Section 3.

SECTION 3.2
FUNDAMENTAL OF DIGITAL TRANSMISSION & PRINCIPLE OF HIGHER ORDER MULTIPLEXING

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FUNDAMENTAL OF DIGITAL TRANSMISSION

1.0 INTRODUCTION
Basically there are two ways in which information of any type can be transmitted over telecommunication media analog or digital. Analog means that the amplitude of the transmitted amplitude signal varies over a continuous range. Digital transmission means that a stream of on/off pulses are sent on the transmission media. The pulses are referred to as bits. Examples of analog signals are human voice, hifi music, temperature reading, etc. While that of digital are data, telegraphy signals. Telecommunication systems started with the transmission of digital signals. In fact, nonelectric signaling systems date back over 2000 years. The Greek General Polybius is known to have used a scheme based on an array of 10 torches in 300 B.C. and Roman armies made extensive use of a form of samaphore signaling. Claude Chappe, Sommering, Wheatstone and Cook were all experimenting with different kinds of Telegraphy till it was perfected by Mores. In all this, only written message was transmitted and message was converted to a coded signal to match the characteristics of a transmission line. Gary, Bandot and others developed other codes which were mainly used in Telegraph network. Thus, we can say, by 1972 most of the basic techniques of digital transmission had been discovered. In 1876, Alexander Graham Bell invented the Telephone and as means of communication, the telephone was fast, personal and convenient. It needed no training in the use of codes and so made electrical communications directly accessible to the general public. Thus, telephone began to dominate the development of communications. Telephony involves the transmission of analog signals and when a practical amplifying service appeared in the form of the thermionic valve, this also proved suitable for dealing with analog signals. Hence, after 1880, the developing Telecom networks were basically designed to handle analog transmissions and to an increasing extent, the digital transmission in the form of telegraphy had to be adopted to fit in with the characteristics of these networks. By 1950s, the world's communications systems were based entirely on analog transmission. However, interest in the digital transmission received an impetus after the publications of classic papers of Nyquist and Shannon. With the invention of pulse code modulation by Reeves in 1938, the basic principles for digitizing analog speech signals were established. However, the technical means for transmitting digitized speech signals were not available at that time. It was not until the transistor came into use that indications of the economic advantages of digital techniques as compared to analog methods became apparent. LSI and VLSI techniques that are now available have made digital communications far more economical as compared to analog methods became apparent. LSI and VLSI techniques that are now available have made digital communications far more economical as compared to analog systems. Digital transmission systems are gaining more acceptance in view of : (1) introduction of digital switching systems, (2) the need to transmit non voice signals which are increasingly becoming important instead of the plain old Telephone service, and (3) the introduction of new media like optical fibres, waveguide which are more suitable for digital transmission systems, will be introduced in the network and by the turn of the century, most of the countries would have gone completely digital.

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2.0

WHY DIGITAL?

Why indeed is communication more and more digitally based? When the stereo shop says its products are digital quality, is this an advertising slogan, or is digital really better? Let us look at the slogans and the reality and see why this revolution is really occurring. There are many solid reasons, and here they are in rough order of importance. i.) Quality Control: Fig.1 shows the qualitative representation of the signal to noise ratio along a transmission line. In both analog and digital systems the signal power P is subject to line attenuation which can be compensated by repeaters..

Fig. 1 Signal to Noise Ratio along a XMission Path However, a main difference exists in the accumulated noise power N. In the transmission of analog signals, this power Na is amplified in linear repeaters by the same factor as the useful signal and the noise contributions from the individual repeater section accumulate. In the digital transmission on the other hand, the signal is practically achieved of the noise Nd with the aid of regenerative repeaters. Residual noise may only become effective in the form of digital errors and jitter due to regeneration, reshaping and retiming (3 Rs.) carrier out section by section, only the digital errors are accumulated while the noise is not. The need to recognize only the presence or absence of a pulse makes the system highly immune from noise. Thus, the transmission quality is almost independent of distance and method of transmission involved. This is of particular value in transmission paths subject to extreme

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interference such as for instance in space flights or in communications with interplanetary probes ii.) Compatibility of different media: Cables, radio links, switching equipment can be interconnected without decoding the digital signals by means of relatively cheap interface equipment which contributes little or no impairment to the signal. There is thus no need to take any consideration of the particularities of the original signal. iii.) Compatibility of different traffic : Any digital media of suitable capacity can carry encoded speech, telephone signaling, telegraphy, digital data, encode visual information or an arbitrary mixture there of. The desperate requirements of these signals can be handled in the terminals and have no mutual interference between different types of traffic. The introduction of ISDN is thus possible. iv.) Multiplexing, demultiplexing, branching of digital signals produce no additional interference as noise in analog communications. Hence, these can be done as often as necessary. Moreover, all bits are subject to same interference and hence all TDM channels are treated equally, i.e. there are no channels of inferior quality as for instance in FDM transmission certain channels at the edges of the transmission bands. v.) Level fluctuations occurring during transmission have no effect on the primary signal recovered in the receiver. In FDM, however, sophisticated equipments are required to maintain the level more or less constant. vi.) Economies in certain applications: PCM is inherently cheaper than the FDM and the investment needed can be made progressively as the traffic growth justifies it. Economies can be achieved by combining services already of a digital nature. Digital signals can be switched by digital exchanges without demodulation. vii.) Possibility of novel facilities: The digital mode lends itself to such things as cryptography, storage and various forms of digital processing not accomplished otherwise. viii.) Cheap Hardware. First and foremost, digital hardware has become very cheap, as we have just stressed. This makes all the other advantages cheap to buy ix.) Applicability to extremely difficult transmission paths. x.) Simpler equipment: There is no need of complicated filter and analog amplifiers for various ranges. xi.) Easy repeatability of design.

