DATA

CONVERSION
Presented By : Donald
Bodollo
Samar State University | 2024
 Data Conversion

The key to digital communication is to convert data in analog

form to digital form. Special circuits are available to do this. Once it is
in digital form, the data can be processed or stored. Data must
usually be reconverted to analog form for final consumption by the
user; e.g., voice and video must be in analog form.

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 Basic Principles of Data
Conversion
Translating an analog signal to a digital signal is called
analog-to-digital (A/D) conversion, digitizing a signal, or
encoding (see Fig. 7-5),. The device used to perform this
translation is known as an analog-to-digital (A/D) converter or
ADC. A modern A/D converter is usually a single-chip IC that takes
an analog signal and generates a parallel or serial binary output.
The opposite process is called digital-to-analog (D/A)
conversion. The circuit used to perform this is called a digital-
to-analog (D/A) converter (or DAC) or a decoder. The input to
a D/A converter may be a serial or parallel binary number, and the
output is a proportional analog voltage level. (see Fig. 7-6)
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 A/D
conversion
An analog signal is a smooth or continuous voltage or current
variation (see Fig. 7-7). It could be a voice signal, a video waveform, or
a voltage representing a variation of some other physical characteristic
such as temperature. Through A/D conversion these continuously
variable signals are changed to a series of binary numbers. A/D
conversion is a process of sampling or measuring the analog signal at
regular time intervals. At the times indicated by the vertical dashed
lines in Fig. 7-7, the instantaneous value of the analog signal is
measured and a proportional binary number is generated to represent
that sample. As a result, the continuous analog signal is translated to a
series of discrete binary numbers representing samples.

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 A key factor in the sampling process is the frequency of sampling f,
which is the reciprocal of the sampling interval t shown in Fig. 7-7. To
retain the high frequency information in the analog signal, a sufficient
number of samples must be taken so that the waveform is adequately
represented. It has been found that the minimum sampling frequency is
twice the highest analog frequency content of the signal. For example, if
the analog signal contains a maximum frequency variation of 3000 Hz,
the analog wave must be sampled at a rate of at least twice this, or
6000 Hz. This minimum sampling frequency is known as the Nyquist
frequency fN. (And fN >= 2fm, where fm is the frequency of the input
signal.) An alias is a signal that is mistakenly sampled when the
sampling frequency is less than twice the input frequency. An
antialiasing filter is used to ensure that the correct signal is used.
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 D/A conversion
To retain an analog signal converted to digital, some form of binary
memory must be used. The multiple binary numbers representing each
of the samples can be stored in random access memory (RAM). Once
they are in this form, the samples can be processed and used as data
by a microcomputer which can perform mathematical and logical
manipulations. This is called digital signal processing (DSP).
At some point it is usually desirable to translate the multiple binary
numbers back to the equivalent analog voltage. This is the job of the
D/A converter, which receives the binary numbers sequentially and
produces a proportional analog voltage at the output. Because the
input binary numbers represent specific voltage levels, the output of
the D/A converter has a stairstep characteristic. 7
 If these binary numbers are fed to a D/A converter, the output is a stairstep
voltage as shown. Since the steps are very large, the resulting voltage is only
an approximation to the actual analog signal. However, the stairsteps can be
filtered out by passing the D/A converter output through a low-pass filter with
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an appropriate cutoff frequency.
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 PULSE MODULATION
It is the process of changing a binary pulse signal to
represent the information to be transmitted. The primary benefits
of transmitting information by binary techniques arise from the
great noise tolerance and the ability to regenerate the degraded
signal. Any noise that gets added to the binary signal along the
way is usually clipped off. Further, any distortion of the signal can
be eliminated by reshaping the signal with a Schmitt trigger,
comparator, or similar circuit.

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 If information can be transmitted on a carrier consisting of binary
pulses, these aspects of binary techniques can be used to improve
the quality of communications. Pulse modulation techniques were
developed to take advantage of these qualities. The information
signal, usually analog, is used to modify a binary (on/off) or pulsed
carrier in some way.
With pulse modulation the carrier is not transmitted continuously
but in short bursts whose duration and amplitude correspond to the
modulation.
There are four basic forms of pulse modulation: Pulse-amplitude
modulation (PAM), pulse-width modulation (PWM), pulse-position
modulation (PPM), and pulse-code modulation (PCM).
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 COMPARING PULSE-
MODULATION METHODS
Fig. 7-28 shows an analog modulating signal and the various
waveforms produced by PAM, PWM, and PPM modulators. In all
three cases, the analog signal is sampled, as it would be in A/D
conversion. The sampling points are shown on the analog
waveform. The sampling time interval t is constant and subject
to the Nyquist conditions. The sampling rate of the analog
signal must be at least two times the highest frequency
component of the analog wave.

