Baseband Formatting Techniques

Formatter Techniques
• Process-------?

• Method-------?
 Formatter
• Output of the source must be converted to a format that can
be transmitted digitally.

• Formatter is not required if source is in..........?

• Sampler,Quantiser, Encoder.

• Some times encoder sub block of formatter and source

encoder block are combined Such as DPCM, DM ,ADM.

• Baseband Processor – Interference of one symbol on the

other.
Formatting
 BASEBAND
FORMATTING
TECHNIQUES
ANALOG-TO-DIGITAL CONVERSION

A digital signal is superior to an analog signal because

- more robust to noise and can easily be
recovered, corrected and amplified.

Waveform Coding techniques,

•Pulse code modulation,
•Differential Pulse Code Modulation,
•Delta modulation and
•Adaptive Delta Modulation
• Temporal Waveform coding Techniques
• Pulse code modulation,
• Differential Pulse Code Modulation,
• Delta modulation and
• Adaptive Delta Modulation

4.7
 Pulse Code
Modulation

Figure The basic elements of a PCM system.

 PCM - Transmitter
• PCM consists of three steps to digitize an analog
signal:
1. Sampling
2. Quantization
3. Binary encoding
 Before we sample, we have to filter the signal to
limit the maximum frequency of the signal as it
affects the sampling rate.
 Filtering should ensure that we do not distort
the signal, i.e. remove high frequency
components that affect the signal shape.
Components of PCM encoder
 Pulse Code Modulation (PCM)
x(t)

0
t

Consider the analog Signal x(t).

 Pulse Code Modulation (PCM)
x[n]

0
n

The signal is first sampled

 Pulse Code Modulation (PCM)

0
n
 Pulse Code Modulation (PCM)

0
n
 Pulse Code Modulation (PCM)

Sample n
 Pulse Code Modulation (PCM)

And Hold n
 Pulse Code Modulation (PCM)

0
n
 Pulse Code Modulation (PCM)
Assign Closest
Level
3

0
n
 Pulse Code Modulation (PCM)

0
n
 Pulse Code Modulation (PCM)

0
n
 Pulse Code Modulation (PCM)

0
n
 Pulse Code Modulation (PCM)

0
n
 Each quantization level corresponds to a unique combination
of bits. The analog signal is transmitted/ stored as a stream of
bits and reconstructed when required.

0
n
 Each quantization level corresponds to a unique combination
of bits. The analog signal is transmitted/ stored as a stream of
bits and reconstructed when required.

0
n
00 01 10 11 10 01 00
 Pulse Code Modulation (PCM)
x(t) Original Signal

0
t
 Pulse Code Modulation (PCM)
x~(t)
Quantized Signal

0
t

It is quite apparent that the quantized signal is not exactly the

same as the original analog signal. There is a fair degree of
quantization error here. However; as the number of quantization
levels is increased the quantization error is reduced and the
quantized signal gets closer and closer to the original signal
 Pulse Code Modulation (PCM)
x~(t)
Quantized Signal

0
t

It is quite apparent that the quantized signal is not exactly the

same as the original analog signal. There is a fair degree of
quantization error here. However; as the number of quantization
levels is increased the quantization error is reduced and the
quantized signal gets closer and closer to the original signal
 Sampling
• Formatting an analog signal proceeds in three steps.
*Discretisation in time, which is sampling.
*Discretisation in amplitude or quantisation.
*Encoding the quantised values or simply encoding.

• The operation of sampling discretises an analog signal in time domain so

that instead of a continuous time waveform we get continuous values of
the signal at discrete points of time.

• Diagram
• Sampling Theorem:
A band-limited signal having no spectral
components above fm Hz can be determined
uniquely by values sampled at uniform
intervals of

• The Nyquist rate of sampling which gives the

minimum sampling frequency needed to
reconstruct the analog signal from sampled
waveforms.
 Sampling
• Analog signal is sampled every TS secs.
• Ts is referred to as the sampling interval.
• fs = 1/Ts is called the sampling rate or sampling
frequency.
• There are 3 sampling methods:
– Ideal - an impulse at each sampling instant
– Natural - a pulse of short width with varying amplitude
– Flattop - sample and hold, like natural but with single
amplitude value
• The process is referred to as pulse amplitude
modulation PAM and the outcome is a signal with
analog (non integer) values
Three different sampling methods for PCM
Note

According to the Nyquist theorem, the

sampling rate must be
at least 2 times the highest frequency
contained in the signal.
Recovery of a sampled sine wave for different sampling rates
Nyquist sampling rate for low-pass and bandpass signals
 Quantisation
• Process of Converting analog sample of the
signal to discrete form.
• Graphically straight line representing the
relation between I/p and o/p is replaced by
stair case approximation process.
• Threshold like appearance.
• Round off Values
 Encoding
• CTS becomes DTS ,but not in the form best suited for
transmission over a line, radio path etc..

• Encoding process to translate the discrete set of

sample values to a more appropriate form of signal.

