transformation or amplification etc operations over this digital signals. The digital output signal from the DSP is given to the DAC. ADVANTAGES OF DSP OVER ASP 1. Physical size of analog systems are quite large while digital processors are more compact and light in weight. 2. Analog systems are less accurate because of component tolerance ex R, L, C and active components. Digital components are less sensitive to the environme ntal changes, noise and disturbances. 3. Digital system are most flexible as software programs & control programs can be easily modified. 4. Digital signal can be stores on digital hard disk, floppy disk or magnet ic tapes. Hence becomes transportable. Thus easy and lasting storage capacity. 5. Digital processing can be done offline. 6. Mathematical signal processing algorithm can be routinely implemented on digital signal processing systems. Digital controllers are capable of performin g complex computation with constant accuracy at high speed. 7. Digital signal processing systems are upgradeable since that are softwar e controlled. 8. Possibility of sharing DSP processor between several tasks. 9. The cost of microprocessors, controllers and DSP processors are continuo usly going down. For some complex control functions, it is not practically feasi ble to construct analog controllers. 10. Single chip microprocessors, controllers and DSP processors are more ve rsatile and powerful. Disadvantages Of DSP over ASP 1. Additional complexity (A/D & D/A Converters) 2. Limit in frequency. High speed AD converters are difficult to achieve in practice. In high frequency applications DSP are not preferred. CLASSIFICATION OF SIGNALS 1. Single channel and Multi-channel signals 2. Single dimensional and Multi-dimensional signals 3. Continuous time and Discrete time signals. 4. Continuous valued and discrete valued signals. 5. Analog and digital signals. 6. Deterministic and Random signals 7. Periodic signal and Non-periodic signal 8. Symmetrical(even) and Anti-Symmetrical(odd) signal 9. Energy and Power signal 1) Single channel and Multi-channel signals If signal is generated from single sensor or source it is called as single chann el signal. If the signals are generated from multiple sensors or multiple source s or multiple signals are generated from same source called as Multi-channel sig nal. Example ECG signals. Multi-channel signal will be the vector sum of signals generated from multiple sources. 2) Single Dimensional (1-D) and Multi-Dimensional signals (M-D) If signal is a function of one independent variable it is called as single dimen

sional signal like speech signal and if signal is function of M independent vari ables called as Multi-dimensional signals. Gray scale level of image or Intensi ty at particular pixel on black and white TV are examples of M-D signals. 3) Continuous time and Discrete time signals. Sr No Continuous Time (CTS) Discrete time (DTS) 1 This signal can be defined at any time instance & they can take all valu es in the continuous interval(a, b) where a can be - & b can be This signal can be defined only at certain specific values of time. These time instance need not be equidistant but in practice they are usually takes at equally spaced interv als. 2 These are described by differential equations. These are described by d ifference equation. 3 This signal is denoted by x(t). These signals are denoted by x(n) or not ation x(nT) can also be used. 4 The speed control of a dc motor using a techogenerator feedback or Sine or exponential waveforms. Microprocessors and computer based systems uses discrete time signals. 4) Continuous valued and Discrete Valued signals. Sr No Continuous Valued Discrete Valued 1 If a signal takes on all possible values on a finite or infinite range, it is said to be continuous valued signal. If signal takes values from a fi nite set of possible values, it is said to be discrete valued signal. 2 Continuous Valued and continuous time signals are basically analog signa ls. Discrete time signal with set of discrete amplitude are called digital s ignal. 5) Analog and digital signal Sr No Analog signal Digital signal 1 These are basically continuous time & continuous amplitude signals. These are basically discrete time signals & discrete amplitude signals. These si gnals are basically obtained by sampling & quantization process. 2 ECG signals, Speech signal, Television signal etc. All the signals gener ated from various sources in nature are analog. All signal representation in com puters and digital signal processors are digital. Note: Digital signals (DISCRETE TIME & DISCRETE AMPLITUDE) are obtained by sampl ing the ANALOG signal at discrete instants of time, obtaining DISCRETE TIME sign als and then by quantizing its values to a set of discrete values & thus generat ing DISCRETE AMPLITUDE signals. Sampling process takes place on x axis at regular intervals & quantization proce ss takes place along y axis. Quantization process is also called as rounding or truncating or approximation process. 6) Deterministic and Random signals Sr No Deterministic signals Random signals 1 Deterministic signals can be represented or described by a mathematical equation or lookup table. Random signals that cannot be represented or des cribed by a mathematical equation or lookup table. 2 Deterministic signals are preferable because for analysis and processing of signals we can use mathematical model of the signal. Not Preferable. The random signals can be described with the help of their statistical propertie s. 3 The value of the deterministic signal can be evaluated at time (past, pr esent or future) without certainty. The value of the random signal can not b e evaluated at any instant of time.

