Digital

Communications
Objectives
On completing this chapter, you should be
able to:

Compare analog and digital communication

techniques and discuss the appropriate use of
each,

Calculate the information capacity of a channel,

Calculate the minimum sampling rate for a

signal and explain the necessity for sampling at
that rate or above,
Objectives
On completing this chapter, you should be
able to:

Describe pulse code modulation and calculate

the number of quantizing levels, the bit rate,
and the dynamic range for PCM systems,

Perform calculations to show the effect of

compression on a PCM signal,

Describe the coding and decoding of a PCM

signal,
Objectives
On completing this chapter, you should be
able to:

Describe and compare the line codes in terms

of frequency components and clock-information
content,

Show how time-division multiplexing can be

used to send multiple digital signals over a
single channel,

Describe the operation of vocoders,

Digital Communications
The process of transmitting digital
pulses directly (wired system), or digitally
modulated analog carrier (wireless system),
from one point to another.
 Analog & Digital Systems
Analog signal with baseband transmission
Analog Analog
Source Baseband Channel Destination

Analog transmission with modulation and demodulation

Analog Modulator Demod. Analog
Source (Tx) Channel (Rx) Destination

Digital signal transmitted on digital channel

Digital Digital Digital
Coder Decoder
Source Channel Destination
 Analog & Digital Systems
Digital signal transmitted by modem
Digital Analog Digital
Modem Modem
Source Channel Destination

Analog signal transmitted digitally

Analog ADC Digital Decoding Analog
Source and Coding Channel DAC Destination

Analog signal digitized and transmitted by modem

ADC Analog Decoding
Modem Modem
and Coding Channel DAC

Analog Analog
Source Destination
 Why Digital Communications?
Advantages Disadvantages

More noise resistant Wide BW requirement

Higher S/N Needs coding and

decoding circuits
Easier to process,
switch, and multiplex Requires precise time
synchronization
Simpler to measure
and evaluate Incompatible w/ existing
 Information Theory

The scientific study of the efficient use of

bandwidth.

Vehicular Highway

Vehicular Traffic
Information Measure

Average Information (Entropy)

Relative Entropy

Redundancy

Rate of Information
 Example
A telephone touch-tone keypad has the
digits 0 to 9, plus the * and # keys. Assume
the probability of sending * or # is 0.005
and the probability of sending 0 to 9 is
0.099 each. If the keys are pressed at a rate
of 2 keys/s, compute the entropy and data
rate for this source.
 Example
From the given table
Symbol Probability of Time required to
Occurrence P(xi) transmit the symbol
xi
x1 0.21 10 s
x2 0.14 15 s
x3 0.09 20 s
x4 0.11 30 s
x5 0.15 25 s
x6 0.18 15 s
Determine thexfollowing: 0.12 25 s
7
a. Information Measure of each symbol in bits, dits and nats

b. Entropy (bits, dits and nats)

c. Relative Entropy (bits, dits and nats)

d. Redundancy

e. Rate of information (bps, dits/sec and nats/sec)

Information Capacity
A measure of how much information can be
transferred through a communications channel at
a particular instant of time.

A physical quantity
measured in bit or
in dit.

I = log2n
Information Highway
I = Amount of information
Note: 1 dit = 3.32 bits n = No. of coding levels
Limits of Information Capacity

Hartleys law (1928)

CBxt C = 2Blog2n

Where:
C = information capacity (bps)
B = bandwidth (Hz)
t = transmission time (s)
n = No. of coding levels
Shannons limit for info. capacity (1948)

C = B log2 (1 + S/N)

C = 3.32Blog (1 + S/N)

Total information (H):

H = Ct
Where:
S/N = signal-to-noise ratio
t = time in seconds
Sample Problems

1. A telephone line has a bandwidth of 3.2

kHz and S/N of 35 dB. A signal is
transmitted down this line using a four-level
code. What is the maximum theoretical
data rate?

