AHMED O.

ABU ELKHAIR – ISLAMIC UNIVERSITY OF GAZA (2010)

Pulse Code Modulation

Introduction
Pulse Code Modulation (PCM) was pioneered by the British engineer Alec
Reeves in 1937. The first transmission of a message using PCM was in 1943
during the World War II.

Pulse code modulation is the heart of technology in communications in today’s

digital world. It’s a process in which analog signals are converted to digital
form. The analog signal is represented by a series of pulses and non-pulses
(1 or 0 respectively). The magnitude of signal is regularly sampled in uniform
intervals, and then quantized in a series of binaries.

PCM has been used in digital telephone systems and 1980s-era electronic
musical keyboards. It is also the standard form for digital audio in computers
and the compact disc "red book" format. It is also standard in digital video,
for example, using ITU-R BT.601. Uncompressed PCM is not typically used
for video in standard definition consumer applications such as DVD or DVR
because the bit rate required is far too high.

Why PCM?
The stream of pulses and non-pulse streams of 1’s and 0’s are not easily
affected by interference and noise. Even in the presence of noise, the
presence or absence of a pulse can be easily determined. Since PCM is digital,
a more general reason would be that digital signals are easy to process by
cheap standard techniques. This makes it easier to implement complicated
communication systems such as telephone networks.

In the following sections we will discuss the stages of PCM in an extended

way with full details.

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 AHMED O. ABU ELKHAIR – ISLAMIC UNIVERSITY OF GAZA (2010)

PCM Implementation
The practical implementation of PCM makes use of other processes. The
processes are carried out in the order in which they appear below:

 Filtering
 Sampling
 Quantizing
 Encoding

Many authors ignore the first step of filtering as they consider it as a

secondary stage. Simply we can say that filtering stage removes frequencies
above the highest signal frequency. These frequencies if not removed, may
cause problems when the signal is going through the stage of sampling. It’s
practically important step, but we can ignore it in theory. Next we will discuss
extensively the other important three processes which together called the
modulation process.

Figure 1 - Block diagram for A/D Converter using PCM

Sampling
Theoretically, it considered as the first step in implementation. PCM mostly
based on the sampling theorem which states that If a signal f(t) is sampled
at regular intervals of time and at a rate higher than twice the highest
significant signal frequency, then the samples contain all the information of
the original signal. The function f(t) may be reconstructed from these samples
by the use of a low-pass filter. Figure (2) briefly describe the functionality of
sampler process. If voice data are limited to frequencies below 4000 Hz, a
conservative procedure for intelligibility, 8000 samples per second would be
sufficient to completely characterize the voice signal. Note, however, that
these are analog samples.

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 AHMED O. ABU ELKHAIR – ISLAMIC UNIVERSITY OF GAZA (2010)

Figure 2 - The Sampling Process

The telephone system, being a worldwide standard 8kHz sampling system,

satisfies Nyquist, as all voice signals are band-limited to 4kHz. When the voice
waveform is sampled, a train of short pulses is produced, each representing
the amplitude of the waveform at the specific instant of sampling. This
process is called Pulse Amplitude Modulation (PAM). The envelope of the PAM
samples replicates the original waveform.

In PAM the successive sample values of the analog signal s(t) are used to
effect the amplitudes of a corresponding sequence of pulses of constant
duration occurring at the sampling rate. No quantization of the samples
normally occurs (Figure 3a, b). In principle the pulses may occupy the entire
time between samples, but in most practical systems the pulse duration,
known as the duty cycle, is limited to a fraction of the sampling interval. Such
a restriction creates the possibility of interleaving during one sample interval
one or more pulses derived from other PAM systems in a process known as
time-division multiplexing (TDM).

Figure 3 - (a) Analog signal, s(t). (b) Pulse-amplitude modulation

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Quantization
The PAM samples still represent the voice signal in analog form. For digital
transmission, further processing is required. Pulse Code Modulation is a
technique used to convert the PAM samples to a binary weighted code for
digital transmission. PCM coding is a two step process performed by the
CODEC. The first step is quantization, where each sample is assigned a
specific quantizing interval. The second step is PCM coding of the quantizing
interval into an 8-bit PCM code word. Each is discussed in the text that
follows. Converting PAM samples to a digital signal involves assigning the
amplitude of a PAM sample one of a whole range of possible amplitude values,
which are divided into quantizing intervals. There are 256 possible quantizing
intervals, 128 positive and 128 negative. The boundaries between adjacent
quantizing intervals are called decision values. Below Figure(4) show the
simple representation of quantization process.

