PCM

PCM PRINCIPLES
BASIC LECTURE SERIES

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MULTIPLEXING TECHNIQUES
There are basically two types of multiplexing
techniques
i. Frequency Division Multiplexing (FDM)
ii Time Division Multiplexing (TDM)
7/18/2014 ALTTC:BASIC LECTURE SERIES
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FDM
Frequency Division Multiplexing Techniques (FDM)
The FDM techniques is the process of translating individual
speech circuits (300-3400 Hz) into pre-assigned frequency
slots within the bandwidth of the transmission medium. The
frequency translation is done by amplitude modulation of
the audio frequency with an appropriate carrier
frequency. At the output of the modulator a filter network
is connected to select either a lower or an upper side
band. Since the intelligence is carried in either side band,
single side band suppressed carrier mode of AM is used.
This results in substantial saving of bandwidth that also
permits the use of low power amplifiers.
FDM techniques usually find their application in analogue
transmission systems. An analogue transmission system is
one which is used for transmitting continuously varying
signals.

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FDM

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TDM
Basically, time division multiplexing involves
nothing more than sharing a transmission
medium by a number of circuits in time
domain by establishing a sequence of time
slots during which individual channels
(circuits) can be transmitted. Thus the entire
bandwidth is periodically available to each
channel. Normally all time slots are equal in
length. Each channel is assigned a time slot
with a specific common repetition period
called a frame interval.

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TDM
Each channel is sampled at a specified rate and
transmitted for a fixed duration. All channels are sampled
one by, the cycle is repeated again and again. The
channels are connected to individual gates which are
opened one by one in a fixed sequence. At the receiving
end also similar gates are opened in unision with the gates
at the transmitting end.
The signal received at the receiving end will be in the
form of discrete
samples and these are combined to reproduce the
original signal. Thus, at a given instant of time, only one
channel is transmitted through the medium, and by
sequential sampling a number of channels can be
staggered in time as opposed to transmitting all the
channel at the same time as in EDM systems. This
staggering of channels in time sequence for transmission
over a common medium is called Time Division
Multiplexing (TDM).

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7/18/2014 Course Name / Topic Name
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PCM
It was only in 1938, Mr. A.M. Reaves (USA)
developed a Pulse Code Modulation (PCM)
system to transmit the spoken word in digital
form. Since then digital speech transmission
has become an alternative to the analogue
systems.
PCM systems use TDM technique to provide a
number of circuits on the same transmission
medium viz open wire or underground cable
pair or a channel provided by carrier,
coaxial, OFC, microwave or satellite system.

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STEPS
To develop a PCM signal from several
analogue signals, the following processing
steps are required
Filtering
Sampling
Quantization
Encoding
Line Coding

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STEPS
FILTERING
Filters are used to limit the speech signal to the frequency band
300-3400 Hz.
SAMPLING
It is the most basic requirement for TDM. Suppose we have an
analogue signal which is applied across a resistor R through a
switch S. Whenever switch S is closed, an output appears across
R. The rate at which S is closed is called the sampling frequency
because during the make periods of S, the samples of the
analogue modulating signal appear across R. We get stream of
samples of the input signal which appear across R. The
amplitude of the sample is dependent upon the amplitude of
the input signal at the instant of sampling. The duration of these
sampled pulses is equal to the duration for which the switch S is
closed. Minimum number of samples are to be sent for any
band limited signal to get a good approximation of the original
analogue signal and the same is defined by the sampling
Theorem
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SAMPLING
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SAMPLING THEOREM
A complex signal such as human speech has a wide range of
frequency components with the amplitude of the signal being different
at different frequencies. To put it in a different way, a complex signal
will have certain amplitudes for all frequency components of which the
signal is made. Let us say that these frequency components occupy a
certain bandwidth B. If a signal does not have any value beyond this
bandwidth B, then it is said to be band limited. The extent of B is
determined by the highest frequency components of the signal.
Sampling Theorem States
"If a band limited signal is sampled at regular intervals of time and at a
rate equal to or more than twice the highest signal frequency in the
band, then the sample contains all the information of the original
signal." Mathematically, if fH is the highest frequency in the signal to be
sampled then the sampling frequency Fs needs to be greater than 2
fH.
i.e. Fs>2fH
Let us say our voice signals are band limited to 4 KHz and let sampling
frequency be 8 KHz.
Time period of sampling Ts = 1 sec
8000
or Ts = 125 micro seconds

