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Course 7 Agenda
• Digital encoding techniques (continuation):
• Biphase (Manchester, differential Manchester)
• Block coding techniques: 4B/5B, HDB-3, B8ZS

• Analogue to digital Techniques:


• Pulse Code Modulation (PCM) - continuation
• Delta Modulation

• Baseband Data Transmission Systems


• General model of the baseband data transmission
systems
• ISI-less data transmission
• Nyquist filters 1

Biphasic codes

• Manchester and differential Manchester

• Principle: supplementary transitions introduced


compared to NRZ, AMI

• Voltage transition in middle of bit period (denoted


biphase).

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Manchester (MPE) encoding (1)

• Voltage transition in middle of bit period (denoted biphase).


• 2 conventions
• 1 = low to high voltage transition (or high to low)
• 0 = high to low voltage transition (or low to high)
• used for the IEEE 802.3 standard for baseband coaxial cable and
twisted-pair LANs .eg 10Mbps Ethernet (10 Base 5, 10 Base 2,
10 BaseT, 10 Base FL.
• good for clock synchronization and recovery.
• no DC component
• Modulation rate = 2 x bit rate results an inefficient use of
bandwidth (10Mbps Ethernet uses a modulation rate of 20M
baud)

Two conventions of Manchester encoding (2)

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Encoding the digital bit stream 10110 with


Manchester MPE (3)

Differential Manchester Encoding (DME)

•Voltage transition in middle of bit period is for


clocking
•1 = no transition at beginning of bit period
•0 = transition at beginning of bit period (low-to-
high or high-to-low, depending on previous output
level)
•used in IEEE 802.5 (Token ring) at specified for the IEEE
802.5 Token Ring LAN, at 4Mbps and 16Mbps, using
shielded twisted pair.
•inefficient use of bandwidth – D=2R
modulation rate = 2 x bit rate
•Slightly better signal detection and clocking in noisy
environment compared to Manchester encoding.

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Biphase codes pros and cons

• Pros
• Predictable transition at the middle of each bit, which
can be used for synchronization (“self-clocking” codes)
• Error detection possible due to the same feature
• No DC component
• Cons
• Reduced bandwidth efficiency (half the bandwidth
efficiency of NRZ)
• Modulation rate is two times the data rate D=2R

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Block coding
• Use sequences with some known transitions to replace blocks
of constant voltage level
• Filling sequence
• Must produce enough transitions in order to make easier the
synchronization
• Must be recognized by receiver and replaced with original
sequence
• Same length as the original sequence (no efficiency
degradation)
• No DC component
• No long sequences of constant line signal
• The data rate is preserved (no overhead, redundancy)
• To provide error detection capability
• Most widely used are HDB-3 and B8ZS, both based on AMI

4B/5B Block coding


• 4B/5B maps groups of four bits onto groups of 5 bits, with at
least one bit of 1 in the output. When , later NRZI-encoded, the
bit of 1 provides necessary clock transitions for the receiver
• A run of 4 bits such as 0000 contains no transitions and that
causes clocking problems for the receiver.
4B/5B solves this problem by assigning to each block of 4
consecutive bits an equivalent word of 5 bits.

• These 5 bit words are pre-determined in a dictionary and they


are chosen to ensure that there will be at least two transitions
per block of bits.

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• With 4 bits there are 24 = 16 different bit patterns.


• With 5-bit 'packets' there are 25 = 32 different bit patterns.
• In conclusion there are 16 combinations not used. As a result,
the 5-bit patterns can always have two '1's in them even if the
data is all '0's a translation occurs to another of the bit
patterns. This enables clock synchronizations required for
reliable data transfer.
• The unused code combinations are either declared invalid or
become transmission control signal, as it can be seen in the
next table.
• 4B/5B was popularized by Fiber Distributed Data Interface
(FDDI) in the mid-1980s, and was later adopted for 100BASE-TX
standard for Fast Ethernet (100 Mbps) defined by IEEE 802.3u in
1995, for twisted pairs Cat 5 or better.