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 Of the four types of pulse modulation, PAM is the simplest and least
expensive to implement. On the other hand, because the pulses vary in
amplitude, they are far more susceptible to noise, and clipping techniques
to eliminate noise cannot be used because they would also remove the
modulation.
PWM and PPM are binary and therefore clipping can be used to reduce
the noise level. Although the techniques of pulse modulation have been
known for decades, they are no longer widely used. Of the three types,
PWM is the most common. One example is for remote-control purposes,
e.g., in model airplanes, boats, and cars.
Pulse-width modulation (PWM) methods are also used in switch mode
power supplies (dc-dc convertors, regulators, etc.), motor speed control,
as well as in class D audio switching power amplifiers. Today pulse-
modulation techniques have been largely superseded by more advanced
digital techniques such as pulse-code modulation (PCM), in which actual
binary numbers representing the digital data are transmitted. 14
 PULSE-CODE MODULATION
The fundamental and most important pulse digital modulation technique is the
pulse code modulation (PCM). This technique is the breakthrough for moving from
analog to digital communication. PCM technique is essentially the result of the
thought process to represent message signal in digital form rather than the original
analog form. The motivation is the merit of digital signal over analog signal for
communication, namely, noise robustness.

PCM may be treated as an extension of PAM. In PAM the time parameter is

discretized, but the amplitude still remains continuous. That is, within the allowable
amplitude limits; the signal value can take on infinite values. However, all these
infinite values may not be distinct from the perception (auditory or visual) point of
view. For instance, in case of speech signal, all amplitude values may not be
important from the auditory perception point of view. Therefore, we may not lose
information by discretizing the amplitudes to some finite values. 15
 What is essentially done is to round or approximate a group of nearby
amplitude values and represent them by a single discrete amplitude value.
This process is termed as quantization. The signal with discretized
amplitude values is termed as quantized signal. There will be error
between the original analog signal and its quantized version which is
measured and represented in terms of quantization noise. What is
preferable is minimum quantization noise and hence more closely quantizing
signal amplitudes. This leads to a greater number of discrete levels. Hence it
is a tradeoff.
The quantization can be carried out either by dividing the whole
amplitude range into uniform or nonuniform intervals. Accordingly, we have
uniform and nonuniform quantization. However, nonuniform
quantization is relatively difficult to implement compared to uniform
quantization. 16
 PCM is also named after the same as uniform or nonuniform PCM. The
nonuniform quantization and hence PCM are based on the observation of
the nonuniform distribution of signal values within the allowable limits. For
instance, in case of speech, most of the signal values are around the zero
level and few will be in the maximum range. Hence benefit can be
achieved in terms of quantization noise by using nonuniform quantization.
The input of sampler block will have signal which is continuous both in
time and amplitude. The output of sampler block will have the signal which
is discrete in time and continuous in amplitude. The output of quantizer
will have signal which is discrete both in time and amplitude. The output of
the encoder will have unique binary code for each discrete amplitude
value. The whole process of sampling, quantizing and encoding is also
termed as analog to digital conversion (ADC) operation. Thus, for any
analog signal, the output of ADC is nothing but PCM signal.
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Sampling-involves measuring the value of the analog signal at regular intervals
of time
Quantization - where the amplitude of each sample is mapped to a discrete
value.
Encoding - After sampling and quantization, the resulting discrete values are
encoded into binary numbers, which can be processed, stored, or transmitted by
digital systems.
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Traditional PCM
In this PCM, the analog signal is sampled and converted to a
sequence of parallel binary words by an A/D converter. The parallel
binary output word is converted to a serial signal by a shift register (see
Fig. 7-29). Each time a sample is taken, an 8-bit word is generated by
the A/D converter. This word must be transmitted serially before
another sample is taken and another binary word is generated. The
clock and start conversion signals are synchronized so that the resulting
output signal is a continuous train of binary words.

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Traditional PCM

Fig. 7-30 shows the timing signals. The start conversion signal
triggers the S/H to hold the sampled value and starts the A/D
converter. Once the conversion is complete, the parallel word from
the A/D converter is transferred to the shift register. The clock pulses
start shifting the data out 1 bit at a time. When one 8-bit word has
been transmitted, another conversion is initiated and the next word is
transmitted. In Fig. 7-30, the first word sent is 01010101; the second
word is 00110011.

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 At the receiving end of the system, the serial data is shifted into a
shift register (see Fig. 7-31). The clock signal is derived from the data
to ensure exact synchronization with the transmitted data. Once one
8-bit word is in the register, the D/A converter converts it to a
proportional analog output. Thus the analog signal is reconstructed
one sample at a time as each binary word representing a sample is
converted to the corresponding analog value. The D/A converter
output is a stepped approximation of the original signal. This signal
may be passed through a low-pass filter to smooth out the steps.