Figure 4 - Quantization Process

If the max and min amplitude values of information signal x(t) are Amax and
Amin, respectively, and if n-digit code words will be used, then the quantizing
interval/pace “a”

Becomes:

In quantizing process, “which quanta region does the sample belong to” is an
important question. The sample value is rounded to the closest quanta level.
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 AHMED O. ABU ELKHAIR – ISLAMIC UNIVERSITY OF GAZA (2010)

Later the quantized signal is encoded and the signal is matched with code
words. In two-word number system, +V volt pulse can be sent for ‘1’s, and
space/no volt is sent for ‘0’s to transmit the code.

As another method, +V volt pulse is sent for ‘1’s, and –V volt pulse is sent
for ‘0’s. A guide gap (tg) is kept between two pulses. An example to the PCM
steps explained up to here is given in Figures 5 and 6 respectively.

Figures 5 and 6 - Sampling & Quantizing of an analog signal

and indication of corresponding PCM waveforms

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Encoding
The last stage is encoding, in which the result of quantization process
converted to bit streams, each sample can be change to a code word. For
example a quantization code of 2 is encoded as 010; 5 is encoded as 101;
and so on. Note that the number of bits for each sample is determined from
the number of quantization levels. If the number of quantization levels is L,
the number of bits is [n = Log2 L].

Pulse-code modulation can be either return-to-zero (RZ) or non-return-to-

zero (NRZ). For a NRZ system to be synchronized using in-band information
there must not be long sequences of identical symbols, such as ones or
zeroes. For binary PCM systems, the density of 1-symbols is called ones-
density.

Ones-density is often controlled using pre-coding techniques such as Run

Length Limited encoding, where the PCM code is expanded into a slightly
longer code with a guaranteed bound on ones-density before modulation into
the channel. In other cases, extra framing bits are added into the stream
which guarantees at least occasional symbol transitions.

Another technique used to control ones-density is the use of a scrambler

polynomial on the raw data which will tend to turn the raw data stream into
a stream that looks pseudo-random, but where the raw stream can be
recovered exactly by reversing the effect of the polynomial. In this case, long
runs of zeroes or ones are still possible on the output, but are considered
unlikely enough to be within normal engineering tolerance.

In other cases, the long term DC value of the modulated signal is important,
as building up a DC offset will tend to bias detector circuits out of their
operating range. In this case special measures are taken to keep a count of
the cumulative DC offset, and to modify the codes if necessary to make the
DC offset always tend back to zero.

Many of these codes are bipolar codes, where the pulses can be positive,
negative or absent. In the typical alternate mark inversion code, non-zero
pulses alternate between being positive and negative. These rules may be
violated to generate special symbols used for framing or other special
purposes.

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Figure 7 - Encoding Process

Demodulation Process
To produce output from the sampled data, the procedure of modulation is
applied in reverse (See Figure 8). After each sampling period has passed, the
next value is read and a signal is shifted to the new value. As a result of these
transitions, the signal will have a significant amount of high-frequency
energy. To smooth out the signal and remove these undesirable aliasing
frequencies, the signal would be passed through analog filters that suppress
energy outside the expected frequency range (that is, greater than the
Nyquist frequency fs / 2). Some systems use digital filtering to remove some
of the aliasing, converting the signal from digital to analog at a higher sample
rate such that the analog filter required for anti-aliasing is much simpler. In
some systems, no explicit filtering is done at all; as it's impossible for any
system to reproduce a signal with infinite bandwidth, inherent losses in the

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system compensate for the artifacts — or the system simply does not require
much precision. The sampling theorem suggests that practical PCM devices,
provided a sampling frequency that is sufficiently greater than that of the
input signal, can operate without introducing significant distortions within
their designed frequency bands.

The electronics involved in producing an accurate analog signal from the

discrete data are similar to those used for generating the digital signal. These
devices are DACs (digital-to-analog converters), and operate similarly to
ADCs. They produce on their output a voltage or current (depending on type)
that represents the value presented on their inputs. This output would then
generally be filtered and amplified for use.

Figure 8 - Demodulator is the reverse of Modulator

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PCM Timing and Synchronization

The PCM receiver must be able to identify the start and finish of each full
sampling sequence and to identify each bit position. The sampling clock needs
to be either sent to, or regenerated at, the receiving side to determine when
each full sequence of sampling begins and ends. The data clock is also needed
to determine exactly when to read each bit of information. A PCM channel is
sampled at 8,000 Hz or once every 125 Ms. If there is one channel or 30 TDM
channels the sampling period is fixed at 125 Ms and this period is known as
a frame. Therefore the frame clock must have a period of 125 Ms. the rising
edge of the frame clock informs the receiver that the next bit will be Bit 1 of
a new sample. The falling edge of the data clock informs the receiver that it
must read the data bit.

