Principles of Communication

Digital Pulse Modulation

Prepared by
Dr Shilpi Gupta
PULSE CODE MODULATION (PCM)
• Pulse code modulation (PCM) is the name given to the class of
baseband signals obtained from the quantized PAM signals by
encoding each quantized sample into a digital word. For baseband
transmission, the codeword bits are transformed to pulse waveforms.
1. Definition
• Pulse-code modulation is known as a digital pulse modulation
technique. In fact, the pulse-code modulation (PCM) is quite complex
compared to the analog pulse modulation techniques (i.e., PAM, PWM
and PPM) in the sense that the message signal is subjected to a great
number of operations.
• When pulse modulation is applied to a binary symbol, the resulting
binary waveform is called a pulse-code modulation (PCM) waveform.
Elements of a PCM system
It consists of three main parts i.e., transmitter, transmission path and receiver. The
essential operations in the transmitter of a PCM system are sampling, quantizing and
encoding as shown in figure.
• The quantizing and encoding operations are usually performed in the
same circuit which is known as an analog-to-digital converter (ADC).
• Also, the essential operations in the receiver are regeneration of
impaired signals, decoding and demodulation of the train of
quantized samples. These operations are usually performed in the
same circuit which is known as a digital-to-analog converter (DAC).
• At intermediate points, along the transmission route from the
transmitter to the receiver, regenerative repeaters are used to
reconstruct (i.e., regenerate) the transmitted sequence of coded pulses
in order to combat the accumulated effects of signal distortion and
noise.
• Thus, it is the combined use of quantizing and coding that
distinguishes pulse code modulation from analog modulation
techniques.
Few Important Points
(i) PCM is a type of pulse modulation like PAM, PWM or PPM but
there is an important difference between them PAM, PWM or PPM are
analog pulse modulation systems whereas PCM is a digital pulse
modulation system.
(ii) PCM output is in the coded digital form (code words). It is in the
form of digital pulses of constant amplitude, width and position. A PCM
system consists of a PCM encoder (transmitter) and a PCM decoder
(receiver).
(iii) It should be understood that the PCM is not modulation in the
conventional sense.
Because in modulation, one of the characteristics of the carrier is varied
in proportion with the amplitude of the modulating signal. Nothing of
that sort happen in PCM.
A PCM Generator or Transmitter
A practical block diagram of a PCM generator.
low-pass filter blocks all the frequency components which are
lying above f Hz. Now the signal x(t) is bandlimited to fm Hz.
m