4B to 5B encoding table

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Bipolar with 8 Zero Substitution (B8ZS)


• An encoding technique based on BAMI, modified to overcome
problems of loss of synchronization due to long strings of zeros.
• commonly used in US/Canada for T1 telephonic communications
(1.544MHz)
• For every string of 8 zeros, bipolar code is substituted according
to the following rules:

• If an octet of all zeros occurs and the last voltage pulse


preceding this octet was positive, then the eight zeros of the
octet are encoded as 000+-0-+.
• If an octet of all zeros occurs and the last voltage pulse
preceding this octet was negative, then the eight zeros of the
octet are encoded as 000-+0+-.
• The receiver recognizes the pattern: two BAMI code violations
and interprets the octet as a string of all zeros.

B8ZS Example

the last preceding voltage pulse positive, the code for 8 zeros 000+-0-+.
the last preceding voltage pulse negative, the code for 8 zeros 000-+0+-.

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HDB3 High-density Bipolar, 3zeros


• Another improvement of the bipolar-AMI encoding
• Rule: 4 consecutive zeros are encoded with a signal which
has at least one transition
• When a group of 4 zeros occurs:
• encode as 000V (violation) if the number of bits of “1”
after the last polarity violation is odd
• B00V (valid bipolar 00 Violation) if the number of bits
of “1” after the last polarity violation is even
• V means the same polarity as the previous one, B
respects AMI pattern
• This rule aims to remove the DC component (polarity
violation always changes its sign)

HDB3 Example 1

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HDB3 Example 2

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HDB3 Example 3

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Block coding pros and cons


• Pros:
• Classical drawbacks of AMI are eliminated
• No redundancy, therefore the same data rate is
maintained
• Error detection capabilities
• Good spectral efficiency
• Cons:
• Extra-processing needed at transmitter and, especially at
receiver side
• B8ZS employed in Canada 1.544 Mbps-T1
• HDB-3 commonly used in Australia, New Zealand,
Europe and Japan 2.048 Mbps-E1

Spectral Power Density for different encoding


techniques

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Bit error rate for different encoding techniques

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Time and frequency domain representations of a


single rectangular pulse.

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Distribution of transmitted signal's average power


through the frequency band.

One way to characterize the modulation rate is to


determine the number of transitions per bit time.
Table compares transition rates for various techniques.
It indicates the signal transition rate in the case of a data
stream of alternating 1s and 0s, and for the data stream
that produces the minimum and maximum modulation
rate.

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One way to characterize the modulation rate is to


determine the number of transitions per bit time.
Table compares transition rates for various techniques.
It indicates the signal transition rate in the case of a data
stream of alternating 1s and 0s, and for the data stream
that produces the minimum and maximum modulation
rate.

Analog Data To Digital Signal


Digitizing Techniques

❑Pulse Code Modulation (PCM)


❑Delta modulation (DM)
❑Once analog data have been converted to digital signals
(digitizing), the digital data:
• can be transmitted using NRZ-L
• can be encoded as a digital signal using a code other
than NRZ-L
• can be converted to an analog signal, using different
digital modulation techniques (ASK, PSK, FSK)

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Digitizing analogue data

The device used for converting analog data into digital


form for transmission, and subsequently , at receiver ,
for recovering the original analogue data from the
digital, is known as a codec (coder-decoder).
So voice data can be treated as digital even if the
transmission requirements (use of microwaves) dictates
that is used an analogue system.