4 ignal

Example Sine or exponential waveforms. Example Noise signal or Speech s

7) Periodic signal and Non-Periodic signal The signal x(n) is said to be periodic if x(n+N)= x(n) for all n where N is the fundamental period of the signal. If the signal does not satisfy above property called as No n-Periodic signals. Discrete time signal is periodic if its frequency can be expressed as a ratio of two integers. f= k/N where k is integer constant. a) cos (0.01 n) per cycle. b) cos (3 n) c) sin(3n) d) cos(n/8) cos( n/8) Periodic N=200 samples Periodic N=2 samples Non-periodic Non-Periodic

8) Symmetrical(Even) and Anti-Symmetrical(odd) signal A signal is called as symmetrical(even) if x(n) = x(-n) and if x(-n) = -x(n) the n signal is odd. X1(n)= cos(n) and x2(n)= sin(n) are good examples of even & odd s ignals respectively. Every discrete signal can be represented in terms of even & odd signals. X(n) signal can be ritten as X(n)=

Rearranging the above terms e have X(n)= +

Thus X(n)= Xe(n) + Xo(n) Even component of discrete time signal is given by Xe(n) = Odd component of discrete time signal is given by Xo(n) = Test hether the follo ing CT aveforms is periodic or not. If periodic find out the fundamental period. a) 2 sin(2/3)t + 4 cos (1/2)t + 5 cos((1/3)t Ans: Period of x (t)= 12 b) a cos(t 2) + b sin(t/4) Ans: Non-Periodi c a) Find out the even and odd parts of the discrete signal x(n)={2,4,3,2,1} b) Find out the even and odd parts of the discrete signal x(n)={2,2,2,2}

9) Energy signal and Po er signal Discrete time signals are also classified as finite energy or finite average po er signals. The energy of a discrete time signal x(n) is given by E= x2 (n) n=- The average po er for a discrete time signal x(n) is defined as Lim 1 P = N 2N+1 x2 (n)

n=- If Energy is finite and po er is zero for x(n) then x(n) is an energy signal. If po er is finite and energy is infinite then x(n) is po er signal. There are som e signals hich are neither energy nor a po er signal. a) Find the po er and energy of u(n) unit step function. b) Find the po er and energy of r(n) unit ramp function. c) Find the po er and energy of an u(n). DISCRETE TIME SIGNALS AND SYSTEM There are three ays to represent discrete time signals. 1) 2) 0 3) n 4 0 Functional Representation 4 x(n)= else here for n=1,3 -2 for n =2

Tabular method of representation -3 5 x(n) 0 0 -2 0 -1 0 0 0 1 4 2 -2 3 4

4) Sequence Representation X(n) = { 0 , 4 , -2 , 4 , 0 ,}

n=0 STANDARD SIGNAL SEQUENCES 1) Unit sample signal (Unit impulse signal) (n) = 1 0 n=0 n=0 i.e (n)=

