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Analog and Digital Communication (ELE-323) Lecture # 11

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 Analog and Digital
Communication (ELE-323)

Lecture # 11
Dr. Uzma Nawaz 8/16/2020

Course Instructor: Dr. Uzma Nawaz

 Contents

• Applications of Sampling
• Digital Communication
• Pulse Code Modulation (PCM)
• Quantization
• Transmission Bandwidth and output SNR

Dr. Uzma Nawaz 8/16/2020

 Applications of Sampling
The sampling theorem is very useful in signal analysis, processing and transmission
because it allows us to replace a continuous time signal by a discrete sequence of
numbers.
 Processing a continuous time signal is therefore equivalent to processing a discrete
sequence of numbers called as digital filtering.
The transmission of continuous time signal reduces to the transmission of a sequence of
numbers i.e. communicating continuous time signal by pulse train.
 The continuous time signal g(t) is sampled, and sampled values are used to modify
certain parameters of periodic pulse train.
 Varying parameters like amplitude, widths, or position of pulses in proportion to
sampled values of signal g(t) for signal modulation. As a result,
 Pulse-amplitude modulation (PCM)
 Pulse-width modulation (PWM)
 Pulse-position modulation (PPM) and
8/16/2020
 Pulse-code modulation (PCM)
Dr. Uzma Nawaz
Applications of Sampling
Pulse Amplitude Modulation (PAM) is an analog modulating scheme in which the
amplitude of the pulse carrier varies proportional to the instantaneous amplitude of the
message signal.

Dr. Uzma Nawaz 8/16/2020

Applications of Sampling
Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM) or Pulse
Time Modulation (PTM) is an analog modulating scheme in which the duration or
width or time of the pulse carrier varies proportional to the instantaneous amplitude of
the message signal.

Dr. Uzma Nawaz 8/16/2020

 Applications of Sampling
Pulse Position Modulation (PPM) is an analog modulating scheme in which the
amplitude and width of the pulses are kept constant, while the position of each pulse,
with reference to the position of a reference pulse varies according to the
instantaneous sampled value of the message signal.

Dr. Uzma Nawaz 8/16/2020

 Applications of Sampling

Basic Structure of PAM / Basic Structure of PCM

PWM / PPM

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Applications of Sampling
PAM / PWM / PPM vs PCM
 PAM/PWM/PPM are systems where the information signal is typically discrete in time but
not necessarily in amplitude (not truly digital). Their properties are;
• Infinite number of waveforms can be sent
• They are Useful for time multiplexing multiple signals
• Noise readily degrades information
• Not particularly common
• PAM is the first step in PCM, thus is useful for study of PCM
 PCM are systems where the information must be discrete in time and amplitude. It has
following properties;
• Finite number of waveforms can be sent (i.e., digital)
• Requires both sampling and quantization
• Can be made more robust to noise
Dr. Uzma Nawaz 8/16/2020
 Applications of Sampling
In all cases of PAM, PWM, PPM instead of transmitting g(t), we transmit the corresponding
pulse-modulated signal. At the receiver, the information of pulse-modulated signal and
reconstruct the analog signal g(t).
 One advantage of pulse modulation is that it permits the simultaneous transmission of
several signals on a time-sharing basis time division multiplexing (TDM).
Pulse-modulated signal occupies only a part of channel time, we can transmit several
pulse-modulated signals on the same channel by interweaving them. In this manner we
can multiplex several signals on the same channel by reducing pulse widths.
 Another method of transmitting several baseband signals simultaneously is frequency
division multiplexing (FDM).
In FDM, various signals are multiplexed by sharing the channel bandwidth. The spectrum
of each message is shifted to a specific band not occupied by any other signal. The
information of various signals is located in non overlapping frequency bands of the
channel.
TDM share the timescale for the different signals; Whereas FDM shares the frequency
scale for the different signals.
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Dr. Uzma Nawaz

Applications of Sampling

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 Digital Communication

The basic function of a communication system is to transfer the information from source to
sink. The transmitter converts the message signal to a format suitable for transmission.
The message sent can be Analog or Digital and Baseband or Band pass.
 An analog message modulates a sinusoidal carrier (band pass).
 Digital signal can be obtained by converting an analog signal to a digital signal for digital
modulation of either a pulse stream (baseband) or a sinusoidal carrier (band pass).
 Analog Communication: The message signal can take on an infinite number of possible
values and directly uses an analog information source to be sent
 Digital Communication: The message signal must be one of a small number of discrete
messages. Must convert analog signals into a sequence of discrete messages.