3.0

MAIN OBSTACLES TO DIGITALISATION

(a) Spectrum width: For example the bandwidth required for 2700 channels is 12 MHz in the case of analog systems where as band width required in the case of 1920 channels is as much as 140 MHz. Thus, band width required is very large in the case of digital signals, this results consequently : (i) (ii) Less efficient use of carrier capacity in terms of telephone channels; Working at very high frequencies;

(iii) Need of multilevel modulation for radio transmission;

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(iv) Voice interpolation required for satellite communication; (v) Higher sensitivity to selective transmissions caused by propagation. (b) Different transmission of TV signals : Digital transmission of TV signals requires a very wideband if redundancy reduction is not used which, however, involve higher cost and quality problems for moving images. (c) Reliability and power consumption : For the same transmitted signals, digital transmission equipments are in general more complex than analog ones. Eqpt. Line repeaters (12 MHz Vs 140 Mb/s) 1+1 Radio repeater (1800 FDM Vs 140 Mb/s) Analog 2W Digital 4W

200 W

600 W

4.0

Analog and Digital Signals

It is electrical, electronic or optical representation of data, which can be sent over a communication medium. Stated in mathematical terms, a signal is merely a function of the data. For example, a microphone converts voice data into voice signal, which can be sent over a pair of wire. Analog signals are continuous-valued; digital signals are discrete-valued. The independent variable of the signal could be time (speech, for example), space (images), or the integers (denoting the sequencing of letters and numbers in the football score). Figure 2 shows an analog signal.

Figure 2 Analog signal Digital signal can have only a limited number of defined values, usually two values 0 and 1, as shown in Fig. 3.

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Figure 3 Digital signal Digital Signals- A digital signal is a discrete signal. It is depicted as discontinuous (Fig. 3) -Discretely variable (on/off) as opposed to an analog signal which is continuously variable (sine wave) A digital signal has the following characteristics: 1.) Holds a fixed value for a specific length of time 2.) Has sharp, abrupt changes 3.) A preset number of values allowed Each pulse (on/off) is known as a bit. Bit is a contraction of the words binary and digit A binary (two-level) signal (1 or 0) is the most common digital signal in the telecommunication industry. The number of bits transmitted per second is the bit rate of the signal. To convert analog signals to digital signals, a coding system called Pulse Code Modulation or PCM is used. This process is also called Analog-to-Digital, or A/D, conversion. When changing a digital signal to an analog signal, the process is called Digital-to-Analog, or D/A, conversion.

5.0

CHANNEL CAPACITY OR INFORMATION RATE

In general, the capacity of a channel for information transfer is proportional to its bandwidth. Two major theories that relate to the amount of data that can be transmitted based upon the bandwidth of a medium are the Nyquist Relationship and Shannon's Law. Prior to discussing these theories, it is important to understand the difference between bit and baud due to the confusion that dominates the use of these terms. 5.1 Bit versus baud The binary digit or bit is a unit of information transfer. In comparison, the term baud defines a signaling change rate, normally expressed in terms of signal changes per second. In a communications system, the encoding of one bit per signal element results in equivalency between bit and baud. That is, an information transfer rate of X bits per second is carried by a signaling change rate of X baud, where each baud signal represents the value of one bit. Now, suppose our communications system was modified so that two bits are encoded into one signal change. This would result in the baud rate being half the bit rate, which obviously makes bit and baud nonequivalent. The encoding of two bits into one baud is known as dibit encoding.
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5.2

Nyquist relationship

In 1928, Harry Nyquist developed the relationship between the bandwidth and the baud rate on a channel as B = 2W where B is the baud rate and W the bandwidth in Hz. The Nyquist relationship was based upon a problem known as inter-symbol interference which is associated with bandlimited channels. If a rectangular pulse is input to a bandlimited channel, the bandwidth limitation of the channel results in a rounding of the corners of the pulse. These rounding results in the generation of an undesired signal in which the leading and trailing edges formed due to signal rounding can interfere with both previous and subsequent pulses. This signal interference is illustrated in Fig.4.