• Each discrete set of values are represented by CODE

• Binary Waveform e.g. NRZ unipolar ,NRZ Polar..etc

Baseband Processing
 Line codes
• In telecommunication, a line code (also called
digital baseband modulation) is a code
chosen for use within a communications
system for baseband transmission purposes.
Line coding is often used for digital data
transport.
 Line codes
• Electrical representation of binary data
stream.
a) Unipolar NRZ
b) Polar NRZ signalling
c) Unipolar RZ signalling
d) Bipolar RZ or AMI
e)Spit phase or Manchester code.
 Example

A complex low-pass signal has a bandwidth of 200 kHz.

What is the minimum sampling rate for this signal?
Solution
The bandwidth of a low-pass signal is between 0 and f,
where f is the maximum frequency in the signal.
Therefore, we can sample this signal at 2 times the
highest frequency (200 kHz). The sampling rate is
therefore 400,000 samples per second.
 Example

A complex bandpass signal has a bandwidth of 200 kHz.

What is the minimum sampling rate for this signal?
Solution
We cannot find the minimum sampling rate in this case
because we do not know where the bandwidth starts or
ends. We do not know the maximum frequency in the
signal.
 Quantization
• Sampling results in a series of pulses of varying
amplitude values ranging between two limits: a
min and a max.
• The amplitude values are infinite between the
two limits.
• We need to map the infinite amplitude values
onto a finite set of known values.
• This is achieved by dividing the distance between
min and max into L zones, each of height 
Step Size  = (max - min)/L
L=2^n
 Quantization Levels

– The midpoint of each zone is assigned a value

from 0 to L-1 (resulting in L values)

– Each sample falling in a zone is then

approximated to the value of the midpoint.
 Quantization Zones
– Assume we have a voltage signal with
amplitudes Vmin=-20V and Vmax=+20V.
– We want to use L=8 quantization levels.
– Zone width = (20 - -20)/8 = 5
– The 8 zones are: -20 to -15, -15 to -10, -10 to -
5, -5 to 0, 0 to +5, +5 to +10, +10 to +15, +15 to
+20
– The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5,
7.5, 12.5, 17.5
 Assigning Codes to Zones
– Each zone is then assigned a binary code.
– The number of bits required to encode the zones, or
the number of bits per sample as it is commonly
referred to, is obtained as follows:
nb = log2 L
– Given our example, nb = 3
– The 8 zone (or level) codes are therefore: 000, 001,
010, 011, 100, 101, 110, and 111
– Assigning codes to zones:
• 000 will refer to zone -20 to -15
• 001 to zone -15 to -10, etc.
Figure - Quantization and encoding of a sampled signal
 Quantization Error
• When a signal is quantized, we introduce an error
- the coded signal is an approximation of the
actual amplitude value.
• The difference between actual and coded value
(midpoint) is referred to as the quantization error.
• The more zones, the smaller  which results in
smaller errors.
• BUT, the more zones the more bits required to
encode the samples -> higher bit rate
 Bit rate and bandwidth requirements of PCM

• The bit rate of a PCM signal can be calculated form the

number of bits per sample x the sampling rate
Bit rate = nb x fs
• The bandwidth required to transmit this signal depends
on the type of line encoding used.
• A digitized signal will always need more bandwidth than
the original analog signal. Price we pay for robustness and
other features of digital transmission.
 Example

Problem

We want to digitize the 4000 Hz

human voice. What is the bit rate,
assuming 8 bits per sample?
 Solution

The human voice normally contains frequencies

from 0 to 4000 Hz. So the sampling rate and bit
rate are calculated as follows:

4.54
4.55
4.56
Encoding
 Combining the process of sampling and Quantization
produces discrete set of values, but not in the form best
suited to transmission over a line, radio path etc..
 Encoding process translate discrete set of values to
suitable form.
 Encoding speech and image
 Represent in code word (no. of bits) i.e. 2n

 Encoding techniques for analog source

 Types of encoder
 Temporal waveform coding- characteristics of source waveform
 Spectral waveform coding- signal waveform is subdivided in to different
frequency band. E.g Sub band coding-SBC
 Model based Coding- Mathematical modeling of the source.

 Binary Waveform
 Line codes
 Effect of Aliasing

 LPF is used to attenuate the High Frequency

components of the signal that are not essential to the
information.

 Filtered signal is sampled at a rate slightly higher than

the Nyquist rate.

4.58
 Transmission Path

Distorted PCM Regenerative Regenerative Regenerated

wave Repeater Repeater PCM wave

Block Diagram of Regenerative Repeater

Regenerated
Distorted PCM Decision
Amplifier PCM wave
wave Equalizer Making Device

Timing
Circuit
Regeneration

 The most important features of PCM

systems lies in the ability to control the
effects of distortion and noise produced by
transmitting a PCM wave through a
channel.
 Reconstructing the PCM wave by means of
a chain of regenerative repeaters located
along the transmission route.
Regeneration….