Dr. Uzma Nawaz 8/16/2020

Pulse-Code Modulation (PCM)
PCM is the most useful of all pulse modulations.
 PCM is a method of converting analog signal into a digital signal (A/D conversion).
An analog signal can be converted into digital by sampling and quantization i.e. rounding off
its value to one of the closet permissible numbers (quantized levels).
 The amplitudes of analog signal m(t) lie in the range (-𝑚𝑝 , 𝑚𝑝 ) which is partitioned into L
subinterval each of magnitude ∆𝑣 = 2𝑚𝑝 /𝐿
 Each sample amplitude is approximated by the midpoint value of the subinterval in which
sample falls 𝐿 = 16.

Dr. Uzma Nawaz 8/16/2020

Pulse-Code Modulation (PCM)
 Each sample is now approximated to one of the L numbers. Thus, the signal is
digitized with samples taking on any one of the L values known as L-ary
digital signal.
 L-ary signal can be converted into a binary signal by using pulse coding. The
code formed by binary representation of 16 decimal digits from 0 to 15 is
known as natural binary code (NBC). Each of 16 levels to be transmitted is
assigned one binary code of four digits. The analog signal m(t) is now
converted into a binary digital signal.
 To transmit binary data, a distinct pulse shape is assigned to each of two
digits, binary 0 for negative pulse and binary 1 for positive pulse.

Dr. Uzma Nawaz 8/16/2020

 Digital Representation of Analog Signals

Analog Signal Sampled Signal Quantized Signal

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Pulse-Code Modulation (PCM)
The audio signal bandwidth is about 15kHz, the intelligibility is not affected if all
components above 3400 Hz are suppressed by low pass filter.
 In Telephone communication, the resulting audio signal is then sampled at a rate
of 8000 samples per second higher than Nyquist sampling rate of 6.8 kHz (to avoid
unrealizable filters for signal reconstruction).
Each sample is quantized into 256 levels (L=256), requires a group of eight binary
pulses to encode each sample 28 = 256, thus a telephone signal requires 8 x 8000 =
64000 binary pulses per second.
 Compact disc (CD) is a recent application of PCM, this is a high-fidelity situation
requiring the audio signal bandwidth to be 15 kHz.
Although Nyquist sampling rate is only 30 kHz, the actual sampling rate of 44.1 kHz.
The signal is quantized into a rather large number of levels (L=65,536) to reduce the
quantizing error. The binary coded samples are then recorded on CD.

Dr. Uzma Nawaz 8/16/2020

Quantization
Quantization is the process of mapping continuous infinite values to a smaller set of discrete finite
values.
 The amplitude of message signal m(t) is limited to the range (−𝑚𝑝, 𝑚𝑝 ) for quantization.
(𝑚𝑝 is not peak amplitude).
 The amplitudes of m(t) beyond ±𝑚𝑝 are chopped off thus, 𝑚𝑝 is not a parameter of the
signal m(t) but a constant of quantizer.
 The amplitude range (−𝑚𝑝, 𝑚𝑝 ) is divided into L uniformly spaced intervals in which it lies.
The quantized samples are coded and transmitted as binary pulses.
 At the receiver, some pulses will be detected incorrectly due to either quantization error or
pulse detection error.
In almost all practical schemes, the pulse detection error is quite small compared to the
quantization error and can be ignored.

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 Quantization
Consider the error in the received signal i.e. caused by quantization.
If 𝑚(𝑘𝑇𝑠 ) is the kth sample of signal m(t) and 𝑚(𝑘𝑇𝑠 ) is the corresponding quantized
sample then from interpolation formula,
𝑚 𝑡 = 𝑚 𝑘𝑇𝑠 𝑠𝑖𝑛𝑐(2𝜋𝐵𝑡 − 𝑘𝜋)
𝑘

𝑚 𝑡 = 𝑚 𝑘𝑇𝑠 𝑠𝑖𝑛𝑐 (2𝜋𝐵𝑡 − 𝑘𝜋)

𝑘

Where 𝑚 𝑡 is the signal reconstructed from quantized samples. The distortion component
q(t) in the reconstructed signal is,
𝑞 𝑡 =𝑚 𝑡 −𝑚 𝑡
𝑞 𝑡 = [𝑚 𝑘𝑇𝑠 − 𝑚 𝑘𝑇𝑠 𝑠𝑖𝑛𝑐(2𝜋𝐵𝑡 − 𝑘𝜋)
𝑘

𝑞 𝑡 = 𝑞(𝑘 𝑇𝑠 )𝑠𝑖𝑛𝑐(2𝜋𝐵𝑡 − 𝑘𝜋)

𝑘

Where q(𝑘𝑇𝑠 ) is the quantization error in the kth sample.

