Module 4: Digital representation of analog signals

Why Digitize Analog sources?

Advantages of the transmission of digital information over analog information are

• Digital systems are less sensitive to noise than analog. For long transmission lengths,
the signal may be regenerated effectively error-free at different points along the path,
and the original signal transmitted over the remaining length.
• With digital systems, it is easier to integrate different services, for example, video and
the accompanying soundtrack, into the same transmission scheme.
• The transmission scheme can be relatively independent of the source. For example, a
digital transmission scheme that transmits voice at 10 kbps could also be used to
transmit computer data at 10 kbps.
• Circuitry for handling digital signals is easier to repeat and digital circuits are less
sensitive to physical effects such as vibration and temperature.
• Digital signals are simpler to characterize and typically do not have the same amplitude
range and variability as analog signals. This makes the associated hardware easier to
design.
Digital techniques offer strategies for more efficient use of all media (e.g., cable, radio waves,
optical fiber)

• Various media sharing strategies, known as multiplexing techniques, are more easily
implemented with digital transmission strategies.
• There are techniques for removing redundancy from a digital transmission, so as to
minimize the amount of information that has to be transmitted. These techniques fall
under the broad classification of source coding.
• There are techniques for adding controlled redundancy to a digital transmission, such
that errors that occur during transmission may be corrected at the receiver without any
additional information. These techniques fall under the general category of channel
coding.
•
• Digital techniques make it easier to specify complex standards that may be shared on a
worldwide basis. This allows the development of communication components with
many different features (e.g., a cellular handset) and their interoperation with a different
component (e.g., a base station) produced by a different manufacturer.
• Other channel compensations techniques, such as equalization, especially adaptive
versions, are easier to implement with digital transmission techniques.

PRODUCT PROPERTY OF IMPULSE FUNCTION

f(t)δ(t−t0) = f(t0) δ(t−t0)

SIFTING PROPERTY OF IMPULSE FUNCTION
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To combat the effects of aliasing in practice, we may use two corrective measures, as described
here:

1 Prior to sampling, a low-pass pre-alias filter is used to attenuate those high-frequency

components of the signal that are not essential to the information being conveyed by the signal.

2. The filtered signal is sampled at a rate slightly higher than the Nyquist rate.

The use of a sampling rate higher than the Nyquist rate also has the beneficial effect of easing
the design of the reconstruction filter used to recover the original signal from its sampled
version.

Consider the example of a message signal that has been pre-alias (low pass) filtered, resulting
in the spectrum shown in Figure a.

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Figure7.4 b, assuming a sampling rate higher than the Nyquist rate.
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specified as follows (see Figure c):
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determined by the pre-alias filter.

• The filter has a transition band extending (for positive frequencies) from W to fs-W, where
fs, is the sampling rate. The fact that the reconstruction filter has a well-defined transition band
means that it is physically realizable.
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Pulse Amplitude Modulation
In pulse-amplitude modulation (PAM), the amplitudes of regularly spaced pulses are varied in
proportion to the corresponding sample values of a continuous message signal. The pulses can
be of a rectangular form or some other appropriate shape.

It is similar to natural sampling. In natural sampling message signal is multiplied by a periodic

train of rectangular pulses. In natural sampling the top of each modulated rectangular pulse
varies with respect to amplitude of message signal where as in PAM it is maintained constant.

There are two operations involved in the generation of the 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

Let h(t) be the standard rectangular pulse of unit amplitude and duration T and its spectrum
H(f)
Let s(t) denote the sequence of flat top samples. We may express the PAM signal as,

(1)

By definition, the instantaneously sampled version of m(t) is given by

(2)

Using the sifting property of the delta function,

(3)

From equations (1) and (3)

Taking fourier transform on both sides

 (4)

(a)Spectrum of sampled signal.

(b)Spectrum of low pass filter

(c)Transmitted spectrum
Reconstruction of m(t) from PAM signal s(t)

To recover the original message signal m(t), s(t) is passed through a reconstruction filter
whose frequency response is shown below
d)Amplitude response of reconstruction filter

It is assumed that the message signal is limited to bandwidth W and the sampling rate fs is
larger than the Nyquist rate 2W.

From eqn (4), the spectrum of the resulting filter output is M(f)H(f) as shown below.

(e) spectrum after receiver filtering

Fourier transform of the rectangular pulse h(t) is given by

By using flat-top samples to generate a PAM signal, we have introduced amplitude distortion
as well as a delay of T/2. The distortion caused by the use of pulse-amplitude modulation 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.

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
amplitude response of the equalizer is given by
Time-Division Multiplexing
The sampling theorem provides the basis for transmitting the information contained in a band-
limited message signal m(t) as a sequence of samples of m(t) taken uniformly at a rate that is
usually slightly higher than the Nyquist rate.

