Download 6.1 Digital Transmission Systems

• 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 loudness
while 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’.
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.
• Quality of signals with analog systems varies as
overall distance of the circuit varies since
used in analog systems
along with the original signal.
• With digital systems only one of certain number of
possible states exist (1 or 0) which if identified can be
used to recreate the original signal from an degraded
input.
is the key benefit of digital
transmission systems.
• By adopting digital systems the noise
performance of a long distance telephone
channel is as good as that of a short distance
channel.
• 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 suited
for radio systems.
• Digital Exchanges and ISDN:
– Digital exchanges have tremendous advantages over
analog exchanges, these can be even enhanced by ISDN
customers (no conversion required)
• Lesser Maintenance
• Extra Security:
– With digital systems, it is possible to use different codes by
‘scrambling’ the signals at transmitter and unscrambling at
the receiver end.
• Space Saving and Economical:
– Digital systems are physically small and relatively cheap
thanks to advancements in LSI circuits
• Higher Capacity
– Digital systems can use the available bandwidth of
the channel much more efficiently as compared to
analog systems. In other words, better
multiplexing schemes are available and being
developed.
• The aim of any transmission system is to produce at
the output, an exact replica of any signal which is
applied to the input.
• In AM or FM, a system carrier is continuously varied
by the signal.
• It is not necessary to continuously send the
information and only “samples” at certain levels are
sufficient to represent it fully.
– Example MOVIE.
• 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 then
encoding it.
• The PCM signal itself is a succession of discrete,
numerically encoded binary values derived from
digitizing the analog signal.
PCM Steps
• Nyquist Sampling Theorem: If a signal is sampled
at 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 of 8 kHz is used which
is easily above twice the highest frequency
component within this range.
• 8000 samples per second, or 8 kHz, sampling
period 125 s
• Within one sampling period, samples of
several telephone channels can be
sequentially accommodated. This process is
called TDM.
• Alias distortion occurs if Shannon’s criterion is
not satisfied
Pulse Amplitude Modulation
• Sampling is the process of determining the
instantaneous voltage at given intervals in time. PAM
is the technique used to produce a pulse when the
signal is sampled.
• The pulse's amplitude is equal to the level at the
time in which the analog signal was sampled. The
amplitude of the pulses in a PAM signal contains the
intelligence or modulating voltage.
• The higher the sampling rate, the closer the recovered
signal 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 bandwidth
limitation on the large amounts of data that would need
to be transmitted.
Quantization
• Instead of transmitting the exact amplitude of the
sampled 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 specified
discrete 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.
Quantization
• Quantizing is a process by which analog samples (from a
pulse amplitude modulated (PAM) signal) are classified into a
number of adjacent quantizing intervals.
• Each interval is represented by a single value called the
Quantized Value.
• This process introduces an error in the magnitude of the
samples resulting in quantizing noise
• However, once the information Is in quantized form, it can be
sent over reasonable distance without further loss in quality
through regeneration of the binary levels involved to counter
distortion.
Quantization
the permitted range of
values of an analog signal divided into
quantizing intervals.
Linear Quantization
.
: the difference between the
input signal and the quantized output signal
Linear Quantization.. Example
Linear Quantization.. Example
• Consider sample 2, the actual amplitude of the
signal is +1.7V.
• This is assigned level 2 (same for any voltage
between 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 at
the encoder)
• This produces an error of 0.2V between original
input and output signals.
Linear Quantization
• Errors occur on every sample except where the
sample 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 complicate
the coding operation and increase bandwidth
requirements.
• Quantizing noise depends on step size and not on
signal amplitude.
Non-Linear Quantization
• With linear quantization, the signal to noise
ratio is large for high levels but small for low
level signals.
• Therefore, non-linear quantization is used.
Non-Linear Quantization
• The quantizing intervals are not of equal size.
• Small quantizing intervals are allocated to small
signal values (samples) and large quantization
intervals to large samples so that the signal-toquantization distortion ratio is nearly independent of
the signal level.
• S/N ratios for weak signals are much better but is
slightly less for the stronger signals.
Non-Linear Quantization
a process in which compression
is followed by expansion.
.
A-Law
• 13 piece-wise linear segments
• A=87.6, for x>0
•
Y
1 ln Ax
1 ln A
Y
Ax
1 ln A
for  x  1
1
A
for 0  x 
where x = normalized input level,
Y = normalized quantized steps,
ln = natural logarithm
1
A
(most significant
bit) tells the distant-end
receiver if that sample is a
positive or a negative
voltage
identify
the segment.
, shown in
the figure as XXXX, identify
where in the segment that
voltage line is located
The 13-segment approximation of the A-law curve used with E1 PCM equipment
µ-Law
• µ-law used in North America and Japan:
Y= sgn(x) ln(1+µx)
---------------ln(1+µ)
where µ=255.
ENCODING
• PCM signal to be transmitted is obtained by
encoding the quantizing intervals.
• Allocation of 8-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 positive
intervals is ‘1’ and for negative intervals is ‘0’
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 by
PCM words of of all other telephone signals arranged
in consecutive order.
• This results in PCM TIME DIVISION MULTIPLEX signal
MULTIPLEXING
• Multiplexing function is carried out fully
electronically.
• A switch moves from one input to other.
• The PCM-TDM signal is then available at the output
of the switch.
• The time interval within which a PCM word is
transmitted is known as Time Slot.
• A bit train containing one PCM word each from all
inputs is known as Pulse Frame.
Receiver Side
• Demultiplexing PCM-TDM signal i.e. the PCM
words of the telephone signals are distributed to
the individual lines.
• Decoding PCM words in the PCM signal i.e. a
signal amplitude is allocated to each word, which
is equal to the midpoint value of the particular
quantizing interval. Result is a PAM signal.
