What is a Signal? (Define a Signal). What is a System?

A SIGNAL is defined as any physical quantity that changes with time, distance, space, speed, position, pressure, temperature or any other independent variable. A SIGNAL consists of informations about the behavior or nature of some phenomenon. Eg:- ECG signal, Speech Signal, Image, x(t) = 10t, f(x,y) = 5x2+20xy+30y etc. A System is a physical device that performs an operations or processing on a signal. Eg:- Filter, Amplifier etc. What are the 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. Analog and digital signals. 5. Deterministic and Random signals 6. Periodic signal and Non-periodic signal 7. Symmetrical(even) and Anti-Symmetrical(odd) signal 8. 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 channel signal. If the signals are generated from multiple sensors or multiple sources or multiple signals are generated from same source called as Multi-channel signal. Example ECG signals. Multichannel signal will be the vector sum of signals generated from multiple sources. 2) Single Dimensional (1-D) and Multi-Dimensional signals (M-D) If the signal is a function of one independent variable it is called as single dimensional signal like speech signal and if the signal is a function of M independent variables then it is called as Multi-dimensional signals. Gray scale level of image or Intensity at particular pixel on black and white TV are examples of M-D signals. 3) Continuous time and Discrete time signals. Sl Continuous Time (CTS) Discrete time (DTS) No. 1 This signal can be defined at any time This signal can be defined only at certain instance & they can take all values in specific values of time. These time instance the continuous interval(a, b) where a need not be equidistant but in practice they can be - & b can be are usually takes at equally spaced intervals. 2 These are described by differential These are described by difference equation. equations. 3 This signal is denoted by x(t). These signals are denoted by x(n) or notation x(nT) can also be used. 4 The temperature recorded over an Microprocessors and computer based systems interval of time, AC power supply and uses discrete time signals. ECG waveforms are continuous time.

5) Analog and digital signal Sl Analog signal

Digital signal

No. 1

These are basically continuous time & These are basically discrete time signals & continuous amplitude signals. discrete amplitude signals. These signals are basically obtained by sampling & quantization process. ECG signals, Speech signal, Television All signal representation in computers and signal etc. All the signals generated from digital signal processors are digital. various sources in nature are analog.

Note: Digital signals (DISCRETE TIME & DISCRETE AMPLITUDE) are obtained by sampling the ANALOG signal at discrete instants of time, (obtaining DISCRETE TIME signals) and then by quantizing its values to a set of discrete values ( generating DISCRETE AMPLITUDE signals). Sampling process takes place on x axis at regular intervals & quantization process takes place along y axis. Quantization process is also called as rounding or truncating or approximation process. 6) Deterministic and Random signals Sl Deterministic signals No. 1 Deterministic signals can be represented or described 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. 3 The value of the deterministic signal can be evaluated at time (past, present or future) without certainty. 4 Example: Sine or exponential waveforms.

Random signals Random signals that cannot be represented or described by a mathematical equation or lookup table. Not Preferable. The random signals can be described with the help of their statistical properties. The value of the random signal can not be evaluated at any instant of time. Example:Noise signal or Speech signal

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 Non-Periodic signals. Sum of discrete time signal is periodic if its fundamental frequencies can be expressed as a ratio of two integers. N0= N1/N2 where N1, N2 are integer constants. 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) then signal is odd. x1(n)= cos(n) and x2(n)= sin(n) are good examples of even & odd signals respectively. Every discrete signal can be represented in terms of even & odd signals. X(n) signal can be written as ( ) ( ) ( ) ( ) ( )

Rearranging the above terms we have ( ) ( ) ( ) ( ) ( )

Thus X(n)= Xe(n) + Xo(n) Even component of discrete time signal is given by ( ) ( ) ( ) Odd component of discrete time signal is given by ( ) ( ) ( )

9) Energy signal and Power signal Discrete time signals are also classified as finite energy or finite average power signals. The energy of a discrete time signal x(n) is given by | ( )|

The average power for a discrete time signal x(n) is defined as | ( )|

If energy is finite and power is zero for x(n) then x(n) is an Energy signal. If power is finite and energy is infinite then x(n) is Power signal. There are some signals which are neither energy nor a power signal. Standard Discrete Signals 1) Unit sample signal (Unit impulse signal) (n) = 1 0 n=0 n=0

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

3) Unit ramp signal r(n) = n 0 n0 n<0

4) Exponential signal x(n) = a n = (re j ) n = r n e j n = r n (cos n + j sin n) 5) Sinusoidal waveform x(n) = A Sin wn