An important feature of the sampling process is a conservation of time. That is, the transmission
of the message samples engages the communication channel for only a fraction of the sampling
interval on a periodic basis, and in this way some of the time interval between adjacent samples
is cleared for use by other independent message sources on a time-shared basis. We thereby
obtain a time-division multiplex (TDM) system, which enables the joint utilization of a
common communication channel by a plurality of independent message sources without
mutual interference among them.

Block Diagram of TDM

Each input message signal is first restricted in bandwidth by a low-pass pre-alias filter to re
move the frequencies that are nonessential to an adequate signal representation.
The low pass filter outputs are then applied to a commutator, which is usually implemented
using electronic switching circuitry.

The function of the commutator is twofold:

(1) to take a narrow sample of each of the N input messages at a rate/s that is slightly higher
than 2IV, where W is the cutoff frequency of the pre-alias filter.
(2) to sequentially interleave these N samples inside the sampling interval Ts.

Following the commutation process, the multiplexed signal is applied to a pulse modulator, the
purpose of which is to transform the multiplexed signal into a form suitable for transmission
over the common channel.

It is clear that the use of time-division multiplexing introduces a bandwidth expansion factor
N, because the scheme must squeeze N samples derived from N independent message sources
into a time slot equal to one sampling interval.
At the receiving end of the system, the received signal is applied to a pulse demodulator, which
performs the reverse operation of the pulse modulator. The narrow samples produced at the
pulse demodulator output are distributed to the appropriate low-pass reconstruction filters by
means of a decommutator, which operates in synchronism with the commutator in the
transmitter. This synchronization is essential for a satisfactory operation of the system.

Pulse width modulation (PWM) or Pulse Duration Modulation (PDM)

In pulse-duration modulation (PDM), the samples of the message signal are used to vary the
duration of the individual pulses. This form of modulation is also referred to as pulse-width
modulation or pulse-length modulation. The modulating signal may vary the time of occurrence
of the leading edge, the trailing edge, or both edges of the pulse.

Pulse position modulation

In PDM, long pulses expend considerable power during the pulse while bearing no additional
information. If this unused power is subtracted from PDM, so that only time transitions are
preserved, we obtain a more efficient type of pulse modulation known as pulse-position
modulation (PPM).
In PPM, the position of a pulse relative to its unmodulated time of occurrence is varied in
accordance with the message signal as shown in above figure (refer figure 7.10 )

Let Ts denote the sample duration. Using the sample m(nTs) of a message signal m(t) to
modulate the position of the nth pulse, we obtain the PPM signal.

(1)

where kp is the sensitivity of the pulse-position modulator and g(t) denotes a standard pulse.
Clearly, the different pulses constituting the PPM signal s(t) must be strictly non overlapping;
A sufficient condition for this requirement to be satisfied is to have

(2)

which, in turn, requires that,

(3)

The closer is to one half the sampling duration Ts, the narrower must the standard
pulse g(t) be in order to ensure that the individual pulses of the PPM signal s(t) do not interfere
with each other, and the wider will the bandwidth occupied by the PPM signal be.

Assuming that Eq. (2) is satisfied, and that there is no interference between adjacent pulses of
the PPM signal s(f), then the signal samples m(nTs) can be recovered perfectly.

GENERATION OF PPM WAVES

Figure 1: Block Diagram of PPM generator.

(a) Message signal. (b) Staircase approximation of the message signal, (c) Sawtooth wave.
(d) Composite wave obtained by adding (b) and (c). (e) Sequence of “Impulses” used to
generate the PPM signal.
The message signal m(t) is first converted into a PAM signal by means of a sample-and-hold
circuit, generating a staircase waveform u(t). The pulse duration T of the sample-and-hold
circuit is the same as the sampling duration Ts. Next, the signal u(t) is added to a sawtooth
wave yielding the combined signal v(t) shown in Figure d. The combined signal v(t) is applied
to a threshold detector that produces a very narrow pulse (approximating an impulse) each time
v{t) crosses zero in the negative-going direction. The resulting sequence of “impulses” i{t) is
shown in Figure (e). Finally, the PPM signal s(t) is generated by using this sequence of impulses
to excite a filter whose impulse response is defined by the standard pulse g(t).

DETECTION OF PPM WAVES

Consider a PPM wave S(t) with uniform Sampling. Assume that the message signal m(t) is
strictly band limited. The function of PPM receiver is as follows

• Convert the received PPM wave into a PDM wave with the same modulation.

• Integrate this PDM wave using a device with a finite integration time, thereby computing the
area under each pulse of the PDM wave.

• Sample the output of the integrator at a uniform rate to produce a PAM wave, whose pulse
amplitudes are proportional to the signal samples m(nTs) of the original PPM wave s(t).

• Finally, demodulate the PAM wave to recover the message signal m(t).

