Digital Communication

Prof. Sandhya Potadar

Course Objectives:

1.Explain functional block of Digital Communication

System
2. Analyze PCM, DPCM, DM, ADM source coding
techniques
3. Explain conversion of digital data to digital signal and
ISI for reliable baseband transmission
4. Classify random processes
5. Describe binary and M-ary digital modulation techniques
6. Explain the optimum filter, correlation receiver and
response of matched filter receiver in presence of noise
7. Describe the principle of spread spectrum modulation
including pseudo noise sequence
 Course Outcomes
After completion of the course, students will be able to
1. Describe waveform coding technique and evaluate bitrate,
bandwidth and signal-to- noise ratio
2. Describe and interpret data formats, multiplexing,
synchronization and Intersymbol Interference for reliable
baseband Transmission
3. Classify random processes in terms of mean, variance and
autocorrelation
4. Describe and analyze bandpass modulation techniques along with
their performance measure - bit period, bandwidth, signal space
representation and Euclidian distance
5. Analyze the error probability of digital modulation techniques
with matched filter and correlator 6. Illustrate the concept of
Direct sequence and Frequency hopped spread spectrum
 Scope of the subject
• Communications is a process by which information is exchanged
between individuals through a common system of symbols, signs,
or behaviour
“It is about communication between people; the rest is technology”

• Communication systems are reliable, economical and efficient

means of communications
– Public switched telephone network (PSTN), mobile telephone
communication (GSM, 3G, 4G...), broadcast radio or television, navigation
systems, ...
•

SYLLABUS

Unit 1: Digital Transmission of Analog Signal

Comparison between analog and digital communication, Block
diagram of digital communication system, Sampling Process, PCM
Generation and Reconstruction, Quantization Noise, Non-uniform
Quantization and Companding, PCM with noise: Decoding noise, Error
threshold, Differential Pulse Code Modulation, Delta Modulation,
Adaptive Delta Modulation, and Delta Sigma Modulation.
 General structure of a communication system

Noise
Transmitted Received Received
Info. signal signal info.
Source
SOURCE Transmitter Channel Receiver User

Transmitter

Source Channel
Formatter Modulator
encoder encoder

Receiver

Source Channel
Formatter Demodulator
decoder decoder
 Block Diagram of Digital
Communication System
remove
redundant symbols are converted
digital to waveforms
symbols information security Reduce (Pe)

Frequency Spread :produce a signal that is less

vulnerable to interference, enhance privacy
Multiple access : similar to Multiplex
 Digital Signal Nomenclature
• Information Source
• Character
• Digital Message
• M – ary
• Digital Waveform
• Bit Rate
– Actual rate at which information is transmitted per second
• Baud Rate
– Refers to the rate at which the signaling elements are transmitted, i.e.
number of signaling elements per second.
• Bit Error Rate
– The probability that one of the bits is in error or simply the probability of
error
 Advantages of Digital Communication system
Immunity to transmission noise & Distortion
Reliability & flexibility of Circuits
Ease of multiplexing
Ease of signalling
Performance moniterability
Ease of encryption
 Digital Communication advantages
•Reliable communication; less sensitivity to changes in
environmental conditions (temperature, etc.)
•Easy multiplexing
•Easy signaling
• Hook status, address digits, call progress information
•Voice and data integration
•Easy processing like encryption and compression
•Easy system performance monitoring
• QOS monitoring
•Integration of transmission and switching
•Signal regeneration, operation at low SNR, superior performance
Disadvantages of digital Communication system
➢ Increased bandwidth (64 KB for a 4 KHz channel, without compression
(However, less with compression)
➢Increased Signalling Rate
➢Increased System Complexity & Need for time synchronization
(Bit, character, frame synchronization needed
➢Analogue to Digital and Digital to Analogue conversions
•Very often non-linear ADC and DAC used, some performance degradation

➢Non graceful Degradation

➢Topologically restricted multiplexing.

➢ Incompatibility with existing analog equipment

.
 Digital versus analog Communications:
Why Digital?
1. Easy to regenerate the distorted signal
– Regenerative repeaters along the transmission path can detect a digital signal and retransmit
a new, clean (noise free) signal
– These repeaters prevent accumulation of noise along the path
This is not possible with analog communication systems
2. Immunity to distortion and interference
– Two-state signal representation
The input to a digital system is in the form of a sequence of bits (binary or M-ary)
❑ Digital communication is rugged in the sense that it is more immune to channel noise and
distortion
3. Hardware is more flexible
– Digital hardware implementation is flexible and permits the use of microprocessors, mini-
processors, digital switching and VLSI
4. Shorter design and production cycle & Low cost
– The use of LSI and VLSI in the design of components and systems have resulted in lower cost
 Why Digital Communications?
5. Can combine different signal types – data, voice, text, etc.
Different kinds of digital signal are treated identically.

