3.8K tayangan

Diunggah oleh Shaji Joseph

- Digital Signal Processing Using MATLAB
- Digital Signal Processing Notes
- DSP
- Digital Signal Processing
- Analog and Digital Signal Processing
- Digital Signal Processing by Ramesh Babu
- Digital Signal Processing Notes
- Digital Signal Processing Notes ( VTU Syllabus) : Prof. S.M.Hattaraki
- Real-Time Digital Signal Processing
- David Crawford Epson
- Foundation of Digital Signal Processing
- Dsp Question Bank With Solutions
- Digital Signal Processing Fundamentals
- Digital Signal Processing Applications
- Wiley.interscience.introduction.to.Digital.signal.processing.and.Filter.design.oct.2005.eBook LinG
- 29883125 Digital Signal Processing
- EEE-VI-DIGITAL SIGNAL PROCESSING [10EE64]-NOTES.pdf
- digital signal processing by nagoor kani
- FIR Filter Design
- Vtu Dsp Notespdf

Anda di halaman 1dari 36

(DSP)

SIGNAL PROCESSING

Topics

• Continuous time (CT) and Discrete time (DT) signals

• Periodic and pulse signals

• Energy and power in signals

• Standard CT and DT signals

• Impulse, step, pulse, ramp, sine and exponential

• Analysis of CT signals

• Fourier Series and Fourier Transforms

• Convolution and Correlation

• Analysis of DT signals

• Discrete Fourier Transform of DT signals

• Fast Fourier Transform

• Z Transform

• Digital filtering in time domain

• Linear Filters

• FIR Filters

• IIR Filters

SIGNAL

• A signal is defined as any physical quantity that varies with time, space or any other

independent variable or variables.

• Mathematically, a signal can be represented as a function of one or more independent

variables.

S1(t) = 7t

S2(t) = 18t2.

Here S1(t) and S2(t) represents two signals- one that varies linearly with time t and the

other varies quadratically with t.

• Complex signals are there which cannot be expressed by simple mathematical

equations. (eg: speech signal, ECG , EEG…)

SIGNAL

Digital

Analog signals

Analog Analog

Input Analog Output

Signal signal processor Signal

Analog System

• Analog signals are continuous function of an independent variable, such as time, space

etc.

• It is defined for every instant of independent variable, so the magnitude is continuous

in the specified range for analog signals.

DIGITAL SIGNAL PROCESSING (DSP)

• The digital systems are either software, hardware or firmware.

• In digital processing of signal on a digital computer, the operation performed on

a signal consists of a number of mathematical operations as specified by a

software program.

• Program represents an implementation of system in software

• Digital hardware (logic circuits) also can be used as system in hardware to

perform the desired operations.

Converter Processor Converter

Analog Digital Digital Analog

Input Input Output Output

Signal Signal Signal Signal

Advantages of DSP:

reconfiguring the DSP operations by simply changing the program

Accuracy : DSP provides better control of accuracy requirements, while

tolerance limits has to be met in the analog counterpart

Easy Storage : Digital signals can be easily stored in magnetic media without

deterioration or loss of signal fidelity. They can also be easily

transported and processed off-time in remote laboratories.

Processing : DSP allows for the implementation of more sophisticated signal

processors than its analog counterparts.

Cost effective : With advancement in VLSI technology the digital

implementation of the signal processing system is cheaper.

Limitations of DSP:

1. The conversion speed of ADC and the processing speed of signal processors

should be very high to perform all real time processing

2. Signals of high bandwidth require fast sampling rate ADCs and fast processors.

Applications of DSP:

system and for transmission & reception of voice signals.

Speech synthesis in message warning systems.

Communication : Elimination of noise by filtering and echo cancellation

by adaptive filtering in transmission channels.

Biomedical : Spectrum analysis of ECG signals to identify various

disorders in heart. Spectrum analysis of EEG signals to

study the malfunctions or disorders in the brain.

Consumer electronics : Music synthesis, Karoake systems, Surround sound

systems, Digital audio & video.