2. A broadcast TV channel has 6 MHz of

bandwidth, what would be the permissible
S/N in dB if a 16 level code is used to
digitally transmit this signal?
 PULSE
MODULATION
Pulse Modulation
Sampling an analog information and
then converting the samples into discrete
pulses.
Common forms of PM
o Pulse Width Modulation

o Pulse Position Modulation

o Pulse Amplitude Modulation

PWM
The width of a
constant-amplitude
pulse is varied in Analog Signal

proportion to the ts ts
amplitude of the analog
signal at the time the
signal is sampled.
Sampling Pulse

It is also called as
pulse duration
modulation (PDM) or
pulse length modulation
PWM
(PLM)
 PWM
Disadvantage:
The transmitter is powerful enough to handle maximum-
width pulses

Advantage:
It still works if synchronization between transmitter and
receiver fails
 PWM
Generation:
Monostable multivibrator
PWM
Demodulation
Using an IC in which a signal emerges whose
amplitude at any time is proportional to the pulse width at
that time similar to class D amplifier
 PPM
Analog Signal
The position of a
ts ts
constant-amplitude,
constant-width
pulse, within a
Sampling Pulse
prescribed time slot
is varied according
to the amplitude of
PPM
the sample of the
analog signal.
 PPM
Disadvantage:
Dependent on transmitter-receiver synchronization

Advantage:
Requires constant transmitter power output
 PPM
Generation:
Generate PWM, then, remove the leading edges and the
bodies of the PWM pulses

Demodulation:
flip-flop or bistable multivibrator
PAM
Analog Signal

The amplitude ts ts
of a constant
position, constant-
width pulse, is Sampling Pulse
varied according to
the amplitude of
the sample of the
analog signal. PAM
 PCM
The only
digitally encoded
Analog Signal modulation
ts ts scheme.

The output
pulses are of fixed
Sampling Pulse

width and
amplitude and is
PCM binary as opposed
to PWM, PPM and
PAM.
 Applications
PAM
Used as an intermediate form of modulation
with PSK, QAM and PCM and is seldom used by
itself
PWM and PPM
Are used in special purpose communications
systems mainly for military but are seldom
used for commercial digital transmission
PCM
Most prevalent form of pulse modulation and
used in public switched telephone network
 Simplified BD of a Single-Channel Simplex PCM
System
PCM Transmitter
PAM Parallel
Data
Sample
BPF & Hold ADC P-S
Input
Sample pulse Conversion clock Line speed clock

Serial PCM Code Regenerative Serial PCM Code

Repeater
PCM Receiver
Parallel PAM
Data
Hold
S-P DAC Circuit LPF
Output

Line speed clock Conversion clock

PCM Sampling
Process of periodically sampling a
continually varying analog input signal, then
converting these samples to a multi-level
PAM signal.

Sampling Techniques
- Natural Sampling

- Flat-topped Sampling
Natural Sampling

The incoming
analog signal is low-
pass filtered and then
multiplied by pulse
train. It is when tops of
the sample pulses
retain their natural
shape during the
sample interval,
making it difficult for an
ADC to convert the
sample to a PCM code
Flat-topped Sampling

A sample-and-hold
circuit is used to keep the
amplitude of the sample
constant for the duration
of the pulse.

Aperture Error
an error introduced
when the amplitude of the
sampled signal changes
during sample pulse train
Natural v.s. Flat-top Sampling
Natural Flat-topped

Analog Signal Analog Signal

Sampling Sampling
Pulse Pulse

Output Output
Waveform Waveform
 Sample and Hold Circuit

Sampling pulse

+ +
Q1 PAM out
Analog input Z1 Z2

C1

Discussion:

Note:C
The FET
Aperture
storage
1acquisition
(Qis1the
) or
time
hold
Acquisition
acts
time
of
circuit.
the
as
should
Time
a%
capacitor
simple
be
the
very
analog
istime
short
called
tha
switch
to
FET
the
reduce
isA/D
and
on
provides distortion%
conversion
aperture a time
low because
impedance it is path
duringtothis
deposit
time that
thethe
analog
ADC
sample the
coverts voltage
sample across
voltage
capacitor
to PCMC1
Input and Output Waveform

