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Chapter 4

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0% found this document useful (0 votes)
33 views63 pages

Chapter 4

Uploaded by

abdi gm
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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Chapter 4

Base Band Pulse Signaling

〉 The sampling theorem


〉 T.D.M.
〉 Pulse Analog Modulation
〉 Quantizing and Encoding
〉 PAM, PCM, DPCM, DM
Sampling
• Process of converting a continuous time signal to an
equivalent discrete time signal.

• Continuous time signal x(t) is applied at input of


multiplier.

• Other input of multiplier is train of impulse.

• At the output of multiplexer we get the sampled version


of x(t)
Sampling
Sampling cont.
• One important property of sampling is that the signal can be fully
represented by their sampled value at some discrete time instant.

• If the sampled values are known the signal can be reconstructed or


recovered as original signal.

• The time interval can be used by other signals if a suitable


transmission is produced.
• The concept of use of time interval by several signal is known as
T.D.M. (Time Division Multiplexing)

• The sampling occurs at regular intervals of time Ts seconds apart.


Thus sampling frequency fs may be represented as
fs =1/Ts
Sampling

x(t ) . . =
x[n] x=
(t ) t =nT x(nT )
T
• Since p(t) is periodic with period T, it can be represented
by its Fourier series

p (t ) ∑=
k∈
jkω s t
ck e , ωs
T
sampling frequency
(rad/sec)
T /2
1

− jkω s t
where ck p (t )e dt , k ∈ 
T −T / 2
T /2
1 1

− jkω s t
= δ (t )e dt
T −T / 2 T
1 jkω s t
• Therefore p (t ) = ∑ e
k∈ T
Sampling – Cont’d
1 1
= (t ) p (t ) ∑ x(=
xs (t ) x= t )e jkω s t
∑ x(t )e jkω s t

k∈ T T k∈

1
X s (ω )
= ∑
T k∈
X (ω − kω s )
Ideal Sampling – Cont’d

X (ω )

1
X s (ω )
= ∑
T k∈
X (ω − kω s )
Sampling frequency
Aliasing effect

10
Sampling frequency
• In the case of voice transmission the standard sampling rate
is 8 KHz, it being just slightly more than twice the highest
significant frequency component.

• This implies a pulse rate of 8KHz or 125microsecond


period.

• Pulse duration of 1 microsecond may be adequate, it is


easy to see that a number of different message could be
multiplexed (TDM) on the channel.
TDM
• Time division multiplexing (TDM) system which enables
joint utilization of a common transmission channel.
TDM
• A low pass anti-aliasing filter remove the frequencies
which are nonessential to an adequate signal representation
at the transmitter.
• Then a commutator (sampler) that takes a narrow sample
of each of the N input signals to be transmitted.
• A similar synchronized rotating machine at the receiver is
used to distribute the N separate signals.
• 8KHz rate required for voice transmissions is achieved
using electronic switching system.
• Then the multiplexed signal is applied to a pulse modulator
which transforms the multiplexed signal into a form
suitable for transmission over the channel.
• At the receiving end of the system, the received signal is
applied a pulse demodulator, which performs the reverse
operation of the pulse modulator.
TDM
• There is a price paid for system gains obtained by pulse
modulation schemes. More important than the greater
equipment complexity is the requirement for greater
channel (bandwidth) size.
• If a maximum 3 KHz signal directly amplitude modulates a
carrier, a 6 KHz bandwidth is required.
• On the other hand, if a 1 microsecond pulse does the
modulating, just allowing its fundamental component of
1/1 micro second or 1MHz to do the modulating means a
2MHz bandwidth is required in AM.
• In spite of the large bandwidth required, TDM is still
preferable (if not the only possible way) to using 100
different transmitters, antennas or transmission lines, and
receivers in cases where large numbers of messages must
be conveyed simultaneously.
Pulse Analog Modulation
• Type of modulation in which the carrier consists of
(discrete) pulse trains, and some parameter of the pulse
train is varied in accordance with the message signal.
• The three types we consider here; pulse-amplitude
modulation (PAM), pulse width modulation (PWM), and
pulse position modulation (PPM).
• The pulse parameter that is varied in step with the analog
signal is varied in direct step with the signal's value at each
sampling interval.
• Notice that the pulse amplitude in PAM and pulse width in
PWM are not zero when the signal is minimum. This is
done to allow a constant pulse rate and is important in
maintaining synchronization in TDM systems.
Pulse Modulation

16
Pulse Amplitude Modulation (Pam)
• PAM can be defined as a process in which the amplitude
of regular spaced rectangular pulse vary in direct
proportion to the instantaneous sample values of
continuous signal.