Seismology : Spectrum Analysis of seismic signals (I.e. signals

generated by movement of rocks) can be used to predict the

Earthquake, Volcanic eruptions, Nuclear explosions and

Earth movement.

Image processing : Two different filtering on images for image

enhancement, finger print matching, identifying hidden

images in the signals received by radars etc.

DISCRETE TIME SIGNALS

A discrete time signal x(n) is a function of an independent variable where the independent

variable is an integer.

Signal x(n) is not defined for non-integer values of n.

A discrete time signal is defined for every integer value of ‘n’ in the range -∞ < n < ∞

1. Functional representation 2. Graphical representation

x(n) = -1 ; n = -2 x(n)

2 ; n = -1 2

1.6

1.5 ; n = 0 1.5 1.4

-0.9 ; n = 1

1.4 ; n = 2 -2 -1 0 1 2 3

1.6 ; n= 3

-1 0.9

0 ; other n

3. Tabular representation

n -2 -1 0 1 2 3

x(n) -1 2 1.5 -0.9 1.4 1.6

4. Sequence representation

the symbol is represented as

x(n) = {… -1, 2, 1.5, -0.9, 1.4, 1.6 …}

1. Digital impulse signal or unit sample sequence δ(n)

1;n≥0 1

Impulse signal, δ(n) =

0;n≠0

0 n

δ1(n)

1;n≥0

Delayed impulse δ1(n) = δ(n – n0) = 1

0;n≠0

n0 n

u(n)

1;n≥0

Unit step signal, u(n) = 1

0;n<0

0 n

The unit step signal is related to digital impulse by the

summation relation Unit step signal

ur(n)

3. Ramp signal

n;n≥0

Ramp signal, ur(n) =

0;n<0

n

Ramp signal

4. Exponential signal

g(n)

a ;n≥0

n

0<a<1

Exponential signal, g(n) =

0;n<0

Exponential signal

Mathematical operations on Discrete Time signals

1. Shifting in time

A signal x(n) may be shifted in time by replacing the independent variable ‘n’ by

‘n-k’, where ‘k’ is an integer. If k is positive integer, the time shift results in a

delay by k units of time. If k is negative integer, the time shift results in an

advance of the signal by mod(k) units in time. The delay results in shifting each

sample of x(n) to right. The advance results in shifting each sample of x(n) to left.

x(n) 3

Eg:-

Let x(n) = 1 ; n = 2 2

1

2; n = 3

3; n = 4 0 1 2 3 4 5 6

Let x1(n) = x(n-2), where x1 is delayed signal

of x(n) x1(n) 3

2

when n = 4; x1(4) = x(4-2) = x(2) = 1 1

when n = 5; x1(5) = x(5-2) = x(3) = 2

when n = 6; x1(6) = x(6-2) = x(4) = 3 0 1 2 3 4 5 6

n

x2(n) 3

of x(n)

2

when n = 0; x2(0) = x(0+2) = x(2) = 1 1

0 1 2 3 4 5 6

when n = 2; x2(2) = x(2+2) = x(4) = 3 n

2. Folding or reflection or Transpose

The folding of a signal x(n) is performed by changing the sign of the time base n

in the signal x(n). The folding operation produces a signal x(-n) which is mirror

of x(n) with respect to time origin n = 0.

Eg:- Let x(n) = n; -3 ≤ n ≤ 3. Now folded signal x1(n) = x(-n) = -n; -3 ≤ k ≤ 3

x 1(n)

x (n)

-3 -2 -1 1 2 3

0 1 2 3 n 0 n

-3 -2 -1

Amplitude scaling of a signal by a constant A is accomplished by multiplying the

value of every signal sample by A.

Let y(n) be amplitude scaled signal of x(n), then y(n) = Ax(n)

Let x(n) = 20 ; n = 0 and A = 0.1, then y(n) = 2.0 ; n=0

36 ; n = 1 3.6 ; n=1

40 ; n = 2 4.0 ; n=2

-15 ; n = 3 -1.5 ; n=3

scaling or down sampling.

