Sanaa University

Faculty of Engineering
Electrical Department
Forth year
Signal Processing

PCM
Worked by :
Majid Mohammed AlZariey

2390/2012

Mahmoud Nasooh AlQousi

460 /2012

Supervised by:
Dr. Abdulsalam G. Alkholidi
Contents
Contents
What might PCM refer to??
Definition

1
2
5
0

History of PCM
The word pulse
Modulation
Angle Modulation
Amplitude Modulation
Pulse Modulation
Demodulation
Analog to Digital Conversion
PCM Parameters
PCM Types
The WAVE File Format
Applications
References

6
13
13
14
17
19
27
28
34
35
37
38
39

What might PCM refer to??

Process Control Monitor

Personal Call
Manager (telephones)

Process Control Management

Personal Computer Magazine

Power Control Module

Please Call Me

Pulse Code
Modulation

Definition
In Brief, What Is PCM?

Pulse-code modulation (PCM) is a data transmission scheme well-suited to

transmitting binary data. Binary data is a sequence of ones and zeroes. In PCM,
these ones and zeroes are translated into pulses of transmitted energy. A pulse
with a positive amplitude corresponds to a binary one, while a negative
amplitude pulse corresponds to a zero. There are several advantages to using
PCM to transmit data.
4

In the digital domain, PCM (Pulse Code Modulation) is the most straightforward
mechanism to store audio. The analog audio is sampled in accordance with the
Nyquest theorem and the individual samples are stored sequentially in binary
format.

PCM samples the signal 8000 times a second; each sample is represented by 8
bits for a total of 64 Kbps. There are two standards for coding the sample level.
The Mu-Law standard is used in North America and Japan while the A-Law
standard is use in most other countries.

History of PCM
In the history of electrical
communications, the earliest reason for
signal was to interlace samples from
multiple telegraphy sources, and convey
single telegraph cable and, it was
as early as 1853, by the American
inventor Moses G. Farmer.

In 1920, the Bartlane cable picture

transmission system, named after its
inventors Harry G. Bartholomew and Maynard
McFarlane, used telegraph signaling of
punched in paper tape to send samples of
quantized to 5 levels,this is considered to be
PCM.

sampling a
them over a
conveyed

D.
characters
images
kind of

In 1926, Paul M. Rainey of Western

patented a facsimile machine which
its signal using 5-bit PCM, encoded by an
mechanical analog-to-digital converter. The
not go into production.

Electric
transmitted
optomachine did

In 1937, Alec Reeves came up with

the idea of
Pulse Code Modulation (PCM). At the time, few, if any, took notice of Reeves
development. Even Reeves was forced to abandon his invention unable to see
how it could be implemented with the technology of the day. In 1965, some 28
years later, the Franklin Institute awarded Alec Reeves the Stuart Ballantine
Medal for his pioneering work on PCM. Labeling it a major communications
invention, the Franklin institutes press release reminded the public that PCM
had made it possible for the Mariner IV spacecraft to transmit its wonderful
images of Mars back to Earth. But in 1965, the true potential of PCM was still
untapped. Today, on the seventy-fifth anniversary of Reeves idea, PCM has
become an indispensable element in our modern communications
infrastructure and a fundamental enabler of modern popular culture. For
example, PCM has very dramatically transformed the way we record, distribute,
and listen to music.
Long Distance Telephony and Noise

Alec Reeves, like other engineers working in telephony, grappled with the
problem of the additive nature of noise when a signal underwent multiple
amplifications along a long distance line. The development of telephony was a
remarkable advance over telegraphy but it also introduced a new challenge.
How was one to transmit an analog signal over long distances? Lee De Forests
invention of the triode vacuum tube in 1906, which he called the Audion, not
only heralded the birth of electronics and the rise of the radio broadcast
technology, it also provided telephony with an important tool to expand the
range of long distance calls: an amplification device. But each time the
telephone signal was amplified, more noise would be introduced. Because of
the dot-dash encoding, telegraphy did not suffer from the same problem. A
telegraph repeater could easily replicate a weak dot or dash into a fresh one
6