Fig. 4 Pulse response through a bandlimited channel. The bandwidth limitation of a channel causes the leading and trailing edges of a pulse to interfere with other pulses as the signal change exceeds twice the bandwidth of a channel. This condition is called inter-symbol interference. The Nyquist relationship states that the rate at which data can be transmitted prior to inter-symbol interference occurring must be less than or equal to twice the bandwidth in Hz. Thus, an analog circuit with a bandwidth of 3000 Hz can only support baud rates at or under 6000 signaling elements per second. Since an oscillating modulation technique such as amplitude, frequency or phase modulation halves the achievable signaling rate, a twisted pair telephone circuit supports a maximum signaling rate of 3000 baud. 5.3 Shannon's law In 1948, Claude E. Shannon presented a paper concerning the relationship of coding to noise and calculated the theoretical maximum bit rate capacity of a channel of bandwidth W Hz. The relationship developed by Shannon is given by C = W log2 (1+S/N) where C = capacity in bits per second,
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W S N 5.4

= = =

bandwidth in Hz, Signal power at the receiver input power of thermal noise = No.W

Symbols, bits and bauds

A symbol is quite apart from a bit in concept although both can be represented by sinusoidal or wave functions. Where bit is the unit of information, the symbol is a unit of transmission energy. It is the representation of the bit that the medium transmits to convey the information. Imagine bits as widgets, and symbols as boxes in which the widgets travel on a truck. We can have one widget per box or we can have more. Packing of widgets (bits) per box (symbols) is what modulation is all about. In communications, the analog signal shape, by pre-agreed convention, stands for a certain number of bits and is called a symbol.

Figure 5 Digital information travels on analog carrier A symbol is just a symbol. It can stand for any number of bits, not just one bit. The bits that it stands for are not being transmitted, what is transmitted is the symbol or actually the little signal packet shown above. The frequency of this packet is usually quite high. The 1 Hz signal shown above is just an abstraction. A baud is same as the symbol rate of a communication system. So if we send 200 bauds, then we are send 200 symbols per second.

6.0

DIGITAL MODULATION TECHNIQUES

In order to transmit digital signals over Radio systems. It is necessary to transfer the information to the Radio frequency carrier. Modulation is the process of facilitating the transfer of information over a medium. Sound transmission in air has limited range for the amount of power your lungs can generate. To extend the range your voice can reach, we need to transmit it through a medium other than air, such as a phone line or radio. The process of converting information (voice in this case) so that it can be successfully sent through a medium (wire or radio waves) is called modulation. Digital, information can be imposed upon the carrier by modifying the amplitude, frequency, phase or a combination of these characteristics, The choice of the modulating scheme is made after considering a number of conflicting requirements, which include susceptibilities to noise interference, fading, non linearities, spectrum efficiency (i.e. Bits/sec/Hz) and equipment complexities with associated
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cost aspect. The spectrum efficiency is a ratio of bit speed (say R bits per second) and band width say B Hz. This ratio i.e, R/B is known as the spectrum efficiency for the particular modulation technique adopted for the purpose of modulation of the RF carrier. There are three basic types of digital modulation techniques as given below: Amplitude-Shift Keying (ASK) Frequency-Shift Keying (FSK) Phase-Shift Keying (PSK)

All of these techniques vary a parameter of a sinusoid to represent the information which we wish to send. A sinusoid has three different parameters than can be varied. These are its amplitude, phase and frequency. Modulation is a process of mapping such that it takes your voice (as an example of a signal) converts it into some aspect of a sine wave and then transmits the sine wave, leaving the actual voice behind. The sine wave on the other side is remapped back to a near copy of your sound. The medium is the thing through which the sine wave travels. So wire is a medium and so are air, water and space. The sine wave is called the carrier. The information to be sent, which can be voice or data is called the information signal. Once the carrier is mapped with the information to be sent, it is no longer a sine wave and we call it the signal. The signal has the unfortunate luck of getting corrupted by noise as it travels. In ASK, the amplitude of the carrier is changed in response to information and all else is kept fixed. Bit 1 is transmitted by a carrier of one particular amplitude. To transmit 0, we change the amplitude keeping the frequency constant. On-Off Keying (OOK) is a special form of ASK, where one of the amplitudes is zero as shown below.

Figure 6 - Baseband information sequence 0010110010 ASK(t) = s(t)sin(2ft)

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Figure 7 - Binary ASK (OOK) signal In FSK, we change the frequency in response to information, one particular frequency for a 1 and another frequency for a 0 as shown below for the same bit sequence as above. In the example below, frequency f1 for bit 1 is higher than f2 used for the 0 bit.

Figure 8 - Binary FSK singnal In PSK, we change the phase of the sinusoidal carrier to indicate information. Phase in this context is the starting angle at which the sinusoid starts. To transmit 0, we shift the phase of the sinusoid by 180. Phase shift represents the change in the state of the information in this case.

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Figure 9 - Binary PSK Carrier (Note the 180 phase shifts at bit edges) ASK techniques are most susceptible to the effects of non-linear devices which compress and distort signal amplitude. To avoid such distortion, the system must be operated in the linear range, away from the point of maximum power where most of the non-linear behavior occurs. Despite this problem in high frequency carrier systems, Amplitude Shift Keying is often used in wire-based radio signaling, both with or without a carrier. ASK is also combined with PSK to create hybrid systems such as Quadrature Amplitude Modulation (QAM) where both the amplitude and the phase are changed at the same time.

