Analog and Digital

Communications
Lavanya Poluboyina
Department of ECE
 UNIT - IV
Pulse Modulation
 Contents
Pulse Analog Modulation
• Types of Pulse analog modulation
• PAM
• PWM
• PPM
• Comparison of FDM and TDM
Pulse Digital Modulation
• PCM Generation and Reconstruction
• Quantization Noise
• Non-Uniform Quantization and Companding
• DPCM
• DM
• Adaptive DM
• Noise in PCM and DM
Applications: Design of E1 and T1 digital-carrier systems
 Types of Modulation

Modulation

Analog Modulation Digital Modulation

Continuous Pulse ASK FSK PSK QAM

AM Angle Pulse Analog Pulse Digital

FM PM PAM PWM PPM PCM DPCM DM

ADM
12/05/24 4
 Introduction
• After continuous wave modulation, let’s enter into
Pulse modulation. The following are the analog
pulse modulation techniques:
• Pulse Amplitude Modulation
• Pulse Width Modulation
• Pulse Position Modulation
 Pulse Amplitude Modulation (PAM)

• In Pulse Amplitude Modulation (PAM) technique, the amplitude of

the pulse carrier varies, which is proportional to the instantaneous
amplitude of the message signal.
• The pulse amplitude modulated signal will follow the amplitude of
the original signal, as the signal traces out the path of the whole
wave. In natural PAM, a signal sampled at Nyquist rate can be
reconstructed, by passing it through an efficient Low Pass Filter (LPF)
with exact cutoff frequency.
• The following figures explain the Pulse Amplitude Modulation.
 Pulse Width Modulation (PWM)

• Refer Electronic Communication Systems by

George Kennedy
 Pulse Position Modulation (PPM)

• Refer Electronic Communication Systems by

George Kennedy
 Time Division Multiplexing
• Time division multiplexing (TDM) is a technique used for transmitting several analog message
signals over a single communication channel, by dividing the time frame in to number of
slots, i.e. one slot for each signal.
• Here there are four input signals; all are band limited to f x by the input Low pass filters, and
all these are sequentially sampled at the transmitter by using a rotary switch i.e.
commutator.
• This commutator makes fs revolutions per second and extracts one sample from each input
during each revolution.
• The out put of the switch is a PAM wave form containing samples of the input signals
periodically interfaced with time.
TDM: Example
•We have four sources, each creating 250 characters per second. If the interleaved
unit is a character and 1 synchronizing bit is added to each frame, find (a) the data
rate of each source, (b) the duration of each character in each source, (c) the frame
rate, (d) the duration of each frame, (e) the number of bits in each frame, and (f) the
data rate of the link.
1. The data rate of each source is 2000 bps = 2 Kbps.
2. The duration of a character is 1/250 s, or 4 ms.
3. The link needs to send 250 frames per second.
4. The duration of each frame is 1/250 s, or 4 ms.
5. Each frame is 4 x 8 + 1 = 33 bits.
6. The data rate of the link is 250 x 33, or 8250 bps
Statistical TDM
 Statistical TDM
• Addressing is required in Statistical TDM
• Slot size: the ratio of the data size to address size must be reasonable to make
transmission efficient
• No synchronization bit: no need for frame-level sync.
• Bandwidth: normally less than the sum of the capacities of each channel
Comparison of FDM and TDM
FDM TDM
Analog Multiplexing technique. Digital Multiplexing technique.
Share the bandwidth of the communication Share the bandwidth of the communication
channel spatially. channel temporally.
Synchronization is not required. Synchronization is required.
Requires complex circuitry at the transmitter Circuitry is not very complex.
and receiver.
Suffers from the problem of crosstalk due to Problem of crosstalk is negligible.
imperfect BPF.
No propagation delays Propagation delays present as signals are
transmitted in multiple time slots.
Channel utilization efficiency is poor. Channel utilization efficiency is good.
Less flexible Relatively more flexible
Expensive Relatively less cost
Model of Digital Communication System

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• Aim of the source coding is to remove the redundancy present in the message so
that bandwidth required for transmission is minimized.