without introducing any noise. In 1937, Reeves had concluded that the best
way to overcome the noise issue in long distance telephony was to transmit a
digitized version of the analog voice signal.
Alec Reeves was born on 10 March 1902, in Redhill, Surrey, U.K. Reevess
father, Edward Ayearst Reeves, had a distinguished career as a geographer. He
was noted author on cartography and the Royal Geographical Societys
Surveyor. In 1918, Alec Reeves went to Imperial College, London, to study
engineering. After graduating in 1921, he did postgraduate study at Imperial
College. In 1923, Reeves joined the London Laboratory of International Western
Electric, a leading manufacturer of radio and telecommunications equipment.
In 1925, after his firm had been taken over by International Telephone and
Telegraph (IT&T), Reeves went to work at IT&T's laboratory in Paris. It was there
that Reeves came up, in 1937, with the idea of using a binary representation of
sound to overcome the noise issue in long distance analog telephone
transmissions. It a sense it was a return to the robustness of telegraphy.
Nearly seventeen years earlier, in 1921, Paul M. Rainey, from Western Electric,
had filed a patent for a machine that would send faxes via telegraphy using a
PCM-like technique to encode the optical scans of the pages. An object of this
invention, claimed Rainey in his patent, is to provide means whereby
facsimile pictures, drawings or the like may be transmitted by means of code
combinations or permutations of electrical impulses. It took five years for the
patent to be granted. Perhaps the patent office had difficulty wrapping its mind
around the idea. Little is known as to whether Western Electric took the idea
seriously and tried to produce a working prototype. Reeves knew nothing of
Raineys PCM technique, which used an opto-mechanical implementation.
Besides, Reeves was interested in an entirely different problem: noise in long
distance telephony, using purely electronic digital techniques.

The First Disclosure of PCM: Paul M. Rainey, "Facsimile Telegraph System," U.S.
Patent 1,608,527. Filed 20 July 1921.Issued 30 November 1926.
In 1938, after obtaining a French patent for his idea, he filed for a U.S. patent in
1939, which was then granted in 1942. His patents characterless title, Electric
Signaling System, stood in sharp contrast to the great import of the patents
contents. Many years later, Reeves recalled that, from the beginning, he
realized that it could be the most powerful tool so far against the effects of
interference on speech especially on long routes with many regenerative
repeaters, since these devices could easily be designed and spaced so as to
make the noise nearly noncumulative. And yet Reeves walked away from this
work. He realized that the PCM was an idea ahead of its time. The state of
electronics at the outbreak of WW II was not up to the task of making PCM a
viable commercial solution for telephony. Time would be needed for digital
electronic hardware to catch up to the demands required by PCM. Finally, with
the outbreak of war, Reevess focus shifted to the war effort and radar. He
became be the Chief Scientist at Britains the Air Ministry Research
Establishment, which had been founded by Watson Watt. During this time he
also invented Oboe a system to for accurate bombing through overcast skies.
Oboe was used in the large bombing raids over Germany and in the Pacific.
Paradoxically, a wartime imperative brought a new impetus to the development
of PCM, but this time from a very different need, one that had little to do with
long distance telephony and noise.
Making Telephone Calls Secret: Bell Labs and SIGSALY

At the start of WW II, the only available technology for secure voice
communication was the A-3 Scrambler system. U.S. military authorities did not
know that the Germans had broken the A-3 Scrambler. Nevertheless top
8

military officers like General George Marshal did not trust A-3 to securely
transmit the most sensitive of information. Very early on in the war, the U.S.
Army asked Bell Labs to come up with a new way of securing voice
communications. It soon became apparent that digitizing the analog voice
signal would allow one to apply cyphering techniques to the message. With
cross-licensing agreements with IT&T, the Bell Labs people turned to Reeves
work on PCM. The resulting speech enciphering system, called SYGSALY,
became the first working example of PCM technology. Under the cloak of
secrecy, Bell Labs made great strides in advancing the state-of-the-art in PCM
techniques. By the war's end, several groups at Bell Labs had worked on PCM.

SIGSALY terminal (1943)

During the 1947-48 period, in numerous articles, the Bell Labs work on PCM
finally became public. H.S. Black and J.O. Edson, who had been key people in
Bells speech encryption efforts, published their account in the AIEE
Transactions. They announced to the world that a radically new modulation
technique for multichannel telephony has been developed which involves the
conversion of speech into coded pulses. They also recognized the importance
of Reeves patent. They concluded this important paper with PCM appears to
have exceptional possibilities from the standpoint of freedom from interference
especially when applied to systems having many repeaters in tandem, but its
full significance in connection with future radio and wire transmission may take
some time to reveal. It is interesting that Black and Edson chose an AIEE and
9

