Principles of Communication

Digital Pulse Modulation

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. 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.
• 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 fm Hz. Now the signal x(t) is
bandlimited to fm Hz.
The sample and hold circuit then samples this signal at
the rate of fs. Sampling frequency fs is selected
sufficiently above Nyquist rate to avoid aliasing
The output of sample and hold circuit is denoted by
x(nTs). This signal x(nTs) is discrete in time and
continuous in amplitude.
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.
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 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.
• Digitizing a signal often results in improved transmission
quality, with a reduction in distortion and an improvement in
signal-to-noise ratio.
Quantization
• Considering 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.
• 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.
• 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, 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

Let m(kTs) be the kth sample of the signal m(t). If 𝑚(kTs)

ෝ is the
corresponding quantized sample, then from the reconstruction formula

and
Where, 𝑚
ෝ t is the signal reconstructed from quantized samples. The distortion
component q(t) in the reconstructed signal is therefore q(t) = 𝑚
ෝ t −m(t). Thus,

where q(kTs) is the kth quantization error sample. The error signal q(t) is the
undesired effect, and, hence, acts as noise, known as quantization noise. To calculate
the power, or the mean square value of q(t), we can use its time-average
Signals sinc (2πBt −mπ) and sinc (2πBt −nπ) are orthogonal

the integrals of the cross-product terms on the right-hand side

From the orthogonality relationship

the right-hand side of Eq. represents the average, or the mean, of

the square of quantization error samples.
each input sample value to the uniform quantizer is approximated
by the midpoint of the subinterval (of height v) in which the sample
falls; the maximum quantization error is ±v/2. Thus, the quantization
error lies in the range (−Δv/2, Δ v/2), where,
Since the signal value m(t) can be anywhere within (−mp, mp), we can assume that
the quantization error is equally likely to lie anywhere in the range (− Δv/2,
Δv/2). Under such assumption, the mean square quantization error 𝑞ሸ2 is given by

Assuming that the pulse detection error at the receiver is negligible, the reconstructed
signal 𝒎
ෝ 𝒕 at the receiver output is

Power of quantized noise

in which the desired signal at the output is m(t), and the (quantization) noise is q(t).
 Because 𝑞 2ሸ(𝑡) is the mean square value or power of the quantization noise, it will
be denoted by No,

Since the power of the message signal m(t) is 𝑚2ሸ(𝑡) , then the signal power and
the noise power within 𝑚(t)
ෝ , respectively, are

Hence, the resulting SNR is simply

SNR is a linear function of the message signal power 𝑚2ሸ(𝑡)

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.
• 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 .
Quantization noise is directly proportional to the square
of step size 𝑁𝑞 =  /12.
2

• 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.
Non-uniform Quantization
• Non-uniform quantization can provide fine quantization of the
weak signals and coarse quantization of the strong signals.
Thus, in case of non-uniform quantization, quantization noise can
be made proportional to signal size.
• The effect is to improve the overall SNR by reducing the noise
for the predominant weak signals, at the expense of an increase in
noise for the rarely occurring strong signals.
• If the quantizer characteristics is non-linear and the step size is
not constant instead if it is variable, dependent on the amplitude
of input signal then the quantization is known as non-uniform
quantization.
• In non-uniform quantization, the step size is reduced with the
reduction in signal level. For weak signals (P < < 1), the step size
is small, therefore the quantization noise reduces, to improve the
signal to quantization noise ratio for weak signals.
Companding
• The step size is thus varied according to the signal level to keep the
signal to noise ratio adequately high. This is non-uniform quantization.
The non-uniform quantization is practically achieved through a
process called companding.
• Companding is non-uniform quantization. It is required to be
implemented to improve the signal to quantization noise ratio of weak
signals. We know that the quantization noise is given by
COMPANDING (i.e., COMPANDED PCM)
In the uniform quantization, once the step size is fixed, the
quantization noise power remains constant. However, the signal
power is not constant. It is proportional to the square of signal
amplitude.
Hence signal power will be small for weak signals, but quantization
noise power is constant. Therefore, the signal to quantization noise
for the weak signals is very poor. This will affect the quality of signal.
The remedy is to use companding.
Companding is a term derived from two words i.e., compression and
expansion as under:
Companding = Compressing + Expanding
In practice, it is difficult to implement the non-uniform quantization
because it is not known in advance about the changes in the signal level.
Therefore, a particular method is used.
The weak signals are amplified and strong signals are attenuated before
applying them to a uniform quantizer. This process is called as
compression and the block that provides it is called as a compressor.
At the receiver exactly opposite is followed which is called expansion. The
circuit used for providing expansion is called as an expander. The
compression of signal at the transmitter and expansion at the receiver is
combined to be called companding.
Compressor Characteristics
The standard telephone technique of handling the large range of possible
input signal levels is to use a logarithmic-compressed quantizer instead of a
uniform one. With such a non-uniform compressor, the output SNR is
independent of the distribution of input signal levels.
The compressor provides a higher gain
to the weak signals and smaller gain to the
strong input signals. Thus, weak signals
are artificially boosted to improve the
signal to quantization noise ratio.
This compressor characteristics has been shown only for the positive input
signal but it can be drawn even for the negative input signals. In fact, the
compressor is included at the PCM transmitter.

For small magnitude signals, the compression characteristic has a much

steeper slope than for large magnitude signals.

Thus, a given signal change at small magnitudes will carry the uniform
quantizer through more steps than the same change at large magnitudes.
Expander Characteristics
This characteristics is exactly the inverse of the compressor characteristics. This
ensures that all the artificially boosted signals by the compressor are brought back
to their original amplitudes at the receiver end.
Compander Characteristic

Due to the inverse nature of compressor and expander, the overall characteristics of the compander is
a straight line (dotted line in figure). This indicates that all the boosted signals are brought back to
their original amplitudes.
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 ‘n’ 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 x(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.
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.
Redundant Information in PCM
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.
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).
When an input analog signal is sampled at a rate higher than the Nyquist
rate, successive samples become highly correlated.
There is very little difference in amplitude between two successive
samples.
 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 𝑋(nTs). The comparator finds out the difference between the
෠
actual sample value X(nT ) and predicted sample value 𝑋(nTs).
s

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

s

defined as,
s s ෠
e(nT ) = X(nT ) – 𝑋(nTs).
Thus, error is the difference between unquantized input sample X(nTs) and prediction
෠
of it 𝑋(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 close 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.
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

Reduced Bit rate compared to PCM

Better Compression
Disadvantages
• Error Propagation (Subsequent difference Error Propagation
is possible)
• Predictor complexity

Applications
• Audio Compression
• Video Compression
• Image Compression
SQNR in DPCM
DELTA MODULATION
Why Delta:
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.
Delta modulation is a simplified, 1-bit version of DPCM where the
difference between consecutive samples is quantized to only a single bit
(representing a positive or negative change).
DPCM is a more general technique that quantizes the difference between
a predicted sample and the actual sample into an n-bit data stream,
allowing for more flexible and higher-fidelity signal representation and
compression.
 Working Principle
• 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.
• 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.
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

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.
 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 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.
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 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. Infact, Adaptive delta
modulation is the modification to overcome these errors.
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
 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 logic for step size control is added in the diagram. 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.
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