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FRAME DURATION CONCEPTS
If we have just one channel, then this can be
sampled every 125 microseconds and the
resultant samples will represent the original
signal. But, if we are to sample N channels
one by one at the rate specified by the
sampling theorem, then the time available for
sampling each channel would be equal to
Ts/N microseconds.
Ts in that case in known as frame duration
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TUTORIAL-1
1. Who has developed Pulse Code Modulations in 1938?
a) A.M.Reevs.
b) Cromton Greevs.
c) Sugreev
2. Speech circuit is limited between
a) 300-3300 Hz
b) 300-3400Hz
c) 0-3400 Hz
3. Name five steps to make PCM?
4. Sampling theorem states that the sampling frequency should
be..to that of highest frequency components of the
signal.
a) Triple
b) Same
c) Double.

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SAMPLING
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SAMPLING TOO MANY CHANNELS
Previous figure shows how a number of channels can
be sampled and combined. The channel gates (a, b ...
n) correspond to the switch S . These gates are opened by
a series of pulses called "Clock pulses". These are called
gates because, when closed these actually connect the
channels to the transmission medium during the clock
period and isolate them during the OFF periods of the
clock pulses. The clock pulses are staggered so that only
one pair of gates is open at any given instant and,
therefore, only one channel is connected to the
transmission medium. The time intervals during which the
common transmission medium is allocated to a particular
channel is called the Time Slot for that channel. The width
of this time slot will depend, as stated above, upon the
number of channels to be combined and the clock pulse
frequency i.e. the sampling frequency.

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SAMPLING
In a 30 channel PCM system. TS i.e. 125 microseconds are
divided into 32 parts. That is 30 time slots are used for 30
speech signals, one time slot for signaling of all the 30
chls, and one time slot for synchronization between
Transmitter & Receiver.
The time available per channel would be Ts/N = 125/32 =
3.9 microseconds. Thus in a 30 channel PCM system, time
slot is 3.9 microseconds and time period of sampling
i.e..the interval between 2 consecutive samples of a
channel is 125 microseconds. This duration i.e. 125
microseconds is called Time Frame.
The signals on the common medium (also called the
common highway)of a TDM system will consist of a series
of pulses, the amplitudes of which are proportional to the
amplitudes of the individual channels at their respective
sampling instants
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PAM

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PAM RECOVERY
The original signal for each channel can be
recovered at the receive end by applying
gate pulses at appropriate instants and
passing the signals through low pass filters.
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QUANTIZATION
In FDM systems we convey the speech signals in
their analogue electrical form. But in PCM, we
convey the speech in discrete form. The sampler
selects a number of points on the analogue
speech signal (by sampling process) and
measures their instant values. The output of the
sampler is a PAM signal . The transmission of PAM
signal will require linear amplifiers at trans and
receive ends to recover distortion less signals. This
type of transmission is susceptible to all the
disadvantages of AM signal transmission.
Therefore, in PCM systems, PAM signals are
converted into digital form by using Quantization
Principles. The discrete level of each sampled
signal is quantified with reference to a certain
specified level on an amplitude scale.

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QUANTIZATION
Quantizing can be defined as a process of breaking
down a continuous amplitude range into a finite number
of amplitude values or steps.
A sampled signal exists only at discrete times but its
amplitude is drawn from a continuous range of
amplitudes of an analogue signal. On this basis, an infinite
number of amplitude values is possible. A suitable finite
number of discrete values can be used to get an.
approximation of the infinite set. The discrete value of a
sample is measured by comparing it with a scale having a
finite number of intervals and identifying the interval in
which the sample falls. The finite number of amplitude
intervals is called the "quantizing interval". Thus, quantizing
means to divide the analogue signal's total amplitude
range into a number of quantizing intervals and assigning
a level to each interval.

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QUANTIZATION
For example, a 1 volt signal can be divided
into 10mV ranges like 10-20mV, 30-40mV and
so on. The interval 10-20 mV, may be
designated as level 1, 20-30 mV as level 2 etc.
For the purpose of transmission, these levels
are given a binary code. This is called
encoding. In practical systems-quantizing
and encoding are a combined process. For
the sake of understanding, these are treated
separately.

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QUANTIZATION PROCESS

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QUANTIZATION PROCESS
Suppose we have a signal which is sampled
at instants a, b, c, d and e. For the sake of
explanation, let us suppose that the signal
has maximum amplitude of 7 volts.
In order to quantize these five samples taken
of the signal, let us say the total amplitude is
divided into eight ranges or intervals. Sample
(a) lies in the 5th range. Accordingly, the
quantizing process will assign a binary code
corresponding to this i.e. 101, Similarly codes
are assigned for other samples also. Here the
quantizing intervals are of the same size. This is
called Linear Quantization.