{1} 2) Unit step signal u(n) = 1 0 n 0 n0 n<0 n0 n<0

3) Unit ramp signal ur (n) =

4) Exponential signal x(n) = a n = (re j ) n = r n e j n = r n (cos n + j sin n) 5) Sinusoi al waveform x(n) = A Sin wn PROPERTIES OF DISCRETE TIME SIGNALS 1) Shifting : signal x(n) can be shifte in time. We can elay the sequence or a vance the sequence. This is one by replacing integer n by n-k where k is integ er. If k is positive signal is elaye in time by k samples (Arrow get shifte o n left han si e) An if k is negative signal is a vance in time k samples (Arr ow get shifte on right han si e)

X(n) = { 1, -1 , 0 , 4 , -2 , 4 , 0 ,} n=0 X(n-2)= { 1, -1 , 0 , 4 , -2 , 4 , 0 ,} n=0 X(n+2) = { 1, -1 , 0 , 4 , -2 , 4 , 0 ,}

Delaye by 2 samples :

A vance by 2 samples :

n=0 2) Fol ing / Reflection : It is fol ing of signal about time origin n=0. In this case replace n by n. Original signal: X(n) = { 1, -1 , 0 , 4 , -2 , 4 , 0} n=0 Fol e signal: X(-n) = { 0 , 4 , -2 , 4 , 0 , -1 , 1} n=0 3) A ition : Given signals are x1(n) an x2(n), which pro uces output y(n) wher e y(n) = x1(n)+ x2(n). A er generates the output sequence which is the sum of i nput sequences. 4) Scaling: Amplitu e scaling can be one by multiplying signal with some consta nt. Suppose original signal is x(n). Then output signal is A x(n) 5) Multiplication : The pro uct of two signals is efine as y(n) = x1(n) * x2(n ). SYMBOLS USED IN DISCRETE TIME SYSTEM 1. Unit elay

x(n) 2. Unit a vance x(n) (n+1) 3. A ition x1(n) ) x2(n) 4. Multiplication x1(n)

) x2(n) 5. Scaling (constant multiplier) A x(n) y(n) = A x(n)

y(n) = x(n-1)

y(n) = x

y(n) =x1(n)+x2(n

y(n) =x1(n)*x2(n

CLASSIFICATION OF DISCRETE TIME SYSTEMS 1) STATIC v/s DYNAMIC Sr No STATIC DYNAMIC (Dynamicity property) 1 Static systems are those systems whose output at any instance of time e pen s at most on input sample at same time. Dynamic systems output epen s u pon past or future samples of input. 2 Static systems are memory less systems. They have memories for memorize all samples. It is very easy to fin out that given system is static or ynamic. Just check t hat output of the system solely epen s upon present input only, not epen ent u pon past or future. Sr No System [y(n)] Static / Dynamic 1 x(n) Static 2 A(n-2) Dynamic 3 X2(n) Static 4 X(n2) Dynamic 5 n x(n) + x2(n) Static 6 X(n)+ x(n-2) +x(n+2) Dynamic 2) TIME INVARIANT v/s TIME VARIANT SYSTEMS Sr No TIME INVARIANT (TIV) / SHIFT INVARIANT TIME VARIANT SYSTEMS / SHIFT VARIANT SYSTEMS (Shift Invariance property) 1 A System is time invariant if its input output characteristic o not cha nge with shift of time. A System is time variant if its input output characteris tic changes with time. 2 Linear TIV systems can be uniquely characterize by Impulse response, fr equency response or transfer function. No Mathematical analysis can be performe . 3 a. Thermal Noise in Electronic components b. Printing ocuments by a printer a. Rainfall per month b. Noise Effect

It is very easy to fin out that given system is Shift Invariant or Shift Varian t. Suppose if the system pro uces output y(n) by taking input x(n) x(n) y(n) If we elay same input by k units x(n-k) an apply it to same systems, the syste m pro uces output y(n-k) x(n-k) y(n-k) LINEAR v/s NON-LINEAR SYSTEMS LINEAR NON-LINEAR (Linearity Property) 1 A System is linear if it satisfies superposition theorem. A System is Non-linear if it oes not satisfies superposition theorem. 2 Let x1(n) an x2(n) are two input sequences, then the system is sai to be linear if an only if T[a1x1(n) + a2x2(n)]=a1T[x1(n)]+a2T[x2(n)] a1 x1(n) y(n)= T[a1x1[n] 3) Sr No

