Introduction to Communication Systems

COURSE TEACHER: PROF. DR. M. A MATIN

MAY 08, 2017

 Outline of lecture

 Introduction to Pulse Code Modulation

 Sampling
 Quantizing
 Encoding
 Analog to Digital Conversion
 Bandwidth of PCM Signals
 DPCM
 Pulse Code Modulation
 DEFINITION: Pulse code modulation (PCM) is essentially analog-to-digital
conversion of a special type where the information contained in the
instantaneous samples of an analog signal is represented by digital words in a
serial bit stream.

 The advantages of PCM are:

 Relatively inexpensive digital circuitry may be used extensively.
 PCM signals derived from all types of analog sources may be merged with data signals
and transmitted over a common high-speed digital communication system.
 In long-distance digital telephone systems requiring repeaters, a clean PCM waveform
can be regenerated at the output of each repeater, where the input consists of a
noisy PCM waveform.
 The noise performance of a digital system can be superior to that of an analog
system.
 The probability of error for the system output can be reduced even further by the
use of appropriate coding techniques.
 Sampling, Quantizing, and Encoding
 The PCM signal can be generated by carrying out three basic operations:

 Sampling
 Quantizing
 Encoding

1. Sampling operation generates a flat-top PAM signal.

2. Quantizing operation approximates the analog values by using a finite
number of levels. This operation focuses the followings

a) Uniform Quantizer
b) Quantization Error
c) Quantized PAM signal output

3. PCM signal is obtained from the quantized PAM signal by encoding each
quantized sample value into a digital word.
 Analog to Digital Conversion
 The Analog-to-digital Converter (ADC)
performs three functions:
 Sampling process
Analog  Makes the signal discrete in time.
Input  If the analog input has a bandwidth of
Signal W Hz, then the minimum sample
frequency such that the signal can be
Sample reconstructed without distortion.

 Quantization process
ADC  Makes the signal discrete in amplitude.
Quantize  Round off to one of q discrete levels.
111
110
101
100
 Encoding process
011
010  Maps the quantized values to digital
Encode
001
000
words that are n bits long.

 If the (Nyquist) Sampling Theorem is

satisfied, then only quantization introduces
distortion to the system.
Digital Output
Signal
111 111 001 010 011 111 011
 Quantization

Quantization is the process of “rounding off” a sample

according to some rule.
 E.g. suppose we must round to the nearest
tenth, then:
3.752 --> 3.8 0.001 --> 0
Illustration of the Quantization Error
 Uniform Quantization

Dynamic Range:  Most ADC’s use uniform

(-8, 8) quantizers.
 The quantization levels of a
Output sample 7
XQ uniform quantizer are
5
equally spaced apart.
 Uniform quantizers are
3

1
optimal when the input
-8 -6 -4 -2 -1 2 4 6 8 distribution is uniform. When
-3
Input sample X
all values within the Dynamic
Range of the quantizer are
equally likely.
-5

-7

Quantization Characteristic

Example: Uniform n =3 bit quantizer

q=8 and XQ = {1,3,5,7}
Quantization Example

Analog signal

Sampling

Quantization levels.
Quantized to 5-levels

Quantization levels
Quantized 10-levels
 PCM encoding example

Levels are encoded

using this table

Table: Quantization levels with belonging code words

M=8

Chart 2. Process of restoring a signal.

Chart 1. Quantization and digitalization of a signal. PCM encoded signal in binary form:
Signal is quantized in 11 time points & 8 quantization segments. 101 111 110 001 010 100 111 100 011 010 101
Total of 33 bits were used to encode a signal
 Encoding
 The output of the quantizer is one of M possible signal
levels.
 If we want to use a binary transmission system, then we need to
map each quantized sample into an n bit binary word.

M  2n , n  log 2 ( M )
 Encoding is the process of representing each quantized
sample by an n bit code word.
 The mapping is one-to-one so there is no distortion introduced by
encoding.
 Some mappings are better than others.
 A Gray code gives the best end-to-end performance.

 The weakness of Gray codes is poor performance when the sign

bit (MSB) is received in error.
 GRAY CODES EXAMPLE

 With gray codes adjacent samples differ only in one bit

position.
 Example (3 bit quantization):
XQ Natural coding Gray Coding
+7 111 110
+5 110 111
+3 101 101
+1 100 100
-1 011 000
-3 010 001
-5 001 011
-7 000 010
 With this gray code, a single bit error will result in an
amplitude error of only 2.
 Unless the MSB is in error.
 Waveforms in a PCM system for M=8
M=8

(a) Quantizer Input output characteristics

(b) Analog Signal, PAM Signal, Quantized PAM Signal

M  2n n  log 2 ( M )
M is the number of Quantization levels
(c) Error Signal n is the number of bits per sample

(d) PCM Signal

PCM Transmission SYSTEMS

PCM
 Practical PCM Circuits

 Three popular techniques are used to implement the

analog-to-digital converter (ADC) encoding operation:

 The counting or ramp, ( Maxim ICL7126 ADC)
 Serial or successive approximation, (AD 570)
 Parallel or flash encoders. ( CA3318)

 The objective of these circuits is to generate the PCM

word.