Eg:- If x(n) = an; n ≥ 0; then x1(n) = x(2n) = an for even values of n

x(n) x1(n)

n n

0 1 2 3 4 5 0 1 2 3 4 5

5. Signal (vector) addition

The sum of two signals x1(n) and x2(n) is a signal y(n), whose value at any instant

is equal to the sum of the samples of these two signals at that instant.

i.e. , y(n) = x1(n) + x2(n) ; -∞ < n < ∞

When n = 0; y(0) = x1(0) + x2(0) = 1 + (-2) = -1

When n = 1; y(1) = x1(1) + x2(1) = 2 + 1 = 3

When n = 2; y(2) = x1(2) + x2(2) = -1 + 3 = 2

When n = 3; y(3) = x1(3) + x2(3) = 2 + 1 = 3

y(n) = x1(n) + x2(n) = {-1, 3, 2, 3}

6. Product or vector multiplication

sample basis. The product of two signals x1(n) and x2(n) is a signal y(n), whose

value at instant is equal to the product of the samples of these two signals at that

instant. The product is also called modulation.

Eg:- Let x1(n) = {1, 2, -1, 2} and x2(n) = {-2, 1, 3, 1}

When n = 0; y(0) = x1(0) x x2(0) = 1 x (-2) = -2

When n = 1; y(1) = x1(1) x x2(1) = 2 x 1 = 2

When n = 2; y(2) = x1(2) x x2(2) = -1 x 3 = -3

When n = 3; y(3) = x1(3) x x2(3) = 2 x 1 = 2

y(n) = x1(n) x x2(n) = {-2, 2, -3, 2}

The discrete time signals are classified depending on their characteristics. Some

ways of classifying discrete time signals are :

2.Energy signals and power signals

3.Periodic and aperiodic signals

4.Symmetric and antisymmetric signals

1. Energy signals and power signals ………… 1

E = Ʃ |x(n)|2 , summation over n= -∞ to ∞

The energy of a signal may be finite or infinite, and can be applied to complex

valued and real-valued signals. If E is finite (0 < E < ∞), then x(n) is called an

energy signal.

P = lim (1/ (2N+1)) Ʃ |x(n)|2 limit is as n tends to infinity; ……….. 1

the summation is over n = -N to N

EN = Ʃ |x(n)|2 ; Where summation is over n = -N to N…………. 2

Then we can express the signal energy E as,

E = lim EN ; as N tends to ∞ ………… 3

Then average power of the signal is

P = lim (1 / 2N+1) EN ……………. 4

non-zero, then the signal is called power signal.

Fourier transform

In mathematics, the continuous Fourier transform is one of the specific forms of Fourier

analysis. As such, it transforms one function into another, which is called the frequency

domain representation of the original function (where the original function is often a

function in the time-domain). In this specific case, both domains are continuous and

unbounded. The term Fourier transform can refer to either the frequency domain

representation of a function or to the process/formula that "transforms" one function into

the other.

There are several common conventions for defining the Fourier transform of a complex-

valued Lebesgue integrable function, x In communications and signal processing, for

instance, it is often the function:

When the independent variable f represents time (with SI unit of seconds), the

transform variable represents ordinary frequency (in hertz). If is Hölder continuous,

then it can be reconstructed from X by the inverse transform:

FOURIER TRANSFORM OF DISCRETE TIME SIGNALS

The Fourier transform (FT) of discrete time signals is called discrete time fourier

transform (DTFT).

X(ɷ) or X(eiɷ) = Fourier transform of x(n)

The fourier transform of a finite energy discrete time signal, x(n) is defined as

The Fourier transform is one of the several mathematical tools that is useful in the

analysis and design of LTI (Linear Time Invariant) system. Another one is the Fourier

Series. These signal representations basically involve the decomposition of the signals in

terms of sinusoidal components. In such a decomposition, the signal is said to be represent

in the frequency domain.

To obtain the Fourier Transform representation, we shall start by finding the Fourier

transformation of sampled Analog signal.

