DSP and Signal Theory …Difference between conventional analog

communication and digital radio?

ASK FSK PSK
– Both analog and digital modulation use
Sampling and Quantization
analog carriers for transmission. In analog
PCM modulation, the transmitted information is
analog, while in digital modulation, the
Plus 2 chapters of teh book information signal is digital.
COMMS 2 REVIEWER Analog communication – analog signal
Electronic communications – is the modulates the carrier signal.
transmission, reception, and processing of
information with the use of electronic circuits.

Information – is defined as knowledge or

intelligence that is communicated (i.e.
transmitted or received) between two or more
points.

Conventional analog communications systems

that use: Amplitude modulation, Frequency Digital communication – digital signal
modulation and Phase modulation modulates the carrier.

replaced by digital radio (or digital modulation) Digital modulation is ideal for various
systems that offer several outstanding communication applications, including:
advantages like: Ease of processing, Ease of
 Low-speed voice band modems
multiplexing and Noise immunity
 High-speed systems like broadband DSL
 Digital microwave and satellite
communications
Digital communications could be:
 Cellular and PCS systems
1. Digital modulation – involves transmitting
digitally modulated analog signals (carriers) in a
communication system, often referred to as Simplified Block diagram of a Digital Radio
"digital radio" when used in wireless systems. It System
conveys information in analog form through
physical mediums or free space. High-frequency
analog carriers are modulated by low-frequency
digital signals.

2. Digital transmission – transport information

in digital form.

- require physical facility between the

transmitter and the receiver.
At the transmitter side: Shannon’s Limit of Information Capacity

Precoder – performs level conversion and then  1948 by Claude E. Shannon of Bell
encode the incoming data into groups of bits Telephone Laboratories
that modulate the analog carrier.
I = 3.32 B log [1 + (S/N)]
Modulator – digitally modulates the analog
Where:
carrier.
I = information capacity in bits per second
BPF and amplifier – shapes( filters) and
amplifies the modulated carrier before the B = bandwidth in hertz
transmission.
S/N = signal-to-noise ratio
Demodulator and decoder – extract the original
source information from the modulated carrier.

Clock and recovery circuit – recovers the analog M-ary encoding

carrier and digital timing (clock) signals from the M-ary – refers to a term derived from the term
incoming modulated wave. (since they are binary.
necessary to perform the modulation process)
M – simply means a digit that corresponds to
the number of conditions, levels, or
Information theory – studies the efficient use of combinations possible for a given number of
bandwidth in electronic communication binary variables.
systems and determines a system's information The number of bits necessary to produce a
capacity. given number of conditions is expressed
Information capacity – measured in bit rate mathematically as:
(bits per second), quantifies the number of N = log2 M , where:
independent symbols a system can transmit in a
given time, depending on bandwidth and N = number of bits necessary
transmission time. M = number of conditions

Or the number of conditions possible with N

Hartley’s Law bits is:

 1928 by R. Hartley of Bell Telephone 2N = M

Laboratories

I = kBt Bit rate – rate of change of a digital information

Where: signal

I = information capacity in bits per second Baud – is the rate at which a signal changes on a
transmission medium after encoding and
B = bandwidth in hertz modulation. It represents the transmission,
t = transmission time in seconds modulation, or symbol rate, measured in
symbols per second. Each signaling element
(symbol) may represent multiple information
bits, and baud is the reciprocal of the duration Note: the actual bandwidth necessary to
of one symbol. propagate a given bit rate depends on several
factors such as: (1) type of encoding (2) type of
Baud = 1/ ts
modulation used (3)types of filters used (4)
Where: system noise (5) desired error performance.

Baud = symbol rate in baud per second

ts = time of one signaling element in seconds The equation is only true if transmission of
signals is done in binary.

In higher-than-binary encoding, more than one

Binary signals are represented as series of 1s or bit may be transmitted at a time( remember
0s and are transmitted one at a time. The rate that a baud may represent a number of bits)
of transmission is called bit rate. and it is possible to propagate a bit rate that
Baud on the other hand is also transmitted one exceeds 2B.
at a time. fb = 2B log2 M
But baud may represent more than one Where:
information bit, thus the baud in data
communications system may be a lot lesser fb = channel capacity in bits per second
than the bit rate.
B = minimum Nyquist Bandwidth
In binary systems (such as FSK and PSK), bit rate
M = number of discrete signal or voltage levels
and baud are equal.
Log2 M = number of bits encoded into each
In higher level systems (such as QPSK and 8-
signaling element = N
PSK), bit rate is always greater than baud rate.

the minimum bandwidth necessary to pass M-

Nyquist Bandwidth, B (Also called Nyquist
ary digitally modulated carriers:
frequency)
B = fb / log2 M
- Generally used for comparison
purposes only. B = fb / N

- H. Nyquist: “Binary digital signals can be Since the baud is the encoded rate of change,
propagated through an ideal noiseless it also equals the bit rate divided by the
transmission medium at the rate equal number of bits encoded into one signaling
to twice the bandwidth of the element.
medium”.
Baud = fb / N
fb = 2B, or B = fb / 2
Note:
Where:
The baud is equal to the minimum Nyquist
fb = is the bit rate in bits per second bandwidth and is equal to fb/N.