+ a2x2(n) ] x2(n) a2 x1(n) 1x1(n)+a2x2(n)] x2(n) a2 a1 y(n)=T[a

hence T [ a1 x1(n) + a2 x2(n) ] = T [ a1 x1(n) ] + T [ a2 x2(n) ] It is very easy to fin out that given system is Linear or Non-Linear. Response to the system to the sum of signal = sum of in ivi ual responses of the system. Sr No 1 2 3 4 5 6 4) Sr No System y(n) Linear or Non-Linear ex(n) Non-Linear x2 (n) Non-Linear m x(n) + c Non-Linear cos [ x(n) ] Non-Linear X(-n) Linear Log 10 ( x(n) ) Non-Linear

CAUSAL v/s NON CAUSAL SYSTEMS CAUSAL NON-CAUSAL (Causality Property) 1 A System is causal if output of system at any time depends only past and present inputs. A System is Non causal if output of system at any time d epends on future inputs. 2 In Causal systems the output is the function of x(n), x(n-1), x(n-2).. an d so on. In Non-Causal System the output is the function of future inputs also. X(n+1) x(n+2) .. and so on 3 Example Real time DSP Systems Offline Systems It is very easy to find out that given system is causal or non-causal. Just chec k that output of the system depends upon present or past inputs only, not depend ent upon future. Sr No System [y(n)] Causal /Non-Causal 1 x(n) + x(n-3) Causal 2 X(n) Causal 3 X(n) + x(n+3) Non-Causal 4 2 x(n) Causal 5 X(2n) Non-Causal 6 X(n)+ x(n-2) +x(n+2) Non-Causal 5) Sr No STABLE v/s UNSTABLE SYSTEMS STABLE UNSTABLE (Stability Property) 1 A System is BIBO stable if every bounded input produces a bounded output . A System is unstable if any bounded input produces a unbounded output. 2 The input x(n) is said to bounded if there exists some finite number Mx such that x(n) Mx < The output y(n) is said to bounded if there exists some finite number My such th at y(n) My <

STABILITY FOR LTI SYSTEM It is very easy to find out that given system is stable or unstable. Just check that by providing input signal check that output should not rise to . The conditi on for stability is given by h( k ) < k= - Sr No System [y(n)] Stable / Unstable 1 Cos [ x(n) ] Stable 2 x(-n+2) Stable 3 x(n) Stable 4 x(n) u(n) Stable 5 X(n) + n x(n+1) Unstable ANALYSIS OF DISCRETE LINEAR TIME INVARIANT (LTI/LSI) SYSTEM 1) CONVOLUTION SUM METHOD 2) DIFFERENCE EQUATION LINEAR CONVOLUTION SUM METHOD 1. This method is po erful analysis tool for studying LSI Systems. 2. In this method e decompose input signal into sum of elementary signal. No the elementary input signals are taken into account and individually given t o the system. No using linearity property hatever output response e get for d ecomposed input signal, e simply add it & this ill provide us total response o f the system to any given input signal. 3. Convolution involves folding, shifting, multiplication and summation ope rations. 4. If there are M number of samples in x(n) and N number of samples in h(n) then the maximum number of samples in y(n) is equals to M+n-1. Linear Convolution states that y(n) = x(n) * h(n) y(n) = x (k) h(n k ) = x (k) h[ -(k-n) ] k= - k= - Example 1: h(n) = { 1 , 2 , 1, -1 } & x(n) = { 1, 2, 3, 1 } Find y(n) METHOD 1: GRAPHICAL REPRESENTATION Step 1) Find the value of n = nx+ nh = -1 (Starting Index of x(n)+ starting ind ex of h(n)) Step 2) y(n)= { y(-1) , y(0) , y(1), y(2), .} It goes up to length(xn)+ length(yn ) -1. i.e n=-1 y(-1) = x(k) * h(-1-k) n=0 y(0) = x(k) * h(0-k) n=1 y(1) = x(k) * h(1-k) . ANSWER : y(n) ={1, 4, 8, 8, 3, -2, -1 } METHOD 2: MATHEMATICAL FORMULA Use Convolution formula y(n) = x (k) h(n k ) k= - k= 0 to 3 (start index to end index of x(n)) y(n) = x(0) h(n) + x(1) h(n-1) + x(2) h(n-2) + x(3) h(n-3) METHOD 3: VECTOR FORM (TABULATION METHOD) X(n)= {x1,x2,x3} & h(n) ={ h1,h2,h3}