8/16/2020

Dr. Uzma Nawaz

The signal q(t) is undesired signal and acts as noise known as quantization noise.
 Quantization
Total power of quantization noise q(t) is:

Sampling interval is 2B, the

total number of samples Average or mean square
over the averaging of quantization error.
intervals T is 2BT.
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Dr. Uzma Nawaz

Quantization
The quantum levels are separated by,
∆𝑣 = 2𝑚𝑝 /𝐿
A sample value is approximated by the midpoint of the subinterval (of height ∆𝑣) in
which the sample falls.
∆𝑣
The maximum quantization error is ± . The quantization error lies in the range
2
∆𝑣 ∆𝑣
− , .
2 2
∆𝑣 ∆𝑣
Assuming that the error is equally likely to lie anywhere in the range − , , the
2 2
quantization error will be:

Dr. Uzma Nawaz 8/16/2020

Quantization
As 𝑞2 (𝑡) is the power of quantization noise and can be denoted as 𝑁𝑞 .
𝑚2 𝑝
𝑁𝑞 = 2
3𝐿
Assuming pulse detection error at the receiver is negligible, the reconstructed signal
𝑚 𝑡 at the receiver output is,
𝑚 𝑡 = 𝑚 𝑡 + 𝑞(𝑡)
The desired signal at the output is m(t), and the quantization noise is q(t).
The power of the message signal m(t) is,

𝑚𝑝 is the peak amplitude and is a

constant of quantizer.
𝑆𝑜
(SNR), is a linear function of the
𝑁𝑜
message signal power.

Dr. Uzma Nawaz 8/16/2020

Quantization
Nonuniform Quantization:
𝑆𝑜 𝑆𝑜
is an indication of the quality of received signal. Ideally (SNR) should be constant for
𝑁𝑜 𝑁𝑜
all values of message signal power.
Unfortunately, SNR is directly proportional to the signal power which varies from talker
to talker by as much as 40 dB
The signal power also varies due different lengths of connecting circuits so SNR can vary
depending on talker and length of circuit.
The quantization steps are of uniform value and quantization noise is directly proportional to
the square of step size. Using smaller steps for smaller amplitudes (non uniform
quantization) i.e. by first compressing signal samples and then using a uniform quantization
The input-output characteristics of compressor are;

On x-axis, is
normalized input and
output on y axis
Dr. Uzma Nawaz 8/16/2020
 Quantization
The compressor maps input signal increments ∆𝑚 into large increments ∆𝑦 for small input
signals and vice versa for large input signals.
The quantization noise is smaller for smaller input signal power.
 An approximately logarithmic compression characteristics yields a quantization noise
nearly proportional to the signal power making SNR practically independent of input
signal power over a large dynamic range.
 Using this approach to equalize SNR, loud talkers and stronger signals are penalized with
higher noise steps ∆𝑣 in order to compensate the soft talkers and weaker signals.
Two compression laws are used -law (North America, Japan)and A-law (Europe and rest of
World).
 Both laws have odd symmetry about vertical axis.
The -law (for positive amplitude) is given by,
1 𝜇𝑚 𝑚
𝑦= ln(1 + ) 0≤ ≤1
ln(1+𝜇) 𝑚𝑝 𝑚𝑝

Dr. Uzma Nawaz 8/16/2020

 Quantization
The A-law (for positive amplitudes) is
𝐴 𝑚 𝑚
0≤ ≤ 1/𝐴
1 + 𝑙𝑛𝐴 𝑚𝑝 𝑚𝑝
𝑦=
1 𝐴𝑚 1 𝑚
1 + 𝑙𝑛 ≤ ≤1
1 + 𝑙𝑛𝐴 𝑚𝑝 𝐴 𝑚𝑝
The compression parameter  or A determines the degree of compression.
𝑆𝑜
To obtain constant ,over an input-signal-power dynamic range of 40 dB, should be greater than 100.
𝑁𝑜

For A-law, the value of A=87.6 gives comparable results and has been standardized by CCITT

Dr. Uzma Nawaz 8/16/2020

 Quantization
The compressed samples must be restored to their original values at the receiver by using an
expander with characteristics complementary to that of compressor.
The compressor and expander together are called compandor.
Generally compression of signal increases bandwidth, but in PCM samples of signal are
compressed as samples doesn’t change so B does not increase.
The output SNR for non uniform quantization is = 255 and for uniform quantization = 0 as a
function of message signal power.