B = ideal Nyquist Bandwidth

This is true for all forms of digital modulation Vm(t) is a normalized binary waveform, where at
except for FSK. logic 1, Vm(t) = 1, thus:

Digital modulation techniques:

1. ASK And for logic 0, Vm(t) = -1, thus:

2. FSK

3. PSK

4. QAM

Amplitude-shift keying, ASK

- Also called digital amplitude

modulation, DAM or on-off keying, Amplitude-shift keying input and output
OOK waveforms
- Simplest digital modulation technique.

- A binary information signal directly

modulates the amplitude of the analog
carrier.

- Very similar to the analog amplitude

modulation except that there are only
two output amplitudes possible.

Mathematically, ASK is:

Note:

 For every change in the input binary

data stream, there is one change in the
Where: analog waveform.

Vask(t) = amplitude shift keying wave  The time of one bit (tb) equals the time
of one signaling element (ts)
Vm(t) = digital information modulating signal
 For the entire time the binary input is
A/2 = unmodulated carrier amplitude high, the output is a constant
amplitude, constant frequency signal
c = analog carrier radian frequency in radians
per second, 2fct  For the entire time the binary signal is
low, the carrier is off.
  The bit time, tb is the reciprocal f the bit Vm (t) – binary input modulating signal, in
rate, fb. Volts

 The time of one signaling element, ts is f – peak change or shift in the analog carrier
the reciprocal of the baud. frequency, in Hertz

 Therefore, the rate of change of the ASK

waveform, its baud, is the same as the
rate of change of the binary input, fb.
From:
 Thus, Bit rate = baud, for an ASK
Vfsk(t) = Vc cos {2[fc + Vm(t)f] t}

Note that:
Note :
1. The peak shift in the carrier frequency
For an ASK, N =1 since there are only 2 possible
f is proportional to the amplitude of
conditions, a logic 0 or a logic 1.
the binary input signal Vm(t)
2N = M, if M = 2 then N =1
2. The direction of the shift is determined
 Thus for an ASK, the bit rate, which is by its polarity.
equal to the baud, is also equal to the
3. The modulating signal is a normalized
minimum Nyquist bandwidth:
binary waveform where a logic 1 = 1V
B = fb/ N =fb/ 1 = fb and a logic 0 = -1 V.

FREQUENCY SHIFT KEYING Thus for a logic 1 input, Vm(t) = 1V.

- Is another relatively simple, low- Vfsk(t) = Vc cos {2[fc + Vm(t)f] t} can

performance type of digital modulation. be written as

- Is a form of constant amplitude angle Vfsk(t) = Vc cos {2[fc + f] t}

modulation like standard frequency
And for a logic 1 input, Vm(t) = -1 V.
modulation (FM) except that the
modulating signal is a binary signal that Vfsk(t) = Vc cos {2[fc - f] t}
varies between two discrete voltage
levels rather than a continuously
changing analog waveform. 4. With BFSK, the carrier center frequency
- Also called binary FSK, BFSK. fc is shifted, or deviated up and down in
the frequency domain by the binary
The general expression is: input signal as:
Vfsk(t) = Vc cos {2[fc + Vm(t)f] t}

Where:

Vfsk(t) – binary FSK waveform

Vc – peak analog carrier amplitude, in Volts

fc – analog carrier center frequency, in Hertz

 5. With FSK, frequency deviation is
defined as the difference between
either the mark or space frequency and
the center frequency, or half the
difference between the mark and space
Note: the use of BFSK is restricted only to low
frequencies.
performance, low cost, asynchronous data
modems that are used for data communications
over analog voice band telephone lines.

where f – frequency deviation in Hertz

FSK BIT RATE, BAUD AND BANDWIDTH
fm - fs - absolute difference between the
mark and space frequencies. Bit rate

The time of one bit, (tb) is the same as the time

the FSK output is a mark (tm) or a space
6. When the binary input fb changes from frequency (ts). Thus the bit time equals the
a logic 1 to a logic 0, and vice versa, the time of an FSK signaling element, and the bit
FSK output frequency shifts from a mark rate equals the baud.
fm to a space frequency fs and vice
versa. Baud

The baud for binary FSK can also be determined

by substituting N = 1.

Baud = fb/ N

= fb

Minimum Bandwidth

The minimum bandwidth for FSK is given as:

B = (fs + fb) – (fm – fb)

 = (fs – fm) + 2fb The formula used for modulation index in FM is
also valid for FSK: thus,
B = 2 (f + fb)
h = f/ fa
Where: B – minimum Nyquist bandwidth, in
Hertz Where:

f – frequency deviation, in Hertz h – FM modulation index called the h factor in

FSK
fb – input bit rate, in bps
f – peak frequency deviation, in Hertz

fa – fundamental frequency of the binary

modulating signal in Hertz

Bessel functions can also be used to determine The worst- case modulation index (deviation
the approximate bandwidth for an FSK wave. ratio) is that which yields the widest bandwidth.

