EC8395 Communication Engineering Department of CSE

UNIT II – PULSE MODULATION

Introduction
In the simplest model of a telephone speech communication there is a direct,
dedicated, physical connection between the two participants in the conversation, and this
link is held for the duration of the conversation. The analogue electrical signal produced
by the telephone at either end is sent on to connection without modification.
In Pulse Amplitude Modulation (PAM), the unmodified electrical signal is not
sent on to the connection. Instead, short samples of the signal are taken at regular
intervals, and these samples are sent on to the connection. The amplitude of each sample
is identical to the signal voltage at the time when the sample was taken. Typically, 8,000
samples are taken per second, so that the interval between samples is 125s, and the
duration of each sample is approximately 4s.
Because each sample is very short (~4s) there is a lot of time between samples
(~121s). Samples from other conversations are put into this “spare time”. Usually, the
samples from 32 separate conversations are put on to a single line. This process is called
Time Division Multiplexing (TDM).
Each sample is very short and will be distorted as it travels across a
communications network. In order to reconstruct the original analogue, signal the only
information the receiver needs to have about a sample is its amplitude, but if this is
distorted then all information about the sample has been lost. To overcome this problem,
the pulse is not transmitted directly, instead its amplitude is measured and converted into
an 8 binary number - a sequence of 1s and 0s. At the receiver end, the receiver merely
needs to detect if a 1 or a 0 has been received so that it can still recover the amplitude of a
PAM pulse even if the 1s and 0s used to describe it have been distorted.
The process of converting the amplitude of each pulse into a stream of 1s and 0s
is called Pulse Code Modulation (PCM)
Note that the process of PAM and PCM (but without the use of TDM) is essentially used
to store music and speech on CDs, but with a higher sample rate, more bits per sample
and complex error correction mechanisms.
Some terms are:
Sampling : The process of measuring the amplitude of a continuous-time signal at
discrete instants. It converts a continuous-time signal to a discrete-time
signal.
Quantizing : Representing the sampled values of the amplitude by a finite set of levels.
It converts a continuous-amplitude sample to a discrete-amplitude sample.
Encoding : Designating each quantized level by a (binary) code.
Sampling and quantizing operations transform an analogue signal to a digital
signal. Use of quantizing and encoding distinguishes PCM from analogue pulse
modulation methods.
The quantizing and encoding operations are usually performed in the same circuit
at the transmitter, which is called an Analogue to Digital Converter (ADC). At the
receiver end the decoding operation converts the (8 bit) binary representation of the pulse
back into an analogue voltage in a Digital to Analogue Converter (DAC)

Low Pass Sampling:

Consider a band-limited signal with no frequency components above a certain
frequency fm. The sampling theorem states that this signal can be recovered completely
from a set of samples of its amplitude, if the samples are taken at the rate of fs > 2fm
samples per second.
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This is often called the uniform sampling theorem for baseband or low-pass
signals.
The minimum sampling rate, 2fm samples per second, is called the Nyquist sampling rate
(or Nyquist frequency); its reciprocal l/(2fm) (measured in seconds) is called the Nyquist
interval.
fs = 2 * fm is called the Nyquist sampling rate.
For telephone speech the standard sampling rate is 8 kHz (or one sample every 125 s).
Sampling Methods
Spectrum of Suppose we have an arbitrary signal (the
Signal(mt)
Unsampled baseband signal m(t)) which has a
Signal M(f) spectrum M(f). Take infinitesimally
Freq short samples of the signal m(t) at a
Sampling pulses uniform rate once every ts seconds i.e. at
fs 2fs 3fs a frequency fs. This is the ideal form of
sampling, it is called instantaneous (or
Spacing = 1/fs Spectrum of Sampling Pulses
impulse) sampling.
Samples In effect the signal m(t) is multiplied by
fs 2fs 3fs a train of impulses giving rise to a train
of pulses as in the lower line of the
Time diagram. The train of sampling impulses
Spectrum of Sampled Signal
Fig 2.1 Sampling has a frequency spectrum consisting of
all harmonics or multiples of fs and all are at the same amplitude.
This sampled signal has a spectrum as shown where M(f) is repeated unattenuated
periodically and appears around all multiples of the sampling frequency (fs = 1/ts).
To recover m(t) from the sampled signal we need only pass the sampled signal through a
low pass filter with a stop frequency of fs/2. All of the higher frequency components will
be dropped. In the diagram, if fs is greater than twice the highest frequency in m(t) the
repetitions of the sampled spectra around the harmonics of the fs do not overlap.
Flat - top Sampling
An Analogue to Digital Converter requires that the sample value be held constant
for a fixed time until the conversion is completed. This requires a flat-top sampled signal.
This has approximately the same repeated frequency spectrum as with the instantaneous
sampling above, but with each repetition slightly spread out.
The simplest and most common sampling method is performed by a functional block
termed a Sample and Hold (S/H) circuit.
The output from the circuit must be held at a
constant level for the sampling duration. V control
- switches the MOSFET ON until the charge on C is
Input signal equal to the amplitude of the sampled voltage. V
+ Sample
Out control then goes LOW, the MOSFET is OFF and
the charge is held by the capacitor. The charge held
on the capacitor puts a voltage across the capacitor,
Vcontrol Gnd and it is held at that value until the next time that V
control switches the MOSFET ON. This is called a
Fig 2.2 Sample and Hold sample and hold circuit and is usually used as the
Aliasing Error input to an ADC.
If a signal is under sampled (sampled at a rate below the
Nyquist rate), the spectrum consists of overlapping
repetitions of the sampled spectrum. Because of the
Spectrum of Sampled Signal overlapping tails a single repetition of the spectrum no
longer has the complete information about the
Fig 2.3 Spectrum of sampled signal