X1

x2

x3 h1 h2 h3

y(-1) = h1 x1 y(0) = h2 x1 + h1 x2 y(1) = h1 x3 + h2x2 + h3 x1 METHOD 4: SIMPLE MULTIPLICATION FORM X(n)= {x1,x2,x3} & x1 x2 x3 y(n) = y1 y2 y3 PROPERTIES OF LINEAR CONVOLUTION x(n) = Excitation Input signal y(n) = Output Response h(n) = Unit sample response 1. X(n) Response = y(n) = x(n) *h(n) h(n) Response = y(n) = h(n) * x(n) 2. Associate La : (Associative Property of Convolution) [ x(n) * h1(n) ] * h2(n) = x(n) * [ h1(n) * h2(n) ] Commutative La : (Commutative Property of Convolution) x(n) * h(n) = h(n) * x(n) h(n) ={ h1,h2,h3}

X(n) Response

X(n)

3 Distribute La : (Distributive property of convolution) x(n) * [ h1(n) + h2(n) ] = x(n) * h1(n) + x(n) * h2(n) CAUSALITY OF LSI SYSTEM The output of causal system depends upon the present and past inputs. The output of the causal system at n= n0 depends only upon inputs x(n) for n n0. The linear convolution is given as y(n) = h(k) x(nk) k=- At n= n0 ,the output y(n0) ill be

Response

y(n0) = h(k) x(n0k) k=- Rearranging the above terms... - y(n0) = h(k) x(n0k) + h(k) x(n0k) k=0 k=-1 The output of causal system at n= n0 depends upon the inputs for n< n0 Hence h(-1)=h(-2)=h(-3)=0 Thus LSI system is causal if and only if h(n) =0 for n<0 This is the necessary and sufficient condition for causality of the system. Linear convolution of the causal LSI system is given by n y(n) = x (k) h(n k ) k=0 STABILITY FOR LSI SYSTEM A System is said to be stable if every bounded input produces a bounded output. The input x(n) is said to bounded if there exists some finite number Mx such tha t x(n) Mx < . The output y(n) is said to bounded if there exists some finite n umber My such that y(n) My < . Linear convolution is given by y(n) = x (k) h(n k ) k=- Taking the absolute value of both sides y(n) = h(k) x(n-k) k=- The absolute values of total sum is al ays less than or equal to sum of the abso lute values of individually terms. Hence y(n) h(k) x(nk) k=- y(n) h(k) x(nk) k=- The input x(n) is said to bounded if there exists some finite number Mx such tha t x(n) Mx < . Hence bounded input x(n) produces bounded output y(n) in the LSI system only if h(k) < k=- With this condition satisfied, the system ill be stable. The above equation sta tes that the LSI system is stable if its unit sample response is absolutely summ able. This is necessary and sufficient condition for the stability of LSI system . SELF-STUDY: Exercise No. 1 Q1) Sho that the discrete time signal is periodic only if its frequency is expr essed as the ratio of t o integers. Q2) Sho that the frequency range for discrete time sinusoidal signal is - to rad ians/sample or - cycles/sample to cycles/sample. Q3) Prove (n)= u(n)= u(n-1). n Q4) Prove u(n)= (k) k=- Q5) Prove u(n)= (n-k) k=0 Q6) Prove that every iscrete sinusoi al signal can be expresse in terms of wei ghte unit impulse.