 Regenerative Repeaters
* Equalization
* Timing
* Decision making device

Equalizer – Shapes the received pulses

so as the compensate for the effects of
amplitude and phase distortion.
Regeneration…

* Timing Circuit – Provide the periodic

pulse train, derived from the received
pulse
* Decision Device – Timing circuit and
amplitude of the equalized pulse – decide
the output.
Regeneration…
 In practice, the regenerated signal departs
from the original signal for two main
reasons.
1) The presence of channel noise and interference
causes the repeater to make wrong decisions
occasionally, there by introducing bit errors in the
regenerated signal.
2) If the spacing between received pulses deviated from
its assigned value a jitter is introduced into the
regenerated pulse position.
 Receiver

Regeneration Decoder Reconstruction Destination

Input
circuit Filter
Decoding

 The first operation in the receiver is to

regenerate the received pulse.
 The clean pulses are the regrouped in to
code-words.
 Then decode into quantized PAM signal.
 Decoding – generating a pulse amplitude
– linear sum of all the pulses in the code
word ( 20, 21, 22 ….)
Reconstruction

 The final operation in the receiver is to

recover the analog signal.
 This done by passing the decode output
through a low pass reconstruciton filter
whose cutoff frequency is equal to the
message bandwidth ‘W’.
Noise in PCM

 Aliasing Noise
 Quantization Noise
 Channel Noise
 Inter symbol Interference
 Quantization Types
 Conversion of CTS to DTS
 Quantization Error is produced in PCM by
doing round off.
 Types -Uniform and Non uniform
 Constant Step size
 Step Size Variation
 Uniform Process has a two effects
 Midtread-Representation Values
 Midrisers- Decision Levels
 Diagram
4.68
Midtread-odd level
 Mid riser Level

4.70
Midtread and Midrise
 Quantization Noise
 Quantization noise is produced by rounding off
the sample values to the nearest levels.

 Evaluate the quantization noise by making

certain assumptions that permit a mathematical
analysis of the problem.

 Consider a memory less quantizer that is both

uniform and symmetric , with a total of L=2^n
levels.
 Zero mean and Variance

4.72
 Quantization Process

4.73
 Derivation for Signal to quantization Noise
 Consider a memory less quantizer, x denote
the quantizer input and y denote the
quantizer output.
 Assume that Quantization noise is uniformly
distributed over the range of -/2to /2
 Probability density function of quantization error
(Uniform distribution).
 Mean zero and variance E[x2] of quantization
noise.
 SQNR
  Derive the expression for signal to
quantization noise ratio for PCM system
that employs linear quantization
techniques. Assume that input to the PCM
system in a sinusoidal signal.

 SQN= 1.8+6n

4.75
 Trade off between channel BW and
Signalling rate

 BW ≥ (Signalling Rate /2)

 BW ≥ (nbfs /2)

 BW ≥ (nbfm)

4.76
Problem-Mid tread

To avoid significant overload distortion -4 σx to 4 σx

• SQN= (6n-7.2 )dB
Channel capacity

 Maximum rate at which information can

be transmitted over AWGN channel.
 SNR >>0 dB, then the channel is Band limited
 SNR <<0 dB, then the channel is power
limited
 For a noisy channel with capacity ‘c’
information transmitted at a rate “R’
bits/sec
 R<C, coding tech./modulation tech.
 R>C, prob. Of error.
Channel capacity
 Suppose that the spectrum of a channel is between 10
MHz and 12 MHz, and an intended capacity of 8 Mbps.
(1) What should be the SNR in order to obtain this
capacity?
(2) What would be the capacity if the environment starts
suffering lesser noise and the SNR goes up to 27 dB.
Answer

 B=2 MHz=2*10^6, C=8 Mbps=8*10^6

bps
1. C=B*log2(1+SNR) <=> 2^(C/B)-
1=SNR<=> SNR=15
2. SNR(dB)=10*log10(SNR)<=>SNR=10^2.7<
=>SNR=501
(approximately),C=18Mbps=18*10^6 bps
 Find the Nyquist rate and Nyquist interval for the
following signals.
Non Uniform Quantizer
• In telephonic communication , it is preferable to use a variable
separation between the levels.
• Loud talk
• Weak Talk
• Able to reduce the quantization noise power for low amplitude signals,
the average SNR improved .
• Step size can be made smaller for smaller signals and larger for larger
signals . Its difficult to implement.
• Practical approach
– Predistort the signals by compression characteristics and put in to
uniform quantizer.
– COMPANDING ( Compressed and Quantized signal )
– Block diagram (i/p, Compressor , Uniform quantizer and Expander)

• Two Compression Laws

• A-law and μ law .
 DPCM
 Value from present sample to next sample does not
differ by large amount.
 The adjacent samples of the signal carry the same
information with little difference.
 Diagram
 Over all bit rate will decrease and number of bits
required to transmit one sample will also be
reduced.
 Based on the principle of prediction.
 Value of present sample is predicted from the past.
 Prediction may not be exact but it is very close to the
actual value
4.88
References:

 [1] Simon Haykin, “Digital

Communication”, John Wiley, 2009
 [2] Farouzan, Data Communication,
TaTaMcGraw Hill, 2006.