The sample and hold circuit then samples this signal at the
rate of fs. Sampling frequency fs is selected sufficiently above
nyquist rate to aviod aliasing i.e.,
The output of sample and hold circuit is denoted by x(nTs).
This signal x(nTs) is discrete in time and continuous in
amplitude.
A q-level quantizer compares input x(nTs) with its fixed digital
levels. It assigns any one of the digital level to x(nTs) with its
fixed digital levels which results in minimum distortion or
error. This error is called quantization error.
• Thus, output of quantizer is a digital level called xq(nTs).
Now, the quantized signal level xq(nTs) is given to binary
encoder. This encoder converts input signal to ‘v’ digits
binary word. Thus xq(nTs) is converted to ‘v’ binary bits.
This encoder is also known as digitizer.
• It may be noted that it is not possible to transmit each bit of
the binary word separately on transmission line. Therefore ‘v’
binary digits are converted to serial bit stream to generate
single baseband signal. In a parallel to serial converter,
usually a shift register does this job. The output of PCM
generator is thus a single baseband signal of binary bits.
• an oscillator generates the clocks for sample and hold circuit
and parallel to serial converter.
 Quantization
• As digital communications have become the dominant form of
communication technology
• To handle the transmission of analog message signals by digital
means, the signal has to undergo an analog-to-digital conversion.
• After sampling an analog signal, the next step in its digital
transmission is the generation of the coded version (digital
representation) of the signal.
• Pulse Code Modulation (PCM) provides one method to meet such a
requirement.
• In PCM, the message signal is sampled and amplitude of each sample
is approximated (rounded off) to the nearest one of a finite set of
discrete levels.
Quantization
• This will enable us to represent both time and amplitude in discrete
form. Hence, it is possible to transmit the message signal by means of
a digital (coded) waveform. This process is called discretization in time
and Amplitude.
• Digitizing a signal often results in improved transmission quality, with
a reduction in distortion and an improvement in signal-to-noise ratio.
• Let us consider an analog signal as shown in figure (a). First of all, we
get samples of this signal according to sampling theorem. For this
purpose, we mark the time-instants t0, t1, t2 and so on, at equal time-
intervals along the time axis.
• At each of these time-instants, the magnitude of the signal is measured
and thus samples of the signal are taken.
QUANTIZATION
• However, since the magnitude of each sample can take any value in a
continuous range, the signal in figure (b) is still an analog signal.
QUANTIZATION
• In quantization, the total amplitude range which the signal may occupy
is divided into a number of standard levels.
• amplitudes of the signal x(t) lie in the range (– mp, mp) which is
partitioned into L intervals, each of magnitude Δ = 2mp/L
• Now, each sample is approximated or rounded off to the nearest
quantized level. Since each sample is now approximated to one of the L
numbers, therefore, the information is digitized. After quantization, the
analog waveform can still be recovered, but not precisely.
• In fact, improved reconstruction fidelity of the analog waveform can be
achieved by increasing the number of quantization levels (requiring
increased system bandwidth).
• The quantized signal is formed by taking any one of L values. Such a
signal is known as L-ary Signal.
• L ary signal is converted into a binary signal by using pulse coding
technique.
• For L=16, code formed by binary representation of 0-15 is known as
Natural Binary Code (NBC). Thus each sample is encoded by 4 bits.
Each sample is transmitted by group of 4 binary pulses (pulse code).
PCM in Audio signal
• The audio signal BW is about 15 kHz,
• Subjective tests show that signal articulation is not affected if all
components above 3400 Hz are suppressed.
• Objective in telecommunication is intelligibility rather than high
fidelity, the components above 3400 Hz are eliminated by a LPF.
• The residual signal is sampled at 8 KHz to avoid unrealizable filter
requirements for signal reconstruction.
• Each sample is finally quantized into 256 levels which requires a group
of 8 binary pulses to encode each sample.
• So a telephone signal requires 8 x 8000 = 64000 binary pulses per
second.
• At the receiver some pulses will be detected incorrectly. There are 2
sources of error in this scheme :
• quantization error and pulse detection error
• In most of the cases pulse detection error is negligible compared to
• quantization error
Because sampling rate is 2B, total no of samples over averaging interval T is 2BT.
Quantizer
Classification of Quantization Process
The quantization process can be classified into two types as under:
(i) Uniform quantization
(ii) Non-uniform quantization.
This classification is based on the step size as defined earlier.
(i) Uniform Quantizer
A uniform quantizer is that type of quantizer in which the ‘step size’ remains same
throughout the input range.
(ii) Non-uniform Quantizer
A non-uniform quantizer is that type of quantizer in which the ‘step-size’ varies
according to the input signal values.
• So/No, the SNR, is an indication of the quality of the received signal.
Ideally same quality (constant SNR) is required for all values of the
message signal power .
• But the SNR is directly proportional to signal power which varies from
talker to talker by as much as 40 dB.
• Signal power can also vary because of different lengths of the
connecting circuits.
• Even for the same talker, the quality of received signal will deteriorate
markedly when the person speaks softly.
• Statistically, it is found that smaller amplitudes predominate in speech
and larger amplitudes are much less frequent.
• Hence SNR will be low most of the time.
• Since quantizing steps are of uniform value v.
• Quantization noise is directly proportional to the square of step size 𝑁𝑞 =
v 2 /12.
• Problem can be solved by using smaller steps for smaller amplitudes (non
uniform quantizing).
• The same result is obtained by first compressing signal samples and then
using a uniform quantization.
• The horizontal axis is the normalized (input signal amplitude m divided by
signal peak value mp). Vertical axis is O/P signal y.
• The compressor maps input signal increments m into larger increments
y for small input signals and vice versa for large input signals.
• Hence m contains a larger no of steps (or smaller step size) when m is
small.
The quantization noise is smaller for smaller input signal power. An approximately
logarithmic compression characteristic yields a quantization noise nearly
proportional to signal power. Hence, making SNR practically independent of the
input signal power over a large dynamic range.
The loud talkers and stronger signals are penalized with higher steps in order to
compensate the soft talkers and weaker signals.
 Signal to quantization noise ratio in PCM
(with and without compression)