• Reproducing the original analog telephone signal
from the PAM signal through a LPF.
Line Code
• A code chosen for use within a
communication systems for transmission
purposes.
• A line code may differ from the code
generated at a user terminal, and thus may
require translation
Line Codes
• Objectives:
– Better spectrum (no DC component)
– Noise immunity
– Error detection
– Clocking capability
• No added complexity
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 1
and negative for 0 and returns to level 0 as well.
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.
Line Codes for PCM
• HDB3 -- used to eliminate series of more than
3 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 V’s to have the same polarity.
0000 ==> X00V, X is so chosen the V’s polarities
alternate.
Line Codes
1
0
1
1
0
0
0
0
1
1
AMI
1
1
0
0
0
0
0
0
0
0
0
0
0
V
B
0
V
B
1
1
0
0
0
0
B8ZS
-
- +0 + 0
HDB3
0
0
V
B
0
0
V
B
0
0
V
0
1
0
Regenerative Repeater
• The advantage of PCM lies chiefly in the fact that
it is a digital process.
• it is much easier for a receiver to distinguish
between a 1 and a 0 than to reproduce faithfully
a continuous wave signal.
• Transmission media carrying PCM signals
employ regenerative Repeaters that are spaced
sufficiently close to each other (approximately
2kms) to prevent any ambiguity in the
recognition of the binary PCM pulses
Regenerative Repeater
• The regenerative repeater conditions the received
(attenuated and distorted) pulses through
preamplifiers and equalizer circuits.
• The signal is then compared against a voltage
threshold
• Above the threshold is a logic 1, and below the
threshold is a logic 0. The resulting signal is said to be
threshold detected.
Regenerative Repeater
• Timing circuits within the regenerative
repeater are synchronized to the bit rate of
the incoming signal.
• The threshold detected signal is sampled at
the optimum time to determine the logic level
of the signal.
• The resulting code is used to regenerate and
retransmit the new equivalent signal
Regenerative Repeater
Formats for 30-channel PCM systems
(E1)
• A time slot: 8 bits
• A frame lasts 125 s and is divided into 32 slots,
numbered slot 0 to slot 31, transmission rate 2.048
Mbps
• Time slot 0: frame alignment and service bits
• Time slot 16 for multiframe alignment and
signaling, the remaining 30 slots for data
transmission (voice channel)
• A multiframe consists of 16 frames (2ms) numbered
frame 0 to 15
Formats for 30-channel PCM systems
(E1)
• TS 0 for even frames: Y0011011
for odd frames: Y1ZXXXX
TS 16 for frame 0: 0000XZXX (0000 multiframe
alignment signal, Y:international use, Z: frame
alignment loss indicator, X: not used)
• TS 16 for frames 1 to 15: signaling for 30
channels
PCM 30 Pulse Frame
Formats for 24 channel PCM systems
(T1)
• Used in North America and Japan (DS1)
• A frame lasts 125 s, 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
Formats for 24 channel PCM systems
(T1)
• A multiframe consisting of 12 frames, frame
alignment word 101010 on odd frames, multiframe
alignment word 001110 on even frames
• Transmission rate (1+24* 8)/125 = 1.544 Mbps
PCM24 Pulse Frame
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).
SUMMARY
Higher-Order Digital Multiplexing
(CCITT)
• T1 and E1 are the primary order of digital
multiplexing
• Higher orders can be formed
• Example: second order multiplexing (2 to 8
Mbps) -- Four low rate bit streams E1
(tributary) are multiplexed in a bit-by-bit
manner to E2
PLESIOCHRONOUS HIERARCHY
5760 Ch
397.2 Mbps
2.048 Mbps
Intentionally Left Blank
PULSE-CODE MODULATION SYSTEM OPERATION
Simplified functional block diagram of a PCM codec or channel bank.
Low-pass filter
The voice channel to be transmitted is passed through a 3.4-kHz low-pass filter.
Sampling circuit
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.
Channel gating
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 μsec, or
3.906 μsec.
Coder
The input to the coder is the PAM highway. The coder accepts a sample of each channel
in sequence and then generates the appropriate 8-bit signal character corresponding to
each sample presented.
Digit combiner
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 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
Timing recovery keeps the receive clock in synchronism with the far-end transmit clock.
receive clock
The receive clock provides the necessary gating pulses for the receive side of the PCM
codec.
Frame-alignment circuit
The frame-alignment circuit senses the presence of the frame-alignment signal at the
correct time interval, thus providing the receive terminal with frame alignment
Decoder
The decoder, under control of the receive clock, decodes the code character signals
corresponding to each channel. The output of the decoder is the reconstituted pulses
making up a PAM highway.
Signaling processor
Responsible for processing the control information associated with the corresponding
voice channels.
Intentionally Left Blank
Transmission Limitations
Transmission medium for PCM could be wire pair, coaxial cable, fiberoptic 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
•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.
The sources of timing jitter may be classified as
or
Systematic jitter sources lead to jitter which degrades the bit
stream in the same way at each repeater in the chain.
Systematic sources include:
•Intersymbol interference
•Clock threshold effects
•Finite pulse width
Nonsystematic jitter sources causes timing degradations which are
random from repeater to repeater.
Nonsystematic jitter sources such as:
• Mistuning
• Crosstalk
Thermal and impulse noise are not serious contributors to timing
jitter
In a long repeater chain, the total accumulated jitter is dominated
by components produced by systematic sources.
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,
circuit
in a long
There are three cable characteristics that
create this distortion:
•Loss
•Amplitude distortion (amplitude–frequency
response)
•Delay distortion
Because of the nature of a digital system, impairments like
thermal noise need only be considered on a per-repeatersection 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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