Input waveform

Aperture time
Conversion time
Sample pulse Q1 Q1
On On
Capacitor Q1 Off Capacitor
charges discharges

Output Droop
waveform
Sample Problem
1. For the sample-hold-circuit, determine the largest
value of capacitor that can be used. Use an
output impedance for Z1 of 10 , an on
resistance for Q1 of 10 , an acquisition time of
10 s, a maximum peak-to-peak input voltage of
10 V, a maximum output current from Z1 of 10
mA and an accuracy of 1%.
Accuracy (%) Charge Time
10 2.3
1 4.6
0.1 6.9
0.01 9.2
Nyquist (1928) Sampling Theorem
The minimum sampling rate (fs) for a
given PCM system must be twice the highest
frequency of the analog input.

fs 2fa

Where:

fs = minimum Nyquist sampling rate (samples/sec)

fa = maximum analog input frequency (Hz)

Aliasing/Fold-Over Distortion
Occurs when the sampling rate is less
than the Nyquist minimum sampling rate.

2fs - fa
3fs - fa
f s - fa f s + fa
W/out Aliasing 2fs + fa
3fs + fa

Audio
0 fa fs 2fs 3fs Frequency

With Aliasing f s - fa f s + fa
2fs + fa 4fs - fa

0 fa fs 2fs - fa 2fs 3fs - fa 3fs 3fs + fa Frequency

Alias Frequency

falias = fs - fa

Where:
falias = the frequency of the aliasing distortion
fs = sampling rate
fa = the modulating (baseband) frequency
Solution to Aliasing:
Band pass filter before the sample-and-
hold circuit.
Effects of Aliasing

Without Aliasing

Original Signal and Reconstructed

Samples Signal
Sampling Times
Effects of Aliasing

With Aliasing

Original Signal and Reconstructed

Samples Signal
Sampling Times
Sample Problems
1. A PCM system uses a sample rate of 20, 000
samples/sec. What would be the highest cut-off of the BPF
used to rid the system from fold-over distortion?

2. Determine the alias frequency for a 14,000 samples/sec

sample rate and an analog input frequency of 8 kHz.

3. For a PCM system with a maximum audio input frequency

of 6kHz, determine the minimum sample rate and the
alias frequency produced if a 7 kHz signal were allowed to
enter the sample-and-hold circuit.
 Analog-to-Digital Conversion (ADC)
Analog signal Digital Signal
Xa(t)

X(n) Xq(n)
Sampler Quantizer Coder

1. Sampling
Conversion of a continuous-time signal into a discrete-
time signal
Taking samples of the continuous-time signal at
discrete-time instants
Sampling interval is T
 Analog-to-Digital Conversion (ADC)
Analog signal Digital Signal
Xa(t)

X(n) Xq(n)
Sampler Quantizer Coder

2. Quantization
Conversion of a discrete time continuous valued signal
into a discrete-time, discrete valued digital signal xq(n)
Digital signal values are infinite set of possible values
The difference between xq(n) and x(n) [xq(n) - x(n)] is
called the quantization error
 Analog-to-Digital Conversion (ADC)
Analog signal Digital Signal
Xa(t)

X(n) Xq(n)
Sampler Quantizer Coder

3. Coding
Each discrete value xq(n) is represented by a b-bit
binary sequence
Sampling of Analog Signals

Analog Xa(t) X(n) = Xa(nT) Discrete-

ADC Time
Signal
Signal

The discrete-time signal x(n) is obtained by taking-

samples of the analog signal xa(t) every T second.
Sampling of Analog Signals
The time interval T is called the sampling period or
sampling interval.
The sampling rate or the sampling frequency is

Relationship between the variable t of the analog

signal and the variable n of discrete-time signal
Sampling of Analog Signals
Consider an analog sinusoidal signal

Sampling frequency is Fs = 1/T, so that

Relative or normalize frequency:

Sample Problem
Consider two analog signals

The sampling rate is Fs = 40 Hz. Find x1(n) and

x2(n).
Sampling Theorem for Sinusoids
Sinusoids of different amplitudes, frequencies and
phase:

Where: N = number of frequency components

A = amplitude
F = frequency
= phase
Sample Problem
Consider an analog signal

a. Find the frequencies of the analog signal.

b. What is the largest frequency?
c. What is the minimum Nyquist rate?
 Trigonometric Identities
sin (A + B) = sinAcosB + cosAsinB
cos (A + B) = cosAcosB sinAsinB
sin (A B) = sinAcosB cosAsinB
cos (A B) = cosAcosB + sinAsinB
cos A = sin (A + 90o)
-cos A = sin (A - 90o )
-sin A = cos (A + 90o )
sin A = cos (A - 90o )
Assignment
The analog signal

a. What is the minimum Nyquist rate for this signal?

b. Using a sampling rate Fs = 5000 samples/s. What is
the discrete-time signal obtained after sampling?
c. What is the analog signal ya(t) we can reconstruct
from the samples if we use ideal interpolation?
d. repeat b and c if Fs = 15000 samples/sec
 Quantization
Process of converting an infinite number
of possibilities to finite number of
conditions.

In essence, quantization is the process

of rounding off the amplitudes of flat-top
samples to a manageable number of levels.
Note:
The number of bits used to represent a sample
determines the number of possible code
combinations.
Quantization Interval or Quantum

The magnitude difference between

adjacent steps.

Overload Distortion (Peak Limiting)

It occurs if the magnitude of the sample

exceeds the highest quantization interval.
Quantizing
Assigning PCM codes to absolute
magnitudes

Resolution

The magnitude of a quantum. It is equal

to the voltage of the minimum step size
which is equal to the least significant bit
(Vlsb) of the PCM code.
 Sign-Magnitude Code

Sign Magnitude Decimal Value

1 11 +3 Quantization
Interval is 1 V
1 10 +2
1 01 +1
1 00 +0
0 00 -0
0 01 -1
0 10 -2
0 11 -3
 Folded Binary Code
Decimal Quantization
Sign Magnitude
Value Range
1 11 +3 + 2.5 V to + 3.5 V

1 10 +2 + 1.5 V to + 2.5 V

1 01 +1 + 0.5 V to + 1.5 V

1 00 +0 + 0 V to + 0.5 V

0 00 -0 + 0 V to - 0.5 V

0 01 -1 - 0.5 V to - 1.5 V

0 10 -2 - 1.5 V to - 2.5 V

0 11 -3 - 2.5 V to - 3.5 V
 Quantization Error (Qe)
A.k.a Quantization Noise (Qn)

Results when a given PAM is quantized.

(magnitude is rounded off to the nearest
available level).

The maximum Qe is one-half the

magnitude of the quantum (the minimum
step size/resolution).
 Illustration of Qe
111 +3V 2.6 V

110 +2V 2V
101 +1V
Analog input
100 +0V
000 -0V
-1V
001 -1V
010 -2V
011 -3V Sample pulse

111 +3V t1 t2 t3

110 +2V
101 +1V
Output PAM
100 +0V
000 -0V
001 -1V
Sample time Sample time Sample time
010 -2V
110 001 111
011 -3V
PCM codes
 Reduced Qe

Analog input

Sample pulse

t1 t2 t3 tN

Output PAM
 Linear Input v.s. Output Transfer Curve

Vout Vout
Analog signal

quantization error

Vin Vin

Maximum positive
Maximum negative
quantizing error
quantizing error

Quantized signal

Qe = LSB
Sample Problems
1. Determine the quantized level, quantization error
and PCM code using Three-bit sign-magnitude
(resolution of 1 V) PCM code for the analog
sample voltage of
a. +1.07 V
b. +2.6 V
c. -3.95 V
 No. of PCM bits per Sample
Max. allowable input amplitude

Resolution

Dynamic Range
Dynamic Range (DR)
The ratio of the largest possible
magnitude (Vmax) to the smallest possible
magnitude (Vmin), other than zero, that can
be decoded by the DAC.
Mathematically:

Vmax
DR DR 2 1
n

Vmin
Where: DR = dynamic range (unitless or can be
expressed in dB) DR(dB) = 20 log (DR)
n = No. of bits in a PCM code, excluding
the sign bit
Vmax = maximum voltage magnitude that can
be discerned by the DACs in the receiver
Vmin = the quantum value (resolution)
Coding Efficiency
A numerical indication of how
efficiently a PCM code is utilized.