Q7) Prove the Linear Convolution theorem. necessary to establish similarity between one set of a we woul like to correlate two processes or ata. Corre to convolution, because the correlation is essentially sequences in which one of the sequences has been revers

Applications are in 1) Images processing for robotic vision or remote sensing by satellite in which ata from ifferent image is compare 2) In ra ar an sonar systems for range an position fin ing in which transmitte an reflecte waveforms are compare . 3) Correlation is also use in etection an i entifying of signals in noise. 4) Computation of average power in waveforms. 5) I entification of binary co ewor in pulse co e mo ulation system. DIFFERENCE BETWEEN LINEAR CONVOLUTION AND CORRELATION Sr No Linear Convolution Correlation 1 In case of convolution two signal sequences input signal an impulse res ponse given by the same system is calculate In case of Correlation, two sign al sequences are just compare . 2 Our main aim is to calculate the response given by the system. Our main aim is to measure the egree to which two signals are similar an thus to extra ct some information that epen s to a large extent on the application 3 Linear Convolution is given by the equation y(n) = x(n) * h(n) & calcula te as y(n) = x (k) h(n k ) k= - Receive signal sequence is given as Y(n) = x(n-D) + (n) Where = Attenu tion F ctor D= Del y (n) = Noise sign l 4 Line r convolution is commut tive Not commut tive. TYPES OF CORRELATION Under Correl tion there re t o cl sses. 1) CROSS CORRELATION: When the correl tion of t o different sequences x(n) nd y(n) is performed it is c lled s Cross correl tion. Cross-correl tion of x( n) nd y(n) is rxy(l) hich c n be m them tic lly expressed s rxy(l) = x (n) y(n l ) n= - OR rxy(l) = x (n + l) y(n) n= -

2) AUTO CORRELATION: In Auto-correl tion lf, hich c n be m them tic lly expressed s rxx(l) = x (n) x(n l ) n= - OR rxx(l) = x (n + l) x(n)

e correl te sign l x(n)

ith itse

CORRELATION: It is frequently ta an another. It means lation is closely relate convolution of two ata e .

n= - PROPERTIES OF CORRELATION 1) The cross-correl tion is not commut tive. rxy(l) = ryx(-l) 2) The cross-correl tion is equiv lent to convolution of one sequence ith folde d version of nother sequence. rxy(l) = x(l) * y(-l). 3) The utocorrel tion sequence is n even function. rxx(l) = rxx(-l) Ex mples: Q) Determine cross-correl tion sequence x(n)={2, -1, 3, 7,1,2, -3} & y(n)={1, -1, 2, -2, 4, 1, -2 ,5} Ans er: rxy(l) = {10, -9, 19, 36, -14, 33, 0,7, 13, -18, 16, -7, 5, -3} Q) Determine utocorrel tion sequence x(n)={1, 2, 1, 1} Ans er: rxx(l) = {1, 3, 5, 7, 5, 3, 1} A/D CONVERSION BASIC BLOCK DIAGRAM OF A/D CONVERTER An log sign l X (t) Digit l sign l x(n)

SAMPLING THEOREM It is the process of converting continuous time sign l into discrete time sign l by t king s mples of the continuous time sign l t discrete time inst nts. X[n]= X (t) here t= nTs = n/Fs .(1)

When s mpling t r te of fs s mples/sec, if k is ny positive or neg tive inte ger, e c nnot distinguish bet een the s mples v lues of f Hz nd sine ve o f (f + kfs) Hz. Thus (f + kfs) ve is li s or im ge of f ve.