Source Encoder
1. It converts the input i.e. symbol sequences into binary sequence of 0’s and 1’s, by
assigning code words to the input symbol sequences.
2. The important parameters of a source encoder are codeword, entropy and
efficiency of the coder.

Source Decoder
Source decoder converts the binary output into symbol sequences.
Channel Encoder/Decoder

1.The channel encoder adds redundant bits to the message.

2.Channel decoder will detect and correct the errors present in the received signal
with the help of added redundant bits.
3.In general, channel encoder divides the input message bits into blocks of k bits
and replaces each k-bit message block by an n-bit code word by introducing (n-k)
check bits to each message block.
Modulator:
1. The Modulator converts the input bit stream into an electrical waveform suitable
for transmission over the communication channel.
2. Modulator should be designed to minimize the effects of channel noise, to match
the frequency spectrum of transmitted signal with channel characteristics.

Demodulator:
1. The extraction of the message from the information bearing waveform produced
by the modulation is accomplished by the demodulator.
2.The output of the demodulator is bit stream.
Channel:
1. The Channel provides the electrical connection between the source and
destination.
2. The different channels are: Pair of wires, Coaxial cable, Optical fiber, Radio
channel, Satellite channel or combination of any of these.
3. The communication channels have only finite bandwidth, non-ideal frequency
response, the signal often suffers from amplitude and phase distortion as it travels
over the channel.
4. Also, the signal power decreases (with distance) due to the attenuation of the
channel.
5. The signal is corrupted by unwanted, unpredictable electrical signals referred to
as noise.
6. The important parameters of the channel are bandwidth, amplitude and phase
response, and the statistical properties of noise.
Advantages of Digital Communication System
• More immune to noise and interference
• Makes communication reliable with added error
detection and correction techniques
• Provides added security
• Easy Multiplexing
• Simpler, easier and cheaper compared to analog systems
(advancements in IC technology made it possible)

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Disadvantages of Digital Communication System

• Requires more transmission bandwidth.

• Needs synchronization in case of synchronous
operation

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 Sampling Theorem
• Sampling Theorem:
“A band limited signal can be reconstructed from it’s
samples if the sampling rate is at least equal to twice
the maximum frequency component present in it.”
Figure 1 shows a signal g(t) that is band limited.

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• Let’s say the maximum frequency component of g(t) is f m. To
recover the signal g(t) from it’s samples, it has to be sampled at a
rate fs ≥ 2fm.
• The minimum required sampling rate fs = 2fm is called Nyquist rate.

Proof: Let g(t) be a band limited signal whose bandwidth is

fm (ωm = 2πfm). See Fig. 2.

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• δT(t) is the sampling signal with fs = 1/T > 2fm. See Fig. 3.

• Let gs(t) be the sampled signal. Its Fourier Transform Gs(ω) is given
by
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30
• If ωs = 2ωm, i.e., T = 1/2fm. Therefore, Gs(ω) is given by

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• To recover the original signal G(ω):
1. Filter with a Gate function, H2ωm (ω) of
width 2ωm.
2. Scale it by T.

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• Aliasing – Aliasing is a phenomenon where the high frequency components of
the sampled signal interfere with each other because of inadequate sampling ωs
< 2ωm.

• Aliasing leads to distortion in recovered signal. This is the reason why sampling
frequency should be atleast twice the bandwidth of the signal.
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• Oversampling – In practice, signals are oversampled,
where fs is significantly higher than Nyquist rate to avoid
aliasing

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Different Sampling Methods

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Nyquist Sampling Rate

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Pulse Code Modulation (PCM)

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• Three processes
– The analog signal is sampled
– The sampled signal is quantized
– The quantized values are encoded as streams of bits

• Components of PCM Encoder

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• Sampling: PAM (Pulse amplitude Modulation)
- According to the Nyquist theorem, the sampling rate must be at least 2
times the highest frequency contained in the signal.