AND

Where:
Note: the fastest rate of change (highest
fundamental frequency) in a non-return to zero, h – h factor, unitless
NRZ binary signal occurs when alternating 1s
and 0s are occurring. fm – mark frequency, in Hertz

fs – space frequency in Hertz

Since it takes a high and low to produce a cycle, fb – bit rate in bps
the highest fundamental frequency present in a
square wave equals the repetition rate of the
square wave, which with binary signal is equal FSK TRANSMITTER
to half the bit rate. The BFSK modulator is very similar to a
fa = fb/2 conventional FM modulator and is very often a
voltage controlled oscillator (VCO). The center
Where: frequency fc is chosen such that it falls halfway
fa – highest fundamental frequency of the between the mark and space frequencies. A
binary input signal, in Hertz logic 1 input shifts the VCO output to the mark
frequency, and a logic 0 input shifts the VCO
fb – input bit rate in bps output to the space frequency.
 The FSK input signal is simultaneously applied to
the inputs of both bandpass filters, through a
power splitter.

The respective filter passes only the mark or

only the space frequency on to its respective
envelope detector.

The envelope detectors, in turn, indicate the

FSK modulator
total power in each pass band,
A VCO- FSK modulator can be operated in the
and the comparator responds to the largest of
sweep mode where the peak frequency
the two powers.
deviation is simply the product of the binary
input voltage and the deviation sensitivity of the This type of FSK detection is referred to as non-
VCO. coherent detection. That is, there is no
frequency involved in the demodulation process
With the sweep mode of modulation, the
that is synchronized either in phase, frequency,
frequency deviation is expressed
or both with the incoming FSK signal.
mathematically as:

f = Vm(t)k
FSK COHERENT RECEIVER
Where:

f – peak frequency deviation in Hertz

Vm(t) – peak binary modulating signal voltage in

Volts

k – deviation sensitivity in Hertz per Volt

FSK RECEIVERS
The incoming FSK signal is multiplied by a
1. Non coherent receiver - there is no recovered carrier signal that has the exact same
frequency involved in the demodulation frequency and phase as the transmitter
process that is synchronized either in reference.
phase, frequency or both with the
incoming FSK signal. The two transmitted frequencies(the mark and
space frequencies) are not generally
2. Coherent receiver continuous, thus not practical to reproduce a
3. FSK NONCOHERENT RECEIVER local reference that is coherent with both.

That’s why, coherent FSK detection is seldom

used.

PHASE-LOCKED LOOP, PLL

 occurs, the demodulator has trouble following
the frequency shift and an error might occur.

A CP-FSK is a BFSK except that the mark and

space frequencies are synchronized with the
input binary bit rate.

The mark and space frequencies are selected

such that they are separated from the center
frequency by an exact multiple of ½ the bit rate,
that is:
PLL - FSK demodulator is the most common
circuit used for demodulating binary FSK signals. fm = fs = n[fb/2], where n is any integer.

PLL- FSK works similarly as a PLL-FM This ensures smooth phase transition in the
demodulator. analog output signal when it changes from
mark to space frequency and vice versa.
As the input to the PLL shifts between the mark
and space frequencies, the dc error voltage at
the output of the phase comparator follows the
frequency shift.

Because there are only two input frequencies,

there are also two output error voltages. One
represents logic 1 and the other represents
logic 0.
Note that when the output frequency changes,
The output is thus a two-level representation of it is a smooth continuous transition, hence
the FSK input. there are no phase discontinuities.
Generally, the natural frequency of the PLL is CP-FSK has better bit error performance than
made equal to the center frequency of the FSK conventional BFSK for a given S/N ratio.
modulator.
The disadvantage of CP-FSK is that it requires
CONTINUOUS PHASE FREQUENCY - SHIFT synchronization circuits and is, therefore, more
KEYING, CP-FSK expensive to implement.

_______________________________________

PHASE SHIFT KEYING

- Another form of angle – modulated,

Non-continuous FSK waveform constant amplitude digital modulation.

Note: when the input changes from logic 1 to - PSK is a M-ary digital modulation
logic 0 and vice versa, there is an abrupt phase scheme similar to conventional phase
discontinuity in the analog signal.When this modulation except that with PSK, the
input is a binary digital signal and there
 are limited number of output phases The balanced modulator acts as a phase
possible. reversing switch.

1. Binary PSK – the simplest form of PSK Depending on the logic condition of the digital
where N =1 and M = 2 input, the carrier is transferred to the output
either in phase or 180 degrees out of phase
2. Quaternary PSK - Is a M-ary encoding
with the reference carrier oscillator.
scheme where N = 2 and M = 4.

3. 8-PSK - Is a M-ary encoding scheme

where N = 3 and M = 8. BALANCED RING MODULATOR

4. 16-PSK - Is a M-ary encoding scheme

where N = 4 and M = 16.

1. BINARY PHASE - SHIFT KEYING

- The simplest form of PSK where N = 1

and M = 2. Therefore BPSK has two
phases, 2N = 21 = 2, are possible for the The balanced ring modulator has two inputs:
carrier. One phase represents logic 0,
1. a carrier that is in phase with the
the other, logic 1.
reference oscillator
- As the input digital signal changes state,
2. and the binary digital data.
the phase of the output carrier shifts
between two angles separated by 180 For the balanced modulator to function
degrees. properly, the digital input voltage must be much
greater than the peak carrier voltage.
- Another name is phase - reversal
keying, PRK and biphase modulation. This ensures that the digital input controls the
on/off state of diodes D1 to D4.
- BPSK is a form of square wave
modulation of a continuous wave, CW
signal.
If the binary input is logic 1 ( a positive voltage),
BPSK TRANSMITTER diodes D1 and D2 are forward biased and ON.
While diodes D3 and D4 are reverse biased and
OFF.