Mathematically:

Min. No. of bits

Coding eff . x100%
Actual No. of bits

Note:
No. of bits include the sign bit
Sample Problems
1. For a PCM system with following parameters,
determine: (a) minimum sample rate, (b)
minimum number of bits used in the PCM code,
(c) resolution, (d) maximum quantization error
and (e) coding efficiency.

maximum analog input frequency = 5kHz

maximum decoded voltage at the receiver = 3.05 V
minimum dynamic range = 50 dB
Signal-to-Quantization Noise Ratio
(SQR)

The signal power to the quantizing

noise power ratio.

The signal power to the quantizing

noise power ratio.

Also called as signal-to-distortion ratio.

Occurs when the input signal is at its

minimum amplitude.
Mathematically:
Resolution
SQR
Qe
v /R 2 v
SQR 10 log 2 SQR 10.8 20 log
(q / 12) / R q
Where:
R = resistance in ohms
q = quantization interval in volts
v = rms signal voltage in voltd
v2/R = average signal power in watts
(q2/12)/R = average quantization noise power in watts
Sample Problems
1. Determine the resolution and quantization
error for an 8-bit linear sign-magnitude PCM
code for a maximum decoded voltage of
1.27 V.

2. Determine the dynamic range for 10-bit

sign-magnitude PCM code.

3. For a resolution of 0.04 V, determine the

voltage of this sign-magnitude PCM code:
0110101.
Assignment

Determine the minimum number of bits used

in the PCM code, resolution, maximum Qe,
coding efficiency and minimum SQR. Then,
find the equivalent PCM code, quantization
error and SQR for -5.01 V, -0.32 V and 11.62
V for the following parameters:
Maximum decoded voltage at the receiver =
10.25 V
Minimum dynamic range = 100 dB
Linear v.s. Non-Linear PCM Codes
Linear Coding
The magnitude change
A between any two successive
M
steps is uniform.
P
L
The resolution for the higher
I
amplitude analog signals is
T
the same as for the lower
U amplitude signals.
D
E
The SQR for the lower
Input level amplitude signals is less
than for the higher
amplitude signals.
 Linear v.s Non-Linear PCM Codes
Non-Linear Coding
The step size of PAM
A
M
increases with the
P
amplitude of the input
signal.
L
I
T There is increase accuracy
U when it is needed.
D
E The Qe for higher amplitude
Input level
signals is higher, hence
decreasing the SQR.
 Idle Channel Noise
The only input to the PAM sampler when
there is no analog input signal.
Uniform code with Uniform code with
midrise quantization midtread
quantization

Idle channel
noise

Decoded noise No decoded noise

 Idle Channel Noise
Midtread Quantization
the first quantization interval is made
larger in amplitude than the rest of the
steps. As a result, the noise is suppressed
during the encoding process
Midrise Quantization
the lowest-magnitude positive and
negative codes have the same voltage
range as all the other codes
Coding Methods
Used to quantized PAM signals into 2n
levels.

Classifications:

Level-at-a-time Coding

Digit-at-a-time Coding

Word-at-a-time Coding
Coding Methods
Level-at-a-time Coding
Compares the PAM signal to a ramp waveform
while the binary counter is being advanced at a
uniform rate
When the ramp waveform equals or exceeds the

PAM sample, the counter contains the PCM code

Requires a very fast clock if the number of bits in
the PCM code is large
Is limited to low-speed applications
Coding Methods
Digit-at-a-time Coding
Determines each digit of the PCM code
sequentially
Is analogous to a balance where known reference

weights are used to determine an unknown weight

An example is a feedback coder that uses a
successive approximation register (SAR)
The entire PCM code word is determined in this

king of coder
Coding Methods
Word-at-a-Time Coding
Word-at-a-time coders are flash coders and are
more complex
Logic circuits sense the highest threshold circuit

sensed by the PAM input signal and produce the

approximate PCM code
Suitable for high-speed applications but
impractical for large values of n
Companding
Process of compressing then expanding a
signal.