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unsampled signal, and it is no longer possible to recover it from the sampled signal. To
recover the original signal at the receiving end the sampled signal is passed through a
lowpass filter with a cut off of fs/2, we get a spectrum that is not the sampled signal but is
a different version due to:
• Loss of the tail of the sampled signal spectrum beyond fs/2
• This same tail appears inverted, or folded, onto the spectrum at the cut-off frequency.
This tail inversion is known as aliasing, (or spectral folding or foldover distortion).
The aliasing distortion can be eliminated by cutting the tail (i.e. filtering) of the sampled
signal beyond f > fs/2 before the signal is sampled. By so doing, the overlap of
successive cycles in the sampled signal is avoided. The only error in the recovery of the
unsampled signal is that caused by the missing tail above fs/2.

It is simpler to consider aliasing by considering a single frequency component of

m(t). We will look at the frequency fm and it is sampled at a rate fs. The diagrams show
the frequencies which will be present in the sampled signal. There will be frequency
components at fm, fs - fm, fs + fm, 2 fs - fm, 2 fs + fm, 3 fs - fm, 3 fs + fm, etc. etc.
In the first case fm is very much less than fs, so that fs - fm is much higher than the cut off
of the filter (fs/2).
In the second case fm is below, but close to fs/2, so that a sharp cut off filter is
required to ensure that fm is passed but fs - fm is stopped.
In the third case fm is higher than fs/2, so that fs - fm is less than fs/2. The low pass filter
with a cutoff of fs/2 will therefore block fm (the actual signal frequency) but will pass a
signal with frequency fs - fm.
This is aliasing
Strictly speaking, a band limited signal does not exist in reality. It can be shown
that if a signal is time limited it cannot be band limited. All physical signals are
necessarily time limited because they begin at some finite instant and must terminate at
some other finite instant. Hence, all practical signals are theoretically non band limited.
A real signal contains a finite amount of energy, therefore its frequency spectrum
must decay at higher frequencies. Most of the signal energy resides in a finite band, and
the spectrum at higher frequencies contributes little. The error introduced by cutting off
the tail beyond a certain frequency B can be made negligible by making B sufficiently
large.
Thus, for all practical purposes a signal can be considered to be essentially band
limited at some value B, the choice of which depends upon the accuracy desired. A
practical example of this is a speech signal. Theoretically, a speech signal, being a finite
time signal, has an infinite bandwidth. But frequency components beyond 3400 Hz
contribute a small fraction of the total energy. When speech signals are transmitted by
PCM they are first passed through a lowpass filter of bandwidth of 3500 Hz. (This filter
is called an anti aliasing filter). Higher sampling rates (i.e. 8000 samples/sec) permits
recovery of the signal from its samples using relatively simple filters i.e. it allows for
guard bands between the repetitions of the spectrum (otherwise recovering signals
sampled at the Nyquist rate would require very sharp cut-off (ideal) filters).
In summary, aliasing distortion produces frequency components in the desired
frequency band that did not exist in the original waveform. Aliasing problems are not
confined to speech digitisation processes. The potential for aliasing is present in any
sample data system.
Motion picture taking, for example, is another sampling system that can produce
aliasing. A common example occurs when filming a rotating wheel. Often the sampling
process (the picture refresh rate) is too slow to keep up with the wheel movements and
spurious rotational rates are produced. If the wheel rotates 3550 between frames, it looks
to the eye as if it has moved backwards 50.

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Sampling:

A message signal may originate from a digital or analog source. If the message signal is
analog in nature, then it has to be converted into digital form before it can transmit by
digital means. The process by which the continuous-time signal is converted into a
discrete – time signal is called Sampling. Sampling operation is performed in accordance
with the sampling theorem.

Sampling Theorem For Low-Pass Signals:-

Statement:- “If a band –limited signal g(t) contains no frequency components for ‫׀‬f‫ > ׀‬W,
then it is completely described by instantaneous values g(kT ) uniformly spaced in time
with period T ≤ 1/2W. If the sampling rate, fs is equal to the Nyquist rate or greater (fs ≥
2W), the signal g(t) can be exactly reconstructed.
g(t)

δ (t)
.
-2Ts -Ts 0 1Ts 2Ts 3Ts 4Ts

gδ(t)

-2Ts -Ts 0 Ts 2Ts 3Ts 4Ts

Fig 2.4 Sampling Process

Proof:-

Part - I If a signal x(t) does not contain any frequency component beyond W Hz, then the
signal is completely described by its instantaneous uniform samples with sampling
interval (or period) of Ts < 1/(2W) sec.

Part – II The signal x(t) can be accurately reconstructed (recovered) from the set of
uniform instantaneous samples by passing the samples sequentially through an ideal
(brick-wall) lowpass filter with bandwidth B, where W ≤ B < fs – W and fs =
1/(Ts).{x(nTs)}≡ xs(t) = Σ x(t).δ(t- nTs)
where x(nTs) = x(t)⎢t =nTs , δ(t) is a unit pulse singularity function and „n‟
is an integer. The continuous-time signal x(t) is multiplied by an (ideal)
impulse train to obtain {x(nTs)} and can be rewritten as,
xs(t) = x(t).Σ δ(t- nTs) -------------1.2
.
Now, let X(f) denote the Fourier Transform F(T) of x(t), i.e.