not IRE journal in which to reveal this work to the world. In 1957, Black went on
to win AIEEs Lamme Medal. In 1948, which, in part, was due to his work in
PCM. In 1948, Oliver, Pierce and Shannon published their landmark The
Philosophy of PCM in the Proceedings of the IRE. Their paper, a rigorous
analysis of PCM, confirmed the merits of Reeves original conception.
PCM offers a greater improvement in signal-to-noise than other systems. By
using binary (on-off) PCM, a high-quality signal can be obtained under
conditions of noise and interference so bad that it is just possible to recognize
the presence of each pulse. Further, by using regenerative repeaters which
detect the presence or absence of pulses and then emit reshaped, respaced
pulses, the initial signal-to-noise ratio can be maintained through a long chain
of repeaters.
Although they saw equipment for PCM as more complex than other forms of
modulation, Oliver, Pierce, and Shannon concluded that in all, PCM seems
ideally suited for multiplex message circuits, where a standard quality and high
reliability are required.
What is striking about these papers, and all the others published by the Bells
Labs group during the late 1940s, is the absence of any reference to speech
encryption, which had been the driving force for Bells entry into PCM. The
transition to civilian applications appears to have been seamless. When it
came to it R&D investment in PCM, Bell Labs never took its eye off the
companys central mission, the telephone communications business. Although
PCM for civilian uses had gotten off to a good start, progress remained slow.
Reeves observed that PCM had been a child with a long infancy, and that, even
in 1965, this technology was still in the adolescent stage. Adequate
miniaturization was still holding back its development. But two decades after
the invention of the transistor at Bell Labs, semiconductor technology was
finally diffusing rapidly through the economy. This accelerated progress was
finally providing the hardware needed to make PCM economically viable for the
wider civilian market. Reeves believed that PCM was going to be essential
enabler for the information society that was appearing on the horizon.
ARPANET, timesharing services, and the rise of cable television pointed to a
demand for technology that could move large volumes of information across
national and international networks. In 1965, Reeves argued that, by the year
2000, transmitting moving pictures would also be an essential part of data
networks. He also felt that the pressures on the transportation infrastructure
would further increase the importance of PCM. In the year 2000 commuters
will refuse to accept the delays and inconveniences that even a moderate
10

journey to and from their place of work would entail. We shall have to transport
the brains, the skills of the staff, not their bodies, to their daily jobs, again
involving not merely ordinary data !inks but a great many private television
channels as well. Reeves concluded his crystal ball gazing by suggesting that
PCM would form the very backbone of the communications systems. He was on
the mark with this prediction, but his suggestion of a revolution in commuting
patterns may need a few more decades before it comes to pass.
Although PCM had advanced considerably during Reevess life time, he never
lived to see it outgrow adolescence. Reeves died in 1971.

The word pulse

Refers to pulses found in transmission lines, which are a natural consequence
of two other almost simultaneously evolved analog methods: pulse width
modulation and pulse position modulation, where each uses discrete signal
pulses of varying widths or positions. Otherwise, PCM has little similarity to
these other forms of signal encoding. These methodologies were introduced to
the U.S. in the early 1960s as telephone companies began converting voice to
digital signals to facilitate transmission between cities.

Modulation:
Modulation is a technique in which message signal is transmitted to the
receiver with the help of carrier signal. Here in modulation, we combine both
carrier signal and message signal. You may get the doubt that what is the need
of modulation. Just imagine that you have a paper which contains the message
and you would like to send it to your friend standing 40 feet from your place.
You cant just through the paper to your friend because paper will not travel
that much distance but if you take small stone and cover the paper with it and
through it to your friend, it will definitely reach the target. In the same way, we
need a carrier signal to transmit our message. Sometimes, message signal is
also called as modulating signal. The exact definition of modulation is given
below:
Modulation is a process of message signal and modulating is varied according
to the carrier signal for transmission purpose. The message signal can varied in
11

accordance to the carrier signal that is in terms of angular or amplitude. So we

are modulating the signal.
Advantages of Modulation:

With the help of modulation, we can increase the quality of reception.

We can also decrease the height of the antenna.
Avoid mixing of different frequency signals and increase the range of
communication i.e. without modulation, we can transmit the message up
to 100 meters and with modulation, we can transmit the message up to
150 meters.
Allow the flexibility for adjusting the bandwidth.
In the definition, we have seen that message signal can be varied according to
the angular or amplitude of the carrier signal. What happens if message signal
is varied in accordance to angular of carrier signal and what happens if
message signal is varied according to the amplitude of carrier signal?

Angle Modulation:
In the angle modulation, again there are two different types of modulations.
Frequency modulation
Phase modulation.
1. Frequency Modulation:

The process of carrier signal frequency is varied according to the message

signal or modulation signal
frequency by
keeping the amplitude
constant is
called frequency
modulation.

Advantages of Frequency

Modulation:

Frequency modulation has more noise resistivity when compared to other

modulation techniques. Thats why they are mainly used in broadcasting
12

and radio communications.

