Chapter 8

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Republic of Yemen

University of Saba Region


Faculty of IT&CS
Course: Communications Technology 1

Analog Pulse Modulation


Digital Pulse Modulation
Lec.8

Dr. Abdullah Alamri


Outline
1. Introduction
2. Sampling Theorem
3. Methods of Sampling
4. Anti-aliasing Filter
5. Quantization
6. Binary encoding
7. PCM

2
Transformations

3
Modulation

Continuous wave (CW)


modulation Pulse Modulation

AM Angle
Analog Pulse Digital Pulse
modulation Modulation Modulation

FM PM
PAM PPM PDM

DM PCM

4
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
Decode Demodulate/
filter Receive
Textual Detect
info.

Digital info.

5
Formatting of baseband signals
◼ Data already in a digital format would bypass the formatting
function.
◼ Textual information is transformed into binary digits by use
of a coder.
◼ Analog information is formatted using three separate
processes:
◼ sampling.
◼ quantization, and
◼ Coding.

6
transmission of baseband signal
through a baseband channel

◼ No channel can be used for the transmission of binary digits


with out first transforming the digits to waveforms that are
compatible with the channel.
◼ For base band channels, compatible waveforms are pulses.
◼ pulse modulate : the conversion from a bit stream to a
sequence of pulse waveforms.

◼ The output of the modulator is typically a sequence of


pulses with characteristics that correspond to the digits
being sent.

7
transmission of baseband signal
through a baseband channel

◼ After transmission through the channel, the pulse


waveforms are recovered (demodulated) and detected
to produce an estimate of the transmitted digits.

◼ The final step, (reverse) formatting, recovers an


estimate of the source information.

8
Components of PCM encoder

9
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

4.10
Three different sampling methods for PCM

4.11
Sampling

◼ How frequent to sample the signal? – “The


sampling theorem”.
◼ The main types of sampling
◼ Ideal sampling (impulse sampling)
◼ Natural sampling
◼ Falt-Top Sampling

12
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.

13
Note

According to the Nyquist theorem, the


sampling rate must be
at least 2 times the highest frequency
contained in the signal.
Ideal (impulse) Sampling
Time domain
Frequency domain
Aliasing effect

LP filter

Nyquist rate

aliasing
Natural sampling
(Sampling with rectangular waveform)

Signal waveform Sampled waveform

0
0 1 201 401 601 801 1001 1201 1401 1601 1801 2001
1 201 401 601 801 1001 1201 1401 1601 1801 2001

N
a
tu
ra
lsa
mp
le
r

0
1 2
014
016
018
011
00
112
011
40
116
011
80
120
01

17
Natural Sampling

18
Flat-Top sampling (Sample & Hold)
◼ Samples with zero width cannot be realized in
practice (esp. at Rx.).
◼ Sample and hold (S/H) to obtain a sequence of
pulses with a fixed amplitude
◼ Obtain an analog Pulse-Amplitude Modulation
(PAM) waveform.

19
Avoid Aliasing
◼ Band-limiting signals (by filtering) before
sampling.
◼ Sampling at a rate that is greater than
the Nyquist rate.
◼ Improving the aliased signal

Anti-aliasing A/D fs(t)


f(t)
filter conversion

Sampling
Aliasing
Increasing sampling rate
Quantization
◼ Amplitude quantizing: Mapping samples of a continuous
amplitude waveform to a finite set of amplitudes.

Out

In
▪Average quantization noise power
Quantized

▪Signal peak power


values

▪Signal power to average


quantization noise power
29
Quantization example
amplitude
x(t)
111 3.5

110 2.5 Quant. levels


101 1.5

100 0.5

011 -0.5 boundaries

010 -1.5

001 -2.5 x(nTs): sampled values


xq(nTs): quantized values
000 -3.5
Ts: sampling time
PCM t
codeword 110 110 111 110 100 010 011 100 100 011 PCM sequence
Sources of corruption of analog
recovered signal
◼ 1. Sampling and quantization effects
◼ Quantization noise
◼ Quantizer saturation
◼ Timing jetter (shifted recovered samples)
◼ 2. Channel effect
◼ Channel noise
◼ Intersymbol interference
◼ 3. S/N Ratio for quantized pulses
32
Quantization error

◼ Quantizing error: The difference between the


input and output of a quantizer e(t)=x
ˆ(t)−x(t)
Process of quantizing noise
Qauntizer
Model of quantizing noise
y = q(x)

x(t ) xˆ (t )
x(t ) xˆ (t )
x
e(t )

+
e(t) =
xˆ(t) − x(t)
33
Quantization error …
◼ Quantizing error:
◼ linear errors happen for inputs within the dynamic
range of quantizer
◼ Saturation errors happen for inputs outside the
dynamic range of quantizer
◼Saturation errors are larger than linear errors

34
Quantization
◼ Scalar Quantizer Block Diagram

◼ Mid-tread
◼ Mid-rise

35
Uniform and non-uniform quant.
◼ Uniform (linear) quantizing:
◼No assumption about amplitude statistics and correlation properties of the
input.
◼Not using the user-related specifications
◼Robust to small changes in input statistic by not finely tuned to a specific
set of input parameters
◼Simply implemented

◼Application of linear quantizer:


◼Signal processing, graphic and display applications, process control
applications
◼ Non-uniform quantizing:
◼Using the input statistics to tune quantizer parameters
◼Larger SNR than uniform quantizing with same number of levels
◼Non-uniform intervals in the dynamic range with same quantization noise
variance
◼Application of non-uniform quantizer:
◼Commonly used for speech
Quantization

Quantization is a non linear transformation which maps elements


from a continuous set to a finite set. It is also the second step
required by A/D conversion.

Analog Signal Sample Quantize Digital Signal


- Continuous time - Discrete time
- Continuous value - Discrete time - Discrete value
- Continuous value
Uniform Quantization
output w2(t)
V

-V V
input w1(t)

-V
Region of operation
For M=2n levels, step size :
 = 2V /2n = V(2-n+1)
Quantization Error, e
output w2(t)
V

-V V
input w1(t)

-V

Error, e
/2
-/2 input w1(t)
Nonuniform Quantizer

Used to reduce quantization error and increase the dynamic


range when input signal is not uniformly distributed over its
allowed range of values.

allowed
values input

values
for most time
of time
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.
Non-uniform quantization
◼ It is done 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
42
Statistical of speech amplitudes

◼ In speech, weak signals are more frequent than strong ones.


Probability density function
1.0

0.5

0.0
1.0 2.0 3.0
Normalized magnitude of speech signal
Using equal step sizes (uniform quantizer) gives low   for weak signals
S

 N q
and high  S  for strong signals.
 N q
◼ Adjusting the step size of the quantizer by taking into account the speech statistics
improves the SNR for the input range.

43
A-law and m−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
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 = log2 (M )
Encoding is the process of representing each quantized sample by
an  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
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.
PCM Transmission System
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

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