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Section 3.2

PRINCIPLE OF HIGHER ORDER MULTIPLEXING

1.0 WHAT IS HIERARCHY?
The term digital hierarchy has been created when developing digital transmission systems. It was laid down when by multiplexing a certain number of PCM primary multiplexers were combined to form digital multiplexers of higher order (e.g. second-order multiplex equipments).

Consequently, a digital hierarchy comprises a number of levels. Each level is assigned a specific bit rate which is formed by multiplexing digital signals, each having the bit rate of the next lower level. In CCITT Rec. G.702, the term digital multiplex hierarchy is defined as follows: A series of digital multiplexes graded according to capability so that multiplexing at one level combines a defined number of digital signals, each having the digit rate prescribed for the next lower order, into a digital signal having a prescribed digit rate which is then available for further combination with other digital signals of the same rate in a digital multiplex of the next higher order. To cope with the demand for ever higher bit rates, a multiplex hierarchy called the plesiochronous digital hierarchy (PDH) evolved. The bit rates start with the basic multiplex rate of 2 Mbit/s with further stages of 8, 34 and 140 Mbit/s. In North America and Japan, the primary rate is 1.5 Mbit/s. Hierarchy stages of 6 and 44 Mbit/s developed from this as shown in fig. 1

Fig. 1 Plesiochronous Digital Hierarchies

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2.0

MULTIPLEXING OF DIGITAL SIGNALS

The functions of digital multiplex equipment are to combine a defined integral number of digital input signals (called tributaries) at a defined digit rate by time division multiplexing and also to carry out the reverse process (demultiplexing). In analogue system, multiplex equipment uses F.D.M. to assemble individual channels into groups, super group etc. Similarly, in digital systems, hierarchical levels have been defined using T.D.M. and are identified by their digit rate measured in bit/sec. Bit rate Mbit/sec. 2.048 8.448 34.368 139.264 No. of channels 30 120 480 1920

The digital signals which are to be multiplexed may be synchronous to one clock (called master clock) or they may not be synchronous (called asynchronous signals).

3.0

MULTIPLEXING OF SYNCHRONOUS DIGITAL SIGNALS

The various tributary bit streams are synchronous and operate at the same rate defined as T bit/sec. To multiplex n such tributaries the rate of multiplex output should be nT bit/s. The method adopted for multiplexing such n signals into one stream may be as follows: (i) Block interleaving/ Word Interleaving : Bunch of information taken at a time from each tributary and fed to main multiplex output stream. The memory required will be very large. (ii) Bit interleaving : A bit of information taken at time from each tributary and fed to main multiplex output stream in cyclic order, a very small memory is required. At the demultiplex end, it is necessary to recognize which bit of information belongs to which tributary. This could be achieved by transmitting a fixed code after a fixed number of information bits called frame. The fixed code is called frame alignment signal. It is recognized first and received frame of information is aligned to this fixed code. This method of multiplexing is easy but not reliable. If any deviation in nominal bit rate of a tributary occurs, it will cause loss of time slot and hence loss of information.

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Fig.2 Block and Bit Interleaving

4.0

MULTIPLEXING OF ASYNCHRONOUS SIGNAL

Here, various tributaries operate at different bit rates.

Two signals are asynchronous at their corresponding significant instant occur at nominally the same rate, any variation in rate being constrained within specified limits. When nominal bit rate of tributaries are within specified limit. It is necessary to synchronize the tributary signal with a common nominal bit rate of multiplexer derived from timing generator of multiplexer. The synchronization is done in such a way that there is no loss of information. The process adopted for such synchronization is called Pulse stuffing or justification. Justification is a process of changing the rate of digital signals in a controlled manner. There are three types of justification processes: (a) (b) (c) Positive justification: Common synchronization bit rate offered at each tributary is higher than the bit rate of individual tributary. Positive-negative justification: Common synchronization bit rate offers is equal to the nominal value. Negative justification: Common synchronization bit rate offered is less than the nominal value.

Fig. 3(a) shows a configuration where the outputs of two PCM transmitters A&B are to be multiplexed in the combiner. If A and B are synchronous, they can be easily multiplexed by the combiner as shown in Fig. 3(b). Generally, however, A&B are clocked by separate clock
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Section 3.2

sources of asynchronous. In this case multiplexing is not successively accomplished simply by the use of combiner owing to the occurrence of pulse phase fluctuations and/or pulse amplitude superposition as can be seen in Fig.3(c).

Fig. 3(a)

Fig. 3(b) When A and B are Synchronous

Fig. 3 (c) When A and B are Asynchronous

5.0

RETIMING ASYNCHRONOUS SIGNALS BY JUSTIFICATION

Figure 4 shows a system for explaining the principle of the multiplexer for successfully multiplexing plural asynchronous signals. The waveforms appearing at various points in Fig.4 are shown in Fig.5. An asynch. input pulse train A is written into MEM I comprising several elements. The writing pulse train C whose bit rate is f is extracted from A at a clock extraction (CLK EXT I). On the other hand, the written information is read out of MEM I with a sufficient phase lag with respect to time of writing in. Through an inhibit gate (INH GATE I), the reading pulse train D is obtained by dividing the output bit rate nf (1+ ) of a common clock generator (CLK GEN) at a bit rate divider (DIV 1). n no. of asynch. signals to be multiplexed. clock increase rate.