A carrier voltage is created across transformer

T2 that is in phase with the carrier voltage
across T1.

Consequently, the output signal is in phase with

the reference oscillator.
 Logic 1 Input

If the binary input is logic 0 ( a negative

voltage), diodes D1 and D2 are reverse biased
and OFF. While diodes D3 and D4 are forward
biased and ON.

A carrier voltage is created across transformer

T2 that is 180 degrees out of phase with the
carrier voltage across T1.

Consequently, the output signal is 180 degrees

out of phase with the reference oscillator.

Phasor diagram

Logic 0 Input

BPSK truth table, phasor diagram and

constellation diagram
constellation diagram ( also called signal state –
space diagram

BANDWIDTH CONSIDERATIONS OF BPSK

A balanced modulator is a product modulator.

The output signal is the product of two input
signals.

Mathematically, the output of a BPSK

modulator is proportional to:

BPSK output = [sin(2fat) X sin(2fct)]

Where:
fa – maximum fundamental frequency of binary Note that the time of one BPSK signaling
input in Hertz element (ts) is equal to the time of one
information bit (tb) which indicates that the bit
fc – reference carrier frequency in Hertz
rate equals the baud.

Solving for the trigonometric identity for the

BPSK RECEIVER
product of two sine functions,

Thus, the minimum double-sided Nyquist

bandwidth, B is:
The input signal may be +sin ct or -sin ct.

The coherent carrier recovery circuit detects

and regenerates a carrier signal that is both
frequency and phase coherent with the original
transmit carrier.

The balanced modulator is a product detector,

And because fa = fb/2, where fb = input bit rate, the output is a product of the two inputs, the
BPSK signal and the recovered carrier.

The low pass filter LPF separates the recovered

binary data from the complex demodulated
signal.
Where B is the minimum double sided Nyquist
bandwidth
For a BPSK input signal of +sin ct, logic 1, the
output of the balanced modulator is:
OUTPUT PHASE VS TIME RELATIONSHIP FOR A
BPSK MODULATOR
Or:

Leaving:
The LPF has a cut off frequency much lower output phases, +45, +135, -45 and -135
than 2c and thus, blocks the second harmonic degrees.
of the carrier and passes only the positive
- For each two bit dibit clocked into the
constant coefficient. A positive voltage
modulator, a single output change
represents a demodulated logic 1.
occurs, and the rate of change at the
output (baud) is equal to one-half the
input bit rate
For a BPSK input signal of –sin ct, logic 0, the
output of the balanced modulator is:

QPSK TRANSMITTER

Or:

Leaving:

Two bits ( a dibit) are clocked into a bit splitter.

After both bits have been serially inputted, they
The output of the balanced modulator contains are simultaneously parallel outputted. One bit is
a negative voltage directed to the I channel and the other to the Q
channel. The I bit modulates a carrier that is in
(-[1/2] V) and a cosine wave at twice the phase with the reference oscillator, hence the
carrier frequency 2c . Again, the LPF blocks the name “I” for in phase channel, and the Q bit
second harmonic of the carrier modulates a carrier that is 90 degrees out of
phase or in quadrature with the reference
carrier, hence the name “Q” for quadrature
QUARTERNARY PHASE - SHIFT KEYING channel. Once a dibit has been split into the I
QPSK, or quadrature PSK, is another form of and Q channels, the operation is the same as
angle modulated, constant amplitude digital the BPSK modulator. A QPSK modulator is
modulation. essentially two BPSK modulators combined in
parallel.
- Is a M-ary encoding scheme where N =
2 and M = 4.

- With QPSK, four output phases are

possible for a single carrier frequency.

- In the modulator, each dibit code

generates one of the four possible
 With QPSK, because the input data are divided
into two channels, the bit rate in either the I or
the Q channel is equal to one-half of the input
data rate, fb/2.

The highest fundamental frequency present at

the data input to the I or the Q balanced
modulator is equal to ¼ of the input data rate.

The output of the I and Q balanced modulators

requires a minimum double-sided Nyquist
For logic 1 = +1 V and logic 0 = -1 V, two phases bandwidth equal to ½ of the incoming bit rate,
are possible at the output of the I balanced fN = twice fb/4 = fb/2
modulator and that is +sin ct and -sin ct .

Two phases are possible at the output of the Q

balanced modulator, +cos ct and -cos ct The output of the balanced modulators can be
expressed as:
When the linear summer combines the two
quadrature signals, there are 4 possible
resultant phasors: +sin ct +cos ct, +sin ct -
cos ct, -sin ct +cos ct and -sin ct -cos ct.

BANDWIDTH CONSIDERATIONS OF QPSK

Where:

Thus:
 At the I channel for an input of –sinct + cos ct,

THE QPSK RECIEVER

The power splitter directs the input QPSK signal

to the I and Q product detectors and the carrier
recovery circuit. At Q Channel

The carrier recovery circuit reproduces the

original transmit carrier oscillator signal. The OFFSET QPSK
recovered carrier must be frequency and phase
coherent with the transmit reference carrier. - Is a modified form of QPSK where the
bit waveforms n the I and Q channels
The QPSK signal is demodulated in the I and Q are offset or shifted in phase from each
product detectors, which generate the original I other by one-half of a bit time.
and Q data bits.