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-------------1.3
Now, from the theory of Fourier Transform, we know that the F.T of
Σ δ(t- nTs), the
impulse train in time domain, is an impulse train in frequency domain:

F{Σ δ(t- nTs)} = (1/Ts).Σ δ(f- n/Ts) = fs.Σ δ(f- nfs) -----1.4

If Xs(f) denotes the Fourier transform of the energy signal xs(t), we can use
convolution property:

Xs(f) = X(f)* F{Σ δ(t- nTs)}

= X(f)*[fs.Σ δ(f- nfs)]
= fs.X(f)*Σ δ(f- nfs)

--------------1.5
This equation, when interpreted appropriately, gives an intuitive proof to
Nyquist‟s theorems as stated above and also helps to appreciate their practical
implications. Let us note that while writing Eq.(1.5), we assumed that x(t) is an energy
signal so that its Fourier transform exists.

With this setting, if we assume that x(t) has no appreciable frequency component greater
than W Hz and if fs > 2W, then Eq.(1.5) implies that Xs(f), the Fourier Transform of the
sampled signal xs(t) consists of infinite number of replicas of X(f), centered at discrete
frequencies n.fs, -∞ < n < ∞ and scaled by a constant fs= 1/Ts

Fig 2.5 Indicates that the bandwidth of this instantaneously sampled wave xs(t) is infinite
while the spectrum of x(t) appears in a periodic manner, centered at discrete frequency
values n.fs.

Part – I of the sampling theorem is about the condition fs > 2.W i.e. (fs – W) > W
and (– fs + W) < – W. As seen from Fig. 1.2.1, when this condition is satisfied, the
spectra . of xs(t), centered at f = 0 and f = ± fs do not overlap and hence, the spectrum of

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x(t) is present in xs(t) without any distortion. This implies that xs(t), the appropriately
sampled version of x(t), contains all information about x(t) and thus represents x(t).

The second part suggests a method of recovering x(t) from its sampled version
xs(t) by using an ideal lowpass filter. As indicated by dotted lines in Fig. 1.2.1, an ideal
lowpass filter (with brick-wall type response) with a bandwidth W ≤ B < (fs – W), when
fed with xs(t), will allow the portion of Xs(f), centered at f = 0 and will reject all its
replicas at f = n fs, for n ≠ 0.
This implies that the shape of the continuous time signal xs(t), will be retained at
the output of the ideal filter. If the sampling rate, fs ≥ 2fu, exact reconstruction is possible
in which case the signal g(t) may be considered as a low pass signal itself.

Sampling of Band Pass Signals:

Consider a band-pass signal g(t) with the spectrum shown in figure:

BB B
Band width = B
Upper Limit = fu
Lower Limit = fl -fu -fl 0 fl fu f

Fig 2.6 Spectrum of sampled signal

Example-1 :
Consider a signal g(t) having the Upper Cutoff frequency, fu = 100KHz and the
Lower Cutoff frequency fl = 80KHz.
The ratio of upper cutoff frequency to bandwidth of the signal g(t) is
fu / B = 100K / 20K = 5.
Therefore we can choose m = 5.
Then the sampling rate is fs = 2fu / m = 200K / 5 = 40KHz

Example-2 :
Consider a signal g(t) having the Upper Cutoff frequency, fu = 120KHz and the
Lower Cutoff frequency fl = 70KHz.

The ratio of upper cutoff frequency to bandwidth of the signal g(t) is

fu / B = 120K / 50K = 2.4
Therefore we can choose m = 2. ie.. m is an integer less than (fu /B).
Then the sampling rate is fs = 2fu / m = 240K / 2 = 120KHz.

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Quantization
This is the process of setting the sample amplitude, which can be continuously variable
to a discrete value. Look at Uniform Quantization first, where the discrete values are evenly
spaced.
Uniform Quantization
We assume that the amplitude of the signal m(t) is confined to
the range (-mp, +mp ). This range (2mp) is divided into L
Output

levels, each of step size , given by

-m +m
 = 2 mp / L
p p

 Input A sample amplitude value is approximated by the midpoint of

the interval in which it lies. The input/output characteristics of
a uniform quantizer is shown.

Fig 2.7 Qualtization

Types of Quantizers:

1. Uniform Quantizer
2. Non- Uniform Quantizer

Uniform Quantizer:

In Uniform type, the quantization levels are uniformly spaced, whereas in non-
uniform type the spacing between the levels will be unequal and mostly the relation is
logarithmic.

Types of Uniform Quantizers: ( based on I/P - O/P Characteristics)

1. Mid-Rise type Quantizer
2. Mid-Tread type Quantizer

In the stair case like graph, the origin lies the middle of the tread portion in Mid –
Tread
type where as the origin lies in the middle of the rise portion in the Mid-Rise type.

Mid – tread type: Quantization levels – odd number.

Mid – Rise type: Quantization levels – even number.

Quantization Noise and Signal-to-Noise:

“The Quantization process introduces an error defined as the difference between the input
signal, x(t) and the output signal, yt). This error is called the Quantization Noise.”

q(t) = x(t) – y(t)

Quantization noise is produced in the transmitter end of a PCM system by rounding off
sample values of an analog base-band signal to the nearest permissible representation levels of
the quantizer. As such quantization noise differs from channel noise in that it is signal

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dependent.

Let „Δ‟ be the step size of a quantizer and L be the total number of quantization levels.
Quantization levels are 0, ± ., ± 2 ., ±3 .......
.
The Quantization error, Q is a random variable and will have its co-sample values
bounded
I
by [-(Δ/2) < q < (Δ/2)]. f is small, the quantization error can be assumed to a
uniformly distributed random variable.