The higher amplitude signals are compressed

(amplified less than the lower-amplitude
signals) prior to transmission, then
expanded (amplified more than the lower-
amplitude signals) at the receiver.

A mean of improving the dynamic range of a

communications system.
 Basic Companding Process
Compression Expansion
25 dB Compressed
+20 dB Dynamic Range +20 dB
+10 dB
+10 dB +10 dB

+5 dB

50dB 0 dB 0 dB 0 dB 50dB
DR DR
- 5 dB

- 10 dB -10 dB
- 10 dB

- 20 dB - 20 dB
- 15 dB
- 30 dB - 30 dB
Transmission
Input Output
Media
Common Forms:
Analog Companding

Compression was implemented using

specially designed diodes inserted in the
analog signal path in a PCM transmitter
prior to the sample-and-hold circuit.

Expansion was implemented with diodes

that were placed just after the LPF in the
PCM receiver.
 PCM System w/ Analog Companding
PCM Transmitter
PAM
Parallel
Data

Analog Sample
BPF Compressor & Hold ADC P-S
Input
Sample pulse Conversion clock Line speed clock

Serial PCM Code Regenerative Serial PCM Code

Repeater
PCM Receiver
PAM
Parallel
Data

Hold
S-P DAC Circuit
LPF Expander
Output

Line speed clock Conversion clock

 Log PCM Codes
-Law Companding
Used in the USA and Japan.

Vmax ln ( 1 Vin /Vmax )

Vout
ln ( 1 )

Where:
Vmax = Maximum uncompressed analog input amplitude
Vin = Amplitude of the input signal at a particular instant of time
= Parameter used to defined the amount of compression
Vout = Compressed output amplitude
 -Law Compression Characteristics
=255
100
40
1.0
15
5
Note:
Relative output amplitude

The parameter determines

the range of signal power in
which the SQR is relatively
constant.

Voice transmission requires

a minimum DR of 40 dB and a
7-bit PCM code, hence 100.
0
1.0
Relative input amplitude
Sample Problems
1.For a compressor with a = 255,
determine
a. the voltage gain for the following relative
values of Vin: Vmax, 0.75 Vmax, 0.5 Vmax
and 0.25 Vmax.
b. The compressed output voltage for a
maximum input voltage of 10 V.
c. Input and Output Dynamic ranges and
compression (in dB).
Note: compression (dB) = DRin(dB) - DRout(dB)
 Log PCM Codes
A-Law Companding
Used in Europe was established by ITU-T. It
has a slightly flatter SQR than -law but is
inferior in terms of idle channel noise.
Sample Problems
1. For a compressor with A = 5, determine
a. the voltage gain for the following relative
values of Vin: Vmax, 0.75 Vmax, 0.5 Vmax
and 0.25 Vmax.
b. The compressed output voltage for a
maximum input voltage of 4 V.
c. Input and Output Dynamic ranges and
compression.
Digital Companding

Involves compression in the transmitter

after the input has been converted to a
linear PCM code and then expansion in the
receiver prior to PCM decoding.
 PCM System w/ Digital Companding
PCM Transmitter
PAM
Parallel Compressed
Data PCM

Sample and Digital

BPF Hold Ckt. ADC Compressor P-S
Input
Sample pulse Conversion clock Line speed clock

Serial PCM Code Regenerative Serial PCM Code

Repeater
PCM Receiver
PAM
Parallel Expanded
Data PCM
Digital Hold
S-P Expander DAC Circuit LPF
Output

Line speed clock Conversion clock

12-bit-to-8-bit Digital Companding
Digital Compression Error
A numerical indication of how
efficiently a PCM code is utilized.