The outputs of the product detectors are fed to

the bit combining circuit, where they are
converted from parallel I and Q data channels
to a single binary output data stream.

MATHEMATICAL DESCRIPTION OF OQPSK block diagram

DEMODULATION PROCESS

The four possible inputs are: +sin ct +cos ct,

+sin ct -cos ct, -sin ct +cos ct and -sin ct -
cos ct.
OFFSET KEYED PSK BIT ALIGNMENT AND during modulation. A disadvantage of OQPSK is
CONSTELLATION DIAGRAM that changes in the output phase occur at twice
the data in either the I or Q channels.
Consequently, with OQPSK the baud and
minimum bandwidth are twice that of
conventional QPSK for a given transmission bit
rate. QPSK is sometimes called Offset-keyed
QPSK (OKQPSK)

OQPSK bit alignment 8 - PSK

With 8 – PSK three bits are encoded, forming

tribits and producing eight different output
phases.

With 8-PSK, n = 3, M = 8 and there are 8

possible output phases. To encode 8 different
phases, the incoming bits are encoded in groups
of 3 called tribits, 23 = 8.

8 PSK TRANSMITTER

Constellation diagram

The incoming serial bit stream enters the bit

splitter, where it is converted to a parallel, three
channel output ( the I or in-phase channel, the
Q or in quadrature channel, and the C or the
Because changes in the I channel occur at the control channel).
midpoints of the Q channel bits and vice versa, The bit rate in each of the channels is fb/3.
there is never more than a single bit change in
the dibit code and therefore there is never The bits in the I and C channels enter the I
more than a 90 degrees sift in the output phase. channel 2-to-4 level converter, and the bits in
the Q and C’ channels enter the Q channel 2-to-
In conventional QPSK, a change in the input 4 level converter.
dibit from 00 to 11 to 10 causes a
corresponding 180 degrees shift in the output
phase. Therefore, an advantage of OQPSK is the
The 8 PSK modulator
limited phase shift that must be imparted
The 2-to-4 level converters are parallel input 1/3 of the binary input data rate, fb/3. The I, Q
digital-to-analog converters. (DACs) and C bits are outputted simultaneously and in
parallel so the 2-to-4 level converters change
With two input bits, four output voltages are
inputs and outputs at a rate equal to fb/3.
possible.
The highest fundamental frequency in the , Q
The algorithm for the DACs is quite simple.
and C channel is equal to 1/6 the bit rate of the
The I and the Q bit determines the polarity of binary input( one cycle in the I, Q or C channel
the output analog signal ( logic 1 = +V and logic takes the same amount of time as six input bits)
0 = -V), whereas the C or C’ bit determines the
Also the highest fundamental frequency in
magnitude (logic 1 = 1.307 V and logic 0 = 0.541
either PAM signal is equal to 1/6 of the input
V).
binary bit rate
With two magnitudes and two polarities, four
different output conditions are possible.
Mathematically, the output of the balanced
modulator is:

Where

BANDWIDTH CONSIDERATION OF AN 8 PSK

And

Thus:

The output frequency spectrum extends from fc

+ fb/6 to fc – fb/6, and the minimum bandwidth
fn is

Because the data are divided into 3 channels,

the bit rate in the I, Q and C channel is equal to
8 PSK RECEIVER 16 PSK TRUTH TABLE AND CONSTELLATION
DIAGRAM

The power splitter directs the input 8 PSK signal

to the I and Q product detectors and the carrier
recovery circuit.

The carrier recovery circuit reproduces the

original reference oscillator signal.

The incoming 8 PSK signal is mixed with the

recovered carrier in the I product detector and
with a quadrature carrier in the Q product
detector.

The outputs of the product detectors are 4-level

PAM signals that are fed to the 4-to-2 level ADC.

The outputs from the I channel 4 to 2 level

converter are the I and C bits.

The outputs from the Q channel 4 to 2 level _______________________________________

converter are the Q and C’ bits.
DSP DEFINED
The parallel to serial logic circuit converts the
Digital Signal Processing (DSP) is used in a wide
I/C and Q/C’ bit pairs to serial I, Q and C.
variety of applications, and it is hard to find a
good definition that is general.

16 PSK  We can start by dictionary definitions of

the words:
- Is a M-ary encoding scheme where N =
4 and M = 16. Digital: operating by the use of discrete signals
to represent data in the form of numbers
- There are 16 different output phases
possible. Signal: a variable parameter by which
information is conveyed through an electronic
- Four bits called quadbits are combined circuit
producing 16 different output phases.
Processing: to perform operations on data
- The minimum bandwidth and baud according to programmed instructions
equal ¼ the bi rate, B = baud = fb/4
Which leads us to a simple definition of:
  Digital Signal processing: changing or  Since the early 1970s when the first DSP
analyzing information which is chips were introduced, the field of
measured as discrete sequences of digital signal processing has evolved
numbers dramatically.