Consider a memory less quantizer that is both uniform and symmetric.

L = Number of quantization
levels X = Quantizer input
Y = Quantizer output

The output y is given by

Y=Q(x)

which is a staircase function that befits the type of mid tread or mid riser quantizer of
interest.

Non – Uniform Quantizer:

In Non – Uniform Quantizer the step size varies. The use of a non – uniform quantizer is
equivalent to passing the baseband signal through a compressor and then applying the
www
compressed signal to a uniform quantizer. The resultant signal is then transmitted.

UNIFORM
COMPRESSOR QUANTIZER EXPANDER

Fig: 2.8 Model of Non-Uniform Quantizer

At the receiver, a device with a characteristic complementary to the compressor called

Expander is used to restore the signal samples to their correct relative level.

The Compressor and expander take together constitute a Compander.

Compander = Compressor + Expander

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Advantages of Non – Uniform Quantization :

1. Higher average signal to quantization noise power ratio than the uniform quantizer when
the signal pdf is non uniform which is the case in many practical situation.
2. RMS value of the quantizer noise power of a non – uniform quantizer is
substantially proportional to the sampled value and hence the effect of the quantizer noise is
reduced.

Companding
In a uniform or linear PCM system the size of every quantization interval is determined by the
SQR requirement of the lowest signal to be encoded. This interval is also for the largest signal -
which therefore has a much better SQR.
Example: A 26 dB SQR for small signals and a 30 dB dynamic range produces a
56 dB SQR for the maximum amplitude signal.
In this way a uniform PCM system provides unneeded quality for large signals. In speech the
max amplitude signals are the least likely to occur. The code space in a uniform PCM system is
very inefficiently utilised.
A more efficient coding is achieved if the quantization intervals increase with the
sample value. When the quantization interval is directly proportional to the sample value (
assign small quantization intervals to small signals and large intervals to large signals) the SQR
is constant for all signal levels.
With this technique fewer bits per sample are required to provide a specified SQR for
small signals and an adequate dynamic range for large signals (but still with the SQR as for the
small signals). The quantization intervals are not constant and there will be a non linear
relationship between the code words and the values they represent.
Originally to produce the non linear quantization the baseband signal was passed through a
non-linear amplifier with input/output characteristics as shown before the samples were taken.
Low level signals were amplified and high level signals were attenuated. The larger the sample
value the more it is compressed before encoding. The PCM decoder expands the compressed
value using an inverse compression characteristic to recover the original sample value. The two
processes are called companding.
compressed signal level

input signal level

Fig 2.9 Companding

There are 2 companding schemes to describe the curve above:
1. -Law Companding (also called log-PCM)
This is used in North America and Japan. It uses a logarithmic compression curve which
is ideal in the sense that quantization intervals and hence quantization noise is directly
proportional to signal level (and so a constant SQR).

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2. A- Law Companding
This is the ITU-T standard. It is used in Europe and most of the rest of the world. It is
very similar to the -Law coding. It is represented by straight line segments to facilitate
digital companding.
Originally the non linear function was obtained using non linear devices such as special diodes.
These days in a PCM system the A to D and D to A converters (ADC and DAC) include a
companding function.

Pulse Amplitude Modulation:

• In fact the pulses in a PAM signal may of Flat-top type or natural type or ideal type.
• The Flat-top PAM is most popular and is widely used. The reason for using Flat-top
PAM is that during the transmission, the noise interferes with the top of the transmitted
pulses and this noise can be easily removed if the PAM pulse as Flat-top.
• In natural samples PAM signal, the pulse has varying top in accordance with the signal
variation. Such type of pulse is received at the receiver, it is always contaminated by
noise. Then it becomes quite difficult to determine the shape of the top of the pulse and
thus amplitude detection of the pulse is not exact.

Generation of PAM:
There are two operations involved in the generation of PAM signal
1. Instantaneous sampling of the message signal m(t) every Ts seconds, where the sampling
rate fs = 1/Ts is chosen in accordance with the sampling theorem.
2. Lengthening the duration of each sample so obtained to some constant value T.

Fig 2.10 PAM Signal

Sample and Hold Circuit for Generating Flat-top sampled PAM

Fig 2.11 Sample and hold circuit for PAM Signal generation
• The sample and hold circuit consists of two Field Effect Transistor switches and a

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capacitor.
• The sampling switch is closed for a short duration by a short pulse applied to the gate
G1 of the transistor. During this period, the capacitor C is quickly charged up to a
voltage equal to the instantaneous sample value of the incoming signal.
• Now, the sampling switch is opened and the capacitor holds the charge. The discharge
switch is then closed by a pulse applied to gate G2 of the other transistor. Due to this,
the capacitor is discharged to zero volts. The discharges switch is then opened and thus
capacitor has no voltage. Hence the output of the sample and hold circuit consists of a
sequence of flat-top samples as shown in figure.