• Sampling Rate

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 x(t)
3

0
t

Consider the analog Signal x(t).

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 x[n]
3

0
n

The signal is first sampled

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3

0
n

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3

0
n

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3

0
Sample n

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3

0
And Hold n

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3

0
n

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 Assign Closest
Level
3

0
n

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3

0
n

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3

0
n

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3

0
n

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3

0
n

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 Each quantization level corresponds to a unique combination of bits.
The analog signal is transmitted/stored as a stream of bits and
reconstructed when required.
3

0
n

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 Each quantization level corresponds to a unique combination of bits.
The analog signal is transmitted/stored as a stream of bits and
reconstructed when required.
3

0
n
00 01 10 11 10 01 00

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 x~(t)
Quantized Signal

0
t
It is quite apparent that the quantized signal is not exactly the same as the original analog
signal. There is a fair degree of quantization error here. However; as the number of
quantization levels is increased the quantization error is reduced and the quantized signal
gets closer and closer to the original signal
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 x~(t)
Quantized Signal

0
t

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PCM Transmitter - Receiver

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• In the PCM generator, the signal is first passed through sampler which is sampled
at a rate of fs ,
where fs ≥ 2fm
• The output of the sampler x(kTs) which is discrete in time is fed to a q level
quantizer. The quantizer compares the input samples x(kTs) with it's fixed
quantization levels. It assigns any one of the quantization level to x(kTs) that results
in minimum distortion or error. This error is called quantization error. The
quantized signal xq(kTs) is given to binary encoder. The encoder converts each
quantized sample to an n bit binary word.
• The receiver starts by reshaping the received pulses, removes the noise and then
converts the binary bits to analog. The received samples are then filtered by a low
pass filter; the cut off frequency is at fc, and fc = fm
where fm is the highest frequency component in the original signal.

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• With PCM, the analog signal is sampled and converted to a serial n-bit binary code
(for each sample). Each code has the same number of bits and requires the same
length of time for transmission.

• PCM is by far the most prevalent form of pulse modulation.

• The term pulse code modulation is somewhat of a misnomer, as it is not really a

type of modulation but rather a form of digitally coding analog signals.

• PCM is the preferred method of communications within the PSTN (public switched
telephone network).

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 Quantization
• Quantization is the process of converting an infinite number of possibilities to a finite
number of conditions.

• PCM requires quantization to convert the analog signals containing an infinite

number of amplitude possibilities to a PCM code with a limited number of
combinations.

• The smaller the magnitude of a quantum (step size), the better the resolution and
the more accurately the quantized signal will resemble the original analog sample.

• The likelihood of a sample voltage being equal to one of the possible quantization
levels is remote. Therefore, each sample voltage is rounded off (quantized) to the
closest available level and then converted to its corresponding PCM code.

• The rounded off error is called the quantization error (ε).

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 The quantizer of above of fig. is known as “mid-riser” type. For such quantizer, a slightly positive and
a slightly negative values of the input signal will have different levels at output. This may be a problem
when the speech signal is not present but a small noise is present at the input of the quantizer. To
avoid such a random fluctuation at the output of the quantizer, the “mid-tread” type uniform quantizer may
be used.
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 Transmission Bandwidth of PCM
• Bit rate = Bits per sample * sampling rate

• Bandwidth required is half of the bit rate.

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 Calculation of SQNR in PCM
• In PCM, the effects of transmission error can be
ignored but there is effect of quantization error on
signal quality.

• There are so many ways to calculate SQNR for PCM

considering different conditions.

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Calculation of Quantization Noise Power(NQ)
• Errors are introduced in the signal because of the quantization process. This error
is called “quantization error” and is expressed as
ε = xq(kTs) - x(kTs)
• Let an input signal x(t) have an amplitude in the range of xmax to –xmax. Then the
total amplitude range is:
Total amplitude = xmax – (–xmax)
= 2xmax
• If the amplitude range is divided into ‘q’ levels of quantizer, the step size Δ is

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• It may be noted from the above expression that this ratio can be increased by
increasing the number of quantizer levels M.