Two compression Laws:

µ -Law used in North America and Japan
A-Law used in Europe and rest of the
world.
Compressed samples must be restored to their original values at the
receiver by using an expander with a characteristic complementary to
that of the compressor.
The compressor and the expander together are called the compandor.
Generally, Compression of signal increases its BW.
But in PCM, signal is not compressed but its samples. Since no of samples
does not change, the problem of BW increase does not arise in PCM.
PCM Transmission Path

• The path between the PCM transmitter and PCM receiver

over which the PCM signal travel, is called as PCM
transmission path. The most important feature of PCM
system lies in its ability to control the effects of distortion and
noise when the PCM wave travels on the channel.
• PCM accomplishes this capacity by means of using a chain of
regenerative repeaters.
• Such repeaters are spaced close enough to each other on the
transmission path. The regenerative repeater performs three
basic operations namely, equalization, timing and decision-
making.
Hence, each repeater actually reproduces the clean noise free PCM
signal from the PCM signal distorted by the channel noise. This
improves the performance of PCM in presence of noise.
 Block Diagram of a Repeater
• The amplitude equalizer shapes the distorted PCM wave so as to compensate for
the effects of amplitude and phase distortions.
• The timing circuit produces a periodic pulse train which is derived from the input
PCM pulses. This pulse train is then applied to the decision-making device. The
decision-making device uses this pulse train for sampling the equalized PCM
pulses. The sampling is carried out at the instants where the signal to noise ratio is
maximum.
The decision device makes a decision about whether the equalized
PCM wave at its input has a 0 value or 1 value at the instant of
sampling.
Such a decision is made by comparing equalized PCM with a reference
level called decision threshold as illustrated in figure. At the output of
the decision device, we get a clean PCM signal without any trace of
noise.
 PCM RECEIVER
The regenerator at the start of PCM receiver reshapes the pulse and removes the
noise. This signal is then converted to parallel digital words for each sample.

Now, the digital word is converted to its analog value denoted as xq(t) with the help
of a sample and hold circuit. This signal, at the output of sample and hold circuit, is
allowed to pass through a lowpass reconstruction filter to get the appropriate original
message signal denoted as x(t).
• It is impossible to reconstruct exact original signal x(t) because of permanent
quantization error introduced during quantization at the transmitter. In fact, this
quantization error can be reduced by increasing the binary levels. This is
equivalent to increasing binary digits (bits) per sample.
• But increasing bits ‘v’ increases the signalling rate as well as transmission
bandwidth. Therefore, the choice of these parameters is made, in such a manner
that noise due to quantization error (i.e., also called as quantization noise) is in
tolerable limits.
Transmission Bandwidth and Output SNR
For a binary PCM, a distinct group of n binary digits (bits) to each of the
L quantization levels. Because a sequence of n binary digits can be
arranged in 2𝑛 distinct patterns.
L = 2𝑛 or n = log2L
Each quantized sample is, thus, encoded into n bits.
Because a signal m(t) bandlimited to B Hz requires a minimum of 2B
samples per second, we require a total of 2nB bits per second (bps), i.e.
2nB pieces of information per second.
A unit BW can transmit a maximum of 2 pieces of information per
second, we require a minimum channel of BW BT Hz, given by
BT = n B Hz
This is the theoretical minimum transmission BW required to transmit
the PCM signal.
Exponential Increase of Output SNR
𝐿2 = 22𝑛
Output SNR can be expressed as
𝑆𝑜
= 𝑐 2 2𝑛
𝑁0
3𝑚2 (𝑡)
𝑚𝑝2
𝑐=
3
ln(1 + 𝜇 2

𝑆𝑜 2𝑛 𝑆𝑜 2𝐵𝑇 /𝐵
Substitute BT = n B Hz ( n = BT/B) into
𝑁0
=𝑐 2 , 𝑁0
=𝑐 2
Hence, SNR increases exponentially with transmission BW BT.
A small increase in BW will result in large benefit in terms of SNR.
𝑆𝑜 𝑆𝑜
= 10 log10
𝑁𝑜 𝑑𝐵 𝑁𝑜

= 10 log10 [𝑐 2 2𝑛 ]
= 10 log10 C + 2n log10 2
= (α + 6n) dB
Increasing n by 1 (one bit in codeword) SNR increases by 6 dB. So in
PCM, SNR can be controlled by transmission BW.