Data Voice
A bit is a bit!
Media

6. Easier and more efficient to multiplex several digital signals

–Digital multiplexing techniques – Time & Code Division Multiple Access -
are easier to implement than analog techniques such as Frequency
Division Multiple Access

7.Encryption and privacy techniques are easier to implement

8. Better overall performance
–Digital communication is inherently more efficient than analog in realizing the exchange of SNR
for bandwidth
–Digital signals can be coded to yield extremely low rates and high fidelity as well as privacy
 Performance Criteria
• Digital (DCS): The probability of error (PE)
• Analog: SNR signal-to-noise ratio
percent distortion
expected mean-square error between
transmitted & received waveforms

▪Nongraceful degradation :if S/N drops below threshold,

quality changes from very good to very poor. Degrade
gracefully in analog
 Goals in Communication SystemDesign
• To maximize transmission rate, R
• To maximize system utilization, U
• To minimize bit error rate, Pe
• To minimize required systems bandwidth, W
• To minimize system complexity, Cx
• To minimize required power, Eb/No
 Bandwidth of signal
• Baseband versus bandpass:

Baseband Bandpass
signal signal
Local oscillator
 Base band system
Transforming the information source to a form compatible with a digital
system

Formatting and transmission of baseband signal

Digital info.

Textual Format
source info.
Pulse
Analog Transmit
Sample Quantize Encode modulate
info.

Pulse
Bit stream waveforms
Channel
Format
Analog
info. Low-pass
filter
Decode Demodulate/
Receive
Textual Detect
sink
info.

Digital info.
 Steps for Formatting Textual data
1.Convert Characters into bit stream
2.Form group of K bits to form new set of symbols
size
M = 2K k=log2M
&
The system using a symbol set size of M is referred to as an M-ary system
– For K= 1 then, M= 2 Hence binary system
– For K= 2 then M=22 = 4 Hence Quaternary system
– For K= 3 then M=23 = 8 Hence 8-ary system & so on
The bits are therefore partitioned into groups of three [K = log 28=3] bits
and forms a symbol

3. Generate wave forms

If symbol set size =8 The transmitter must generate 8 waveforms
Si(t) , where i=0, 1 , 2 …..7 to represent each symbol
 Formatting analog signals
• To transform an analog waveform into a form
that is compatible with a digital
ccommunication system, the following steps
are taken:
1. Sampling
2. Quantization and encoding
3. Baseband transmission
 Sampling
• The problem is how to choose the sampling
interval Ts so that the original analog signal can
be reconstructed
• The sampler takes a snapshot of the x(t) for
every Ts
Sampling
 Sampling theorem
Analog Sampling Pulse amplitude
signal process modulated (PAM) signal

• Sampling theorem: A bandlimited signal

with no spectral components beyond , can be
uniquely determined by values sampled at
uniform intervals of

– The sampling rate, is called

Nyquist rate.
Time domain Frequency domain
Impulse
Sampling
 Natural
Sampling
 Aliasing
Eliminating aliasing using antialiasing filters Sharp cutoff filter

prefiltered

Continuous spectrum

Sampled signal spectrum

Postfiltering (signal structure wellknown)

Continuous spectrum

Sampled signal spectrum

Sampling
 Practical Sampling Rates
• Speech
Telephone quality speech has a bandwidth of 4 kHz (300 to 3300Hz)
Most digital telephone systems are sampled at 8000 samples/sec

• Audio:
The highest frequency the human ear can hear is approximately
15kHz
- CD quality audio are sampled at rate of 44,000 samples/sec