Mathematically:
Sample Problems
1. Determine the 12-bit linear code, the
quantization error, the 8-bit compressed
code, the decoded 12-bit code, analog
output signal, the compression error,
digital compression error and total error for
a resolution of 0.008 V and analog sample
voltages of (a) +0.064 V, (b) -0.418 V, (c)
+9.726 V, and (d) -3.623 V.
Two Main Categories of Data Compression

1. Lossless Compression
Involves transmitting all the data in the original signal
but using fewer bits

2. Lossy Compression
Allows for some reduction in the quality of the
transmitted signal
Lossless Compression
Look for redundancies in the data which is a
technique called run-length encoding
Useful for facsimile (fax) transmission

Example: A string of zeros can be replaced with a code

that tells the receiver the length of the string.
Lossy Compression
Can involve reducing the number of bits per
sample or reducing the sampling rate
Involves: first reducing the signal-to-noise ratio
and the second limits the high-frequency response
of the signal
An example is a vocoder
Vocoder (voice coder)
is a special voice encoder/decoder used for
digitizing speech signals
Is designed to reproduce only the short-term
power spectrum
typically produce unnatural sounding speech and
is generally used for recorded information
Its purpose is to encode the minimum amount of
speech information necessary to reproduce a
perceptible message with fewer bits than those
needed by a conventional encoder/decoder
Encoding Techniques
1. CHANNEL VOCODER

was developed by Homer Dudley in 1928

Compressed conventional speech
waveforms into an analog signal with a total
bandwidth of approximately 300 Hz
It uses bandpass filters to separate speech
waveform into narrower sub-band
Encoding Techniques
2. FORMANT VOCODER

Takes advantage of the formants which are

the peak frequencies of speech signal in
which most speech energy concentrates
It simply determines the location of these
peaks and encodes and transmits only the
information with the most significant short-
term components
Encoding Techniques
3. LINEAR PREDICTIVE CODER

Extracts the most significant portions of speech

information directly from the time waveform
rather than from the frequency spectrum
Produces a time-varying model of the vocal tract
excitation and transfer function directly from the
speech waveform
At the receiver end, a synthesizer produces the
speech by passing the specified excitation through
a mathematical model of the vocal tract
Encoding Techniques
3. LINEAR PREDICTIVE CODER

Two main ways of generating the excitation signal:

1. Pulse Excited Linear Predictive (PELP)
uses a white noise generator for unvoiced sounds, and a
variable-frequency pulse generator for voiced sounds
the pulse generator creates a tone rich in harmonics, as
is the sound produced by human vocal cords.
Electronic Switch

Pulse Generator

Filter

Noise Generator
Encoding Techniques
Two main ways of generating the excitation signal:
2. Residual Excited Linear Predictive (RELP)
apply the inverse of the filter that will be used at the
receiver to the voice signal
the output of this filter is a signal that, when applied to
the receiver filter, will reproduce the original signal exactly
one method to represent values is using a codebook and
transmit the number if the closest codebook entry
the receiver looks up the codebook entry, generates the
corresponding signal, and uses it instead of the pulse and
noise generators
Inverse Lookup
Voice signal Filter Residual signal Table Excitation signal
PCM Line Speed
Is simply the data rate at which serial PCM bits
are clocked out onto the transmission lines
It is dependent on the sample rate and the
number of bits in the compressed PCM code.

Where: line speed = the transmission rate in bps

samples/sec = sample rate (fs)
bits/sample = number of bits in the
compressed PCM code
Sample Problem
1. For a single-channel PCM system with a
sample rate fs = 8000 samples per second
and an eight-bit compressed PCM code,
determine the line speed.
DELTA Modulation
Uses a single-bit PCM code to achieve digital
transmission of analog signals.
With delta modulation, rather than transmit a
coded representation of the sample, only a single
bit is transmitted, which simply indicates whether
that sample is larger or smaller than the previous
sample
If the current sample is smaller that the previous,
a logic 0 is transmitted. If the current is larger
than the previous, a logic 1 is transmitted
Delta Modulation Transmitter
Delta Modulation Encoder
Delta Modulation Receiver
Problems with Delta Modulation
a. SLOPE Overload
The slope of the analog signal is greater than
the Delta Modulator can maintain
Solutions:

Increase the clock frequency

Increase the magnitude of the minimum

step size
Problems with Delta Modulation
b. Granular Noise
Happens when the original analog input signal
has a relatively constant amplitude, the
reconstructed signal has a variations that were
not present in the original signal
Solutions:

Decrease the step size

Adaptive Delta Modulation
Is a delta modulation system where the step size
of the DAC is automatically varied, depending on
the amplitude characteristics of the analog input
signals
Adaptive Delta Modulation

Problem with Delta Modulation when the

output of the transmitter is a string of
consecutive 1s or 0s
Solution: when three consecutive 1s or 0s
occur, the step size of the DAC is increased
or decreased by a factor of 1.5
Differential PCM (DPCM)
Is designed specifically to take advantage of the
sample-to-sample redundancies in typical speech
waveforms
With DPCM, the difference in the amplitude of two
successive samples is transmitted rather than the
actual sample, thus, fewer bits are required for
DPCM than conventional PCM
DPCM Transmitter
DPCM Receiver
Pulse Transmission
Practical digital systems utilize filters with
bandwidths that are approximately 30% or more
in excess of the ideal Nyquist bandwidth.

Secondary Lobe aka ringing tail

Result of bandlimiting a pulse causing this energy to
spread over a significantly longer time
Pulse Transmission
Output frequency spectrum:

Where: f() = rad/s

= 2f (rad/s)
T = pulse width (seconds)
Example:
Find the output frequency at 1 kHz at a period of 0.8
ms.
Pulse Transmission
Approximately 90% of the signal power is
contained within the first spectral null.

Nyquist Rate:

R = 2B
Where: R = signaling rate = 1/T
Intersymbol Interference (ISI)
Happens when the ringing tails of several pulses
overlapped, thus interfering the with the major pulse lobe.
Energy in the form of spurious responses from the third
and fourth impulses from one pulse appears during the
sampling instant (T = 0) of another pulse.
Rectangular pulses will not remain rectangular in less than
an infinite bandwidth. The narrower the bandwidth, the
more rounded the pulses.
It causes crosstalk between channels that occupy
adjacent time slots in a time-division-mukltiplexed carrier
system. Special filters are called equalizers are inserted in
the transmission path to equalize the distortion for all
the frequencies, creating a uniform transmission medium
and reducing transmission impairments.
Intersymbol Interference (ISI)
Four Primary Causes
1. Timing Inaccuracies
Causes if the rate of
transmission does not
conform to the ringing
frequency designed into the
channel
2. Insufficient Bandwidth
When the bandwidth is
reduced, ringing frequency is
reduced, and ISI is more
likely to occur
 Intersymbol Interference (ISI)
Four Primary Causes of ISI
3. Amplitude Distortion
Pulse Distortion results when the frequency
characteristics of a communications channel depart from
the normal or expected values. It occurs when the peaks are
reduced, causing improper ringing frequencies in the time
domain.
Amplitude Equalization compensation for pulse distortion
4. Phase Distortion
Occurs when frequency components undergo different
amounts of time delay while propagating through the
transmission medium
Special Delay Equalizers are placed in tranmission path to
compensate for the varying delays
Eye Pattern or Eye Diagram
is a convenient technique for determining the
effects of the degradations introduced into the
pulses as they travel to the regenerator
All waveforms are superimposed over adjacent
signaling intervals
 Sample Eye Diagram

Vertical Hairs
represent decision time
Horizontal Hairs
represent decision level

Data Transition Jitter

the overlapping signal pattern does not cross the
horizontal zero line at exact integer multiples of the
symbol clock
 ISI Degradation

Where: H = ideal vertical opening (cm)

h = degraded vertical opening (cm)

Example:
If the opening of the eye pattern is 90% as shown in
the figure, find the ISI degradation.
 Signal Power in Binary Digital Signals
Binary Digital Signals
a. when /T < 0.5

b. When /T = 0.5

Average Power Effective RMS value