PCM PRINCIPLES

Need for PCM

To cater to the increased demand of traffic between two stations or between two subscribers at the same station.

Increasing number of pairs to the under ground cable is uneconomical and leads to maintenance problems.

Multiplexing
The technique used to provide a number of circuits using a single transmission link is called Multiplexing.

Multiplexing techniques
Frequency Division Multiplexing (FDM). Time Division Multiplexing (TDM).

Frequency Division Multiplexing

It is the process of translating individual speech circuits(300-3400 Hz) into pre assigned frequency slots within the transmission bandwidth. At input, 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 upper side band. Find their applications in the analogue transmission systems.
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Frequency division multiplexing

CHL
VF I/P F1 +VF or F1-VF

carrier

F1

BPF F2 +VF or F2-VF BPF F3 +VF or F3-VF

CHL
VF I/P CHL VF I/P carrier F3 FDM PRINCIPLE carrier F2

Transmission
link

BPF
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Time division multiplexing

Sharing a transmission medium by a number of circuits in time domain. Establishing a sequence of time slots during which individual channels (circuits) can be transmitted. Entire bandwidth is completely available to each channel. Each channel is assigned a time slot with a specific common repetition period called a frame interval. Each channel is sampled at a specified rate and transmitted for a fixed duration.
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Time division multiplexing

CH1 TRANS RECEIVE

GATE

GATE

CH1

CH2

medium CH2

CH3

CH3

Pulse code modulation systems

In 1938 , Mr. A. H. Reaves (USA) developed a Pulse Code Modulation (PCM) system to transmit the spoken words in the digital form. PCM systems use TDM technique to provide a number of circuits on the same transmission medium.
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Basic requirements for PCM systems To develop a PCM signal from several analogue signals, the following processing steps are required. 1. Filtering 2. Sampling 3. Quantization 4. Encoding 5. Line coding
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Filtering
Filters are used to limit speech signal to the frequency band of 300-3400 Hz.

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Sampling process

SWITCH S Wm R

V1

V2

(a)
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MAKE(S)

BREAK(S)

V1 V1

V2 V2

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 When the switch S is closed , an output appears across R. The rate at which switch S is closed is called the SAMPLING FREQUENCY. The amplitude of the sample is dependent upon the amplitude of the input signal at the instant of sampling.

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Sampling theorem
If a band limited signal is sampled at a 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. Fs 2Fh If our voice signals are band limited to the 4 KHz and let the sampling frequency be 8 KHz then the time period of the sampling, Ts = 1sec/8000. or Ts = 125 secs. 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.
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Sampling & Combining channels

CH1 a CH2 TRANSMISSION PATH a CH2 b CH3 c CH1

b
CH3 c

a
b c
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Sampling & Combining channels

The channels gates (a , b, c,n) correspond to the switch S. These gates are opened by series of pulses called CLOCK PULSES. These are called gates because they connect the channels to the transmission medium during clock period and isolate them during the OFF periods. The time interval during which the common transmission medium is allocated to a particular channel is called the TIME SLOT for that Channel. The width of the Time Slot depends on number of channels to be combined and the clock pulse frequency i.e. sampling frequency.
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30 CHANNEL PCM SYSTEM

125 microseconds are divided into 32 parts (Time Slots). 30 Time Slots are used for 30 speech channels. One time slot for signalling of all the 30 channels. One time slot for synchronization between transmitter and receiver. Time available per channel would be 125/32=3.9 seconds. Time period of sampling or the interval between two consecutive samples of a channel is 125 sec this duration is called TIME FRAME.
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PAM output signals

CHANNEL 1 A

CHANNEL 2 B

CHANNEL 3
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CONTINUE ->

MULTIPLEX SIGNAL D

PULSE TRAIN E SEPARATING CHANNEL 2

CHANNEL 2 SEPARATED F

RECONSTRUCTION OF ORIGINAL SIGNAL

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Quantization
The process of measuring the numerical values of the samples and giving them a table value in a suitable scale is called Quantising. In other words, Process of breaking continuous amplitude range into a finite number of amplitude values or steps. Used for converting PAM signals into digital form to avoid distortion.
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Quantization signal with

+Ve and Ve values

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DECIMAL

BINARY
1111 111O 1101 1100 1011 1010 1001 1000 0000 0001 0010 0011 0100 0101 0110 0111
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7 6 5 4 3 2 1 0 0 1 2 3 4 5 6 7

Illustration of Quantization Distortion

Analog Signal Amplitude Range 0-10 mV 10-20 mV 20-30 mV 30-40 mV Quantizing Interval (Mid Value) 5mV 15mV 25mV 35mV Quantizing Level 0 1 2 3 Binary Code

1000 1001 1010 1011

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Quantization Distortion
As in the Quantization in one level (step size) there are several amplitudes (one range) which are given one binary Code . Hence

The process of quantization leads to approximation of the input signals. The detected signal have some deviations from the actual values.
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 If v represents the step size & e represents the difference in amplitude between the actual signal level and its quantized equivalent, then it can be proved that Mean square quantizing error is equal to v/12. i.e. the error depends upon the size of the step. In linear quantization , equal step means equal degree of error for all input amplitudes. The S/N ratio for weaker signals will be poor.
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.
To reduce quantization distortion We need to reduce step size or in other words, increase the number of steps in the given amplitude range . B = Fm log L, where L is the number of quantum steps hence when we increase the no. of steps, bandwidth (B) increases.
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Non uniform quantization

Probability of occurrence of small amplitude is more than the probability of occurrence of large one. Providing more quantum levels in the small amplitude range. Providing few quantum levels in higher amplitudes.