Digital signal processing is concerned with (1)  With the tremendous rapid increase in
the representation of signals in digital form, (2) the speed of DSP processors, along with
and with the processing of these signals and the the corresponding increase in their
information that they carry. sophistication and computational
power, digital signal processing has
Although DSP began to flourish in the 1960s,
become an integral part of many
some of the important and powerful processing
commercial products and applications
techniques that are in use today may be traced
and is becoming a common place term.
back to numerical algorithms that were
proposed and studied centuries ago.

Since the early 1970s when the first DSP chips DSP VS ORDINARY DIGITAL PROCESSING
were introduced, the field of digital signal
Note two unique features of Digital Signal
processing has evolved dramatically.
processing as opposed to plain old ordinary
With the tremendous rapid increase in the digital processing:
speed of DSP processors, along with the
 signals come from the real world - this
corresponding increase in their sophistication
intimate connection with the real world
and computational power, digital signal
leads to many unique needs such as the
processing has become an integral part of many
need to react in real time and a need to
commercial products and applications, and is
measure signals and convert them to
becoming a common place term.
digital numbers

 signals are discrete - which means the

 Digital Signal processing: changing or information in between discrete
analyzing information which is samples is lost
measured as discrete sequences of
numbers
ADVANTAGES OF DSP
 Digital signal processing is concerned
with (1) the representation of signals in The advantages of DSP are common to many
digital form, (2) and with the processing digital systems and include:
of these signals and the information
that they carry. 1. Versatility:

 Although DSP began to flourish in the  digital systems can be reprogrammed

1960s, some of the important and for other applications (at least where
powerful processing techniques that are programmable DSP chips are used)
in use today may be traced back to  digital systems can be ported to
numerical algorithms that were different hardware (for example a
proposed and studied centuries ago. different DSP chip or board level
product)
2. Repeatability: distance), x(n) is generally referred to as
a function of time.
 digital systems can be easily duplicated
- Undefined for non-integer values of n
 digital systems do not depend on strict
component tolerances - a real valued signal x(n) will be
represented graphically by a “lollipop
 digital system responses do not drift
plot”
with temperature

3. Simplicity:

 some things can be done more easily

digitally than with analogue systems

APPLICATIONS OF DSP

DSP is used in a very wide variety of

applications.
In some problems and applications, it is
convenient to view x(n) as a vector. Thus the
sequence values x(0) to x(N-1) may often be
considered as the elements of a column vector
as follows:

but most share some common features: Discrete time signals are often derived by
Sampling a continuous time signal, such as
 they use a lot of math (multiplying and
speech, with an analog to digital converter.
adding signals)
For example, a continuous time signal xa(t) that
 they deal with signals that come from
is sampled at a rate of fs = 1/Ts samples per
the real world
second produces the sampled signal x(n) which
 they require a response in a certain is related to xa(t) as follows:
time

DISCRETE TIME SIGNAL

- an indexed sequence of real or complex Not all discrete time signals are obtained in
numbers. this manner. Some signals may be considered
as naturally occurring discrete time sequences
- Is a function of an integer-valued like the daily stock market prices, population
variable , n that is denoted by x(n). statistics, warehouse inventory and the Wolfer
Although the independent variable n sunspot numbers.
need not necessarily represent “time”
(n may represent spatial coordinate or
COMPLEX SEQUENCES

In general, a discrete time signal may be 2. Unit step

complex-valued.

A number of important applications such as

digital communications, complex signals arise
naturally.

A complex signal may be expressed either in

terms of its real and imaginary parts.
3. Exponential sequence
z(n) = a(n) + j b(n)
x(n) = an,
Or in polar form as z(n) = c(n) Ө(n), where c(n)
= a(n)2 + b(n)2 where a may be real or complex number

If z(n) is a complex sequence, the complex When a = ejӨ, then an = ejnӨ

conjugate, denoted by z*(n) is formed by: And following Euler’s identity: ejnӨ = cos nӨ + j
negating the sign of the imaginary sin nӨ
component as z*(n) = a(n) - j b(n)

or simple negating the sign of the The unit sample and unit step are related as:
argument as z*(n) = c(n) -Ө(n)

FUNDAMENTAL SEQUENCES

Although most information bearing signals of

practical interest are complicated function of
time, there are three simple yet important - the unit sample
discrete time signals that are frequently used in
the representation and description of more And the unit sample can be expressed as the
complicated signals. difference of two steps:

1. Unit sample

Or unit impulse

the unit samples

If we add the unit samples at different n’s, we

get the unit step
And the unit sample can be expressed as the PERIODIC AND APERIODIC SIGNALS
difference of two steps:
Periodic – if for some positive real integer N,

This is equivalent to saying that the sequence

repeats every N samples.
Other important sequences:
N – is called the fundamental period,
1. Signum function, sgn the smallest possible integer that will satisfy
Sgn = { 1 for n  0, 0 for n = 0, -1 for n  0 x(n) = x( n + N)