As the bit rate of the reading pulse train D is set at (f+ f) which is higher than any value of f, the time of read out (D) gradually approaches that of write in . The phase difference between C&D is monitored by a phase comparator of COMP I and just before the difference reaches zero, a pulse is applied to the inhibit input of INH GATE I from a control circuit (CONT I) to inhibit the gate. At this moment, with one bit of the reading pulse train D being removed, the reading operation pauses and an information less pulse (or justification
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Section 3.2

pulse) is inserted into the read out pulse train E. the time of read out (D) at the same time is again set to a sufficient lag with respect to time write in (G). As all the signals read out of the respective memories are now retimed by timing pulses derived from the common CLK GEN, they are now easily multiplexed as F in Fig.5 at the combiner (COMB). The information pulses inserted into E (those hatched in Fig. 5) and this sort of retiming method are respectively called justification pulses and justification. The information whether or not justification has been performed, is inserted into F and COMB and transmitted to the receiving side.

6.0

RECOVERING ORIGINAL SIGNALS BY DEJUSTIFICATION

The justification pulses have to be removed at the receiving side to perfectly recover the original signals. This operation is called dejustification. The transmitted pulse train F from the line is received and demultiplexed at distributor (DIST). One of the demultiplexed signal E that corresponds to A, is written into memory MEM 2. The writing pulse train G whose bit rate is is obtained through an ingibit grate (INH GATE 2) by dividing the output bit rate nf(1+ f) of clock extractor (CLK EXT2). On the other hand, the written information is read out of MEM 2 with a slight phase lag with respect to the time of write in. The reading pulse train H, whose bit rate is f, is applied from voltage controlled oscillator (VCO). As the bit rate of the reading pulse train H is lower than that of the writing pulse train G, the time of read out (H) gradually drifts away from that of write in (G). Just before a justification pulse in E (ONE of these hatched in Fig.5) is written into MEM 2, the information, telling that the justification has been performed is applied from DIST to a control circuit (CONT 2). Then a pulse is applied to the inhibiting input of INH GATE 2 from CONT 2 to inhibit the gate. At this moment, with one bit of the writing pulse train G being removed, the writing operation pauses and the justification pulse is removed or dejustified. At the same time, the time of read out (H) again set to be very close to the time of write in (G). As the reading operation does not pause, the original signal is recovered as A. The phase difference between G and H is monitored by a phase comparator (COMP 2), and the low frequency components of the output voltage of COMP 2 are applied to VCO through a low pass filter (LPF). Thus, the jitter introduced due to dejustification into the read out pulse train A is sufficiently suppressed. The loop formed by VCO, COMP and LPF is called a Phase controlled loop.

7.0

JUSTIFICATION CONTROL SIGNAL

Justification control signal indicates at demultiplexer the presence of justifiable bit in the frame. To avoid errors present in the justification control bit, more than one bit is transmitted as control bit and majority decision is taken at demultiplexer. Normally 3 or 5 bits (3 bits in case of 8 and 34 Mbits systems and 5 bits in 140M bits system) are transmitted per tributary per frame as justification control bits and 2 or 3 bits present at demultiplexer out of 3 or 5 bits transmitted are taken as majority decision and it is assumed that justifiable bit is present in the frame. These 3 or 5 bits of justification control bits per tributary per frame are distributed in the frame. Two or three digital errors are required to cause false information of justification (loss of one digit or addition of one digit) which results in a loss of frame alignment in lower hierarchical levels.
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8.0

TRANSMIT TRIBUTARY

The information from tributary is written in an elastic memory with tributary clock derived from incoming signal. Elastic memory is read out by a clock which is faster than the clock of its own. Reading clock is derived from common transmit clock (Common synchronization clock). The reading clock is of rate F2/n (where n is the no. of tributaries and F2 is output frequency of multiplexer for ex. 34,368/4 for 34M bit system) with gaps where non information bit occurs in the frame structure (i.e., for frame alignment signal and justification control bit, service digits). Since read clock always operates faster than write clock, it is required to stop read clock for a bit and insert non-information bit-justification bit. The information which read out from memory contains information bit, justification control bit and justifiable bit. The decision when to insert the justifiable bit is taken when linearly increasing phase difference crosses a threshold level. The threshold value is selected in such a way that average rate of read clock is equal to the write clock rate. In the demultiplexer, the clock timing of the input multiplex signal enables a control on the timing of operations. The detection of frame alignment signals enables the receive frame to be aligned with the transmit frame which enables the receiver to demultiplex the tributary information. This tributary information is written in elastic memory as in transmit tributary by the clock derived from receive clock. A phase locked oscillator is used to read the elastic memory with a timing rate equal to the average write clock and, therefore, equal to that of the corresponding tributary signal at the input of the multiplexer. output.