Mathematical Representation of PAM

We may express the PAM signal as

where Ts = sampling period

m(nTs) = sample value of m(t) obtained at t = nTs
h(t) = standard rectangular pulse of unit amplitude and duration T and it is defined as The
spectrum of flat-top PAM signal is

The spectrum of flat-top PAM signal is

Transmission Bandwidth of PAM:

In PAM signal the pulse duration τ is assumed to be very small compared to time period Ts
between the two samples i.e τ< Ts
• If the maximum frequency in the modulating signal x(t) is fm then sampling frequency
fs is given by fs>=2fm Or 1/Ts >= 2fm or Ts <= 1/2fm Therefore, τ < < Ts <= 1/2fm
• If ON and OFF time of PAM pulse is equal then maximum frequency of PAM pulse
will be fmax = 1/ τ+ τ = 1/2 τ
Therefore,
transmission bandwidth >=fmax
But fmax=1/2 τ

B.W>=1/2 τ

B.W>=1/2 τ >>fm

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Demodulation of PAM:

Fig 2.12 PAM Reconstruction

• The distortion caused using PAM to transmit an analog information bearing signal is
referred to as the aperture effect. This distortion may be corrected by connecting an
equalizer in cascade with the low-pass reconstruction filter as shown in fig.
• The equalizer has the effect of decreasing the in-band loss of the reconstruction filter as
the frequency increases in such a manner as to compensate for the aperture effect.

Ideally, the magnitude response of the equalizer is given by

The amount of equalization needed in practice is usually small.

Advantages of PAM :

• It is the simple and simple process for modulation and demodulation

• Transmitter and receiver circuits are simple and easy to construct.

CLASSIFICATION OF LINECODES
Line coding:

Line coding, refers to the process of representing the bit stream (1‟s and 0‟s) in the
form of voltage or current variations optimally tuned for the specific properties of the physical
channel being used.
The selection of a proper line code can help in so many ways: One possibility is to aid in
clock recovery at the receiver. A clock signal is recovered by observing transitions in the
received bit sequence, and if enough transitions exist, a good recovery of the clock is
guaranteed, and the signal is said to be self-clocking.
Another advantage is to get rid of DC shifts. The DC component in a line code is called the
bias or the DC coefficient. Unfortunately, most long-distance communication channels cannot
transport a DC component. This is why most line codes try to eliminate the DC component
before being transmitted on the channel. Such codes are called DC balanced, zero-DC, zero-
bias, or DC equalized. Some common types of line encoding in common-use nowadays are
unipolar, polar, bipolar, Manchester, MLT-3 and Duobinary encoding. These codes are
explained here:

Unipolar (Unipolar NRZ and Unipolar RZ):

Unipolar is the simplest line coding scheme possible. It has the advantage of being compatible
with TTL logic. Unipolar coding uses a positive rectangular pulse p(t) to represent binary 1,
and the absence of a pulse (i.e., zero voltage) to represent a binary 0. Two possibilities for the
pulse p(t) exist3: Non-Return-to-Zero (NRZ) rectangular pulse and Return-to-Zero (RZ)

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rectangular pulse. The difference between Unipolar NRZ and Unipolar RZ codes is that the
rectangular pulse in NRZ stays at a positive value (e.g., +5V) for the full duration of the logic 1
bit, while the pule in RZ drops from +5V to 0V in the middle of the bit time.
A drawback of unipolar (RZ and NRZ) is that its average value is not zero, which means it
creates a significant DC-component at the receiver (see the impulse at zero frequency in the
corresponding power spectral density (PSD) of this line code.

Fig 2.13 Unipolar Waveform

The disadvantage of unipolar RZ pared to unipolar NRZ is that each rectangular pulse in
RZ is only half the length of NRZ pulse. This means that unipolar RZ requires twice the
bandwidth of the NRZ code.

Polar (Polar NRZ and Polar RZ):

In Polar NRZ line coding binary 1‟s are represented by a pulse p(t) and binary 0‟s are
represented by the negative of this pulse -p(t) (e.g., -5V). Polar (NRZ and RZ) signals .Using
the assumption that in a regular bit stream a logic 0 is just as likely as a logic 1,polar signals
(whether RZ or NRZ) have the advantage that the resulting Dc component is very close to zero.

Fig 2.14 Polar waveform

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The RMS value of polar signals is bigger than unipolar signals, which means that polar
signals have more power than unipolar signals, and hence have better SNR at the receiver.
Actually, polar NRZ signals have more power pared to polar RZ signals. The drawback of
polar NRZ, however, is that it lacks clock information especially when a long sequence of
0‟s or 1‟s is transmitted.

Non-Return -to-Zero, Inverted (NRZI):

NRZI is a variant of Polar NRZ. In NRZI there are two possible pulses, p(t) and –p(t). A
transition from one pulse to the other happens if the bit being transmitted is a logic 1, and no
transition happens if the bit being transmitted is a logic 0.

Fig 2.15 NRZI waveform

This is the code used on pact discs (CD), USB ports, and on fiber-based Fast Ethernet at 100-
Mbit/s .
Manchester encoding:

In Manchester code each bit of data is signified by at least one transition. Manchester
encoding is therefore considered to be self-clocking, which means that accurate clock recovery
from a data stream is possible.

In addition, the DC component of the encoded signal is zero. Although transitions allow the
signal to be self-clocking, it carries significant overhead as there is a need for essentially twice
the bandwidth of a simple NRZ or NRZI encoding
.