2. Constant sequence If the equation is not satisfied by any positive

integer N, then the sequence is called Aperiodic

SUM AND PRODUCT OF PERIODIC SIGNALS

SIGNAL DURATION If x1(n) is a sequence that is periodic with a

period of N1, and x2(n) is a sequence that is
Discrete time signals may be classified in terms periodic with a period N2, then the sum x(n) =
of their duration or extent. x1(n) + x2(n), and the product x(n) = x1(n)x2(n), is
1. Finite-length sequence – if it is equal to always periodic and period is:
zero for all values of n outside a finite N = N1N2/ gcd of N1 and N2
interval [N1, N2]
Where gcd is the greatest common divisor
2. Infinite-length sequence – such as the
unit step and the complex exponential Note: Given any sequence x(n), a periodic
signal may always be formed by replicating
a. Right sided sequence – is any infinite x(n) as:
length sequence that is equal to zero, x(n) = 0,
for all values of n  n0 for some integer n0

b. Left sided sequence - is any infinite

length sequence that is equal to zero, x(n) = 0,
for all values of n  n0 for some integer n0

SYMMETRIC SEQUENCES
c. Two sided sequence - is neither right A discrete time signal will often possess some
sided nor left sided, like the complex form of symmetry that may be exploited in
exponential solving problems.
 1. Even or Odd Symmetry (used for real SIGNAL MANIPULATIONS
sequences)
The study of discrete time signals (and systems)
2. Conjugate symmetric and is concerned with manipulation of signals.
antisymmetric (used for complex
1. Transformations of the independent
sequences)
variable n

2. Transformations of the amplitude of

EVEN AND ODD SYMMETRY x(n), the dependent variable

Even symmetry - A real-valued signal is said to

be even if, for all n
1. Transformations of the independent
x(n) = x(-n) variable n

Odd symmetry – if for all n Sequences are often altered and manipulated
by modifying the index n as follows:
x(n) = -x(-n)

Any signal x(n) may be decomposed by the sum

of its even part, xe(n) and its odd part xo(n) as

x(n) = xe(n) + xo(n)

Where f(n) is some function of n.
xe(n) = ½ [x(n) + x(-n)]
If for some value of n, f(n) is not an integer, y(n)
xo(n) = ½ [x(n) - x(-n)] = x(f(n)) is undefined.

Determining the effect of modifying n may

always be accomplished using a tabular
CONJUGATE SYMMETRIC and CONJUGATE
approach of listing, for each value of n and f(n)
ANTISYMMETRIC
and then setting y(n) = x(f(n)).
- a complex sequence is said to be
However, for many index transformations, this
conjugate symmetric (or also called
is not necessary. And the sequence may be
hermitian) if for all n
determined or plotted directly.
x(n) = x*(-n)
The most common transformation include (a)
- a complex sequence is said to be shifting, (b) reversal and (c) time scaling.
conjugate antisymmetric if for all n
a. SHIFTING
x(n) = - x*(-n)
Shifting – this is the transformation defined by
Note: x*(n) – conjugate of x(n) f(n) = n – n0.

Any complex signal may always be decomposed If y(n) = x(n - n0), x(n) is shifted to the left by n0
as sum of a conjugate symmetric signal and a samples if n0 is negative. This is called an
conjugate antisymmetric signal. advance.

The conjugate of a particular signal x(n) is:

xe(n) = ½ [x(n) + x*(-n)]

 c. (TIME) SCALING

Time Scaling – This transformation is defined by

f(n) = Mn or f(n) = n/N where M and N are
positive integers.

For f(n) = Mn, the sequence x(MN) is formed by

taking every Mth sample of x(n). This is called
A discrete time signal down-sampling.

With f(n) = n/N, the sequence y(n) = x(f(n)) is

defined as:
If y(n) = x(n - n0), x(n) is shifted to the right by n0
samples if n0 is positive. This is called delay.

This is called up-sampling.

A delay by n0 = 2

b. REVERSAL

Reversal – The transformation is given by f(n) =

-n and simply involves flipping the signal x(n)
with respect to the index n.

A discrete time signal

A discrete time signal

Down-sampling by a factor of 2

Time reversal
Up-sampling by a factor of 2
Note: Shifting, reversal and time scaling are Where each term in the sum x(k) (n-k) is a
order-dependent. signal that has an amplitude of x(k) at time n = k
and a value of zero for all other values of n.

DISCRETE TIME SYSTEMS

A discrete time system is a mathematical

operator or mapping that transforms one signal
( the input) into another signal (the output) by
means of a fixed set of rules or operations.

The notation T[.] is used to represent a general

2. Transformations of the amplitude of x(n)
system.
The most common types of amplitude
transformation are addition, multiplication and
scaling.

(a) Addition – The sum of two signals y(n) =

x1(n) + x2(n), -  n   is formed by the The input signal x(n) is transformed into the
pointwise addition of the signal values. output signal y(n) through the transformation
T[.].
(b) Multiplication - The multiplication of two
signals y(n) = x1(n) x2(n),   n   is formed by
the pointwise product of the signal values.
PROPERTIES OF DISCRETE-TIME SYSTEMS
(c) Scaling – amplitude scaling of the signal x(n)
1. Memoryless system
by a constant c is accomplished by multiplying
every signal value by c, as y(n) = c x(n), -  n  2. Additivity
. This operation may also be considered as the
3. Homogeneity
product of two signals x(n) and f(n) = c.
4. Linear systems

5. Shift invariance
SIGNAL DECOMPOSITION
6. Causality
The unit sample may be used to decompose an
arbitrary signal x(n) into a sum of weighted and 7. Stability
shifted unit samples as folows:
8. Invertibility