Fig. 4 System for Multiplexing & Demultiplexing Asynchronous Signals

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10.0 JITTER ASPECT OF MULTIPLEX EQUIPMENT

While considering the jitter aspect of the multiplex system, different types of jitter introduced in the systems are taken into account such as : (a) (b) (c) Jitter introduced due to the routine insertion of the frame alignment words and of the service digits and justification instructions. Justification jitter. Waiting time jitter.

The first two jitter components are at high frequencies in relation to the pass band of the P.L.L. and hence filtered out, whereas waiting time jitter which is due to phase difference between write and read clock and varies from frame to frame, has a low frequency component and cannot be jittered out by P.L.L. at the demultiplexer output.

Fig. 5 Waveforms at respective points

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Fig. 6 Higher-Order Multiplex Frame Structure (a) 8448 Kb/s, (b) 34,368 Kb/s, (c) 139 264 Kb/s
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Section 3.2

LINE CODES
1.0 INTRODUCTION
The digital output of a PCM equipment contains "1s and 'O's. For transmission of the digital signals between two points, the '1' s and 'O' s contained by the signal are transmitted in the form of pulses as shown in Fig. 1.

Fig. 1 Pulse representation of digital signals The transmission medium normally used for transmitting PCM signals is the VF cable pair. If the stream of pulses shown in Fig. 1 is transmitted as it is, the signal undergoes high frequency attenuation distortion and also suffers from other kinds of distortion such as cross talk etc. This is because of the electrical characteristics of the VF pair. Moreover the .signal passed through the cable pair has strong DC content. This is because of the characteristics of the signal and those of the medium do not match. For distortion free transmission, the PCM output should be converted into a suitable code which will match the characteristics of the medium. This code is called the "line code" and the signal converted to the line code is called a line signal. This handout briefly describes the basic requirements of a line code, the different types of line codes and the operation of an HDB3 code decoder.

2.0

REQUIREMENTS OF A LINE CODE.

The line code used for transmission of PCM signals should meet the following requirements: (i) The total band width of the signal should be as small as possible. (ii) The energy in the upper part of the signal spectrum should be small so that the attenuation distortion caused by the high transmission losses at high frequencies is very low. The energy in the lower part of the spectrum should also be low to reduce the interference (cross talk) from VF circuits in the same cable. This would minimize interference from the PCM signals to the other VF circuits as well. (It may be recalled that a narrow pulse has. a wide frequency spectrum, the energy distribution, i.e. the levels of the various frequency components of the spectrum should be such that the major chunk of the signal power is around the centre of the spectrum. The frequency components in the lower and higher limits of the spectrum should have low levels). (iii) There should not be any DC component in the line coded signal (line signal) so that transformers can be used for coupling purposes. (iv) The line code should permit easier designs of repeaters.

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Section 3.2

(v) The line code should contain adequate timing information since this is vital for regenerating the signals at repeater stations and at the receiving station for the purpose of synchronization. (vi) The line code must have an in-built error monitoring capability. Since the invention of PCM by A.M. Reeves in 1938, a number of line' codes has been designed. A few of them will be discussed in the following paragraphs.

3.0

NRZ BINARY CODE

NRZ stands for "Non-return to Zero" code.

Suppose we have a code 100111011001 In Pulse form this would appear as in Fig. 2.

Here it may be seen that whenever a' 1' is continuously transmitted, the output continues at 'V level for a duration equal to the number of bits transmitted. In a30 chl. PCM system, the bit duration is 0.488 micro second. If three '1 ' 1 s are transmitted, the output signal is a pulse which is 3x0.488 micro seconds wide. In the example the signal has only one polarity. A 'O is 0 volt and a ' 1 ' is negative (say -5v). Sometimes, however, a '1 ' is denoted by a positive voltage and 0 is denoted by a negative voltage. A pulse stream 100110111001 in such a case can be represented graphically as in Fig.3.

Fig. 3 Bi-Polar NRZ Signal Here also when there is a string of 1s to be transmitted, the output continues at 1' for as many bits as are continuously transmitted. In both cases, the output does not return to zero after every 'V bit when a number of 1's are transmitted- for this reason, this type of code is called a non return to zero or "NRZ" binary. When the signal has only one polarity, as in the first example, the code is called uni-polar or unbalanced NRZ binary and when the signal has dual polarity, as in the second example it is called a balanced NRZ binary or bipolar NRZ. 3.1 Limitation of NRZ Binary Code. Fig 4 shows the spectrum of an NRZ binary signal. (i) From the spectrum for the NRZ signal, it can be seen that there is a strong DC component.

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Fig. 4 Spectrum for NRZ Signal (ii) There is a large low frequency content. This may result in cross talk. (iii) There is no frequency component at 1/T, 2/T etc. It means that there is no component corresponding to the clock frequency. This makes efficient recovery of timing pulses very difficult. (iv) (v) The high DC component does not permit the use of transformers for coupling. Thus the simple NRZ binary code does not satisfy the requirements of a line code.