 Parallel digital output obtained (from one of the

above techniques) needs to be serialized before
sending over a 2-wire channel

 This is accomplished by parallel-to-serial converters

[Serial Input-Output (SIO) chip]
 UART,USRT and USART are examples for SIO’s
 Bandwidth of PCM signals

 The spectrum of the PCM signal is not directly related to the spectrum of the
input signal.
 The bandwidth of (serial) binary PCM waveforms depends on the bit rate R
and the waveform pulse shape used to represent the data.
 The Bit Rate R is
R=nfs

Where n is the number of bits in the PCM word (M=2n) and fs is the sampling
rate.
 For no aliasing case (fs≥ 2B), the MINIMUM Bandwidth of PCM Bpcm(Min) is:

Bpcm(Min) = R/2 = nfs//2

 The Minimum Bandwidth of nfs//2 is obtained only when sin(x)/x pulse is used
to generate the PCM waveform.

 For PCM waveform generated by rectangular pulses, the First-null Bandwidth

is:

 Bpcm = R = nfs
 PCM Noise and Companding

 Quantization Noise
 Signal to Noise Ratio
 PCM Telephone System
 Nonuniform Quantization
 Companding
 Quantization Noise

 The process of quantization can be interpreted as an

additive noise process.
Signal Quantized
X Signal
XQ
Quantization Noise
nQ

 The signal to quantization noise ratio (SNR)Q=S/N is given

as:

Average Power{ X }
( SNR)Q 
Average Power{nQ }
 Effect of Noise in PCM
 Two main effects produce the noise or distortion in the PCM output:
 Quantizing noise that is caused by the M-step quantizer at the PCM transmitter.
 Bit errors in the recovered PCM signal, caused by channel noise and improper filtering.

 If the input analog signal is band limited and sampled fast enough so that the
aliasing noise on the recovered signal is negligible, the ratio of the recovered
analog peak signal power to the total average noise power is:

 The ratio of the average signal power to the average noise power is

 M is the number of quantized levels used in the PCM system.

 Pe is the probability of bit error in the recovered binary PCM signal at the receiver DAC
before it is converted back into an analog signal.
 Effect of Quantization Noise

 If Pe is negligible, there are no bit errors resulting from channel noise and no ISI,
the Peak SNR resulting from only quantizing error is:

 The Average SNR due to quantizing errors is:

 Above equations can be expresses in decibels as,

Where, M = 2n
α = 4.77 for peak SNR
α = 0 for average SNR
 DESIGN OF A PCM SIGNAL FOR TELEPHONE SYSTEMS

 Assume that an analog audio voice-frequency(VF) telephone signal occupies a band from 300
to 3,400Hz. The signal is to be converted to a PCM signal for transmission over a digital
telephone system. The minimum sampling frequency is 2x3.4 = 6.8 ksample/sec.
 To be able to use of a low-cost low-pass antialiasing filter, the VF signal is oversampled with
a sampling frequency of 8ksamples/sec.
 This is the standard adopted by the Unites States telephone industry.
 Assume that each sample values is represented by 8 bits; then the bit rate of the binary
PCM signal is

 This 64-kbit/s signal is called a DS-0 signal (digital signal, type zero).
 The minimum absolute bandwidth of the binary PCM signal is

R nf s
BPCM  
2 2
This B is for a sinx/x type pulse sampling
 DESIGN OF A PCM SIGNAL FOR TELEPHONE SYSTEMS

 If we use a rectangular pulse for sampling the first null bandwidth is given by

 We require a bandwidth of 64kHz to transmit this digital voice PCM signal, whereas the
bandwidth of the original analog voice signal was, at most, 4kHz.

 We observe that the peak signal-to-quantizing noise power ratio is:

Note:
1. Coding with parity bits does NOT affect the quantizing noise,
2. However coding with parity bits will improve errors caused by channel or ISI,
which will be included in Pe ( assumed to be 0).
 Nonuniform Quantization
 Many signals such as speech have a nonuniform distribution.
 The amplitude is more likely to be close to zero than to be at higher levels.
 Nonuniform quantizers have unequally spaced levels
 The spacing can be chosen to optimize the SNR for a particular type of signal.
Output sample
XQ 6

2 Example: Nonuniform 3 bit quantizer

-8 -6 -4 -2 2 4 6 8

-2
Input sample
X
-4

-6
 Companding
 Nonuniform quantizers are difficult to make and expensive.
 An alternative is to first pass the speech signal through a
nonlinearity before quantizing with a uniform quantizer.