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 Delta
Modulation
Delta modulation (DM) is obtained by simplifying the quantization and
encoding process of PCM. To enable this, the signal is sampled at much higher
than the required Nyquist rate. This oversampling process will result in the
sequence of samples which are very close and hence high correlation among

two successive samples are different by an amplitude of 𝛿. That is, the current
successive samples. Under this condition, it may be safe to assume that any

sample is either larger or smaller than the previous value by 𝛿. If it is larger,

then it is quantized as +𝛿 and as −𝛿 in smaller case. Since it is decided a
priori, only its sign is important. The sign information can be coded using one-
bit binary word, say; l represents + and 0 represent − . The quantization and
encoding blocks therefore become very simple. Thus, if we have the first signal
value and 1 bit quantization information we can reconstruct the complete
quantized signal. 25
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 The block diagram of delta modulator is given in Fig. 5.8 drawn
by referring to the block diagram of PCM given in Fig. 5.7. The
sampler block remains 'same as in the PCM, except that the
sampling frequency is much higher than in PCM case (say 4 times
or more). According to the principles of DM, the quantizer needs to
discretize the amplitude value by referring to the previous value
and say whether it is larger or smaller. Hence an accumulator is

and producing output into two discrete levels as +𝛿 and −𝛿 . The

needed to store previous sample, a summer as a comparing device

encoder is trivial which directly maps the signs of 𝛿 into I or 0. The

sequence of 1's and 0's at the output of encoder constitutes the
DM wave.
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Differential Pulse Code Modulation
Differential pulse code modulation (DPCM) first estimates the
predictable-part from the signal and then codes the unpredictable or error
signal in terms of unique binary words as in PCM and hence the name. The
motivation for the same is that most message signals have high correlation.
Therefore, it is possible to classify the information present in them into
predictable and the unpredictable parts.
The main merit in this approach is the significantly less variance among
the samples in the unpredictable version of the signal compared to the
original. Roughly the variance among the samples will be about half of that
of the original signal. As a result, binary words of smaller length are
sufficient for coding unpredictable part Hence the saving in the bandwidth
requirement, measured as bit rate defined as number of kilobits per second
(kbps). For instance, if64 kbps is required for PCM, then DPCM requires 28
about 48 kbps.
 The block diagram of DPCM modulator is given in Fig. 5.9 drawn
again by referring to the PCM block diagram in Fig. 5. 7. The input
analog signal is passed through the predictor block whose function is to
segregate the information into predictable and unpredictable parts.
The unpredictable part is passed through sampler, quantizer and
encoder blocks to get PCM corresponding to it. The predictable part is
directly passed through the encoder to get the codes. Both these are
combined to get the DPCM wave representing sequence of binary
words corresponding to both the parts

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Demodulation of Pulse Digital Modulated
Signals
The demodulation of PCM is straightforward. Figure 5.10 shows
the block diagram for the reconstruction of analog signal in case of
PCM. For obtaining PCM from analog signal, ADC was employed.
Therefore, for obtaining analog signal from PCM, the reverse of ADC
namely, digital to analog conversion (DAC) is required. Thus, the
binary words are applied one at a time to a DAC circuit to obtain
equivalent analog value. How close the reconstructed analog value to
the original depends on the amount of approximation errors
introduced due to ADC and DAC conversion. By the proper choice of
binary word length, it has been found that the errors are indeed
negligible from the perception point of view.

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 The block diagram of demodulation in case of DM is given in Fig. 5.
11. The DM needs to transmit the first sample and then the OM wave.
By combining both, the analog signal can be reconstructed from the

the first sample by adding to ±𝛿. The second sample is then stored in
DM wave in the following way: The second sample is constructed from

from the second sample using ±𝛿. The process continues till the last
the accumulator for future reference. The third sample is constructed

sample is reconstructed.

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The reconstruction of analog signal in case of DPCM is more involved
and is illustrated in Fig. 5.12. The approximate analog signal of the
unpredictable part is reconstructed by DAC as in PCM. This signal is
used as input to a block constructed using the predictable part and
the approximate version of the original analog signal is obtained at
the output of this block.

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 Benefits of
PCM
PCM is reliable, inexpensive, and highly resistant to noise. In PCM, the
transmitted binary pulses all have the same amplitude and, like FM signals, can
be clipped to reduce noise. Further, even when signals have been degraded
because of noise, attenuation, or distortion, all the receiver has to do is to
determine whether a pulse was transmitted. Amplitude, width, frequency, phase
shape, and so on do not affect reception. Thus PCM signals are easily recovered
and rejuvenated, no matter what the circumstances. PCM is so superior to other
forms of pulse modulation and multiplexing for transmission of data that it has
virtually replaced them all in communication applications.

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 Samar State University

THANK YOU FOR

LISTENING
Presented By : Donald Bodollo