Analog Data to Digital Signal

• Hardware: codec (coder/decoder), ADC/DAC

• Sound card (could be built in) on your PC: ADC (Analog


to Digital Converter) for your microphone. And a DAC
(Digital to Analog Converter) to play music or sound files

Used in:

• VoIP: IP-PBX, Skype

• Video Conferencing

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Short history of pulse coded modulation

◼ A problem of PSTN analog techniques (eg SSB-FDM)


was that transmitting multiple channels was difficult due
to non-linearities, resulting channel cross-talk
◼ 1937 Reeves and Delorane at ITT labs. tested TDM-
techniques by using electron-tubes
◼ 1948 PCM tested in Bell Labs: Using this method it is
possible to represent a 4 kHz analog telephone signal as
a 64 kbit/s digital bit stream ( 2 x 4 kHz x 8 bits
= 64 kbit/s) (using a code with 8 bits)
◼ TDM was taken into use in 1962 with a 24 channel
PCM link
◼ The first 30-channel PCM system installed in Finland
1969

Pulse Code Modulation

• Based on the sampling theorem which states:


If a signal x(t) is sampled at regular intervals of time
and at a rate higher than ! twice the highest
significant signal frequency, then the samples
contain all the information of the original signal.
The function x(t) may be reconstructed from these
samples by the use of a low-pass filter.
Ex for voice data considering B=4kHz a data rate of
R=8kbps is sufficient to characterize the voice signal

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Quantization
• Original signal
has continuous
amplitudes in its
Analog dynamic range
signal to be • PAM - signal is a
transmitted discrete pulse train
with constant
amplitude values
Continuous which periodically
PAM-pulse samples the
train analogue signal
• Quantized PAM
signal has only the
values that can be
Quantized quantized by the
PAM-pulse words available
train (here by 3 bits
words)
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• Analog samples are referred to as pulse amplitude


modulation (PAM) samples
• To each analog sample is assigned a binary code
• The digital signal consists of a block of n bits, where
each n-bit number is the amplitude of a PCM pulse
With a code having 2n bits there will be obtained n
quantization levels
2 bits – 4 quantization levels
3 bits – 8 quantization levels
4 bits -16 quantization levels

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PCM is a method by which an analog message can be


transformed into a digital format, encoded and then decoded at
the receiver

LP-filtering sampling quantization PAM/HDB-3


Channel
receiver

LP-filtering PAM/HDB-3 Pulse


regeneration

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Examples
1.Good voice reproduction can be obtained with a 7-bit
coding (128 quantizing levels).
Voice bandwidth occupies 4 kHz, so in order to satisfy
the Nyquist theorem samples should be taken at least
8000 samples per second.
This implies a data rate of 8000x7bits=56 kbps for the
PCM encoded digital data. This would require conform
to Nyquist criteria a low pass filter with a bandwidth of
at least R/2=B=56/2=28 kHz.

2. Color television uses a 10 bits code for a video signal


with a bandwidth of 4,6 MHz. This implies that
samples should be taken at at 2.4,6 Mbps samples
per second and a data rate of 2.4,6.10 bits=92 Mbps.
Needed bandwidth of at least 92/2=46 MHz.
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Example 3
• What bandwidth should be needed to satisfy the
Nyquist criteria to transmit voice over a channel
using for quantization 256 levels?

Consider that voice bandwidth occupies 4 kHz, so in


order to satisfy the Nyquist theorem samples should
be taken at least 8000 samples per second.

Answer : 32kHz

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Pulse Code Modulation

Example of Pulse Code Modulation using NRZ-L with


a negative voltage for 1 36

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Quantizing noise of PCM

• By quantizing the PAM pulse, original signal is only approximated. This


leads to quantizing noise or quantization error
• Signal-to-noise ratio for quantizing noise

• Thus, each additional bit increases SNR by 6 dB, or a factor of 4.


SNR dB = 20 log 2 + 1.76 dB = 6.02 n + 1.76 dB
n

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2 ways of reducing quantization error

Vertical axis: 32 quantization levels (5 bits) = 25


Quantization error/noise: unavoidable, regenerated analog data will not be 100%
the same as the original.
To reduce quantization error: we have to increase the quantization
number of levels, to 64 quantization levels (6 bits), and/or to increase the
sampling rate (decrease sampling time interval) 38

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PCM refined using nonlinear encoding

• The problem with equal spacing (the same step of


quantization –uniform quantization) is that the mean
absolute error for each sample is the same, regardless
of the signal level. Consequently, lower amplitude
values are relatively more distorted.
• By using a greater number of quantizing steps for
signals of low amplitude, and a smaller number of
quantizing steps for signals of large amplitude, an
important reduction in the overall signal distortion is
achieved (see the following figure)