Ex mple: C se 1: i.e t= n/Fs C se 2: i.e t= n/Fs

X1(t) = cos 2 (10) t x1[n]= cos 2(n/4)= cos (/2)n X1(t) = cos 2 (50) t

Fs= 40 Hz

Fs= 40 Hz

x1[n]= cos 2(5n/4)= cos 2( 1+

)n

Thus S mpling Theorem sign l is Fm x nd the sign l ex ctly recovered from its s r te of s mpling. The im ging ncy is fs/2. If the frequency ed properly.

st tes th t if the highest frequency in n n log is s mpled t the r te fs > 2Fm x then x(t) c n be mple v lues. This s mpling r te is c lled Nyquist or li sing st rts fter Fs/2 hence folding freque is less th n or equ l to 1/2 it ill be represent

sign l

sign l

Discrete time

Qu ntized

cos (/2)n t the

Ex mple: = 1 Hz

x[n] = 5(0.9)n u(n)

here 0 <n <

&

N [n] Xq [n] Rounding Xq [n] Trunc ting eq [n] 0 5 5.0 5.0 0 1 4.5 4.5 4.5 0 2 4.05 4.0 4.0 -0.05 3 3.645 3.6 3.6 -0.045 4 3.2805 3.2 3.3 0.0195 Qu ntiz tion Step/Resolution : The difference bet een the t o qu ntiz tion level s is c lled qu ntiz tion step. It is given by = XM x xMin / L-1 here L indic te s Number of qu ntiz tion levels. CODING/ENCODING E ch qu ntiz tion level is ssigned unique bin ry code. In the encoding oper t ion, the qu ntiz tion s mple v lue is converted to the bin ry equiv lent of th t qu ntiz tion level. If 16 qu ntiz tion levels re present, 4 bits re required. Thus bits required in the coder is the sm llest integer gre ter th n or equ l t o Log2 L. i.e b= Log2 L Thus S mpling frequency is c lcul ted s fs=Bit r te / b. ANTI-ALIASING FILTER When processing the n log sign l using DSP system, it is s mpled t some r te d epending upon the b nd idth. For ex mple if speech sign l is to be processed the frequencies upon 3khz c n be used. Hence the s mpling r te of 6khz c n be used. But the speech sign l lso cont ins some frequency components more th n 3khz. H ence s mpling r te of 6khz ill introduce li sing. Hence sign l should be b n d limited to void li sing. The sign l c n be b nd limited by p ssing it through filter (LPF) hic h blocks or ttenu tes ll the frequency components outside the specific b nd id th. Hence c lled s Anti li sing filter or pre-filter. (Block Di gr m) SAMPLE-AND-HOLD CIRCUIT: The s mpling of n n logue continuous-time sign l is norm lly implemented using device c lled n n logue-to- digit l converter (A/D). The continuous-time si gn l is first p ssed through device c lled s mple- nd-hold (S/H) hose func tion is to me sure the input sign l v lue t the clock inst nt nd hold it fixed for time interv l long enough for the A/D oper tion to complete. An logue-to-digit l conversion is potenti lly slo oper tion, nd v ri tion of the input volt ge during the conversion m y disrupt the oper tion of the converter. The S/H prevents such disruption by ke eping the input volt ge const nt during the conversion. This is schem tic lly il lustr ted by Figure.

After continuous-time sign l h s been through the A/D converter, the qu ntized output m y differ from the input v lue. The m ximum possible output v lue fter

QUANTIZATION The process of converting discrete time continuous mplitude sign l into it l sign l by expressing e ch s mple v lue s finite number of digits is ed qu ntiz tion. The error introduced in representing the continuous v lues l by finite set of discrete v lue levels is c lled qu ntiz tion error or tiz tion noise.