Dr. Uzma Nawaz 8/16/2020

Quantization
The Compandor:
A logarithmic compressor can be realized by a semiconductor diode, because the V-I
characteristics of such diode is of desired form,
𝐾𝑇 𝐼
𝑉= ln 1 +
𝑞 𝐼𝑠
Two matched diodes in parallel with opposite polarity provide approximate characteristics .
In practice, adjustable resistors placed in series with each diode and a third variable
resistor is added in parallel. By adjusting various resistors, the resulting characteristics are
made to fit a finite number of points (usually 7) on the ideal characteristics.
 An alternate approach is to use a piecewise linear approximation to the logarithmic
characteristics. A 15-segmented approximation to the eight bit (L = 256) with  = 255
law is widely used and is marginally inferior in terms of SNR
=255 compressor with =25 expander will be superior to piecewise linear devises (digital
terminal devices)
Combination of differing characteristics is inferior to either of the characteristics obtained
when compressor and expander are operating using same law. 8/16/2020

Dr. Uzma Nawaz

 Quantization
The Encoder:
The multiplexed PAM output is applied at the input of the encoder which quantizes and
encodes each sample into a group of n binary digits.
 A digit-at-a-time encoder makes n sequential comparisons to generate an n-bit code
word.
 The sample is compared with a voltage obtained by a combination of reference voltages
proportional to 27 , 26 , 25 , … . 20 .
 The reference voltages are conveniently generated by a bank of resistors R, 2R,
22 𝑅,…., 27 R.
The encoding involves answering successive questions beginning with whether or not the
sample is in the upper or lower half of the allowed range.
 The first code digit 1 or 0 is generated, depending on whether the sample is in the
upper or lower half of the range.
 In second step, another digit 1 or 0 is generated depending on whether the sample is in
the upper or lower half of subinterval in which it has been located.
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The process continuous until last binary digit in the code is generated.
Dr. Uzma Nawaz
Quantization
 Decoding is the inverse of encoding.
The kth digit is applied to a resistor 2𝑘 𝑅.
The currents in all the resistors are added. The sum is proportional to the quantized
sample value.
For a binary code word of 10010110, the current proportional to 27 + 0 + 0 + 24 + 0 +
22 + 21 + 0 = 150
This is the whole Digital to analog conversion complete process.

Dr. Uzma Nawaz 8/16/2020

Transmission Bandwidth & Output SNR
For a binary PCM, assign a distinct group of n binary digits (bits) to each of the L
quantization levels.
A sequence of n binary digits can be arranged in 2𝑛 distinct patterns,
𝐿 = 2𝑛 or 𝑛 = 𝑙𝑜𝑔2 𝐿
Each quantized sample is thus encoded into n bits.
 As a signal m(t) is band-limited to B Hz requires a minimum of 2B samples per
second, therefore a total of 2nB bits per second (bps), that is 2nB pieces of
information per second.
The unit of bandwidth (1 Hz) can transmit a maximum of two pieces of information per
second. The minimum channel bandwidth is given by;
𝐵𝑇 = 𝑛𝐵 𝐻𝑧
This is theoretical minimum transmission bandwidth, practically higher transmission
bandwidth is used.

Dr. Uzma Nawaz 8/16/2020

 Transmission Bandwidth & Output SNR
𝐿2 = 22𝑛 . The output SNR will be;
𝑆𝑜
= 𝑐(2)2𝑛
𝑁𝑜
Where,

The SNR increase exponentially with transmission bandwidth so there is a trade off between them. A
small increase in bandwidth yields a larger benefit In terms of SNR i.e.

Where 𝛼 = 10𝑙𝑜𝑔10 𝑐, i.e. increasing n by 1 (increase in one bits), quadruples the output SNR (6-dB)
Thus increasing n from 8 to 9, the SNR quadruples but transmission bandwidth increase from 32 to 36 kHz
i.e. 12.5% increase.
This shows that in PCM, SNR can be controlled by transmission bandwidth this makes PCM is superior to
FM or PM.
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Example 6.3

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