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Effect of nonlinear coding

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Companding

• The same effect can be achieved by using


uniform quantizing but companding
(compressing-expanding) the input analogue
signal. A typical companding
function
▪Companding is a process that compresses the intensity
range of a signal by amplifying more weak signals than
strong signals .
▪At receiver, the reverse operation is performed.
▪Nonlinear encoding can significantly improve the PCM S/N
ratio
▪Ex: For voice signals, improvements of 24 to 30 dB have
been achieved. 41

Typical companding functions

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Sampling Rate
• Chose the right sampling rate, not too high, and not too low.
• Remember to satisfy the Nyquist sampling theorem: the PCM
sampling rate must be at least double the highest frequency of
the original analog waveform to ensure a reasonable
reproduction.
Sampling Rate - telephones
• Highest voice frequency is 3.4KHz, so we want at least a
sampling rate of 6800 samples/sec. (Nyquist rate)

• Currently the telephone system allocates 4KHz for a voice signal,


giving a sampling rate of 8000 samples/sec (so it is okay)

• 8 bits precision (256 quantising levels): recovered voice quality


is comparable to that achieved via analog transmission.

• Data rate of 8000 samples per second x 8 bits/sampLe = 64 kbps


is needed for a voice channel. The bandwidth needed at
reception for original signal recovery is B=64/2= 32 kHz 43

Audio - CDs and DVDs

• we speak between 85Hz to 255Hz, can hear up to 20KHz+ ...


• Higher sampling rate: to accurately process higher frequencies
of sound
Examples
• Audio CD: sample rate of 44100 Hz
• Number of digits of each sample: Audio CD has a precision of 16
bits.

• DVD Audio: sample rates of 96 and 192 KHz, but most people
can't hear the difference.
• DVD might have a precision of 24 bits

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Codec translation

• PSTN and VoIP: codec conversion from G711a/u


(64kbps) to (e.g.) G729 (8kbps)

• Uses of narrowband codecs for Internet (or low


bandwidth networks).
• Uses of broadband codecs for high bandwidth networks,
e.g. LAN.
• Too many codec translations (or
compression/decompression) degrades voice quality.

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Delta Modulation

• Analog input is approximated by the staircase function


• Moves up or down by one quantization level () at each sampling interval
• The bit stream approximates the derivate of the analog signal (rather than
amplitude)
• 1 is generated if function goes up
• 0 otherwise

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Example of Delta Modulation 47

Delta modulation schemes at


emitter and receiver.

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• The codec determines if the waveform has risen or


dropped one delta step and it transmits a 1 or 0
accordingly.
• Delta Modulation requires only 1 bit sample.

• The output of the DM process is therefore a binary


sequence that it can be used at the receiver to
reconstruct the staircase function.
• The staircase function can then be smoothed by
some type of integration process or by passing it
through a low-pass filter to produce an analog
approximation of the analog input signal.

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Delta Modulation Parameters


• Two important parameters
• Size of step assigned to each binary digit ()
• Sampling rate
• Two types of errors
• Slope overload noise: if signal waveform quickly
changing, the codec is not able to keep up with the
change.
• Slope-overload noise increases as δ is decreased.
• Quantising noise Slowly changing or constant
waveform: a pattern of 01010101..
• quantizing noise increases as δ is increased
•  must be a balanced choice, between the two types of
errors
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• Accuracy improved by increasing sampling rate


• However, this increases the data rate and thus the
necessary bandwidth for transmission

• Delta modulation is faster and simpler than PCM


• PCM is better from the S/N ratio than DM at the same
data rate

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Reasons for Growth of Digital Techniques

• Growth in popularity of digital techniques for sending analog data


• Repeaters are used instead of amplifiers
• No additive noise
• TDM is used instead of FDM
• No intermodulation noise
• Conversion to digital signaling allows use of more efficient digital switching
techniques

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