dig c ll sign qu n fs

Thus the frequency 50 Hz, 90 Hz , 130 Hz re li s of the frequency 10 Hz s mpling r te of 40 s mples/sec

n log sinusoid l re s mpled ith the fs=40Hz. Find out c ls nd comment on them

Q) Sign l x1(t)=3 cos 600t+ 2cos800t. The link is oper ted t 10000 bits/sec nd e ch input s mple is qu ntized into 1024 different levels. Determine Nyquist r te , s mpling frequency, folding frequency & resolution. DIFFERENCE EQUATION Sr No Finite Impulse Response (FIR) Infinite Impulse Response (IIR) 1 FIR h s n impulse response th t is zero outside of some finite time int erv l. IIR h s n impulse response on infinite time interv l. 2 Convolution formul ch nges to M y(n) = x (k) h(n k ) n= -M For c us l FIR systems limits ch nges to 0 to M. Convolution formul ch n ges to y(n) = x (k) h(n k ) n= - For c us l IIR systems limits ch nges to 0 to . 3 The FIR system h s limited sp n hich vie s only most recent M input sig n l s mples forming output c lled s Windo ing. The IIR system h s unlimited sp n. 4 FIR h s limited or finite memory requirements. IIR System requires infi nite memory. 5 Re liz tion of FIR system is gener lly b sed on Convolution Sum Method. Re liz tion of IIR system is gener lly b sed on Difference Method. Discrete time systems h s one more type of cl ssific tion. 1. Recursive Systems 2. Non-Recursive Systems Sr No Recursive Systems Non-Recursive systems 1 In Recursive systems, the output depends upon p st, present, future v lu e of inputs s ell s p st output. In Non-Recursive systems, the output dep ends only upon p st, present or future v lues of inputs. 2 Recursive Systems h s feedb ck from output to input. No Feedb ck.

Q) Sign l x1(t)=10cos2(1000)t+ 5 cos2(5000)t. Determine Nyquist r te. If the sign l is s mpled t 4khz ill the sign l be recovered from its s mples.

Q) C lcul te Nyquist R 1) x(t)= 4 cos 50 t + 2) x(t)= 2 cos 2000t+ 3) x(t)= 4 cos 100t z Q) The follo ing four orresponding time sign X1(t)= cos 2(10)t X2(t)= cos 2(50)t X3(t)= cos 2(90)t X4(t)= cos 2(130)t

te for the n log sign l x(t) 8 sin 300t cos 100t 3 sin 6000t + 8 cos 12000t

Fn=300 Hz Fn=12KHz Fn=100 H

the qu ntiz tion process could be up to h lf the qu ntiz tion level q bove or q belo the ide l output v lue. This devi tion from the ide l output v lue is c lled the qu ntiz tion error. In order to reduce this effect, e incre ses the nu mber of bits.

First order Difference Equ tion

here

y(n) = Output Response of the recursive system x(n) = Input sign l = Sc ling f ctor y(n-1) = Unit del y to output.

No e ill st rt t n=0 n=0 y(0) = x(0) + y(-1) .(1) n=1 y(1) = x(1) + y(0) .(2) = x(1) + [ x(0) + y(-1) ] = 2 y(-1) + x(0) + x(1) .(3) Hence n

Zero st te response (Forced response) : Consider initi l condition re zero. (Sy stem is rel xed t time n=0) i.e y(-1) =0 Zero Input response (N tur l response) : No input is forced s system is in nonrel xed initi l condition. i.e y(-1) != 0 Tot l response is the sum of zero st te response nd zero input response. Q) Determine zero input response for y(n) 3y(n-1) 4y(n-2)=0; (Initi l Conditions re y(-1)=5 & y(-2)= 10) Ans er: y(n)= 7 (-1)n + 48 (4)n

1) 2)

The first p rt (A) is response depending upon initi l condition. The second P rt (B) is the response of the system to n input sign l.

y(n) =

n+1 y(-1) +

k x (n -k) k= 0

y(n) = x(n) +

y(n-1)

Ex mples y(n) = x(n) + y(n-2)

Y(n) = x(n) + x(n-1)

n 0