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ENCODING
Assigning an interval of 5 for sample 1, 7 for 2
etc. is the quantizing process. Giving, the
assigned levels of samples, the binary code
are called coding of the quantized samples.

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QUANTIZATION

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NUMBER OF LEVELS
Because the quantized samples are coded in
binary form, the quantization intervals will be
in powers of 2. If we have a 4 bit code, then
we can have 2" = 16 levels. Practical PCM
systems use an eight bit code with the first bit
as sign bit. It means we can have 2
8
= 256
(128 levels in the positive direction and 128
levels in the negative direction) intervals for
quantizing.

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QUANTIZATION DISTORTION
Analogue
Signal
Amplitude
Range
Quantizing
Interval
(mid value)
Quantizing Level Binary Code
0-10 mv 5 mv 0 1000
10-20mv 15mv 1 1001
20-30 mv 25 mv 2 1010
30-40 mv 35 mv 3 1011
40-50 mv 45 mv 4 1100
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QUANTIZATION ERROR
To reduce error due to quantization, we, need to reduce
step size or in other words, increase the number of steps in
the given amplitude range. This would however,
increase the transmission bandwidth because
bandwidth B = fm log L. where L is the number of quantum
steps and fm is the highest signal frequency.
But as we knows from speech statistics that the probability
of occurrence of a small amplitude is much greater than
large one, it seems appropriate to provide more quantum
levels (V = low value) in the small amplitude region and
only a few (V = high value) in the region of higher
amplitudes.
In this case, provided the total number of specified levels
remains unchanged, no increase in transmission
bandwidth will be required. This will also try to bring about
uniformity in signal to noise ratio at all levels of input signal.
This type of quantization is called non-uniform
quantization.

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NON UNIFROM QUANTIZATION
In practice, non-uniform quantization is
achieved using segmented quantization (also
called companding).
There is equal number of segments for both
positive and negative excursions. In order to
specify the location of a sample value it is
necessary to know the following:
The sign of the sample (positive or negative
excursion)
The segment number
The quantum level within the segment

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SEGMENTS

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ENCODING

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TDM SCHEMATIC

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CONCEPT OF FRAME
Since Ts is much larger as compared to the pulse width x
of one channel hence a number of channels can be
sampled each for a duration of x within the time Ts. As
per the previous figure, the first sample of the first channel
is taken by pulse 'a', encoded and is passed on to
combiner.
Then the first sample of the second channel is taken by
pulse 'b' which is also encoded and passed on to the
combiner, Likewise the remaining channels are also
sampled sequentially and are encoded before being fed
to the combiner. After the first sample of the Nth channel
is taken and processed, the second sample of the first
channel is taken, this process is repeated for all channels.
One full set of samples for all channel taken within the
duration Ts is called a "frame".
Thus the set of all first samples of all channels is one frame;
the set of all second samples is another frame and so on.

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CONCEPT OF FRAME
For a 30 chl PCM system, we have 32 time
slots.
Thus the time available per channel would be
3.9 microsecs.
Thus for a 30 chl PCM system,
Frame = 125 microseconds
Time slot per chl = 3.9 microseconds.

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STRUCTURE OF THE FRAME
A frame of 125 microseconds duration has 32
time slots. These slots are numbered Ts 0 to Ts
31. Information for providing synchronization
between trans and receive ends is passed
through a separate time slot. Usually the slot
Ts 0 carries the synchronization signals. This slot
is also called Frame alignment word (FAW).
The signaling information is transmitted
through time slot Ts 16. Ts 1 to Ts 15 are utilized
for voltage signal of channels 1 to 15
respectively. Ts 17 to Ts 31 are utilized for
voltage signal of channels 16 to 30
respectively.