• Its is quite similar to Amplitude Modulation.


• The difference is that here a Pulse Train acts as carrier
rather than high freq. sinusoidal wave.

• There are mainly 2 types of PAM signals


• PAM with Natural Sampling
• Pam with Flat Top

• PWM and PPM use constant amplitude pulses and


provide superior noise performance.
PAM with Natural Sampling
• Width of pulse do not have flat top.
• The top of pulse varies in accordance
with the shape of modulating signal.

• With the Natural Sampling, a signal


sampled at Nyquist rate may be
reconstructed exactly by passing
through LOW PASS FILTER with
cut off frequency fm .
• where fm is the highest frequency component

• If N signals are to be multiplied, the max


sampled duration is
T=Ts/N
PAM Natural Sampling Modulator

Pulse amplitude modulator, natural sampling.


• The FET is the switch used as a sampling gate.
• When the FET is on, the analog voltage is shorted to
ground; when off, the FET is essentially open, so that the
analog signal sample appears at the output.
• Op-amp 1 is a noninverting amplifier that isolates the
analog input channel from the switching function.
PAM-Natural Sampling Modulator

Op-amp 2 is a high input-impedance voltage follower


capable of driving low-impedance loads.

The resistor R is used to limit the output current of op-


amp 1 when the FET is “on” and provides a voltage
division with rd of the FET. (rd, the drain-to-source
resistance, is low but not zero)
Flat Top PAM
• The top of pulses of this PAM is flat.
• Noise interference at the top of transmitted pulse can be
easily removed. Due to this it is widely used.

• Better than Natural PAM


• Because in case of Natural PAM, the varying top signal is
when received at receiver, it becomes quite difficult to
determine shape of top of pulse due to noise (which is always
present).
• Thus errors are introduced in the receiving signals due to
wich we prefer a flat top PAM.
Flat Top PAM cont.
• Sampled and hold circuit consist of
2 flat switches and a capacitor.
• The sampling switch is closed for a
short duration by a short pulse
applied to gate G1 of transistor.
• During this period the capacitor is
charged up to a voltage equal to the
instantaneous value of input signal x(t)
• When the sampling switch is opened
the capacitor hold the charge.
• The discharge switch is then closed
by a pulse applied to the gate G2 pf
the other transistor.
• Due to this capacitor discharge to zero
volts.
• Hence the output sampled and hold
circuit consist of Flat Top Samples.
Pulse Time Modulation(PTM)
• There are two types of Pulse Time Modulation
• Pulse Width Modulation (PWM)
• Pulse Phase Modulation (PPM)

• In both PWM and PPM some time parameters of the pulse


is modulated.
• In PWM the width of pulse is varied.
• In PPM position of the pulse is varied.

• Amplitude of pulses remain constant for both PWM and


PPM
Pulse Time Modulation(PTM)
Pulse-Position Modulation (PPM)
• In pulse position modulation, the position of the pulse
relative to its unmodulated time of occurrence is varied
in accordance with the message signal.
• The information content is not contained in either the
pulse amplitude or width as in PAM and PWM, which
means the signal now has a greater resistance to any
error caused by noise.
• when PPM modulation is used to amplitude-modulate a
carrier, a power savings results since the pulse width
can be made very small unlike the case for PWM,
where long pulses expend considerable power while
bearing no additional information.
Demodulation of P.W.M.
• The transistor T1 acts as an inverter.
• During the time interval when the signal is high, the input of
transistor T2 is low.
• Thus during this interval T2 is in
cut-off stage.
• Thus capacitor C is charged through
RC combination.
• During the time interval when the
signal is low, the input of transistor is
high.
• Thus it get saturated during thus time.
• The capacitor gets discharge very rapidly through transistor T2.
• Hence the waveform at the collector transistor T2 is more or less a
saw-tooth waveform whose envelop is modulating signal.
• When this is passed through 2nd order Op-amp low pass filter,
desired demodulated signal is obtained.
Demodulation of P.P.M.
• This circuit makes use of the fact that the gaps between the pulses of
PPM signal contains information regarding the Modulating signal.
• During the gap between pulses
• The transistor is in cut-off.
• The capacitor gets charged through
RC combination.
• During the pulse duration
• The transistor is in saturation.
• Capacitor is discharged through
transistor.
• Thus the collector voltage becomes low.
• Hence the waveform at the collector is approximately a saw-tooth
waveform whose envelop is Modulating signal.
• When this is passed through 2nd order low pass filter, the desired
demodulated output is obtained.
Difference
PAM PWM PPM
• Amplitude of pulse is  Width of pulse is  Relative position of
proportional to proportional to pulse is proportional
amplitude of modulating amplitude of to amplitude of
signal. modulating signal. modulating signal.
• Band width of  Band width of  Band width of
transmitting channel
transmitting channel transmitting channel
depends on width of
pulse. depends on rise time depends on rise time
of the pulse. of the pulse.
• Instantaneous power of
transmitter varies.  Instantaneous power  Instantaneous power
of transmitter varies. remains constant.
• Noise interference is
high.  Noise interference is  Noise interference is
minimum. minimum.
• Complex system.
 Simple to implement.  Simple to implement.
• Similar to A.M.
 Similar to F.M.  Similar to P.M.
Pulse Digital Modulation
Pulse Code Modulation
• PAM is still an analog pulse modulation. It is not
completely digital because the amplitudes of the pulses
takes analog value.
• In analog pulse modulation, information is transmitted
in analog form, but the transmission takes place at
discrete times.
• If the message signal is represented in a form that is
discrete in both time and amplitude, then we have
digital pulse modulation.
• In digital pulse modulation, the signal transmission is
in digital form, as a sequence of coded pulses.
• In digital pulse modulation, PAM signals need to be
further digitized and then encoded for transmission.
Pulse Code Modulation
• Pulse-code modulation (PCM) is the most common
technique used today in digital communications,
representing an analog signal by a digital word.
• telephone system, digital audio recording (DAT or digital audio
tape), CD laser disks, digitized video special effects, voice mail,
and many other applications.
• PCM techniques and applications are a primary building
block for many of today's advanced communications
systems.
• Operations performed in PCM
• Sampling –flat top PAM signal respecting the Nyquist rule.
• Quantizing—discrete by rounding of continuous amplitude values
• Encoding—each level encoded to using binary representation—
and transmitted as pulses.
• Binary PCM (where the pulses have only two
permissible values) is the most common.