4.0

RZ Binary

This is a modification of the NRZ code and stands for "Return to zero" binary. In this '1' bit is represented by a pulse of half the bit duration as shown in Fig. 5(a) (b) The spectrum for this code is shown in Fig. 5 (b).

Fig. 5 (a)- RZ Binary Waveform

FIG. 5(b)- RZ Binary Spectrum Here the '1' bits pulse have only 50% duration. From the spectrum we can seen that there is a strong component at 1/T, the clock frequency. Hence clock recovery is possible. But still, because of the strong DC component and low frequency content, this code is also not suitable for transmission.

5.0

BIPOLAR CODING (AMI Code)

AMI stands for "Alternate Mark Inversion" This code solves the DC content problem,. Here, a logic 'O' is represented by o volt and logic '1' is alternately encoded with positive & negative voltages. Therefore, the average voltage is maintained very close to zero and hence there is no DC component. Under steady state conditions a low DC of the order of 0.4 to 0.9 volts only remains. The waveform for an AMI code is shown in Fig. 6.

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FIG. 6 AMI Code Signal Waveform From the AMI wave form is can be seen that this code has a built in error monitoring facility. Since alternate marks (or 1's) are to be inverted, any deviation from this would mean an error. This can be practically achieved by having a comparator network which will check the polarity of the ' 1' s received. The spectrum of an AMI code signal (after doing Fourier analysis and plotting the various frequency components of the signal) has a shape as shown in Fig. 7.

FIG. 7 Spectrum of AMI Signal From the spectrum For the AMI code it can be seen that the maximum power is centered around the half bit rate i.e. 1/2T and that there is no DC component. Although the AMI code satisfies most of the line code requirements, a series of O's is encountered, the timing information is likely to be lost. This is a limitation of the AMI code. The AMI code is the one specified for 24 channel PCM systems.

6.0

HDB-3 CODE

To overcome the timing difficulties in the AMI code another code called the HDB3 code has been devised. The abbreviation HDB stands for HIGH-DENSITY BIPLOAR code. (i) The HDB3 code is actually a code from a family of codes derived from what is called binary N zero substitution or BNZS method. (ii) In this method, the PCM signal is usually transmitted according to the AMI code; but when a string of N zeroes is encountered, the N zeroes are replaced by a special code, which will deliberately introduce a bipolar deviation or violation. (Normally in the AMI code, if there are N zeroes, they will be transmitted as such. But in the BNZS method, a ' 1 ' pulse is introduced deliberately. The polarity of this " 1 ' depends upon the polarity of the previous mark encountered. This additional' 1 ' pulse introduced in place of a '0' is called a "violation"). (iii) When the substitution of a zero by a violation pulse is done for 4 zeroes, (i.e. N = 4) the BNZS code is called the B4ZS code. Since this code precludes strings of zeroes greater than

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three, it is also referred to as a HDB3 code. Here when the number of zeroes is more than 3, the fourth bit position is filled with a violation pulse. (iv) Consecutive violations are made to be of opposite polarity so that these violations themselves do-not produce any DC component. (v) The violation pulse is always placed in the last bit position. Suppose there are 4 zeroes coming in a row. Then the HDB3 code for this would be BOOV in general where V is the violation pulse. The polarity of this depends on the polarity of the last '1' and the number of 1'encountered prior to the four zeroes. (vi) The first bit of the code was shown as B in (v) above. B is set to '0' if the number of '1' s encountered prior to the violation is ODD. If it is EVEN or ZERO then the "B" bit is filled with a T whose polarity is in accordance with the AMI code. i.e. if the previous ' 1' was positive + then B is '1' with negative polarity and vice versa. (vii) The substitution rules stated above are summarized in table 1. Table 1 HDB3 Code - Substitution Rules Number of ' 1 ' since last Violation Polarity of preceding Odd '1' Negative 000 VPositive 000 V + Even B+OOV+ B- 00 V-

From the above Table it can be seen that when the number of Ts is even, the HDB3 substitution is BOOV; in this, B follows the AMI code and V follows B. If 'B' is positive, then V is also a positive pulse. Thus consecutive violations are made to be of opposite polarity so that there is no DC component added by the violations themselves. Examples of HDB3 Code Conversion Condition 1: (Number of 1 's preceding violation is ODD) Example 1: Consider the NRZ binary wave form given in Fig 8 (a). Assume that there is no previous violation.

Fig. 8 Example of HDB-3 Code Conversion

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[No. of 1 = ODD] Fig 8 (b) is the RZ binary form for Fig 8 (a). Fig 8 (c) is the corresponding HDB3 Code. (i) (ii) (iii) (iv) (v) (vi) (vii) Notice that upto pulse Z, the HDB3 Code follows the AMI Code. After pulse Z, we have four consecutive zeroes. This calls for a violation. Prior to the arrival of these zeroes, three 1 's were encountered i.e. number of 1 's preceding the violation is ODD. This means that the HDB3 substitution for the 4 zeroes will be of the form 000V Also the polarity of the last ' 1 ' before the arrival of the zeroes is positive. Therefore the violation pulse will also be a positive pulse, as shown shaded in Fig 8(c) Then the fourth pulse P arrives which is converted according to AMI code as shown.