Its Fourier transform is given by ,

F{xa*(t)} = - ʃ xa*(t) e-jωt dt (limit is from –infinity to +infinity)

= ʃ Σ xa (nT) δ(t-nT) e-jω(nT) dt (limit is from –infinity to +infinity)

F{xa*(t)} when T = 1

i.e., X(ejω) = F{x(n)} = Σ x(n) e-jωn at T = 1

From the above equation, we find that X(ejω) is a complex no., and it consists of real part

and imaginary part.

1. Linearity

If x1(n) is Fourier transformed to X1(ejω)

Then ,

ax1(n) + bx2(n) ----------------- aX1(ejω) + bX2(ejω)

Proof:

F{ax1(n) + bx2(n)} = Σ(ax1(n) + bx2(n)) e-jωn

= Σ ax1(n) e-jωn + Σ bx2(n) e-jωn

= a X1(ejω) + b X2(ejω)

2. Time Shifting Property

Then ,

x(n-k) ----------------- e-jωk X(ejω)

Proof:

Let n-k = p

(limit is from p=-infinity to + infinity). Hence proved.

3. Time Reversal

Then ,

x(-n) ----------------- X(e-jω)

Proof:

Let -n = p

F{x(p)} = X (e-jω)

(limit is from p=-infinity to + infinity). Hence proved.

4. Convolution Theorem

Then ,

x1(n) * x2(n) ----------------- X1(ejω) * X2(ejω)

Proof:

(summation is from k=-infinity to + infinity)

Fourier transform,

(summation is from n=-infinity to + infinity)

(first summation is from k=-infinity to + infinity, second is over the same limit for n.)

Put n-k = p

(limit is for p = -infinity to + infinity)

5. Differentiation in frequency domain

Then,

Proof:

Results:

Linear Time Invariant Systems

linear time-invariant system are, of course, linearity and time invariance

•Linearity means that the relationship between the input and the output of the system

satisfies the superposition property. If the input to the system is the sum of two

component signals:

where yn(t) is the output resulting from the sole input xn(t).

•Time invariance means that whether we apply an input to the system now or T seconds

from now, the output will be identical, except for a time delay of the T seconds. If the

output due to input x(t) is y(t), then the output due to input x(t − T) is y(t − T). More

specifically, an input affected by a time delay should effect a corresponding time delay in

the output, hence time-invariant.

The fundamental result in LTI system theory is that any LTI system can be characterized

entirely by a single function called the system's impulse response. The output of the

system is simply the convolution of the input to the system with the system's impulse

response. This method of analysis is often called the time domain point-of-view. The same

result is true of discrete-time linear shift-invariant systems, in which signals are discrete-

time samples, and convolution is defined on sequences.

Impulse Response

The impulse response of a system is its output when presented with a very brief input

signal, an impulse. An impulse represents the limiting case of a pulse made very short

in time while maintaining its area or integral (thus giving an infinitely high peak).

While this is impossible in any real system, it is a useful concept as an idealization.

A system in the class known as LTI systems (linear, time-invariant systems) is

completely characterized by its impulse response.

The Impulse response from a simple audio system. Showing the original impulse, with high

frequencies boosted, then with low frequencies boosted.

Practical applications of Impulse response

• Loudspeakers

A very useful real application that demonstrates this idea was the development of

impulse response loudspeaker testing in the 1980s which led to big improvements in

loudspeaker design. Loudspeakers suffer from phase inaccuracy, a defect unlike

normal measured properties like frequency response.

• Digital Filtering

Impulse response is a very important concept in the design of digital filters for

audio processing, because these differ from 'real' filters in often having a pre-echo,

which the ear is not accustomed to.

• Electronic processing

Impulse response analysis is a major facet of radar, ultrasound imaging, and many

areas of digital signal processing. An interesting example would be broadband

internet connections. Where once it was only possible to get 4 kHz speech signal

over a local telephone wire, or data at 300 bit/s using a modem, it is now

commonplace to pass 2 Mb/s over these same wires, largely because of 'adaptive

equalisation' which processes out the time smearing and echoes on the line.