How to produce a PCM signal?

A PCM signal is produced by an analog-to-digital conversion process.


A complete Pulse code modulation (PCM) system

Quantization
• Quantization is the process of transforming the
sampled amplitude of a message signal into a discrete
level taken from a finite set of possible amplitudes.
How to perform the quantization?
1. The amplitudes of signal m(t) lie in the range (- mp, mp),
which is partitioned into L intervals, each of magnitude
∆ν = 2mp/L.
2. Each sample amplitude is approximated by the midpoint
value of the interval in which the sample falls.
m(t) Quantized samples of m(t) = mq(t)
mp (at the middle point of ∆ν)

∆ν

t
0
∆ν = 2m p/L
m(t)
L = number of intervals
- mp mq(t)
q(t)
Quantization of sampled analog signal
Encoding
• After sampling and quantization, the analog message signal
becomes discrete in values, but it is still not in the form
best suited for transmission.

• In order that the signal is best suited for transmission, i.e.


more robust to noise and interference, an encoding process
is required to translate the discrete samples to a more
appropriate form, such as the binary digits.
Encoding
Binary code
mp m(t)
111
110
101
100
2m p/L 011
010
001
L=8
000
-m p
Signal quantization and binary code assignment
7
6
Encoding
5

PAM signal

0 T T t

7 6 5
{

{
{
1 1 1 1 1 0 1 0 1
Coded samples

0 T T t

Binary coding of samples


• In quantization process introduces an error q(t), defined
as the difference between the message signal m(t) and
the corresponding quantized sample mq(t),
q(t) = m(t) – mq(t)
• This error is called the quantization noise.
• Quantization error depends on the step size. Hence if
the steps are uniform in size, small amplitude signals
will have a poorer signal-to-quantization noise ratio
than large-amplitude signals.
• To remedy this, non-uniform step size is used, small
step sizes for low level signals and larger step size for
large amplitudes.
• Non-uniform quantizer results in fewer steps than that
would be needed if uniform quantizer was used.
• A compression process can be used to compress large
amplitude signals before the quantizer.
• An inverse process, expanding, is used at the receiver to
reverse the compression. Companing!