• Video

The human eye requires samples at a rate of at least 20 frames/sec

to achieve smooth motion

36
 Pulse Code Modulation
1. An analog message signal is converted to discrete form in both
time and amplitude and then represented by a sequence of coded
pulses
2. PCM is a digital transmission system with an analog-to-digital
converter (ADC) at the input and a digital-to-analog converter
(DAC) at the output.
• The analog input waveform x(t) is lowpass filtered and sampled
to obtain x(kTs).
• A quantizer rounds off the sample values to the nearest discrete
value in a set of q quantum levels.
• The resulting quantized samples xq (kTs) are discrete in time
(sampling) discrete in amplitude(quantizing)
PULSE CODE MODULATION
PCM is the most commonly used technique in digital communications
Used in many applications:
Telephone systems
Digital audio recording
CD laser disks
voice mail
digital video etc.
 PCM Generation
• The analog input waveform x(t) is lowpass
filtered and sampled to obtain x(kTs) .
• A quantizer rounds off the sample values to
the nearest discrete value in a set of q
quantum levels.
• The resulting quantized samples xq(kTs) are
discrete in time (sampling)
discrete in amplitude(quantizing)
PCM Generation
 Uniform Quantizers

Two types of quantization: (a) midtread and (b) midrise.

In uniform quantization , quantization step size is constant

Uniform Quantization & Problem Quantization Noise

Signal/Noise Ratio (SNR) = Good @ High level signal

Signal/Noise Ratio (SNR) = Bad @ Low level signal
 PCM receiver

Q-PAM
Note that quantization error
amplitude is limited to  k  1/ q
( q − 1) / q
Q-PAM
(quantized signal
amplitude)
2/ q

PAM (analog signal

TS
amplitude)
− ( q − 1) / q 44
7/16/2020
reconstructed waveform Prof.Anuradha Fukane
Refer to Quantization noise word document
Speech signals are Non-Uniform: The amplitude is more likely to be
close to zero than to be at a high level.
 Non-uniform quantization
• It is achieved by uniformly quantizing the “compressed” signal.
• At the receiver, an inverse compression characteristic, called “expansion” is
employed to avoid signal distortion.

compression+expansion companding

y=C(x) x̂
x(t ) y (t ) yˆ (t ) xˆ (t )

x ŷ
Compress Qauntize Expand
Transmitter Channel Receiver
 48
Non-uniform Quantization (Companding)

Compressor Uniform Quantizer Expander

m(t ) mˆ (t )

The 3 stages combine to give

the characteristics of a Non-
uniform quantizer.
Companding
 Companding Law
Audio compression schemes defined by Consultative Committee for
International Telephony and Telegraphy (CCITT) G.711 which compress 16-
bit linear PCM data down to 8 bits of logarithmic data.
◼ There are in fact two standard logarithm based companding techniques
❑ US standard called µ-law companding
❑ European standard called A-law companding
 µ Law

m & v are normalized input & output voltages

µ = positive constant (µ = 0 : linear quantizer)
A Law
 UNIFORM and NONUNIFORM QUANTIZATION:

A –law and µ-law are audio compression schemes defined by Consultative

Committee for International Telephony and Telegraphy(CCITT) G.711 which
compress 16-bit linear PCM data down to eight bits of logarithmic data.
Compression laws

m -law A-law.

PCM telephone systems in PCM telephone systems

USA, Canada and Japan in Europe
 Differential Pulse Code Modulation (DPCM)

• For the signals which does not change rapidly from one sample to next sample, the
PCM scheme is not preferred.
• When such highly correlated samples are encoded the resulting encoded signal
contains redundant information.
• By removing this redundancy before encoding an efficient coded signal can be
obtained.
• One of such scheme is the DPCM technique.
• By knowing the past behavior of a signal up to a certain point in time, it is possible
to make some inference about the future values.

USED in:
lossy compression techniques,
JPEG and in adaptive DPCM (ADPCM),
➢ The sampling frequency is selected to be higher than the Nyquist rate.
Samples taken at 4Ts, 5Ts, and 6Ts are encoded to the same value of (110). This
information can be carried only by one sample value. But three samples are carrying the
same information means redundant.
If the redundancy is reduced, then the overall bitrate will decrease and the number of
bits required to transmit one sample will also reduce.
The signals at each point are named as −
•m(n) - is the sampled input for every Ts secs.
** Or some books write it as {m(n) or m(nTs) or x(kTs)}
^
• m(n) - is the predicted sample

•e(n) - is the difference of sampled input and predicted output, often called
as prediction error

•eq(n) - is the quantized output

• mq[n] - is the predictor input which is actually the summer output of the
predictor output and the quantizer output
 Prediction Gain ( Gp):
The output signal-to-quantization noise ratio of a signal coder is