FIR Systems

In FIR (Finite duration Impulse response) systems, the impulse response consists of

finite number of samples. The convolution formula for FIR system is given by,

Where h(n) = 0; n < 0 and n ≥ N

From eqn 1 , it can be concluded that the impulse response selects only N samples of the

input signal.

Thus, the system acts as a window that views only the most recent N input

signal samples in forming the output. It neglects all prior input samples. So a FIR

system has a finite memory of length N samples.

Where summation is over k = 0 to N-1

On taking Z transform of eqn 2, we get,

IIR Systems

In IIR (Infinite duration Impulse Response) systems, the impulse response has infinite

number of samples. The convolution formula for IIR systems is given by,

Since, this sum involves the present and all the past input sample, we can say that the

system has an infinite memory.

Where the first summation is from k=1 to N and the second summation is over k=0 to M

Let Z{y(n)} = Y(z) ; Z{y(n-k)} = z-k Y(z)

Let Z{x(n)} = X(z) ; Z{x(n-k)} = z-k X(z)

Y(z) + Σ ak z-k Y(z) = Σ bk z-k X(z)

(1 + Σ ak z-k) Y(z) = Σ bk z-k X(z)

….……… 5

= (b0 + b1 z + b2 z + .. + bM z ) / (1 + a1 z + a2 z + .. + aN z )

-1 -2 -M -1 -2 -N

Infinite Impulse Response (IIR)

Infinite impulse response (IIR) is a property of signal processing systems. Systems

with that property are known as IIR systems or when dealing with electronic filter

systems as IIR filters. They have an impulse response function which is non-zero over an

infinite length of time. This is in contrast to finite impulse response filters (FIR) which

have fixed-duration impulse responses.

The simplest analog IIR filter is an RC filter made up of a single resistor (R)

feeding into a node shared with a single capacitor (C). This filter has an exponential

impulse response characterized by an RC time constant.

A finite impulse response (FIR) filter is a type of a digital filter. The impulse response,

the filter's response to a Kronecker delta input, is 'finite' because it settles to zero in a

finite number of sample intervals. This is in contrast to infinite impulse response filters

which have internal feedback and may continue to respond indefinitely. An Nth order

FIR filter has a response to an impulse that is N+1 samples in duration.

Z - transform

In mathematics and signal processing, the Z-transform converts a discrete time-domain

signal, which is a sequence of real or complex numbers, into a complex frequency-

domain representation.

transform name) by E. I. Jury in 1958 in Sampled-Data Control Systems (John Wiley &

Sons). The idea contained within the Z-transform was previously known as the

"generating function method".

The Z-transform, like many other integral transforms, can be defined as either a one-

sided or two-sided transform.

Bilateral Z-transform

The bilateral or two-sided Z-transform of a discrete-time signal x[n] is the function X(z)

defined as

z = Aejφ

where A is the magnitude of z, and φ is the complex argument (also referred to as angle

or phase) in radians.