Example 2 Consider the RZ binary wave form shown in Fig 9. In this the first pulse is a violation pulse resulting from the occurrence of 4 zeroes just before pulse X.

Fig. 9 (i) Here, the first pulse is positive violation pulse. The next pulse (i.e. pulse X) is converted in accordance with AMI code and is therefore shown as a negative pulse in Fig 9. (ii) After pulse X we get 4 Zeroes. (iii) Now, the total number of '1's SINCE the last violation is one, i.e. ODD. Therefore the substitution is of the form 000V. (iv) As the polarity of the last M' before the arrival of zeroes is negative, the violation pulse is also negative which is shown as a shaded pulse. (V-). Example 3 Consider the RZ wave form shown in fig. 10(a), assume that there was no previous violation.

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Fig. 10 (i) Here after pulse X, we have 4 zeroes and again after pulse Y, 4 more zeroes. (ii) The first pulse goes as a positive pulse. The next four zeroes are substituted by 000 V because the polarity of the last '1' is positive and the total number is ODD. (iii) Then pulse is converted into a negative pulse according to AMI code. (iv) The next 4 zeroes are substituted by 000V since in this case the total number of '1' s is again ODD and the polarity of the last '1' is negative. Condition 2: (Number of 1s preceding violation is EVEN) Example 1 Consider the RZ binary shown in Fig. 11 and assume that there was no previous violation.

Fig. 11 (i) In this case the pluses X and Y are converted according to AMI Code as shown in Fig 11. (ii) Four zeroes are encountered after pulse Y. Here the number of' 1s prior to these zeros is EVEN and therefore the substitution is of the form BOOV. (iii) (iv) (v) Since the last "1' is a negative pulse, from Table 1 the substitution BOOV. The HDB3 substitution for the 4 zeroes is shown in Fig 13 (b) as shaded positive pulses. The next pulse Z is converted as a negative pulse in accordance with AMI code. Example 2 Consider the RZ wave form shown in Fig 12 (a). Assume there was no previous violation.

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Fig 12 (i) In this case, the wave form begin with 4 zeroes. There are no previous violations. The number of "1's preceding the string of "0" is zero i.e. EVEN. (ii) Therefore, The substitution is of the form B00V. (iii) The (AMI) coding network is so designed that the very first bit is always a positive pulse. Hence the HBD3 code for Fig 11 would be B00V which is as shown in Fig 11(b). (iv) The following " 1's X, Y and Z are converted according to AMI code. Example 3 Consider the RZ wave from in Fig 13(a). Assume a positive violation pulse to start with.

Fig. 13 HDB2 code conversion (i) Here, as shown in Fig 15 (b), a positive violation pulse is assumed. Then we have pulses X and Y which are converted according to AMI code. (ii) After pulse Y, we get 4 zeroes. Theses should be substituted by BOOV the number of' 1 's is EVEN. since

(iii) Further since the polarity of the last ' 1 ' is positive, the code would be B 00V. This is shown as shaded pulsed in Fig 13 (b). Then pulse Z is converted according to AMI code. Example 4 Consider the RZ Wave form in Fig 14 (a)

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Fig. 14 (i) (ii) (iii) Here after pulses X and Y, we get eight consecutive zeroes. X and Y follow AMI code. Number of T s is Even in this case. The first four zeroes are substituted by B 00 V since the last' 1' was negative. (iv) After the first 4 zeroes, we have another 4 zeroes coming number of' 1' in this case zero i.e. EVEN again. (v) Hence the second set of zeroes is also converted as BOOV. But the polarity of the last '1' (although it was a violation pulse), was positive.Hence, the second set of zeroes is converted, as B 00 V (vi) (vii) Pulse Z, then follows AMI code, The substitution pulses are shown in shaded areas in Fig 16 (b). The spectrum for the HDB3 code is shown in Fig. 15

Fig. 15 Spectrum of HDB-3 Code From the spectrum it can be seen that there is no DC component and that maximum power is around 0.46/T. It means that the power in the lower and upper limits of the spectrum is low. This would minimize high frequency alternation and cross talk. Although this spectrum also has mulls at 1/T2/T etc, because of the violation pulse introduced, timing is not lost when a long string of zeroes is encountered. The HDB3 code satisfies ad the requirements of a line code and is therefore specified by the CCCTT for 30 channel PCM systems having 2048 Kbits/sec, clock.

7.0 CMI CODE (CODED MARK INVERSION)

This is a 2 level NRZ code in which a binary '0' is coded as '01' and binary 1's are coded alternatively as a logic '0' or T. In case of a binary '0' the two CMI bits '0' and '1' are for half clock duration whereas for binary Ts the 'O1 and '1' are for full clock duration. This is illustrates in Fig.16.

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Fig. 16 Example of CMI Coded Signal This is basically a binary code and the bit rate of the code is twice the bipolar AMI code. For this reason CMI code is grouped with 1B2B family of line codes. The CMI code has a high clock content and for this reason. The CMI code is recommended by CCITT for 140 Mb/s multiplex equipment (not a line code).

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