• Also note that Si is the power of x (t) at input of the sampler and hence, may not
represent the SQNR at the output of the low pass filter in PCM decoder. However,
for large N, small δ and ideal and smooth filtering (e.g. Nyquist filtering) at the
PCM decoder, the power So of desired signal at the output of the PCM decoder
can be assumed to be almost the same as Si i.e., So ≈ Si

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69
 Non-uniform Quantization
• In a linear or uniform quantizer, the quantization error in the k-th sample is
ek =|x (kTs) – xq (kTs)|
and the maximum error magnitude in a quantized sample is,
± δ/2.
• So, if in x(t), small amplitudes are more probable in the input signal than
amplitudes closer to ‘± V’, then the quantization noise of such an input signal will
be significant compared to the power of x(t). This implies that SQNR of usually
low signal will be poor and unacceptable. In a practical PCM codec, it is often
desired to design the quantizer such that the SQNR is almost independent of
the amplitude distribution of the analog input signal x (t).

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• We know that

• Assume that full range voltage is 16v. Then max. quantization error will be 1v. For the low signal
amplitudes like 2v, 3v etc., the max. quantization error of 1v is quite high. But for large signal
amplitudes near 15v, 16v etc., it can be considered to be small.
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This is achieved by using a non-
uniform quantizer. In non-uniform
quantization, the step size is not
fixed. It varies as per input signal
amplitude. From the fig. it can be
observed that δ is small for low
input signal levels, whereas it is
higher for high input levels.
Therefore, SQNR will be improved
for low signal levels and keeping
the SQNR almost same throughout
range of the input signal.

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• Normally, we don’t know how the signal level varies in advance. Therefore, non-
uniform quantization becomes difficult to implement.
• So, a non-uniform quantizer can be considered to be equivalent to an amplitude
pre-distortion process [denoted by y = c (x) in fig.] followed by a uniform
quantizer with a fixed step size ‘δ’.

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• Mathematically, c (x) should be a
monotonically increasing function of ‘x’ with
odd symmetry shown in fig. The monotonic
property ensures that c(x) × c-1(x) = 1.

• Note that the operation of c-1(x) is necessary in

the PCM decoder to get back the original signal
undistorted. The range ‘± V’ of x (t) further
implies the following:
c (x) = + V , for x = +V;
= 0 , for x = 0;
= - V , for x = - V

6-75
• From the fig. it can be observed that the signal is amplified for low signal levels and
attenuated for high signal levels.
• After this process, uniform quantization is used.
• At the receiver, a reverse process is done, that is, the signal is attenuated for low
signal levels and amplified for high signal levels.
• Thus the compression of signal at transmitter and expansion at receiver is called
combinedly as companding.
• There are two popular standards for non-linear quantization known as
(a) The μ - law companding
(b) The A – law companding
• The μ - law has been popular in the US, Japan, Canada and a few other countries
while the A - law is largely followed in Europe and most other countries, including
India, adopting ITU-T standards.

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μ - law companding
• The compression function z(x) for μ - law companding is

• μ is a constant here. It ranges from 0 to 255 and μ = 0 corresponds to linear quantization. The typical value of μ
is 100.

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A - law companding
• The compression function z(x) for A - law companding is

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• As approximately logarithmic compression function is used for linear
quantization, a PCM with non-uniform quantization scheme is also referred as
“Log PCM” or “Logarithmic PCM” scheme.

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 Differential PCM
• In a typical PCM-encoded speech waveform, there are often successive samples
taken in which there is little difference between the amplitudes of the two
samples.

• This necessitates transmitting several identical PCM codes, which is redundant.

• Differential pulse code modulation (DPCM) is designed specifically to take

advantage of the sample-to-sample redundancies in typical speech waveforms.

