6 Digital Transmission Systems

8/7/2019 6 Digital Transmission Systems

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A continuous representation of a continuous event. An analog connection is one which continuously varies in

amplitude and frequency.

The amplitude of the signal is representation of its loudnesswhile the frequency represents its tone or pitch.

A digital signal is defined at discrete times only represented

by fixed states digits.

Most usually it is represented by a binary signals with 1s or

0s represented by a positive voltage or zero voltage, or by

two different carrier frequencies or phases. In optical fiber

system 1s or 0s may be represented by light on or light

off.


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There are three notable advantages to digital transmission that make it

extremely attractive to the telecommunication system engineer when

compared to its analog Counterpart.

Noise does not accumulate on a digital system as it does on an

analog system. Noise accumulation stops at each regenerative

repeater where the digital signal is fully regenerated. Noise

accumulation was the primary concern in analog network design.

The digital format lends itself ideally to solid-state technology and, in

particular, to integrated circuits.

It is theoretically compatible with digital data, telephone signaling,

and computers.


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Quality of signals with analog systems varies asoverall distance of the circuit varies since

used in analog systemsalong with the original signal.

With digital systems only one of certain number ofpossible states exist (1 or 0) which if identified can beused to recreate the original signal from an degradedinput.


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is the key benefit of digitaltransmission systems.

By adopting digital systems the noise

performance of a long distance telephonechannel is as good as that of a short distancechannel.


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Higher Data Rate Possible: With digital computers around, digital lines can transfer

data at much higher rates than analog lines.

Improvements using Digital Radio Systems: Since digital systems free from noise, they are best suitedfor radio systems.

Digital Exchanges and ISDN:

Digital exchanges have tremendous advantages overanalog exchanges, these can be even enhanced by ISDNcustomers (no conversion required)


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Higher Capacity

Digital systems can use the available bandwidth of

the channel much more efficiently as compared to

analog systems. In other words, bettermultiplexing schemes are available and being

developed.


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The aim of any transmission system is to produce atthe output, an exact replica of any signal which isapplied to the input.

In AM or FM, a system carrier is continuously varied

by the signal.

It is not necessary to continuously send theinformation and only samples at certain levels aresufficient to represent it fully. Example MOVIE.


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Initially invented by A.H. Reeves in 1937

Pulse Code Modulation is the representation of a

signal by a series of digital pulses firstly by

sampling the signal, quantizing it and thenencoding it.

The PCM signal itself is a succession of discrete,

numerically encoded binary values derived from

digitizing the analog signal.


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PCM Steps


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Nyquist Sampling Theorem: If a signal is sampledat a rate that is at least twice the highest

frequency that it contains, the original signal can

be completely reconstructed.

Since the bandwidth of the telephone lines is 300

to 3400 Hz, a sampling rate of8 kHz is used which

is easily above twice the highest frequency

component within this range.


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8000 samples per second, or 8 kHz, samplingperiod 125 Qs

Within one sampling period, samples ofseveral telephone channels can besequentially accommodated. This process iscalled TDM.

Alias distortion occurs if Shannons criterion isnot satisfied


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Pulse Amplitude Modulation

Sampling is the process of determining theinstantaneous voltage at given intervals in time. PAMis the technique used to produce a pulse when the

signal is sampled.

The pulse's amplitude is equal to the level at thetime in which the analog signal was sampled. The

amplitude of the pulses in a PAM signal contains theintelligence or modulating voltage.


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The higher the sampling rate, the closer the recoveredsignal approaches the original signal.

Ideally, an infinite sampling rate would be desirable in

terms of reproducing the original signal.

This is not practical, however, due to the bandwidthlimitation on the large amounts of data that would needto be transmitted.


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Quantization

Instead of transmitting the exact amplitude of thesampled signal, only certain discrete value closest to

the true one is transmitted.

At the receiving end the signal value will have a

value slightly different from any of the specifieddiscrete steps due to noise and distortions

encountered in the transmission channel.

If the disturbance is negligible, it will be possible to

tell accurately which discrete value was transmitted

and the original signal can be approximately

reconstructed.


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Quantization

Quantizing is a process by which analog samples (from apulse amplitude modulated (PAM) signal) are classified into anumber of adjacent quantizing intervals.

Each interval is represented by a single value called theQuantized Value.

This process introduces an error in the magnitude of thesamples resulting in quantizing noise

However, once the information Is in quantized form, it can besent over reasonable distance without further loss in quality

through regeneration of the binary levels involved to counterdistortion.


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Quantization

the permitted range of

values of an analog signal divided intoquantizing intervals.


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Linear Quantization

.

: the difference between the: the difference between the

input signal and the quantized output signalinput signal and the quantized output signal


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Linear Quantization.. Example


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Linear Quantization.. Example

Consider sample 2, the actual amplitude of thesignal is +1.7V.