This decomposition may be written as:

1. Memoryless System

- A system is said to be memoryless if the

output at any time n = n0 depends only
on the input at time n=n0.
 - Or if for any n0, the value of y(n0) can be 2. y(n) = x(n) + x*(n-1)
determined using only the value of x(n0)
Determine if the system is:
EXAMPLE:
a. additive or not
Determine if the system is memoryless or not:
b. homogeneous or not.
2
a. y(n) = x (n)
ANSWER:
b. y(n) = x(n) + x(n-1)
1. y(n) = x2(n)/ x(n-1)
ANSWER:
a. The system is not additive because:
- Memoryless, because y(n0) depends
only on the value of x(n) at time n0.

- Not memoryless, because the output at

time y(n0) depends on the value of the is not the same as:
input both at time n0 and at time n0-1.

2. Additivity
b. The system is homogeneous because
- an additive system is one for which the
response to a sum of two inputs is equal
to the sum of the inputs individually.

T[x1(n) + x2(n)] = T[x1(n)] + T[x2(n)] 2. y(n) = x(n) + x*(n-1)

For any signals x1(n) and x2(n) a. The system is additive.

3. Homogeneity b. The system is not homogeneous.

- A system is said to be homogeneous if is not equal to

scaling the input by a constant results in
a scaling of the output by the same
amount. Specifically:

T[cx(n)] = cT[x(n)]

For any complex constant c and for any input 4. Linear Systems
sequence x(n). - A system that is both additive and
homogeneous is said to be linear.

EXAMPLE: T[a1x1(n)+a2x2(n)] = a1T[x1(n)] + a2T[x2(n)]

Consider the system: For any two inputs x1(n) and x2(n) and for any
complex constants a1 and a2
1. y(n) = x2(n)/ x(n-1)
Linearity greatly simplifies the evaluation of the - Let y(n) be the response of a system to
response of a system to a given input. an arbitrary input x(n). The system is
said to be shift invariant if for any delay
n0, the response to x(n – n0) is y(n –
EXAMPLE: n0).

Using the decomposition for x(n): - A system that is shift invariant do not
change its characteristics with time.

- A system that is not shift invariant is

said to be shift-varying.

- To test if the system is shift invariant,

compare y(n – n0) with T[x(n-n0). If
And using the additive property, the output y(n) they are the same for any input x(n),
may be written as: and for all shift n0, then the system is
shift invariant.

EXAMPLE:
Because the coefficients x(k) are constants, we
The system is defined by:
may use the homogeneity property to write:
y(n) = x2(n)

Determine if it is shift invariant.

ANSWER:
Letting hk(n) be the response of the system to a
unit sample time n=k, If y(n) = x2(n) is the response of the system to
x(n), then the response of the system to x’(n) =
x( n- n0) is:

y’(n) = [x’(n)]2 = x2(n-n0)

Then
Note that:

y’(n) = y(n-n0)

Therefore, the system is shift-invariant

Which is known as the superposition

6. Causality
summation.
- A system is said to be causal if for any
n0, the response of the system at time
5. Shift invariance n0 depends only on the input up to time
n = n0.
- If a system has the property that a shift
(delay) in the input by n0 results in a 7. Stability
shift in the output by n0, the system is
said to be shift invariant
 - A system is said to be stable in the Sampling: Discretizing the Time Dimension
bounded input-bounded sense if, for
• Definition: Sampling is the process of
any input that is bounded, x(n)  A
taking measurements of the analog
, the output will be bounded:
signal at regular intervals, creating a
y(n)  A  sequence of samples.

8. Invertibility • Sampling Theorem (Nyquist-Shannon):

- A system is said to be invertible if the • To avoid aliasing, the sampling

input to the system may be uniquely rate (fs) must be at least twice
determined from the output. the maximum frequency (fmax)
of the signal: fs≥2⋅fmax.
- In order for a system to be invertible, it
is necessary for distinct inputs to • Aliasing: If the sampling rate is
produce distinct outputs. too low, higher frequencies in
the signal can appear as lower
_______________________________________
Practical Example:
Sampling and Quantization
• In audio applications, the maximum
Analog vs. Digital Signals
frequency humans can hear is around
• Analog signals: Continuous in time (or 20 kHz, so the typical sampling rate for
space) and amplitude. audio is at least 40 kHz (often 44.1 kHz
or 48 kHz).
• Digital signals: Discrete in time and
amplitude, often easier to process and
more robust to noise.
Sampling a Signal
• Converting analog to digital is essential
in many systems, especially for
interfacing with sensors and actuators.

Analog-to-Digital Conversion Process

• Overview of the steps in converting an

analog signal to a digital signal:

• Sampling – Converting the

continuous-time signal to
discrete-time.

• Quantization – Converting Sampling Process

continuous amplitude values to
discrete levels. An analog signal is sampled at equally spaced
intervals, picking off the signal's value at specific
• Encoding (in PCM) – times.
Representing quantized values
in binary form. The interval between samples is T, so sample
times are nT where n = . . ., −2, −1, 0, 1, 2, . . ..
The sampled signal, x[n], is related to the
continuous signal by x[n] = x(nT).