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 Differential PCM
• With DPCM, the difference in the amplitude of two
successive samples is transmitted rather than the
actual sample. Because the range of sample
differences is typically less than the range of
individual samples, fewer bits are required for DPCM
than conventional PCM.

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 Differential Pulse Code Modulation (DPCM)
• The standard sampling rate for PCM of telephone grade band limited speech signal
is fs = 8 Kilo samples per sec with a sampling interval of 125 μ sec. Samples of such
speech signal are usually correlated as amplitude of speech signal does not change
much within 125 μ sec.
• So, if the difference between samples is quantized instead of
sample itself, then the number of
quantization levels can be
decreased and there by the number
of bits required to encode. This
type of digital pulse modulation
scheme, in which differential
quantization is used, is called
DPCM.
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• It works on the principle of prediction.
• The value of present sample is predicted from the past samples.
• Prediction may not be exact but it is very close to the actual sample value.

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……. (1)

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• Decoder first reconstructs the quantized output from incoming binary signal.
• Prediction filter output and quantized output 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.

Schematic diagram of DPCM Demodulator 85

Advantages:
• Improved SQNR
• Reduced Bandwidth requirement

Disadvantages:
• Complexity

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 DELTA MODULATION (DM)
• Delta modulation is a single-bit PCM.

• With conventional PCM, each code is a binary representation of both the sign and the
magnitude of a particular sample. Therefore, multiple-bit codes are required to
represent the many values.

• With DM, rather than transmitting a coded representation of the sample, only a single
bit is transmitted.

• The algorithm is quite simple.

• If the current sample is smaller than the previous sample, a logic 0 is transmitted.

• If the current sample is larger than the previous sample, a logic 1 is transmitted.
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• The input analog signal is sampled and converted to a PAM signal, which is
compared with the output of the DAC.

• The output of the DAC is a voltage equal to the regenerated magnitude of the
previous sample, which was stored in the up-down counter as a binary number.

• The up/down counter is incremented or decremented depending on whether the

previous sample is larger or smaller than the current sample.

• The up/down counter is clocked at a rate equal to the sample rate.

• Initially, the up/down counter is zeroed, and the DAC is outputting 0 V.

• The output of the comparator is a logic 1 condition (+V), indicating that the
current sample is larger in amplitude than the previous sample. On the next clock
pulse, the up/down counter is incremented to a count of 1. 89
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Limitations/Problems of DM
Slope overload distortion - when the analog input signal changes at a faster rate
(slope of the analog signal is greater) than the DAC can track, a kind of distortion
called “slope overload distortion occurs”.
• Increasing the clock frequency reduces the probability of slope overload
occurring.
• Another way to prevent slope overload is to increase the step size.
Granular noise - when the analog input signal has a relatively constant amplitude,
the reconstructed signal has variations that were not present in the original
signal. This is called granular noise.
• It is analogous to quantization noise in conventional PCM.
• It can be reduced by decreasing the step size.

Therefore, to reduce the granular noise, a small resolution (step size) is needed, and
to reduce the possibility of slope overload occurring, a large resolution (step size)
is required. Obviously, a compromise is necessary. 92
Granular noise is more prevalent in analog signals that have gradual slopes and whose
amplitudes vary only a small amount. Slope overload is more prevalent in analog signals
that have steep slopes or whose amplitudes vary rapidly. 93
SQNR of DM

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Advantages:
• Transmission bandwidth is quite small
• Transmitter and receiver implementation is very much
simple

Disadvantages:
• Slope overload distortion (startup error)
• Granular noise (hunting)