PROPERTIES OF DISCRETE TIME SIGNALS 1) Shifting : signal x(n) can be shifted in time. We can delay the sequence or advance the sequence. This is done by replacing integer n by n-k where k is integer. If k is positive signal is delayed in time by k samples (Arrow get shifted on left hand side) And if k is negative signal is advanced in time k samples (Arrow get shifted on right hand side)

X(n) = { 1, -1 , 0 , 4 , -2 , 4 , 0 ,} n=0 Delayed by 2 samples : X(n-2)= { 1, -1 , 0 , 4 , -2 , 4 , 0 ,} n=0 X(n+2) = { 1, -1 , 0 , 4 , -2 , 4 , 0 ,}

Advanced by 2 samples :

n=0 2) Folding / Reflection : It is folding 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 Folded signal: X(-n) = { 0 , 4 , -2 , 4 , 0 , -1 , 1} n=0 3) Addition : Given signals are x1(n) and x2(n), which produces output y(n) where y(n) = x1(n)+ x2(n). Adder generates the output sequence which is the sum of input sequences. Let x1(n)={1,3,2,1}; x2(n)={1,-2,3,2}; then y(n)= {2,1,5,3} 4) Amplitude Scaling: Amplitude scaling can be done by multiplying signal with some constant. Suppose original signal is x(n). Let A be a constant, then output signal is y(n) = A* x(n) let x(n)={1,2,1} and A=2; Then y(n)= {2,4,2} 5) Multiplication (Modulation) : Consider the two discrete signals x1(n) and x2(n). The resulting signal y(n) where y(n) = x1(n) * x2(n), is called the modulation of x1(n) and x2(n) and where y(n) in a discrete signal found by multiplying the sample values of x1(n) and x2(n) at every instant. Let x1(n)={1,2,-2,3} ; x2(n)={1,0.5,0.5,1} then y(n)={1,1,-1,3} What is correlation ? Ans : Correlation gives a measure of similarity between two data sequences. In this process, two signals are compared and the degree to which the two signals are similar is computed. What are the applications of Correlation ? Ans : Typical applications of correlation include speech processing, image processing and radar systems. In a radar system, the transmitted signal is correlated with the echo signal to locate the position of the target. Similarly, in speech processing systems, different waveforms are compared for voice recognition How many complex multiplications and additions are required to find DFT ? Ans : By direct DFT method (i) Complex Multiplications = N 2 (ii) Complex Additions = N*(N 1) By FFT method (i) Complex Multiplications (ii) Complex Additions = N log2 N What is the DFT of [n] ? Ans : DFT { [n] } = 1 What is the DFT of N pt signal u[n] ?

Ans : DFT {u[n] } = N [k] What is the DFT of 4 pt x[n] where x[n] = [n] + u[n] ? Ans : X[k] = 1+ 4 [k] = { 5, 1, 1, 1 } Find DFT of x[n] where x[n] = u[n] + 2 u[n-2] 3 u[n-4] Ans : Here x[n] = { 1, 1, 3, 3 } By DFT X[k] = {8, -2+2j, 2, -2-2j } What is the length of linearly convolved signals ? Ans : Length of linearly convolved signal is always equal to N = L + M 1 where L is length of first signal and M is length of second signal. FFT is faster than DFT . Justify. Ans : FFT produces fast results because calculations are reduced by decomposition technique. In FFT, N pt DFT is decomposed into two N/2 pt DFTs, N/2 pt DFT is decomposed into N/4 pt DFTs and so on Decomposition reduces calculations. FFT algorithms are implemented using parallel processing techniques. Because calculations are done in parallel, FFT produces fast results. Complex Multiplications: N 2 for DFT N/2 log2 N for FFT What are the applications of FFT. ? Ans : (i) Linear Filtering i.e. to find output of digital filter for any given input sequence x[n]. (ii) Spectral Analysis i.e. to find magnitude spectrum and phase spectrum (iii) Circular Correlation ie to find degree of similarity between two signals. Describe the relation between DFT and DTFT. Ans : DFT is frequency sampling of DTFT. When DTFT is sampled in frequency domain with frequency spacing of =2/N we get DFT coefficients. i.e. [ ] ( )| Comparison between Digital and Analog signal processing Digital Signal Processing 1. More Versatile (Digital Systems can be reprogrammed for other applications & can be ported to different hardware) 2. More Repeatability 3. More accuracy 4. Such signals can be easily stored 5. Mathematical processing can be easily implemented 6. Upgradation can be done easily 7. Requires more power consumption Analog Signal Processing Less Versatile