- Digital Signal Processing Using MATLABDiunggah olehTurtogtokh Tumenjargal
- Digital Signal Processing NotesDiunggah olehRevathy Perumalsamy
- DSPDiunggah olehRasha Hassan
- Digital Signal ProcessingDiunggah olehAmarnath M Damodaran
- Analog and Digital Signal ProcessingDiunggah olehnowayjose666
- Digital Signal Processing by Ramesh BabuDiunggah olehajay
- Digital Signal Processing NotesDiunggah olehAkshansh Chaudhary
- Digital Signal Processing Notes ( VTU Syllabus) : Prof. S.M.HattarakiDiunggah olehAchyutha Hosahalli
- Real-Time Digital Signal ProcessingDiunggah olehHyungjoon Lim
- David Crawford EpsonDiunggah olehapi-3826975
- Foundation of Digital Signal ProcessingDiunggah olehVarun Prashant Dikshit
- Dsp Question Bank With SolutionsDiunggah olehazhagumuruganr
- Digital Signal Processing FundamentalsDiunggah olehapi-3826975
- Digital Signal Processing ApplicationsDiunggah olehPramod Putta
- Wiley.interscience.introduction.to.Digital.signal.processing.and.Filter.design.oct.2005.eBook LinGDiunggah olehvoltase
- 29883125 Digital Signal ProcessingDiunggah olehaloove66
- EEE-VI-DIGITAL SIGNAL PROCESSING [10EE64]-NOTES.pdfDiunggah olehLakshmi Oruganti
- digital signal processing by nagoor kaniDiunggah olehMurughesh Murughesan
- FIR Filter DesignDiunggah olehSuman Ahuja
- Vtu Dsp NotespdfDiunggah olehAmit Mb
- Digital Signal ProcessingDiunggah olehurvesh_patel44
- understanin_digital_signal_processing (1)Diunggah olehjobinvcm
- 0470014954Real-Time Digital Signal BDiunggah olehAkkari Zarzis
- Digital Signal Processing by J.S. Katre (Tech Max)-wikiforu.com.pdfDiunggah olehkifle hailu
- Digital Signal ProcessingDiunggah olehPECMURUGAN
- Digital Signal Processing by S. Salivahanan.pdfDiunggah olehApoorva
- dspDiunggah olehdeepak2386
- Eee-Vi-digital Signal Processing [10ee64]-NotesDiunggah olehsakthimaha
- 17912060 Dsp Ec1302 Parta Part b Questions and Answers11Diunggah olehsudhashreetn
- Digital Signal Processing (DSP)Diunggah olehDilpreet Singh

- REST PERIOD ACTIVITY JAN 2011Diunggah olehIvan Brcic
- ATA State Telemedicine Physician Practice Standards Licensure.pdfDiunggah olehiggybau
- Director Manager Internal Audit in Washington DC Resume Jeffrey BarnesDiunggah olehJeffreyBarnes2
- ECG 3Diunggah olehHêny Carlênic
- second lesson planDiunggah olehapi-285000116
- Survey of Potential Health Risk of Rubbish Collectors From the Garbage Dump Sites in Kelantan, MalaysiaDiunggah olehSai
- Zwitterion's Complete MDMA ..Diunggah olehinvitacions
- the mitten-reading-lesson 2Diunggah olehapi-273142130
- Book ListDiunggah olehnulluzer
- Empircal TranslationDiunggah olehSofiane Douifi
- Dedicated StorageDiunggah olehJunaidi Kun
- Linear AlkylbenzeneDiunggah olehmark_59
- Deflagration Flame Arrester Cat Rev6Diunggah olehmusaveer
- Angewandte Chemie International Edition Volume 14 Issue 11 1975 [Doi 10.1002%2Fanie.197507451] Prof. Dr. Rudolf Criegee -- Mechanism of OzonolysisDiunggah olehSubramanya Byndoor
- deconstructing derridaDiunggah olehapi-58165914
- Skum Semi Subsurface Hsss_fds14337 0214 LrDiunggah olehbrujula27
- DAFTAR PUSTAKADiunggah olehdedi irawan
- Usos Diagnósticos de La SalivaDiunggah olehMarcelo Anibal Alvarez
- KAMSTRUP 685-351Diunggah olehsourcekhi
- Adhesive Cementation of Etchable Ceramic Esthetic RestorationsDiunggah olehGuadalupe García Figueroa
- Prepking 310-220 Exam QuestionsDiunggah olehdenmark77
- Struts 1Diunggah olehBabacar Ngom
- APPROACHES TO CURRICULUM.pptxDiunggah olehathira
- Literate-programming.pdfDiunggah oleh3187265
- rocket writingDiunggah olehapi-281447357
- petrel manual.pdfDiunggah olehNwabisa RainDrops Funiselo
- Wikis & Wikipedia New Media Life Cycle AnalysisDiunggah olehAndrew Davis
- American Becomes an Industrial GiantDiunggah olehSmoony
- 13[1].Quality of Work Life and Quality CirclesDiunggah olehAmit Sharma
- shenderson casestudy ece3350 fall2016 docDiunggah olehapi-307757852