When the bit stream is transmitted along a line the pulses become distorted
and the rise and fall times become significant. Ideally, a 1 will be “high” for
15.625 Ms. In practice, the pulse may only be above the “high” threshold for
a few Ms so it is very important that the bit is read within a certain time limit
of the clock pulse.

Figure 9 - PCM Timing and Synchronization

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The simplest way to synchronize a PCM sender to a PCM receiver is to send

the clock signals on different circuits to the data this would be done in a self-
contained system such as private branch exchange (PBX). Telephony is full
duplex so that there is a coder and a decoder at each port, but each would
use the same clock.

To minimize the number of circuits it is possible to use a line-coding scheme

which allows the receiver to extract the clocks from the PCM signal. In this
case the receiver will have free running clocks that lock (using a PLL) to the
phase and frequency of the transitions in the data stream. The line-coding
scheme ensures that there is a transition for every data bit.

Another technique was recently developed which is related to PCM in

functionality but different in complexity as it too simple compared to PCM.
This technique will be discussed in the next section.

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Delta Modulation

Delta modulation (DM or Δ-modulation) is an analog-to-digital and digital-to-

analog signal conversion technique used for transmission of voice information
where quality is not of primary importance. DM is the simplest form of pulse-
code modulation (PCM) where the difference between successive samples is
encoded into n-bit data streams. Another name for DM is pulse width
modulation (PWM).

The pulse code modulation(PCM) can transmit all the bits which are used to
code sample. Hence signaling rate and transmission channel bandwidth are
large in PCM. To overcome this problem delta modulation is used. Delta
modulation transmits only one bit per sample. That is the present sample
value is compared with original analog wave, whether the amplitude is
increased or decreased is sent. Input signal is approximated to step signal by
the delta modulator. The step size is fixed(in adaptive delta modulation is not
fixed). If the difference is positive, then the approximated signal is increased
by one step. If the difference is negative is reduced by one.

When the step is reduced, 0 is transmitted and if the step is increased, 1 is

transmitted. Thus for each sample, only one binary bit is transmitted. Now
let us discuss the advantages and disadvantage of delta modulation.

The advantages of delta modulation are as follows. The delta modulation

transmits only one bit for one sample. Thus the signaling rate and
transmission channel bandwidth is quite small for delta modulation. The
transmitter and receiver implementation is very much simple for delta
modulation. There is no analog to digital converter involved in delta
modulation. The disadvantages of delta modulation are slope overload
distortion and granular noise. And the error in it is greater than (PCM).

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Figure 10 - Block diagram of a Δ-modulator/demodulator

It is known as a single integration
modulator. The input signal is compared to the integrated output pulses and
the delta (difference) signal is applied to the quantizer. The quantizer
generates a positive pulse when the difference signal is negative, and a
negative pulse when the difference signal is positive. This difference signal
moves the integrator step by step closer to the present value input, tracking
the derivative of the input signal. The demodulator is simply an integrator
(like the one in the feedback loop) whose output rises or falls with each 1 or
0 received. The integrator itself constitutes a low-pass filter.

Figure 11 - Delta Modulation

Derived forms of delta modulation are continuously variable slope delta

modulation, delta-sigma modulation, and differential modulation. The
Differential Pulse Code Modulation is the super set of DM.

Applications of Delta Modulation

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1) Telecommunications 3) Audio Delay

Lines

2) Secure Communications

References

1. Bates and Gregory. Voice & Data Communications Handbook. 5th edition. McGraw-Hill Publishing,
2006.

2. William Stalling. Data and Computer Communications. 7th edition. Prentice Hall, 2004.

3. Wikipedia, http://en.wikipedia.org/wiki/ Pulse-code_modulation

4. William N. Waggener. Pulse Code Modulation Systems Design. Artech House Publishers, 1998.

5. Behrouz A. Forouzan. Data Communications and Networking. 4th edition. McGraw Hill Higher
Education, 2007.

6. Washington University website, http://cbdd.wsu.edu/kewlcontent/cdoutput/TR502/page13.htm

7. Shared Paper in PCM, Research by Intersil Inc., 1997.

8. Answers, http://www.answers.com/topic/pulse-modulation-2

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