Particular forms of compression laws that are used in


practice are the so-called μ-law and A-law. (Read about μ-
law and A-law).
Advantages of PCM:

• PCM signals derived from all types of analog sources


may be merged with data signals and transmitted over a
common high-speed digital communication system.
• In long-distance digital telephone systems requiring
repeaters, a clean PCM waveform can be regenerated at
the output of each repeater, where the input consists of a
noisy PCM waveform.
• The probability of error for the system output can be
reduced even further by the use of appropriate coding
techniques.
• Secure communication through the use of special
modulation schemes or encryption.
Transmission bandwidth
• Digital signals use much more bandwidth than analog
signals. This can be explained as follows:
• The binary digits must be transmitted in the sampling
interval originally allotted to one sample, the binary
pulse widths are correspondingly narrower and therefore
occupy a larger bandwidth according to the inverse
time–bandwidth relationship.
• The transmission bandwidth increases proportionately to
the number of binary pulses needed.
• For a binary PCM, a distinct group of binary digits (bits)
is assigned to each of the L quantization levels.
Transmission BW
• Each quantized sample is thus encoded into n bits.
• As n binary digits can be arranged in 2n distinct
patterns,
L ≤ 2n or n ≥ log2L
• According to sampling theorem, a signal m(t) band-
limited to B Hz requires a minimum of 2B samples per
second, a total of 2nB bits per second (bps) is required,
that is, 2nB piece of information per second.
• Because a unit bandwidth (1 Hz) can transmit a
maximum of two pieces of information (1 or 0) per
second, a minimum channel bandwidth is given by.
BT = nB Hz
• This is the theoretical minimum transmission BW
required to transmit the PCM signal as stated by Shanon.
Example:
Plain-old-telephone system (POTS)

Voice bandwidth limited to: 3.4 kHz


Babdwidth including guard band: 4 kHz
Sampling frequency: 8 kHz
Sampling rate: 8,000 samples/s
Coding: binary
8 bits per sample
(L = 28 = 256 quantization levels)
samples bits
Bit rate : 8000 ⋅8 = 64,000 bits / s .
s sample

Transmission Bandwidth = nB = 32kHz


What is the price paid for advantages gained by PCM

• Increased system complexity


• wider bandwidths requirement.

• The first issue of increased complexity is being dealt


with by cost efficient VLSI chips.
• Use of wideband communication satellites and fiber
optics has eased issue of bandwidth.
• Data compression techniques can be used to reduce
redundancy inherently present in PCM signal thereby
reducing the bit rate of transmitted data without
serious degradation in system performance.
Differential Pulse Code Modulation (DPCM)
• When audio or video signals are sampled, it is usually
found that adjacent samples are close to the same value.
• the difference signal is much less in amplitude than
the actual sample  less number of quantization
levels are needed. the number of bits per code is
reduced
• resulting in a reduced bit-rate the bandwidth
required in this case is less than the one required in
PCM.
Differential pulse code modulation (DPCM)
• to transmit PCM signals corresponding to the difference
in adjacent sample values.
If m[k] is the kth sample, we
Signal range
transmit the difference
d[k] = m[k] – m[k-1].

Difference
signal range

Sampled signal
m[k] + d[k] dq[k]
m(t) S Quantizer Encoder DPCM output
-
Delay m[k-1]
Sampling T Delayed samples
pulse train

At the receiver, knowing d[k] and the previous sample value m[k-1]
gives m[k].
DPCM encoder/decoder operation
Encoder
• The previous digitized sample of the analog input signal
is held in the register(temporary storage facility).
• The difference signal is computed by subtracting the
current contents of the register from the new digitized
sample output by the ADC.
• The value in the register is then updated by adding to
the current register contents the computed difference
signal output.
Decoder
• It operates by simply adding the received difference
signal(DPCM) to the previously computed signal held
in the register(PCM).

47
Differential Pulse Code Modulation (DPCM)
• The reconstructed signal may suffer from one or two
possible problems; Accumulation of quantization noise
and Effect of transmission errors.
• The first problem can be solved by using predictive
DPCM signal encoder and decoder.
• If we know the past behaviour of a signal up to a certain
point of time, it is possible to make some inference about
its future values based on the previous values.
• Such a process is known as prediction. The possibility
of prediction of future values of a signal is the basis of
differential pulse code modulation (DPCM) scheme
which involves encoding of the difference of successive
sample values.
Differential Pulse Code Modulation (DPCM)
• The reconstructed signal may suffer from one or two
possible problems; Accumulation of quantization noise
and Effect of transmission errors.
• The first problem can be solved by using predictive
DPCM signal encoder and decoder.
• Instead of quantizing the difference signal d[k] between
x[k] and its previous sample x[k–1], or
d [k ] = x [k ]− x [k −1],
• Quantizing the difference between x[k] and the previous
sample of its quantized form xq[k–1]
g[k] = x[k] − xq[k-1]
• Such a process is known as prediction. The possibility
of prediction of future values of a signal is the basis of
differential pulse code modulation (DPCM) scheme
• Lets follow the relationship of signals in the diagram
gq[k] = g[k]+q[k]
gq[k]= x[k] - xq[k-1] + q[k]
From the block diagram
xq[k]= gq[k] + xq[k-1]
=x[k]-xq[k-1]+q[k]+ xq[k-1]
=x[k]+q[k]
• This shows that xq[k] is quantized version of x[k]. Prediction!
50
DPCM decoder

• We can use the predictor block in the transmitter since


its input is the DPCM output gq[k] and its output is the
desired signal xq[k].