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Companding
Also called segmented quantization. Provides non uniform quantization. Equal no. of segments for both +Ve and Ve excursions. To specify the location of the sample it is necessary to know the following: 1. Sign of the sample. 2. The segment number. 3. The quantum level within the segment.
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Segmented coding curve

128

112 QUANTIZATION LEVELS 96 80 64 48 32 16 1/8 1/4 1/2 1

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1/64

1/32 1/16

Encoding
Conversion of quantized analogue levels to binary signal is called Encoding. To represent 256 steps ,8 level code is used & is called an eight bit word. 8 bit word appears in the form P ABC WXYZ Polarity bit Segment code linear encoding in the segment 1 for +Ve 0 for -Ve
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Encoding Curve with Compression 8 Bit Code

N 128 112 7 96 80 5 6 64 4 48 32 16 -Vm/2 -Vm/4 Vm/4 Vm/2 Vc +Vm 16 V 32 MAXIMUM VOLTAGE 48 4 5 64 6 80 HERE Vc IS ENCODED AS 96 1 111 0101 112 128 SIGN SEGMENT POSITION BINARY NUMBER

-Vm

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From previous figure

The quantization and encoding are done by a circuit called CODER . CODER converts PAM signals to 8 bit binary signal. The curve has following characteristics The function N=f(V) is non linear. It is symmetrical about origin. Zero level corresponds to zero voltage to encoded. It is logarithmatic function. Have 13 straight segments 0 to 7 in +Ve direction 0 to 7 in Ve direction 4 segments 0,1(+) and 0,1(-) lying between +Vm/64 to Vm/64 being collinear are taken as one segment
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Time division multiplexing

COMBINER

SAPARATOR

a
CH2 b CH3

ENCODER

DECODER

CH1

TRANS SAMPLING SIGNALLING GATES PAM

REC SAMPLING GATES PAM CH1 a CH2 b CH3 n

SYNC a

Ts

b
c d SAMPLING PULSES FROM CHANNEL A TO N

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Concepts of frame
Ts in a 30 channel PCM system is 125 Secs. Signalling information of all the channels is transmitted through a separate time slot. To maintain synchronization between receive end and transmit end synchronization data is transmitted through another Time Slot. For a 30 channel PCM system, we have 32 time slots. Time available per channel is 3.9 secs. Frame duration is 125 seconds.
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Structure of frame
A frame of 125 seconds duration has 32 time slots. These time slots are numbered Ts 0 to Ts 31. Time slot Ts 0 carries synchronization signal. This slot is also called Frame Alignment Word (FAW). Time slot Ts 16 carries signalling information. Rest of all time slots carry speech signals.
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Synchronization
The receiver looks for FAW and once it is detected it knows that the information for channel 1 will be there and so on. The FAW is transmitted in the Ts 0 of every alternate frame. Frame not containing FAW, are used for carrying supervisory and alarm signals.

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Signalling in PCM systems

Signalling used for: Proper routing of call between two subscribers. Providing certain information like dial tone ,busy tone, ring back , NU tone, metering pulse, trunk offering signal etc.

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Characteristics of signalling information

Are in the form of DC pulses or multifrequency pulses. Signalling pulses retain their amplitude for a much longer period than the pulses carrying speech information. Signalling channel can be digitized with less number of bits than voice channel.

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Multi Frame Formation

Tb Tb=0.488s 1 time slot(8 bits) 3.9s 1 2 3 4 5 6 7 8

----------------- 15

1 frame(32 time slots)=125 s 16 ---------------------------- 29 30 31

--------------

One multi frame(16 frames) ----------------------------12 13 14 15 2mSecs

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Multiframe
Time slots 16 of each frame carries the signalling data corresponding to two VF channels only. For 30 channels we transmit 15 frames, each having 125 seconds duration. For synchronization one extra frame is used. These group of 16 frames is called multiframe The duration of multiframe is 2 msecs. Each of these frame has 32 time slots . Each time slot carrying the encoded samples of all the channels plus the signalling and synchronization data.
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TIME SLOTS FRAME 0 FRAME 1

0
FA

4 -----------------16 17
MFA S1/S16

---------------------------

31

FA

-----------------------S2/S17

FA

-----------------------S15/S30

FRAME
2 -----------------------------------------------FA FRAME 15 FA : Frame alignment MFA : Multiframe alignment Sn =Signalling for channel(4 bits)
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------------------------

Frame duration 125us

2.048 Mb/s PCM MULTIFRAME

PCM multiframe
We have 32 time slots in a frame, each slot carries a 8 bit word. The total number of bits per frame 32 x 8 = 256 bits. The total number of frames per seconds is 8000. The total number of bits per second are 256 x 8000 = 2048 Kbps. Thus, a 30 channels PCM systems has 2048Kbits.
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Multiframe Structure
In the time slot 16 of F0, the first four bits (positions 1 to 4) contain the multiframe alignment signal which enables the receiver to identify a multiframe. 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 F15 are used for carrying the signalling information. Each frame carry signalling data for 2 VF channels. As each multiframe includes 16 frames, each with a sampling rate 8000 per second, the signalling of each channel will occur at a rate of 500 per second.
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Thanks

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