Fig 2.16 Manchester coding

POWER SPECTRA OF LINE CODES:

Fig 2.17 Power spectra of Line codes

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• Unipolar most of signal power is centered on origin and there is waste of power due to
DC component that is present.
• Polar format most of signal power is centered on origin and they are simple to
implement.
• Bipolar format does not have DC component and does not demand more bandwidth, but
power requirement is double than other formats.
• Manchester format does not have DC component but provides proper clocking.
Pulse Code Modulation

Pulse Code Modulation (PCM) is an extension of PAM wherein each analogue sample
value is quantized into a discrete value for representation as a digital code word.
Thus, as shown below, a PAM system can be converted into a PCM system by adding a
suitable analogue-to-digital (A/D) converter at the source and a digital-to-analogue (D/A)
converter at the destination.
Modulator PCM is a true digital process
Analogue PCM as compared to PAM. In
Input Parallel Digital Output PCM the speech signal is
A to D Binary
Sampler to Serial Pulse
Converter Coder converted from analogue to
Converter Generator
digital form.
Demodulator
PCM is standardised for
PCM Analogue telephony by the ITU-T
Serial to
Input D to A Output
Parallel LPF (International
Converter
Converter
Telecommunications Union -
Fig 2.18 Pulse code Modulation Telecoms, a branch of the
UN), in a series of recommendations called the G series. For example the ITU-T
recommendations for out-of-band signal rejection in PCM voice coders require that 14 dB of
attenuation is provided at 4 kHz. Also, the ITU-T transmission quality specification for
telephony terminals require that the frequency response of the handset microphone has a sharp
roll-off from 3.4 kHz.
In quantization the levels are assigned a binary codeword. All sample values falling between
two quantization levels are considered to be located at the centre of the quantization interval. In
this manner the quantization process introduces a certain amount of error or distortion into the
signal samples. This error known as quantization noise, is minimised by establishing a large
number of small quantization intervals. Of course, as the number of quantization intervals
increase, so must the number or bits increase to uniquely identify the quantization intervals.
For example, if an analogue voltage level is to be converted to a digital system with 8 discrete
levels or quantization steps three bits are required. In the ITU-T version there are 256
quantization steps, 128 positive and 128 negative, requiring 8 bits. A positive level is
represented by having bit 8 (MSB) at 0, and for a negative level the MSB is 1.

Differential pulse coding schemes

PCM transmits the absolute value of the signal for each frame. Instead we can transmit
information about the difference between each sample. The two main differential coding
schemes are:
• Delta Modulation
• Differential PCM and Adaptive Differential PCM (ADPCM)

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Delta Modulation
Delta modulation converts an analogue signal, normally voice, into a digital signal.
The analogue signal is sampled as in the PCM
granular noise
1 1 process. Then the sample is compared with the
1 previous sample. The result of the comparison
1 0 0 1
1 1 0
is quantified using a one bit coder. If the
0 0 1 sample is greater than the previous sample a 1
0 0 0
0
is generated. Otherwise a 0 is generated. The
advantage of delta modulation over PCM is its
Fig 2.19 Delta Modulation Waveform simplicity and lower cost. But the noise
performance is not as good as PCM.
To reconstruct the original from the quantization, if a 1 is received the signal is increased by a
step of size q, if a 0 is received the output is reduced by the same size step. Slope overload
occurs when the encoded waveform is more than a step size away from the input signal. This
condition happens when the rate of change of the input exceeds the maximum change that can
be generated by the output. Overload will occur if:
dx(t)/dt  q /T = q * fs
where: x(t) = input signal, q = step size, T = period between samples, fs = sampling frequency
Assume that the input signal has maximum amplitude A and maximum frequency F. The most
rapidly changing input is provided by x(t) = A * sin (2 *  * F * t).
For this dx(t)/dt = 2 *  * F * A * sin (2 *  * F * t).
This slope has a maximum value of 2 *  * F * A
Overload occurs if 2 *  * F * A > q * fs
To prevent overload we require q * fs > 2 *  * F * A
Example A = 2 V, F = 3.4 kHz, and the signal is sampled 1,000,000 times per second,
requires q > 2 * 3.14 * 3,400 * 2 /1,000,000 V > 42.7 mV
Granular noise occurs if the slope changes more slowly than the step size. The reconstructed
signal oscillates by 1 step size in every sample. It can be reduced by decreasing the step size.
This requires that the sample rate be increased. Delta Modulation requires a sampling rate
much higher than twice the bandwidth. It requires oversampling in order to obtain an accurate
prediction of the next input, since each encoded sample contains a relatively small amount of
information. Delta Modulation requires higher sampling rates than PCM.
Differential PCM (DPCM) and ADPCM
DPCM is also designed to take advantage
Differentiator Encoded
Analogue of the redundancies in a typical speech
Difference
Input Band Limiting Quantiser
Filter + Enoder
Sampleswaveform. In DPCM the differences
ADC
- between samples are quantized with fewer
bits that would be used for quantizing an
Accumulator
DAC
individual amplitude sample. The sampling
rate is often the same as for a comparable
Fig 2.20 DPCM & ADPCM PCM system, unlike Delta Modulation.
Adaptive Differential Pulse Code Modulation ADPCM is standardised by ITU-T
recommendations G.721 and G.726. The method uses 32,000 bits/s per voice channel, as
compared to standard PCM’s 64,000 bits/s. Four bits are used to describe each sample, which
represents the difference between two adjacent samples. Sampling is 8,000 times a second. It
makes it possible to reduce the bit flow by half while maintaining an acceptable quality. While
the use of ADPCM (rather than PCM) is imperceptible to humans, it can significantly reduce
the throughput of high-speed modems and fax transmissions.

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The principle of ADPCM is to use our knowledge of the signal in the past time to predict the
signal one sample period later, in the future. The predicted signal is then compared with the
actual signal. The difference between these is the signal which is sent to line - it is the error in
the prediction. However this is not done by making comparisons on the incoming audio signal -
the comparisons are done after PCM coding.
To implement ADPCM the original (audio) signal is sampled as for PCM to produce a code
word. This code word is manipulated to produce the predicted code word for the next sample.
The new predicted code word is compared with the code word of the second sample. The result
of this comparison is sent to line. Therefore we need to perform PCM before ADPCM.
The ADPCM word represents the prediction error of the signal, and has no significance itself.
Instead the decoder must be able to predict the voltage of the recovered signal from the
previous samples received, and then determine the actual value of the recovered signal from
this prediction and the error signal, and then to reconstruct the original waveform.
ADPCM is sometimes used by telecom operators to fit two speech channels onto a single
64 kbit/s link. This was very common for transatlantic phone calls via satellite up until a few
years ago. Now, nearly all calls use fibre optic channels at 64 kbit/s.