THE QUANTIZATION PROCESS

Amplitude quantization is defined as the process of transforming the sample amplitude m(nTs)
of a message signal m(t) at time t = nTs into a discrete amplitude v(nTs) taken from a finite set
of possible amplitudes.

quantization process is assumed to be memoryless and instantaneous, which means that the
transformation at time t = nTs is not affected by earlier or later samples of the message signal.
This simple form of quantization, though not optimum, is commonly used in practice.

the signal amplitude m is specified by the index k if it lies inside the interval.

where L is the total number of amplitude levels used in the quantizer.

The amplitudes mk, k = 1, 2,..., L, are called decision levels or decision thresholds. At the
quantizer output, the index k is transformed into an amplitude vk that represents all amplitudes
of the interval ik The amplitudes vk, k = 1, 2,...,..L, are called representation levels or
reconstruction levels, and the spacing between two adjacent representation levels is called a
quantum or step-size. Thus, the quantizer output v equals vk if the input signal sample m
belongs to the interval ik.

Quantizers can be of a uniform or nonuniform type. In a uniform quantizer, the representation

levels are uniformly spaced; otherwise, the quantizer is non-uniform.

The quantizer characteristic can also be of midtread or midrise type.

Quantization Noise:

The use of quantization introduces an error defined as the difference between the input signal
m and the output signal v. This error is called quantization noise. Figure below illustrates a
typical variation of the quantization noise as a function of time, assuming the use of a uniform
quantizer of the midtread type.

Let the quantizer input m be the sample value of a zero-mean random variable M. A quantizer
g(.) maps the input random variable M of continuous amplitude into a discrete random variable
V. Let the quantization error be denoted by the random variable Q of sample value q.

Let us find mean square value of the quantization error Q.

Consider then an input m of continuous amplitude in the range (-Vmax, Vmax). Assuming a
uniform quantizer of the midrise type illustrated in Figure b, we find that the step-size of the
quantizer is given by

where Lis the total number of representation levels. For a uniform quantizer, the quantization
error Q will have its sample values bounded by .
The probability density function of the quantization error Q as follows:
The variance of a quantization process Q is 𝜎x2 is given by
 2
For a sinusoidal modulating signal, average power 𝑃 = 𝑉𝑟𝑚𝑠 /𝑅
2
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Assume a load of 1Ω, 𝑃 = = 𝑉𝑟𝑚𝑠 =( ) =
𝑅 √2 2
Pulse Code modulation:

Figure 1: Block Diagram of PCM

In pulse-code modulation (PCM) a message signal is represented by a sequence of coded
pulses, which is accomplished by representing the signal in discrete form in both time and
amplitude. The basic operations performed in the transmitter of a PCM system are sampling,
quantizing, and encoding, as shown in Figure 1.

1. SAMPLING

The incoming message signal is sampled with a train of narrow rectangular pulses so as to
closely approximate the instantaneous sampling process. In order to ensure perfect
reconstruction of the message signal at the receiver, the sampling rate must be greater than
twice the highest frequency component W of the message signal in accordance with the
sampling theorem. In practice, a pre-alias (low-pass) filter is used at the front end of the sampler
in order to exclude frequencies greater than W before sampling.

2. QUANTIZATION

The sampled version of the message signal is then quantized, thereby providing a new
representation of the signal that is discrete in both time and amplitude.

Quantization may follow uniform law( mid thread and mid rise type) or in certain applications
it is preferable to use non-uniform quantizer.

The use of a nonuniform quantizer is equivalent to passing the baseband signal through a
compressor and then applying the compressed signal to a uniform quantizer.

e.g., µ law quantizer and A law quantizer

In order to restore the signal samples to their correct relative level, we must, use a device in the
receiver with a characteristic complementary to the compressor. Such a device is called an
expander. The combination of a compressor and an expander is called a compander.
3. ENCODING

Encoding process is used to translate the discrete set of sample values to a more appropriate
form of signal. A particular arrangement of symbols used in a code to represent a single value
of the discrete set is called a code word or character. Suppose that, in a binary code, each code
word consists of R number of bits per sample. Then, using such a code, we may represent a
total of 2R distinct numbers.

4. LINE CODES

It is in a line code that binary stream of data takes on an electrical representation.

e.g: 1) Unipolar Nonreturn-to-Zero (NRZ) Signaling.

2) Polar Nonreturn-to-Zero (NRZ) Signaling.

3)Unipolar Return-to-Zero (RZ) Signaling.

4)Bipolar Retum-to-Zero (BRZ) Signaling.
5) Split-Phase (Manchester Code)

5. REGENERATION

The most important feature of any digital system lies in the ability to control the effects of
distortion and noise produced by transmitting a digital signal through a channel.