 X2
( SNR) O = 2 variance of the
Q prediction error Prediction
error-to-quantization noise
  X2   E2 
( SNR) O =  2  2  = G P ( SNR) P ratio
 
E   Q 
Prediction gain
The prediction gain is maximized by minimizing the variance of the prediction error.
The prediction gain is defined by
1
GP =
(
1 − 12 )
 E2 =  X2 (1 − 12 ) ρ1 – Autocorrelation function of the
message signal
 PROBLEM:
•Consider a DPCM system whose transmitter uses a first-
order predictor optimized in the minimum mean-square
sense. Calculate the prediction gain of the system for the
following values of correlation coefficient for the message
signal:
R x (1) Rx (1)
(i ) 1 = = 0.825 (ii )  1 = = 0.950
R x (0) Rx (0)
1
GP =
Solution:
For ρ1= 0.825, Gp = 3.13 In dB , Gp = 5dB (
1 − 12 )
For ρ2 = 0.95, Gp = 10.26 In dB, Gp = 10.1dB
Implementation of DPCM
Quantization error is not accumulated.
• The name delta modulation reflects the fact
that each input sample x(k) has been encoded
as a single pulse of height +∂ or -∂ .
• The resulting binary waveform with signaling
rate rb = fs, or one bit per sample.
• For this reason DM is sometimes called “1-bit
PCM.” The corresponding transmission
bandwidth requirement is
BT >= rb/2 or fs/2
 DELTA MODULATION

+

−
Prof.Anuradha Fukane 68
 Delta modulation systems are subject to two types of quantization error:

1. slope –overload distortion,

2. granular noise.

or 
stepsize

To avoid slope
overload noise

Prof.Anuradha Fukane 69
To avoid slope overload distortion, slope of the staircase approximated
signal must be greater than or equal to rate of change of analog signal.
 Advantages & Disadvantages of DM
Advantages
◼ Low signaling Rate
◼ Low Transmission Bandwidth

Disadvantages
◼ Two distortion Occurs 1.slope Overload
2.Granular Noise
◼ Practically signaling rate is much higher than PCM due
to slope Overload distortion
◼ when the input signal is noisy then , noise can cause
accumulative distortion (errors) in demodulated signal at
the receiver.
Prof.Anuradha Fukane 73
 Adaptive Delta Modulation:
.
• Performance of a DM improved by making the step size of the modulator
assume a time-varying form.
• During a steep segment of the input signal the step size is increased.
Conversely, when the input signal is varying slowly, the step size is reduced.
• The size is adapted to the level of the input signal.
In practical implementations of the system, the step size (nTs ) or 2 (nTs )
is constrained to lie between minimum and maximum values.
controls the amount of slope- Inside these limits, the adaptation rule for  (nTs )
overload distortion
δmax δ(nTs) = g(nTs). δ(nTs – Ts)

time-varying multiplier
depends on the present binary outputb(nTs ) of the delta
δmin
controls the amount of idle modulator and the M previous values
b(nTs − Ts ), ....... b(nTs − MTs )
channel noise
g(nTs) = K if b(nTs) = b(nTs – Ts)
g(nTs) = K-1 if b(nTs) ≠ b(nTs – Ts)
ADM Transmitter.

ADM Receiver

75
 Delta-Sigma modulation
The  −  modulation which has an integrator can relieve the draw back of delta
modulation (differentiator)
Beneficial effects of using integrator:
1. Pre-emphasize the low-frequency content
2. Increase correlation between adjacent samples
(reduce the variance of the error signal at the quantizer input )
3. Simplify receiver design

Because the transmitter has an integrator , the receiver consists simply of a low-
pass filter.
(The differentiator in the conventional DM receiver is cancelled by the
integrator )
 Delta Sigma Modulation system
Noisy data cause cumulative errors in
demodulated signals and also due to presence
derivative of DC component

DM system

+integrator,pre-emphasizes the low

frequencies

DSM system Compensate,+differentiator

Feedback path-eliminate the

feedback integrator.
 COMPARISION OF VOICE ENCODING METHODS.

Encoding Sampling Bits per Bit rate

method rate sample Kbps
DM 64-128 1 64-128
PCM 8 7-8 56-64
ADM 48-64 1 48-64
DPCM 8 4-6 32-48
ADPCM 8 3-4 24-32
LPC .04-.1 80 3-8
These PPTs are for Internal Circulation only .

Thank You