Digital Signal Processing - UNIT 1
Digital Signal Processing - UNIT 1
Digital Signal Processing - UNIT 1
Signal:
A Signal is defined as any physical quantity that changes with time, distance, speed,
position, pressure, temperature or some other quantity. A Signal is physical quantity that
consists of many sinusoidal of different amplitudes and frequencies.
Ex x(t) = 10t
1) ASP (Analog signal Processing) : If the input signal given to the system is analog then
system does
If the input signal given to the system is digital then system does digital signal processing.
Ex Digital Computer, Digital Logic Circuits etc. The devices called as ADC (analog to digital
Converter) converts Analog signal into digital and DAC (Digital to Analog Converter) does
vice-versa.
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 environmental 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 magnetic tapes. Hence
becomes transportable. Thus easy and lasting storage capacity.
7. Digital signal processing systems are upgradeable since that are software controlled.
9. The cost of microprocessors, controllers and DSP processors are continuously going
down. For some complex control functions, it is not practically feasible to construct analog
controllers.
10. Single chip microprocessors, controllers and DSP processors are more versatile and
powerful.
2. Limit in frequency. High speed AD converters are difficult to achieve in practice. In high
frequency applications DSP are not preferred.
Digital Signal processing_UNIT 1
CLASSIFICATION OF 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.
Multi-channel signal will be the vector sum of signals generated from multiple sources.
2 Continuous Valued and continuous time signals Discrete time signal with set
are basically analog signals of discrete amplitude are
called digital signal.
5)Analog and digital 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(ωn) and x2(n)= sin(ωn) are good examples of even & odd signals respectively.
Every discrete
Problems:
2) Find out the even and odd parts of the discrete signal x(n)={2,4,3,2,1}
3) Find out the even and odd parts of the discrete signal x(n)={2,2,2,2}
Discrete time signals are also classified as finite energy or finite average power signals.
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.
PROBLEMS:
Shifting : signal x(n) can be shifted in time. We can delay the sequence or advance the
sequence.
Folding / Reflection : It is folding of signal about time origin n=0. In this case replace n by
–n. Original signal:
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.
Scaling: Amplitude scaling can be done by multiplying signal with some constant. Suppose
original signal is x(n). Then output signal is A x(n)
It is very easy to find out that given system is static or dynamic. Just check that output of
the system solely depends upon present input only, not dependent upon past or future.
Response to the system to the sum of signal = sum of individual responses of the 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 ∞.
2. In this method we decompose input signal into sum of elementary signal. Now the
elementary input signals are taken into account and individually given to the system. Now
Digital Signal processing_UNIT 1
using linearity property whatever output response we get for decomposed input signal, we
simply add it & this will provide us total response of the system to any given input signal.
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.
CORRELATION:
It is frequently necessary to establish similarity between one set of data and another. It
means we would like to correlate two processes or data. Correlation is closely related to
convolution, because the correlation is essentially convolution of two data sequences in
which one of the sequences has been reversed.
Applications are in
1) Images processing for robotic vision or remote sensing by satellite in which data from
different image is compared
2) In radar and sonar systems for range and position finding in which transmitted and
reflected waveforms are compared.
TYPES OF CORRELATION
1) CROSS CORRELATION:
When the correlation of two different sequences x(n) and y(n) is performed it is called as
Cross correlation. Cross-correlation of x(n) and y(n) is rxy(l) which can be mathematically
expressed as
AUTO CORRELATION:
PROPERTIES OF CORRELATION
rxy(l) = ryx(-l)
rxx(l) = rxx(-l)
Examples:
Answer: rxy(l) = {10, -9, 19, 36, -14, 33, 0,7, 13, -18, 16, -7, 5, -3}
A/D CONVERSION
It is the process of converting continuous time signal into a discrete time signal by taking
samples of the continuous time signal at discrete time instants.
Thus Sampling Theorem states that if the highest frequency in an analog signal is Fmax and
the signal is sampled at the rate fs > 2Fmax then x(t) can be exactly recovered from its
sample values. This sampling rate is called Nyquist rate of sampling. The imaging or
aliasing starts after Fs/2 hence folding frequency is fs/2. If the frequency is less than or
equal to 1/2 it will be represented properly.
Example:
= cos (∏/2)n
Thus the frequency 50 Hz, 90 Hz , 130 Hz … are alias of the frequency 10 Hz at the
sampling rate of 40 samples/sec
2.QUANTIZATION
The process of converting a discrete time continuous amplitude signal into a digital signal
by expressing each sample value as a finite number of digits is called quantization. The
error introduced in representing the continuous values signal by a finite set of discrete
value levels is called quantization error or quantization noise.
3.CODING/ENCODING
Each quantization level is assigned a unique binary code. In the encoding operation, the
quantization sample value is converted to the binary equivalent of that quantization level.
If 16 quantization levels are present, 4 bits are required. Thus bits required in the coder is
the smallest integer greater than or equal to Log2 L. i.e b= Log2 L Thus Sampling frequency
is calculated as fs=Bit rate / b.
4.ANTI-ALIASING FILTER
When processing the analog signal using DSP system, it is sampled at some rate depending
upon the bandwidth. For example if speech signal is to be processed the frequencies upon
3khz can be used.Hence the sampling rate of 6khz can be used. But the speech signal also
contains some frequency components more than 3khz. Hence a sampling rate of 6khz will
introduce aliasing. Hence signal should be band limited to avoid aliasing.
The signal can be band limited by passing it through a filter (LPF) which blocks or
attenuates all the frequency components outside the specific bandwidth. Hence called as
Anti aliasing filter or prefilter.
5.SAMPLE-AND-HOLD CIRCUIT:
After a continuous-time signal has been through the A/D converter, the quantized output
may differ from the input value. The maximum possible output value after the quantization
process could be up to half the quantization level q above or q below the ideal output value.
This deviation from the ideal output value is called the quantization error. In order to
reduce this effect, we increases the number of bits.