Adaptive Delta Modulation:

The performance of a delta modulator can be improved significantly by making the step size of
the modulator assume a time-varying form. In particular, during a steep segment of the input
signal the step size is increased. Conversely, when the input signal is varying slowly, the step
size is reduced. In this way, the size is adapted to the level of the input signal. The resulting
method is called adaptive delta modulation (ADM). There are several types of ADM,
depending on the type of scheme used for adjusting the step size. In this ADM, a discrete set
of values is provided for the step size.

Fig 2.21 Block Diagram of ADM Transmitter

Fig 2.22 Block Diagram of ADM Receiver

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Prediction filtering & Linear Predictive Coding.

The speech signal is filtered to no more than one half the system sampling frequency
and then A/D conversion is performed. The speech is processed on a frame by frame basis
where the analysis frame length can be variable. For each frame a pitch period estimation is
made along with a voicing decision. A linear predictive coefficient analysis is performed to
obtain an inverse model of the speech spectrum A (z). In addition a gain parameter G,
representing some function of the speech energy is computed. An encoding procedure is then
applied for transforming the analyzed parameters into an efficient set of transmission
parameters with the goal of minimizing the degradation in the synthesized speech for a
specified number of bits. Knowing the transmission frame rate and the number of bits used for
each transmission parameters, one can compute a noise-free channel transmission bit rate.
At the receiver, the transmitted parameters are decoded into quantized versions of the
coeifficent analysis and pitch estimation parameters. An excitation signal for synthesis is then
constructed from the transmitted pitch and voicing parameters. The excitation signal then
drives a synthesis filter 1/A (z) corresponding to the analysis model A (z). The digital samples
s^(n) are then passed through an D/A converter and low pass filtered to generate the synthetic
speech s(t). Either before or after synthesis, the gain is used to match the synthetic speech
energy to the actual speech energy. The digital samples are the converted to an analog signal
and passed through a filter similar to the one at the input of the system.

Linear predictive coding (LPC) of speech

The linear predictive coding (LPC) method for speech analysis and synthesis is based
on modeling the Vocal tract as a linear All-Pole (IIR) filter having the system transfer function:

Fig 2.23 Simple speech production

Where p is the number of poles, G is the filter Gain, and a[k] are the parameters that
determine the poles. There are two mutually exclusive ways excitation functions to model

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voiced and unvoiced speech sounds. For a short time-basis analysis, voiced speech is
considered periodic with a fundamental frequency of Fo, and a pitch period of 1/Fo, which
depends on the speaker. Hence, Voiced speech is generated by exciting the all pole filter model
by a periodic impulse train. On the other hand, unvoiced sounds are generated by exciting the
all-pole filter by the output of a random noise generator. The fundamental difference between
these two types of speech sounds comes from the way they are produced. The vibrations of the
vocal cords produce voiced sounds. The rate at which the vocal cords vibrate dictates the pitch
of the sound. On the other hand, unvoiced sounds do not rely on the vibration of the vocal
cords. The unvoiced sounds are created by the constriction of the vocal tract. The vocal cords
remain open and the constrictions of the vocal tract force air out to produce the unvoiced
sounds

Given a short segment of a speech signal, lets say about 20 ms or 160 samples at a
sampling rate 8 KHz, the speech encoder at the transmitter must determine the proper
excitation function, the pitch period for voiced speech, the gain, and the coefficients ap[k]. The
block diagram below describes the encoder/decoder for the Linear Predictive Coding. The
parameters of the model are determined adaptively from the data and modeled into a binary
sequence and transmitted to the receiver. At the receiver point, the speech signal is the
synthesized from the model and excitation signal.

The parameters of the all-pole filter model are determined from the speech samples by
means of linear prediction. To be specific the output of the Linear Prediction filter is
^ p
s ( n) = − a p ( k ) s ( n − k )
k =1

^
and the corresponding error between the observed sample S(n) and the predicted value s (n) is

^
e(n) = s(n) − s(n)

by minimizing the sum of the squared error we can determine the pole parameters a p (k ) of
the model. The result of differentiating the sum above with respect to each of the parameters
and equation the result to zero, is a sep of p linear equations

a
k =1
p (k )rss (m − k ) = −rss ( m ) where m=1,2,….p

where rss (m) represent the autocorrelation of the sequence s (n ) defined as

N
rss ( m ) =  s(n) s(n + m)
n =0

the equation above can be expressed in matrix form as

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Rss a = −rss(m)

where Rss a is a pxp autocorrelation matrix, rss is a px1 autocorrelation vector, and a is a px1
vector of model parameters.

The gain parameter of the filter can be obtained by the input-output relationship as follow

p
s(n) = − a p (k ) s(n − k ) + Gx(n)
k =1

where X(n) represent the input sequence.