Less Repeatability Less accuracy It is difficult to store analog signals It is difficult to implement mathematical processing algorithm. Upgradation is difficult. Requires less power consumption

Analysis equation Discrete Time Fourier Transform (DTFT) Discrete Fourier Transform (DFT) Z-Transform ( ) [ ]

Synthesis equation [ ] [ ] [ ] ( ) [ ] ( )

[ ] ( )

[ ] [ ]

Sl No 1

Fourier Transform (FT) FT x() is the continuous function of x(n). The range of is from - to or 0 to 2.

Discrete Fourier Transform (DFT) DFT x(k) is calculated only at discrete values of . Thus DFT is discrete in nature. Sampling is done at N equally spaced points over period 0 to 2. Thus DFT is sampled version of FT. DFT is given by equation, ( ) ( )

FT is given by equation ( ) ( )

4 5

FT equations are applicable to most of infinite sequences. In DSP processors & computers applications of FT are limited because x() is continuous function of .

DFT equations are applicable to causal, finite duration sequences In DSP processors and computers DFTs are mostly used. APPLICATION a) Spectrum Analysis b) Filter Design

Sl No 1

Linear Convolution

Circular Convolution

The input sequence need not be periodic. Atleast one of the input sequence should be Output sequence is non periodic. periodic or should be periodically extended. The output sequence is periodic. Linear Convolution is given by the Circular Convolution is calculated as equation y(n) = x(n) * h(n) & calculated as N-1 y(m) = x1(n). x2((m-n))N y(n) = x (k) h(n k ) n=0 k= - Linear Convolution of two signals returns Circular convolution returns a sequence that L+N-1 elements where L and N are has same number of elements that of number of elements in the two sequences. maximum of two signals. Analog Filter Analog filters are used for filtering analog signals. Analog filters are designed with various Digital Filter Digital filters are used for filtering digital sequences. Digital Filters are designed with digital

Sl No 1 2

components like resistor, inductor and capacitor Analog filters less accurate & because of component tolerance of active components & more sensitive to environmental changes. Less flexible

5 6

Filter representation is in terms of system components. An analog filter can only be changed by redesigning the filter circuit.

hardware like FF, counters shift registers, ALU and softwares like C or assembly language. Digital filters are less sensitive to the environmental changes, noise and disturbances. Thus periodic calibration can be avoided. Also they are extremely stable. These are most flexible as software programs & control programs can be easily modified. Several input signals can be filtered by one digital filter. Digital filters are represented by the difference equation. A digital filter is programmable, i.e. its operation is determined by a program stored in the processor's memory. This means the digital filter can easily be changed without affecting the circuitry (hardware).

Sl No 1 2

Finite Impulse Response (FIR)

Infinite Impulse Response (IIR)

FIR has an impulse response that is zero IIR has an impulse response on infinite time outside of some finite time interval. interval. Convolution formula changes to Convolution formula changes to ( ) ( ) ( ) ( ) ( ) ( )

For causal FIR systems limits changes to 0 For causal IIR systems limits changes to 0 to . to M. ( ) 3 ( ) ( ) ( ) ( ) ( )

4 5

The FIR system has limited span which views only most recent M input signal samples forming output called as Windowing. FIR has limited or finite memory requirements. Realization of FIR system is generally based on Convolution Sum Method. Impulse Invariance In this method IIR filters are designed having a unit sample response h(n) that is sampled version of the impulse response of the analog filter. Mapping is many to one. (In this method small value of T is selected to minimize the effect of aliasing.)

The IIR system has unlimited span.

IIR System requires infinite memory. Realization of IIR system is generally based on Difference Method. Bilinear Transformation This method of IIR filters design is based on the trapezoidal formula for numerical integration. Mapping is one to one. (The bilinear transformation is a conformal mapping that transforms the j axis into the unit circle in the z plane only once, thus avoiding aliasing of frequency components.) For designing of LPF, HPF and almost all

Sl No 1

They are generally used for low frequencies

like design of IIR LPF and a limited class of bandpass filter Frequency relationship is linear.

Poles are transferred by using the relation,

types of Band pass and band stop filters this method is used. Frequency relationship is non-linear. Frequency warping or frequency compression is due to non-linearity. All poles are mapped using the equation, [
-

].