 The nonlinearity causes the signal amplitude to be

Compressed.

 The input to the quantizer will have a more uniform distribution.

 At the receiver, the signal is Expanded by an inverse to the
nonlinearity.

 The process of compressing and expanding is called

Companding.
 µ-law Companding

 Telephones in the U.S., Canada

and Japan use -law
companding:
Output |y(t)|

l n (1   | x ( t )|)
| y ( t ) |
ln (1   )

 Where  = 255 and |x(t)| < 1

0 1
Input |x(t)|
 Non Uniform Quantizing
 Voice signals are more likely to have amplitudes near zero than at extreme
peaks.
 For such signals with non-uniform amplitude distribution quantizing noise
will be higher for amplitude values near zero.
 A technique to increase amplitudes near zero is called Companding.

Effect of non linear quantizing

can be obtained by first passing
the analog signal through a
compressor and then through a
uniform quantizer.

x x’ x’ y
Q(.)
C(.)
Compressor Uniform Quantizer
 Example: µ-law Companding

x[n]=speech /song/
0 .5

-0 .5

-1
0 1 0 0 0 2 0 0 0 3 0 0 0 4 0 0 0 5 0 0 0 6 0 0 0 7 0 0 0 8 0 0 0 9 0 0 0 1 0 0 0 0

0 .5

y[n]=C(x[n]) 0

Companded Signal -0 .5

-1
0 1 0 0 0 2 0 0 0 3 0 0 0 4 0 0 0 5 0 0 0 6 0 0 0 7 0 0 0 8 0 0 0 9 0 0 0 1 0 0 0 0

0 .5
Close View of the Signal
Segment of x[n] 0

-0 .5

-1
2 2 0 0 2 3 0 0 2 4 0 0 2 5 0 0 2 6 0 0 2 7 0 0 2 8 0 0 2 9 0 0 3 0 0 0

Segment of y[n]
0 .5

Companded Signal -0 .5

-1
2 2 0 0 2 3 0 0 2 4 0 0 2 5 0 0 2 6 0 0 2 7 0 0 2 8 0 0 2 9 0 0 3 0 0 0
 A-law and µ-law Companding
 These two are standard companding methods.
 u-Law is used in North America and Japan
 A-Law is used elsewhere to compress digital telephone signals
 SNR of Compander
 The output SNR is a function of input signal level for uniform quantizing.

 But it is relatively insensitive for input level for a compander

 SNR Performance of Compander

 The output SNR is a function of input signal level for uniform quantizing.
 But it is relatively insensitive for input level for a compander.

α = 4.77 - 20 Log ( V/xrms) for Uniform Quantizer

where, V is the peak signal level and xrms is the rms value

α = 4.77 - 20 log[Ln(1 + μ)] for μ-law companding

α = 4.77 - 20 log[1 + Ln A] for A-law companding
 V.90 56-Kbps PCM Computer modem
 The V.90 PC Modem transmits data at 56kb/s from a PC via an analog
signal on a dial-up telephone line.

 A μ law compander is used in quantization with a value for μ of 255.

 The modem clock is synchronized to the 8-ksample/ sec clock of the

telephone company.

 7 bits of the 8 bit PCM are used to get a data rate of 56kb/s (
Frequencies below 300Hz are omitted to get rid of the power line noise
in harmonics of 60Hz).

 SNR of the line should be at least 52dB to operate on 56kbps.

 If SNR is below 52dB the modem will fallback to lower speeds ( 33.3
kbps, 28.8kbps or 24kbps).
 Differential Pulse Code Modulation (DPCM)

DPCM = differential PCM, ie take the difference and transmit that.

A model for a signal is s(n) = sˆ(n) + e(n) (1)

Where, s(n) is the signal sample at instant n,

sˆ(n) is an estimation (or approximation) of
signal, and
e(n) is an error term dependent on the
accuracy of the model.

The sˆ(n) term may be viewed as the deterministic component, with

the error e(n) the stochastic component. The deterministic
component is uniquely described by the model parameters, whilst the
sto chas tic co mponent i s descr ibe d in st atis tical ter m s .
 DPCM encoder and decoder

Figure 1: A differential PCM (DPCM) encoder. Figure 2: A differential PCM (DPCM) decoder.
The prediction is based upon the quantized The prediction (based on past quantized
prediction error eˆ(n) together with past predictions outputs) is added to the received error signal
s˜(n). eˆ(n) to generate each output sˆ(n).
 References

1. B P Lathi, Modern Digital and Analog Communication Systems, Oxford

University Press
2. M A Matin, Communication Systems for Electrical Engineers, Springer
3. Couch, Leon W. , Digital and Analog Communication Systems, New Jersey:
Prentice-Hall.
4. B Skalar, Digital Communications-Fundamentals and Applications, Pearson