Ex: A signal m(t) of BW B = 4 KHz is transmitted using a binary companded PCM

with µ = 100. Compare the case of L=64 with the case of L= 256 from the point
of view of transmission BW and output SNR.
APPLICATIONS of PCM
i) With the advent of fibre optic cables, PCM is used in telephony.
(ii) In space communication, space craft transmits signals to earth.
Here, the transmitted power is quite small (i.e., 10 or 15 W) and the
distances are very large (i.e., a few million km).
However, due to the high noise immunity, only PCM systems can
be used in such applications.

ADVANTAGES OF PCM : SALIENT FEATURES OF PCM

(i) PCM provides high noise immunity.
(ii) Due to digital nature of the signal, we can place repeaters
between the transmitter and the receivers. Infact, the repeaters
regenerate the received PCM signal. This can not be possible in
analog systems. Repeaters further reduce the effect of noise.
ADVANTAGES OF…….
(iii) We can store the PCM signal due to its digital nature.
(iv) We can use various coding techniques so that only the
desired person can decode the received signal.

DRAWBACKS OF PCM
(i) The encoding, decoding and quantizing circuitry of PCM is
complex.
(ii) PCM requires a large bandwidth as compared to the other
systems.
Differential Pulse Code Modulation
PCM is not a very efficient system because it generates so many bits
and requires so much bandwidth to transmit.
Many different ideas have been proposed to improve the encoding
efficiency of A/D conversion.
In general, these ideas exploit the characteristics of the source signals.
DPCM is one such scheme.
Reason to use DPCM
It may be observed that the samples of a signal are highly correlated
with each other. This is due to the fact that any signal does not change
fast. This means that its value from present sample to next sample does
not differ by large amount.
 • The adjacent samples of the signal carry the same information with a
little difference. When these samples are encoded by a standard PCM
system, the resulting encoded signal contains some redundant
information. Figure illustrates this redundant information.

A continuous time signal x(t) shown by dotted line.

This signal is sampled by flat top sampling at
intervals Ts, 2Ts, 3Ts ... nTs.
The sampling frequency is selected to be higher
than nyquist rate. The samples are encoded by
using 3 bit (7 levels) PCM.
The sample is quantized to the nearest digital level
as shown by small circles in the figure. The encoded
binary value of each sample is written on the top of
the samples.
Redundant Information in PCM
We can observe from figure that the samples taken at 4Ts, 5Ts and 6Ts
are encoded to same value of (110). This information can be carried
only by one sample.
But three samples are carrying the same information means that it is
redundant.
samples taken at 9Ts and 10 Ts. The difference between these samples
only due to last bit and first two bits are redundant, since they do not
change.
• If this redundancy is reduced, then overall bit rate will decrease and
number of bits required to transmit one sample will also be
reduced.
• This type of digital pulse modulation scheme is known as Differential
Pulse Code Modulation (DPCM).
 Working Principle
The differential pulse code modulation works on the principle of
prediction. The value of the present sample is predicted from the past
samples. The prediction may not be exact but it is very close to the actual
sample value.

The sampled signal is denoted by x(nT ) and the predicted signal is

s

denoted by xˆ(nT ) . The comparator finds out the difference between the
s

actual sample value x(nT ) and predicted sample value xˆ(nT ) .

s s

This is known as Prediction error and it is denoted by e(nT ). It can be

s

defined as,
e(nT ) = x(nT ) – xˆ(nT )
s s s
Thus, error is the difference between unquantized input sample x(nTs) and prediction
of it xˆ(nTs ).
The predicted value is produced by using a prediction filter. The quantizer output
signal eq(nTs) and previous prediction is added and given as input to the prediction
filter.