This capability is accomplished by reconstructing the signal by means of a chain of

regenerative repeaters located at sufficiently close spacing along the transmission route. As
illustrated in Figure 7.27, three basic functions are performed by a regenerative repeater:
equalization, timing, and decision making. The equalizer shapes the received pulses so as to
compensate for the effects of amplitude and phase distortions produced by the transmission
characteristics of the channel. The timing circuitry provides a periodic pulse train, derived from
the received pulses, for sampling the equalized pulses at the instants of time where the signal-
to-noise ratio is a maximum. The sample so extracted is compared to a predetermined threshold
in the decision-making device. In each bit interval a decision is then made whether the received
symbol is a 1 or a 0 on the basis of whether the threshold is exceeded or not. If the threshold is
exceeded, a clean new pulse representing symbol 1 is transmitted to the next repeater.
Otherwise, another clean new pulse representing symbol 0 is transmitted. In this way, the
accumulation of distortion and noise in a repeater span is completely removed, provided that
the disturbance is not too large to cause an error in the decision-making process.

6. DECODING
The first operation in the receiver is to regenerate (i.e., reshape and clean up) the received
pulses one last time. These clean pulses are then regrouped into code words and decoded (i.e.,
mapped back) into a quantized PAM signal. The decoding process involves generating a pulse
the amplitude of which is the linear sum of all the pulses in the code word, with each pulse
being weighted by its place value (20 , 21 , 22 … … … … . 2𝑅−1 ) in the code, where R is the number
of bits per sample.

7. FILTERING

The final operation in the receiver is to recover the message signal wave by passing the decoder
output through a low-pass reconstruction filter whose cutoff frequency is equal to the message
bandwidth.
 8. MULTIPLEXING

In applications using PCM, it is natural to multiplex different message sources by time division,
whereby each source keeps its individuality throughout the journey from the transmitter to the
receiver. This individuality accounts for the comparative ease with which message sources may
be dropped or reinserted in a time-division multiplex system.

Differential Encoding:

This method is used to encode information in terms of signal transitions. In particular, a

transition is used to designate symbol 0 in the incoming binary data stream, while no transition
is used to designate symbol 1, as illustrated in Figure (a). In figure (b) we show the differentially
encoded data stream for the example data shown in figure(a). The waveform of the
differentially encoded data is shown in figure (c), assuming the use of unipolar nonreturn to-
zero signaling. The original binary information is recovered simply by comparing the polarity
of adjacent binary symbols to establish whether or not a transition has occurred. Note that
differential encoding requires the use of a reference bit before initiating the encoding process.
In the given example, symbol 1 is used as the reference bit.

Polarity + - - - + + - + +

Recovered input data 0 1 1 0 1 0 0 1

Non-Uniform Quantization
The use of a nonuniform quantizer is equivalent to passing the baseband signal through a
compressor and then applying the compressed signal to a uniform quantizer. A particular form
of compression law that is used in practice is the so called µ-law defined by

where m and v are the normalized input and output voltages, and µ is a positive constant.
The case of uniform quantization corresponds to µ = 0. For a given value of µ, the reciprocal
slope of the compression curve, which defines the quantum steps, is given by the derivative of
|m| with respect to |v|. µ-law is neither strictly linear nor strictly logarithmic, but it is
approximately linear at low input levels corresponding to µ|m|<< 1, and approximately
logarithmic at high input levels corresponding to µ|m|<< 1.

Another compression law that is used in practice is the so-called A-law defined by

which is shown plotted in figure below.

 The case of uniform quantization corresponds to A=1. The reciprocal slope of this compression
curve is given by the derivative of |m| with respect to |v|, as shown by

Thus the quantum steps over the central linear segment, which have the dominant effect on
small signals, are diminished by the factor A/(l + log A). This is typically about 25 dB in
practice, as compared with uniform quantization.

In order to restore the signal samples to their correct relative level, we must use a device in the
receiver with a characteristic complementary to the compressor. Such a device is called an
expander. Ideally, the compression and expansion laws are exactly inverse so that, except for
the effect of quantization, the expander output is equal to the compressor input. The
combination of a compressor and an expander is called a compander.

Line codes
The electrical representation of the encoded binary streams produced by their individual
transmitters so as to facilitate transmission of the binary streams across the communication
channel.

The lines codes often used the terminology nonreturn-to-zero (NRZ) or return-to-zero (RZ).
Return-to-zero implies that the pulse shape used to rep resent the bit always returns to the 0
volts or the neutral level before the end of the bit. Nonreturn-to-zero indicates that the pulse
does not necessarily return to the neutral level before the end of the bit.

Most commonly used line codes are

1. Unipolar NRZ(UNRZ)
2. Unipolar RZ(URZ)
3. Polar NRZ(PNRZ)
4. Polar RZ(PRZ)
5. Bipolar NRZ(BNRZ)
6. Bipolar RZ(BRZ)
7. Manchester or split phase

Bipolar signaling is also called AMI (alternate mark inversion) signaling format