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ALARM AND SYNCH
Frame Remark
Numbers B1 B2 B3 B4 B5 B6 B7 B8
FO X 0 0 1 1 0 1 1 FAW
F1 X 1 Y Y Y 1 1 1 ALARM
F2 X 0 0 1 1 0 1 1 FAW
F3 etc X 1 Y Y Y 1 1 1 ALARM
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SYNCHRONIZATION
The output of a PCM terminal will be a continuous stream of bits. At the
receiving end, the receiver has to receive the incoming stream of bits and
discriminate between frames and separate channels from these. That is, the
receiver has to recognize the start of each frame correctly. This operation is
called frame alignment or Synchronization and is achieved by inserting a fixed
digital pattern called a "Frame Alignment Word (FAW)" into the transmitted bit
stream at regular intervals. The receiver looks for FAW and once it is detected, it
knows that in next time slot, information for channel one will be there and so on.
The digits or bits of FAW occupy seven out of eight bits of Ts 0 in the following
pattern.
Bit position of Ts 0 B1 B2 B3 B4 B5 B6 B7 B8
FAW digit value

X 0 0 1 1 0 1 1
The bit position B1 can be either '1' or '0'. However, when the PCM system is to
be linked to an international network, the B1 position is fixed at '1'.
The FAW is transmitted in the Ts 0 of every alternate frame.
Frame which do not contain the FAW, are used for transmitting supervisory
and alarm signals. To distinguish the Ts 0 of frame carrying supervisory/alarm
signals from those carrying the FAW, the B2 bit position of the former are fixed at
1. The FAW and alarm signals are transmitted alternatively
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ALARMS
In frames 1, 3, 5, etc, the bits B3, B4, B5
denote various types of alarms. For example,
in B3 position,
if Y = 1, it indicate Frame synchronization
alarm.
If Y = 1 in B4, it indicates high error density
alarm.
When there is no alarm condition, bits B3 B4
B5 are set 0.
An urgent alarm is indicated by transmitting
"all ones". The code word for an urgent alarm
would be of the form.X= 1111111

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SIGNALING
In a telephone network, the signaling information is
used for proper routing of a call between two
subscribers, for providing certain status information
like dial tone, busy tone, ring back, NU tone, metering
pulses, trunk offering signal etc. All these functions are
grouped under the general terms "signaling" in
PCM systems
The time slot 16 of each frame carries the
signaling data corresponding to two VF channels
only. Therefore, to cater for 30 channels, we must
transmit 15 frames, each having 125 microseconds
duration. For carrying synchronization data for
all frames, one additional frame is used. Thus a
group of 16 frames (each of 125 microseconds) is
formed to make a "multi-frame". The duration of a
multi-frame is 2 milliseconds. This is signaling speed as
well for each of the channel.
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MULTIFRAME FORMATION

MICROSECOND
3.9 MICROSECOND
125 MICROSECOND
MILLISECOND
8
42 42 42
MULTIFRAME FORMATION

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ESSENCE
We have 32 time slots in a frame; each slot
carries an 8 bit word.
The total number of bits per frame = 32 x 8 =
256
The total number of frames per seconds is
8000
The total number of bits per second is 256 x
8000 = 2048 K/bits.
Thus, a 30 channel PCM system has 2048 K
bits/sec.

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MUTLIFRAME STRUCTURE
In the time slot 16 of FO, the first four bits (positions 1
to 4) contain the multi-frame alignment signal which
enables the receiver to identify a multi-frame. The
other four bits (no. 5 to 8) are spare. These may be
used for carrying alarm signals. Time slots 16 of frames
F1 to FT5 are used for carrying the signaling
information. Each frame carries signaling, data for
two VF channels. For instance, time slot Ts 16 of frame
F1 carries the signal data for VF channel 1 in the first
four bits. The next four bits are used for carrying
signaling information for channel 16. Similarly, time
slot Ts16 of F2 carries signalling data of chls 2 and 17.
Thus in multi-frame structure, four signaling bits are
provided for each VF channels. As each multi-frame
includes 16 frames, so the signaling of each channel
will occur at a rate of 500 per sec.

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TUTORIAL-2
1. What is the frame duration in PCM( in Microsecond)?
a) 125
b) 225
c) 3.9
2. What is the multiframe duration in Milliseconds
a) 5
b) 500
c) 2
3. What is the signaling speed in bits per second?
a) 200
b) 500
c) 300
4. Each channels is of ..Kb/s
a) 32
b) 16
c) 64
5. Why non linear quantization is used?
a) To avoid burst
b) To lessen Quantization error
c) To increase latency

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TUTORIAL-2
6. What is the time taken by one channel in the PCM frame (in
Microsecond)?
a) 125
b) 3.6
c) 3.9
7.What is the time taken by one bit in the PCM frame(in Nanoseconds)
a) 587
b) 409
c) 488
8. FAW is transmitted in
a) Every Frame
b) Alternate Frame
c) Once in 16 Frame
9. Signaling is transmitted in . Time slot of the frame.
a) 1
st

b) 32
nd

c) 16
th

THANK YOU! THANK YOU!

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PCM PRINCIPLES

BASIC LECTURE SERIES

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