This is assigned level 2 (same for any voltagebetween 1 & 2), which is transmitted as line code

101. At the receiving end 101 is converted to a pulse of

+1.5V (the middle value of the decision level atthe encoder)

This produces an error of 0.2V between original

input and output signals.


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Linear Quantization

Errors occur on every sample except where thesample size exactly coincides the mid-point of the

decision level.

If smaller steps are taken the quantization error will

be less. However, increasing the steps will complicatethe coding operation and increase bandwidth

requirements.

Quantizing noise depends on step size and not on

signal amplitude.


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Non-Linear Quantization

With linear quantization, the signal to noiseratio is large for high levels but small for low

level signals.

Therefore, non-linear quantization is used.


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The quantizing intervals are not of equal size.

Small quantizing intervals are allocated to smallsignal values (samples) and large quantization

intervals to large samples so that the signal-to-quantization distortion ratio is nearly independent ofthe signal level.

S/N ratios for weak signals are much better but isslightly less for the stronger signals.


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a process in which compressionis followed by expansion.

.


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A-Law

13 piece-wise linear segments

A=87.6, for x>0

where x = normalized input level,

Y = normalized quantized steps,

ln = natural logarithm

1 ln 1

1 ln

1

1 ln

1

0

Ax

A A

AxA A

Y for xY for x

!

!


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-Law

-law used in North America and Japan:

Y= sgn(x) ln(1+x)----------------

ln(1+)

where =255.


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ENCODING

PCM signal to be transmitted is obtained by

encoding the quantizing intervals.

Allocation of8-bit word is done to each

individual sample.

An 8-digit binary code is used for 128 positive

and 128 negative quantizing intervals.

First bit used for all PCM words for all positiveintervals is 1 and for negative intervals is 0


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MULTIPLEXING

The 8-bit PCM words of a number of telephone

signals can be transmitted consecutively in repeated

cycles.

A PCM word of one telephone signal is followed byPCM words of of all other telephone signals arranged

in consecutive order.

This results in PCM TIME DIVISION MULTIPLEX signal


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MULTIPLEXING

Multiplexing function is carried out fullyelectronically.

A switch moves from one input to other.

The PCM-TDM signal is then available at the outputof the switch.

The time interval within which a PCM word istransmitted is known as Time Slot.

A bit train containing one PCM word each from allinputs is known as Pulse Frame.


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Line Codes

Objectives:

Better spectrum (no DC component)

Noise immunity

Error detection

Clocking capability

No added complexity


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Line Codes for PCM

Unipolar NRZ -- Stays positive and does not return

to level 0 during binary 1 cell.

Bipolar NRZ -- 2 non zero voltages i.e. positive for

1 and negative for 0 and does not return to 0.

Unipolar RZ -- there is always a return to level 0

between individual bits during binary 1 cell.

Bipolar RZ -- 2 non zero voltages i.e. positive for 1and negative for 0 and returns to level 0 as well.


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Line Codes for PCM

CMI -- 1s represented by alternate + and -

states and 0s always represented by a - state

during first half and + in second half of bit

interval.

AMI -- 1s represented by alternate + and -

states and 0s always represented by zero

voltage.


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Line Codes for PCM

HDB3 -- used to eliminate series of more than3 0s in the AMI.

The last zero of 4 consecutive zeroes is replaced

by a violation (V) pulse that violates the AMI rule. The first zero may be replaced by a 1 to prevent

two Vs to have the same polarity.0000 ==> X00V, X is so chosen the Vs polarities

alternate.


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Line Codes

- - +0

+ -


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Regenerative Repeater

The advantage of PCM lies chiefly in the fact thatit is a digital process.

it is much easier for a receiver to distinguishbetween a 1 and a 0 than to reproduce faithfully

a continuous wave signal. Transmission media carrying PCM signals

employ regenerative re-

Repeaters that are spaced sufficiently close to

each other (approximately 2kms) to prevent anyambiguity in the recognition of the binary PCMpulses


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The regenerative repeater conditions the received

(attenuated and distorted) pulses through

preamplifiers and equalizer circuits.

The signal is then compared against a voltagethreshold

Above the threshold is a logic 1, and below the

threshold is a logic 0. The resulting signal is said to be

threshold detected.


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Timing circuits within the regenerativerepeater are synchronized to the bit rate of

the incoming signal.

The threshold detected signal is sampled atthe optimum time to determine the logic levelof the signal.