This signal is called xq(nTs). This makes the prediction more and more closer to the
actual sampled signal.
Quantized error signal eq(nTs) is very small and can be encoded by using small number
of bits. Thus number of bits per sample are reduced in DPCM.
 The quantizer output can be written as,
eq(nTs) = e(nTs) + q(nTs)
Here, q(nTs) is the quantization error. The prediction filter input xq(nTs) is obtained by sum
xˆ(nTs ) and quantizer output i.e.,
x (nT ) = xˆ(nT ) + e (nT )
q s s q s

Substituting the value of e (nT ) in the

q s

above equation, we get,

x (nT ) = xˆ(nT ) + e(nT ) + q(nT )
q s s s s
 e(nT ) = x(nT ) – xˆ(nT )
s s s

x (nT ) = xˆ(nT ) + x(nT ) – xˆ(nT ) + q(nT )

q s s s s s

x (nT ) = x(nT ) + q(nT )

q s s s

• Therefore, the value of e(nT ) + xˆ(nT ) from above equation into

s s

equation, we get,
x (nT ) = x(nT ) + q(nT )
q s s s
Reception of DPCM Signal : Reconstruction of DPCM Signal
The decoder first reconstructs the quantized error signal from incoming
binary signal.
The prediction filter output and quantized error signals are summed up
to give the quantized version of the original signal. Thus the signal at
the receiver differs from actual signal by quantization error q(nTs),
which is introduced permanently in the reconstructed signal.
Advantage of DPCM : Salient Features
(i) As the difference between x(nTs) and xˆ(nTs ) is being
encoded and transmitted by the DPCM technique, a small
difference voltage is to be quantized and encoded.
(ii) This will require less number of quantization levels and
hence less number of bits to represent them.
(iii) Thus signaling rate and bandwidth of a DPCM system will
be less than that of PCM.
 DELTA MODULATION
In PCM that it transmits all the bits which are used to code a sample.
Hence, signaling rate and transmission channel bandwidth are quite large in
PCM. To overcome this problem, Delta Modulation is used.
Sample correlation is used in DPCM is further exploited in Delta Modulation
(DM) by oversampling (typically 4 times the Nyquist rate) the baseband
signal.
This increase the correlation between adjacent samples, which results in
small prediction error that can be encoded using only 1 bit.
Delta modulation transmits only one bit per sample. Here, the present
sample value is compared with the previous sample value and this result
whether the amplitude is increased or decreased is transmitted.
 Working Principle
• Input signal x(t) is approximated to step signal by the
delta modulator. This step size is kept fixed.
• The difference between the input signal x(t) and staircase
approximated signal is confined to two levels, i.e., +  and
– .
• Now, if the difference is positive, then approximated
signal is increased by one step, i.e., ‘’. If the difference is
negative, then approximated signal is reduced by ‘’.
• When the step is reduced, ‘0’ is transmitted and if the
step is increased, ‘1’ is transmitted. Hence, for each
sample, only one binary bit is transmitted.
 Transmitter
The summer in the accumulator adds quantizer output (± ) with the previous sample
approximation. This gives present sample approximation. i.e.,

The previous sample approximation u[(n – 1)

T ] is restored by delaying one sample period
s

T . The sampled input signal x(nT ) and staircase

s s

approximated signal x^(nT ) are subtracted to get

s

error signal e(nTs).

Thus, depending on the sign of e(nT ), one bit
s

quantizer generates an output of +  or – .

If the step size is + , then binary ‘1’ is
transmitted and if it is – , then binary ‘0’ is
transmitted.
 Mathematical Expressions
The error between the sampled value of x(t)
and last approximated sample is given as,

Assume u(nTs) as the present sample approximation of staircase output,

Let us define a quantity b(nTs) in such a way that,

This means that depending on the sign of error e(nTs), the sign of step size  is
decided. In other words, we can write
Receiver
At the receiver end, the accumulator and low-pass filter (LPF) are used.
The accumulator generates the staircase approximated signal output and is
delayed by one sampling period T . It is then added to the input signal.
s