We can further manipulate this equation and in terms of the error sequence we have

p
Gx(n) = s(n) +  a p (k ) s(n − k ) = e(n)
k =1

then

N −1 N −1
G 2
 x ( n) =  e ( n)
n =0
2

n =0
2

if the input excitation is normalized to unit energy by design, then

N −1 N −1 p
G 2
 x ( n) =  e ( n) = r
n =0
2

n =0
2
ss (0) +  a p (k )rss (k )
k =1

where G^2 is set equal to the residual energy resulting from the least square optimization .

once the LPC coefficients are computed, we can determine weather the input speech
frame is voiced, and if it is indeed voiced sound, then what is the pitch. We can determine the
p

pitch by computing the following sequence in matlab. re (n) =  r (k )r

k =1
a ss (n − k )

where ra (k ) is defined as follow

p
ra (n) =  aa (k )a p (i + k )
k =1

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which is defined as the autocorrelation sequence of the prediction coefficients. The pitch id
r ( n)
detected by finding the peak of the normalized sequence e In the time interval
re (0)
corresponds to 3 to 15 ms in the 20ms sampling frame. If the value of this peak is at least 0.25,

the frame of speech is considered voiced with a pitch period equal to the value of n = Np ,
re ( N p )
where is a maximum value.
re (0)

If the peak value is less than 0.25, the frame speech is considered unvoiced and the
pitch would equal to zero.

The value of the LPC coefficients, the pitch period, and the type of excitation are then
transmitted to the receiver. The decoder synthesizes the speech signal by passing the proper
excitation through the all-pole filter model of the vocal tract.

Typically the pitch period requires 6 bits, the gain parameters are represented in 5 bits after the
dynamic range is compressed logarithmically, and the prediction coefficients require 8-10 bits
normally for accuracy reasons. This is very important in LPC because any small changes in the
prediction coefficients result in large change in the pole positions of the filter model, which
cause instability in the model. This is overcome by using the PARACOR method.

Is speech frame Voiced or Unvoiced ?

Once the LPC coefficients are competed, we can determine weather the input speech frame is
voiced, and if so, what the pitch is.

If the speech frame is decided to be voiced, an impulse train is employed to represent it, with
nonzero taps occurring every pitch period. A pitch-detecting algorithm is used in order to
determine to correct pitch period / frequency. The autocorrelation function is used to estimate
the pitch period as. However, if the frame is unvoiced, then white noise is used to represent it
and a pitch period of T=0 is transmitted. Therefore, either white noise or impulse train becomes
the excitation of the LPC synthesis filter

Two types of LPC vocoders were implemented in MATLAB

Plain LPC Vocoder diagram is shown below :

Fig 2.24 LPC Vocoder

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Time Division Multiplexing (TDM) - Principle

When sending samples of a signal instead of the signal itself there is time available
between each of the samples. Samples from other analogue signals can be put into this space.
Receiver
The process of splitting up
Transmitter
Timing Timing the time into slots and
putting different signals into
Ch1 Ch1 the time slots is known as
i/p Buffer LPF1 o/p
Ch1 Transmission Line Ch1
Time Division Multiplexing
i/p Buffer LPF2 o/p (TDM). A basic real TDM
SW1 SW2
Ch1 Ch1
i/p Buffer LPF3 o/p system interleaves 32 signals
and uses electronic switches.
Fig 2.25 Time Division Multiplexing This is a diagram of a 3
channel PAM-TDM system.
This diagram shows the waveforms produced during the operation of the PAM-TDM system
The switches connect the transmitter and the receiver to each of the channels in turn for a
Ch1 specific interval of time. In effect each channel is sampled
and the sample is transmitted
When the switches are in the channel 1 position, channel 1
Ch2
forms a PAM channel with an LPF for reconstruction, and
so on for channels 2 and 3. The result is that the
Ch3
amplitudes samples from each channel share the line
sequentially, becoming interleaved to form a complex
PAM wave, as shown above.
TDM A major problems in any TDM system is the
Signal
synchronisation of the transmitter and receiver timing
Guard Slots
circuits. The transmitter and receiver must switch at the
Time
Slots 1 2 3 1 2 3 1 2 3 same time and frequency. Also SW1 must be in the
channel 1 position when SW2 is in the channel 1 position,
Fig 2.26 TDM Waveform so that the switches must be synchronised in position also.
In a system that uses analogue modulation (PAM) the time slots are separated by guard slots to
prevent crosstalk between channels.

Frequency Division Multiplexing

• Frequency Division Multiplexing (FDM) Frequency-division multiplexing is a form of
signal multiplexing which involves assigning non-overlapping frequency ranges to
different signals or to each "user of a medium.
• FDM achieves the combining of several signals into one medium by sending signals in
several distinct frequency ranges over a single medium.
• Frequency division multiplexing involves translation of the speech signal from the
frequency band 300-3400 Hz to a higher frequency band. Each channel is translated to a
different hand and then all the channels are combined to form a frequency division multiplexed
signal.
In FDM, the speech channels are stacked at intervals of 4 kHz to provide a guard band between
adjacent channels.

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FDM can be applied when the bandwidth of a link (in hertz) is greater than the combined
bandwidths of the signals to be transmitted.
A demultiplexer applies a set of filters that each extract a small range of frequencies near one
of the carrier frequencies.

Fig 2.27 FDM

Fig 2.28 FDM Process

Advantage of FDM:
1. The senders can send signals continuously.
2. FDM support full duplex information flow
3. Works for analog signals too
4. Noise problem for analog communication has lesser effect
5. AM and FM radio broadcasting and Television broadcasting

Disadvantage of FDM:
1. Separate frequency for each possible communication
2. Inflexible, one channel idle and other one busy
3. The initial cost is high
4. A problem for one user can sometimes affect others
5. Each user requires a precise carrier frequency.

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