The resulting code is used to regenerate andretransmit the new equivalent signal


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Formats for 30-channel PCM systems

(E1)

A time slot: 8 bits

A frame lasts 125 Qs and is divided into 32 slots,numbered slot 0 to slot 31, transmission rate 2.048Mbps

Time slot 0: frame alignment and service bits

Time slot 16 for multiframe alignment andsignaling, the remaining 30 slots for datatransmission (voice channel)

A multiframe consists of 16 frames (2ms) numberedframe 0 to 15


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Formats for 30-channel PCM systems

(E1)

TS 0for even frames: Y0011011

for odd frames: Y1ZXXXX

TS 16 for frame 0: 0000XZXX (0000 multiframealignment signal, Y:international use, Z: frame

alignment loss indicator, X: not used)

TS 16 for frames 1 to 15: signaling for 30

channels


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PCM 30 Pulse Frame


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Formats for 24 channel PCM systems (T1)

Used in North America and Japan (DS1)

A frame lasts 125 Qs, 24 time slots each

having 8 bits The 8th bit in every six frames is used for

signaling.

1 bit at the start of every frame included for

frame and multiframe alignment purpose


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Formats for 24 channel PCM systems (T1)

A multiframe consisting of 12 frames, framealignment word 101010 on odd frames, multiframealignment word 001110 on even frames

Transmission rate (1+24* 8)/125 = 1.544 Mbps


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T1 Format for CCS

Pulse Frames are not combined to for

multifranmes.

The first bits in every even pulse frames are

used for Signaling (S-bits).


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SUMMARY


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PLESIOCHRONOUS HIERARCHY

5760 Ch

397.2 Mbps

2.048 Mbps


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Intentionally Left Blank

PULSE CODE MODULATION SYSTEM OPERATION


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PULSE-CODE MODULATION SYSTEM OPERATION

Simplified functional block diagram of a PCM codec or channel bank.


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The voice channel to be transmitted is passed through a 3.4-kHz low-pass filter.

Low-pass filter

The output of the filter is fed to a sampling circuit. The sample of each channel

of a set of n channels is released in turn to the pulse amplitude modulation

(PAM) highway.

Sampling circuit

The release of samples is under control of a channel gating pulse derived from the

transmit clock. To accommodate the 32channels, the gate is open 125/32 sec, or

3.906 sec.

Channel gating

The input to the coder is the PAM highway. The coder accepts a sample of each channelin sequence and then generates the appropriate 8-bit signal character corresponding to

each sample presented.

Coder


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The coder output is the basic PCM signal that is fed to the digit combiner where

framing-alignment signals are inserted in the appropriate time slots, as well as the

necessary supervisory signaling digits corresponding to each channel

Digit combiner

Digit separator

On the receiver side, it delivers the PCM signal to four locations to carry out the

following processing functions:

(1) Timing recovery

(2) Decoding

(3) Frame alignment

(4) Signaling (supervisory)

Timing recovery keeps the receive clock in synchronism with the far-end transmit clock.

Timing recovery

The receive clock provides the necessary gating pulses for the receive side of the PCM

codec.

receive clock


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Transmission Limitations

Transmission medium for PCM could be wire pair, coaxial cable, fiber-optic cable, and wideband radio media.

Each medium has transmission limitations brought about by

impairments. In one way or another each limitation is a function of

and

As loss increases signal-to-noise ratio suffers, directly impacting bit error

performance

Dispersion is another impairment that limits circuit length over a

particular medium, especially as transmission rate increases


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Displacement of the ideal sampling instant. This leads to a

degradation in system error performance

Slips in timing recovery circuits manifesting itself in degraded

error performance.

Distortion of the resulting analog signal after decoding at the

receive end of the circuit.


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The sources of timing jitter may be classified as or

Systematic jitter sources lead to jitter which degrades the bitstream in the same way at each repeater in the chain.

Systematic sources include:

Intersymbol interference

Finite pulse widthClock threshold effects


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Nonsystematic jitter sources causes timing degradations which arerandom from repeater to repeater.

Nonsystematic jitter sources such as:

Mistuning

Crosstalk

In a long repeater chain, the total accumulated jitter is dominated

by components produced by systematic sources.

Thermal and impulse noise are not serious contributors to timingjitter


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Wire-pair systems have repeaters every

Coaxial cable has repeaters approximately

Fiber-optic systems, depending on design and bit rate, have

repeaters every

Microwave radio, may have repeaters every

Satellite links have the least repeaters, in a long

circuit


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Because of the nature of a digital system, impairments like

thermal noise need only be considered on a per-repeater-section basis, because noise does not accumulate due to

the regenerative process carried out at repeaters and

nodes.

Crosstalk is a major impairment in PCM wire-pair systems,

particularly when go and return channels are carried in the

same cable sheath.

Echo is caused by impedance discontinuities in the

transmission line, including repeaters and terminations

(MDFs, codecs, switch ports)


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