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 Adaptive DM (ADM)
• 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 signal.
• Figure shows how an adaptive delta modulator works.
• When the output of the transmitter is a string of consecutive 1s or 0s, this indicates
that the slope of the DAC output is less than the slope of the analog signal in either
the positive or the negative direction. Essentially, the DAC has lost track of exactly
where the analog samples are, and the possibility of slope overload occurring is
high.
• With ADM, after a predetermined number of consecutive 1s or 0s, the step size is
automatically increased. After the next sample, if the DAC output amplitude is still
below the sample amplitude, the next step is increased even further until
eventually the DAC catches up with the analog signal.
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101
• When an alternative sequence of 1s and 0s is occurring, this indicates that the
possibility of granular noise occurring is high. Consequently, the DAC will
automatically revert to its minimum step size and, thus, reduce the magnitude of
the noise error.
• A common algorithm for an adaptive delta modulator is when three consecutive
1s or 0s occur, the step size of the DAC is increased by a factor of 1.5.
• Various other algorithms may be used for adaptive delta modulators, depending
on particular system requirements.

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103
 PCM for Telephone signals
• PCM is widely used in transmission of speech signals in fixed line telephone
systems.

• T1 carrier system, an example of PCM, which was developed at Bell labs in the
U.S. and is still in use today in the US and Japan.

• A similar scheme called the E1 is used in Europe and Pakistan.

• These schemes are used to multiplex the speech from multiple subscribers and
transmit them to their destinations over a common “Time Shared” channel.
Hence the name time division multiplexing (TDM).

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• A voice signal is essentially limited to a band from 300 to 3400 Hz in the sense that
frequencies outside this band do not contribute much to articulation efficiency and
the telephone circuits that respond to this range of frequencies give quite
satisfactory service.
• Accordingly, it is customary to pass the voice signal through a low-pass filter with a
cutoff frequency of about 3.4 kHz prior to sampling. Hence, the nominal value of the
Nyquist rate is 6.8 kHz.
• The filtered voice signal is usually sampled at a slightly higher rate, namely, 8 kHz,
which is the standard sampling rate in digital telephony.
• For telephone grade speech signal with 8-bits per sample and 8-Kilo samples per
second, a typical SQNR of 38.4 dB is achieved in practice.

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• spectral components of speech signals are 300-3400 Hz
Nyquist Rate = 2x3400 = 6.8 kHz
Practical Sampling Rate fs= 8 kHz
Number of quantization levels = 256
Number of Bits/Sample n = 8 (log2256 )
Data Rate of 1 voice channel = nfs = 8x8000 = 64 kbps

• Therefore, Sampling Interval = 1/8000 = 125µs

– This means that the time between two consecutive samples (from the same source) is 125µs.
TDM systems exploit this fact and utilize this interval to sample signals from other subscribers.
In T1 systems the signals from 24 subscribers is sampled in 125µs.
– The samples are quantized and then converted into a bit stream for transmission over the
channel.

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Bandwidth Requirement
Communication theory tells us that we can transmit errorfree at most two pieces of
information per second per hertz bandwidth (lathi)
Therefore the minimum required bandwidth for transmission of a PCM
speech signal BWmin = 64/2 = 32 kHz
Recall that for analog techniques such as AM, FM etc the bandwidth of the
order of 8 kHz, 16 kHz etc.
We may require more bandwidth but the signal is now digital and we now
have the ability to manipulate, store, regenerate the data
repeaters not amplifiers - no additive noise
TDM not FDM - no intermodulation noise
allow for (superior) digital switching

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• In the T1 digital carrier system 24 voice channels are multiplexed in time.
• A segment containing one codeword (corresponding to one sample) from each of the 24 channels
is called a FRAME.
• Each frame has 24x8 = 192 data bits and takes 125µs.
• Therefore, data rate should be = 24x64kbps = 1.536 Mbps
However, the actual data rate = 1.544Mbps
The extra 8000 bps (1.544-1.536=.008Mbps) result from the overhead bits which are inserted
alongside the data.
• At the receiver it is also necessary to know where a frame starts in order to separate information
bits correctly. For this purpose, a Framing bit is added at the beginning of each frame.
Therefore,
Total number of bits/ frame = 193

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T-1 Line for Multiplexing Telephone Lines

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T-1 Frame Structure

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Digital Hierarchy

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DS and T Line Rates

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 E Line Rates
• European use a version of T lines called E lines

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THANK YOU
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