51
Delta Modulation
Delta Modulation
• In DPCM, if the signal change is represented by just one
bit, that bit being used for the sign of the sample
difference, then we have delta modulation (DM).
• In DM, since only 1 bit/sample is employed, it transmits
information to only indicate whether the analog signal it
encodes is to “go up” or “go down”.
• A single-bit PCM code to achieve digital transmission of
analog uses only 1 bit either logic ‘1’ or ‘0’.
• Logic ‘0’ is transmitted if current sample is smaller than
the previous sample.
• Logic ‘1’ is transmitted if current sample is larger than the
previous sample.
Delta modulation (DM)

Clock pulses

m(t) mq[k]

dq[k]

Illustration of delta modulation


Operation of Delta Modulation

55
Operation of Delta Modulation

56
Delta Modulation - Example

EKT343-Principles of Communication Engineering 57


Noise in Delta Modulation

58
Slope overload distortion occurs granular noise occurs when ∆ is
when ∆ is too small, the staircase too large relative to the local slope
approximation mq(t) can't follow characteristics of m(t).
closely the actual curve of the granular noise is similar to
message signal m(t ). quantization noise in PCM
large ∆ is needed for rapid variations small ∆ is needed for slowly varying
of m(t) to reduce the slope-overload m(t) to reduce the granular noise
distortion

59
Questions
1. How to sample a bandlimited analog signal to ensure distortion-
free reconstruction?
2. What is Nyquist rate and Nyquist interval?
3. What is time division multiplexing (TDM)?
4. How to perform quantization in a PCM system?
5. What is quantization noise?
6. What is the difference between PAM and PCM? What type of
signal PAM signal is? What type of signal PCM signal is?
7. What are the advantages of PCM?
8. What is non-uniform quantization?
9. To maintain the same S/N, which system requires smaller channel
bandwidth, PCM or DPCM?
10.How to obtain delta modulated (DM) signal?
Exercise Problems (Pulse Modulation)
1. For a given signal f(t) = cosω1t + cos2ω1t,
a) Draw the time waveform and the spectrum of the signal;
b) Determine the minimum sampling frequency.
2. If the signal f(t) = 10cos20πt cos200πt, and fs= 450Hz,
a) Determine the spectrum of the sampled signal;
b) If an ideal low-pass filter is used to recover f(t) from the
sampled signal, determine the bandwidth of the low-pass filter
required;
c) What is the Nyquist sampling rate for f(t)?
3. Five signals are combined to be transmitted in a TDM system, the
combined signals will pass through a low-pass filter. Three
channels are used to transmit the signals of frequency range
between 300 to 3300 Hz and the rest two channels transmit the
signals of 50 Hz to 10 kHz range
a) What is the minimum sampling rate required?
b) What is the minimum bandwidth of the low-pass filter required
corresponding to the minimum sampling rate?
4. A TDM system consists of 24 transmission channels and a
synchronization channel, a sampling rate of 8kHz is used. The
bandwidth of the signal for each channel is below 3.3kHz.
Determine the minimum channel bandwidth required to transmit
TDM signal in the system.
5. The information in an analog voltage waveform is to be
transmitted over a PCM system with a ±0.1% accuracy (full scale).
The analog waveform has an absolute bandwidth of 100 Hz and an
amplitude range of –10 to 10 V.
a) Determine the minimum sampling rate needed;\
b) Determine the number of bits needed in each PCM word;
c) Determine the minimum bit rate required in PCM signal;
d) Determine the minimum channel bandwidth required for
transmission of this PCM signal.
6. For a signal f(t) = 9 + Amcosωmt, with Am ≤ 10, to be quantized
into exactly 41 binary levels, with one level set at the smallest
value of f(t).
a. Determine the number of bits needed in each PCM word;
b. What are the values of extreme quantized levels Vmax and
Vmin if the quantized levels are centered to [f(t)max + f(t)min] /
2;
c. If Am = 10 V, find the signal to quantized noise ratio.
7. A speech signal with frequency range between 50 to 3300 Hz,
the sampling rate used is 8 kHz, the sampled signal is transmitted
through a PAM or PCM system.
a. Determine the minimum bandwidth required by PAM
system;
b. If a binary PCM system is used and the number of
quantization level is 8, determine the transmission channel
bandwidth;
c. If the number of quantization level is now 128, recalculate
the transmission channel bandwidth.

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