If input is binary ‘1’ then it adds +  step to the previous output (which is
delayed). If input is binary ‘0’ then one step ‘’ is subtracted from the
delayed signal.
Also, the low-pass filter has the cutoff frequency equal to highest frequency
in x(t).
This low-pass filter smoothens the
staircase signal to reconstruct original
message signal x(t).
Advantages of Delta Modulation : Salient Features of Delta Modulation
(i) Since, the delta modulation transmits only one bit for one sample,
therefore the signaling rate and transmission channel bandwidth is quite
small for delta modulation compared to PCM.
(ii) The transmitter and receiver implementation is very much simple for
delta modulation. There is no analog to digital converter required in delta
modulation.
Drawbacks of Delta Modulation
The delta modulation has two major drawbacks as under:
(i) Slope overload distortion,
(ii) Granular or idle noise
 Slope Overload Distortion
• This distortion arises because of large dynamic range of the
input signal. The rate of rise of input signal x(t) is so high that
the staircase signal cannot approximate it, the step size ‘’
becomes too small for staircase signal u(t) to follow the step
segment of x(t).
• Hence, there is a large error between the staircase
approximated signal and the original input signal x(t). This
error or noise is known as slope overload distortion.
• To reduce this error, the step size must be increased when slope of
signal x(t) is high. Since the step size of delta modulator remains fixed,
its maximum or minimum slopes occur along straight lines.
Therefore, this modulator is also known as Linear Delta Modulator
(LDM).
 Granular or Idle Noise
Granular or Idle noise occurs when the
step size is too large compared to small
variations in the input signal.
This means that for very small variations
in the input signal, the staircase signal is
changed by large amount () because of
large step size.
when the input signal is almost flat, the
staircase signal u(t) keeps on oscillating
by ±  around the signal.
 The error between the input and approximated signal is called granular
noise. The solution to this problem is to make step size small.

Therefore, a large step size is required to accommodate wide dynamic

range of the input signal (to reduce slope overload distortion) and
small steps are required to reduce granular noise. In fact, Adaptive
delta modulation is the modification to overcome these errors.

Bit Rate (i.e., Signaling Rate) of Delta Modulation

Delta modulation bit rate (r) = Number of bits transmitted/second
= Number of samples/sec × Number of bits/sample
= fs × 1 = fs
Delta modulation, in comparison to PCM (and DPCM), it is a very simple and
inexpensive method of A/D conversion.
A 1-bit codeword in DM makes word framing unnecessary at the
transmitter and the receiver. In fact, this strategy allows us to use fewer bits
per sample for encoding a baseband signal.

The delta modulation bit rate is (1/N) times the bit rate of a PCM system,
where N is the number of bits per transmitted PCM codeword.
Hence, we can say that the channel bandwidth for the Delta modulation
system is reduced to a great extent as compared to that for the PCM
system.
ADAPTIVE DELTA MODULATION
• To overcome the quantization errors due to slope overload
and granular noise, the step size () is made adaptive to
variations in the input signal x(t). Particularly in the steep
segment of the signal x(t), the step size is increased.
• Also, if the input is varying slowly, the step size is reduced.
Then, this method is known as Adaptive Delta Modulation
(ADM). The adaptive delta modulators can take continuous
changes in step size or discrete changes in step size.
• The step size increases or decreases according to a specified
rule depending on one bit quantizer output.
• As an example, if one bit quantizer output is high (i.e., 1), then step
size may be doubled for next sample.
• If one bit quantizer output is low, then step size may be reduced by
one step.
Receiver Part
In the receiver of adaptive delta modulator, there are two portions.
The first portion produces the step size from each incoming bit.
Exactly the same process is followed as that in transmitter. The
previous input and present input decides the step size.
It is then applied to an accumulator which builds up staircase
waveform. The low-pass filter then smoothens out the staircase
waveform to reconstruct the original signal.
the staircase waveforms of adaptive delta modulator and the
sequence of bits to be transmitted.
Advantages of Adaptive Delta Modulation : Salient Features
Adaptive delta modulation has certain advantages over delta
modulation as under:
(i) the signal to noise ratio becomes better than ordinary delta
modulation because of the reduction in slope overload distortion and
idle noise.
(ii) because of the variable step size, the dynamic range of ADM is
wider than simple DM.
(iii) utilization of bandwidth is better than delta modulation.
End of Unit