And we are all well aware that radio communication use mainly frequency
modulation for transmission. We know that noise will occur mainly to the
amplitude of the signal. In frequency modulation, amplitude is made
constant and only frequency is varied, so we can easily find out the noise
in the amplitude by using a limiter.
The frequency modulation is having greater resistance to rapid signal
strength variation, which we will use in FM radios even while we are
travelling and frequency modulation is also mainly used in mobile
communication purposes.
For transmitting messages in frequency modulation, it does not require
special equipments like linear amplifiers or repeaters and transmission
levels or higher when compared to other modulation techniques. It does
not require any class C or B amplifiers for increasing the efficiency.
Transmission rate is good for frequency modulation when compared to
other modulation that is frequency modulation can transmit around 1200
to 2400 bits per second.
Frequency modulation has a special effect called capture effect in which
high frequency signal will capture the channel and discard the low
frequency or weak signals from interference.

Disadvantages of Frequency Modulation:

In the transmission section, we dont need any special equipment but in

the reception, we need more complicated demodulators for demodulating
the carrier signal from message or modulating signal.
Frequency modulation cannot be used to find out the speed and velocity of
a moving object. Static interferences are more when compared to phase
modulation. Outside interference is one of the biggest disadvantages in
the frequency modulation. There may be mixing because of nearby radio
stations, pagers, construction walkie-talkies etc.
To limit the bandwidth in the frequency modulation, we use some filter
which will again introduce some distortions in the signal.
Transmitters and receiver should be in same channel and one free channel
must be there between the systems.
Spectrum space is limit for the frequency modulation and careful
controlling the deviation ration.
Applications of Frequency Modulation (FM):

Frequency modulation is used in radios which is very common in our daily

life.
13

 Frequency modulation is used in audio frequencies to synthesize sound.

For recording the video signals by VCR systems, frequency modulation is
used for intermediate frequencies.
Used in applications of magnetic tape storage.
2. Phase Modulation:

In the phase modulation, we vary

signal in accordance with the phase
modulating signal or message
keeping the frequency constant. If
of message or modulating signal is
phase shift will also be greater.

the carrier
of the
signal by
the amplitude
huge then the

Advantages and Disadvantages of Phase

Modulation:

The main advantage of phase modulation is that it has less interference

from static, which is why we use this type of modulation in finding out the
speed or velocity of a moving object. In frequency modulation, we cannot
find out the velocity of moving object.
Disadvantages of Phase Modulation:

The main disadvantage is phase ambiguity comes if we increase the phase

modulation index, and data loss is more and we need special equipment
like frequency multiplier for increasing the phase modulation index.
Applications of Phase Modulation:

Phase modulation application is not different from frequency modulation.

Phase modulation is also used in communication systems.
It may be used in binary phase shift keying.

Amplitude Modulation:
In the amplitude modulation, amplitude of carrier signal wave is varied in
accordance with the modulating or message signal by keeping the phase and
frequency of the signals constant. The carrier signal frequency would be greater
14

than the modulating signal

frequency. Amplitude
modulation is first type of
modulation used for
transmitting messages for
long
distances by the mankind.
The AM
radio ranges in between 535
to 1705
kHz which is great. But when
compared to frequency
modulation, the Amplitude
modulation is weak, but still
it is
used for transmitting
messages. Bandwidth of
amplitude modulation should
be
twice the frequency of
modulating signal or
message signal. If the modulating signal frequency is 10 kHz then the
Amplitude modulation frequency should be around 20 kHz. In AM radio
broadcasting, the modulating signal or message signal is 15 kHz. Hence the AM
modulated signal which is used for broadcasting should be 30 kHz.
Advantages of Amplitude Modulation:

Because of amplitude modulation wavelength, AM signals can propagate

longer distances.
For amplitude modulation, we use simple and low cost circuit; we dont
need any special equipment and complex circuits that are used in
frequency modulation.
The Amplitude modulation receiver will be wider when compared to the FM
receiver. Because, atmospheric propagation is good for amplitude
modulated signals.
Bandwidths limit is also big advantage for Amplitude modulation, which
doesnt have in frequency modulation.
Transmitter and receiver are simple in Amplitude modulation. When we
take a demodulation unit of AM receiver, it consists of RC filter and a diode
which will demodulate the message signal or modulating signal from
modulated AM signal, which is unlike in Frequency modulation.
Zero crossing in Amplitude modulation is equidistant.

15

Disadvantages of Amplitude Modulation:

Adding of noise for amplitude modulated signal will be more when

compared to frequency modulated signals. Data loss is also more in
amplitude modulation due to noise addition. Demodulators cannot
reproduce the exact message signal or modulating signal due to noise.
More power is required during modulation because Amplitude modulated
signal frequency should be double than modulating signal or message
signal frequency. Due to this reason more power is required for amplitude
modulation.
Sidebands are also transmitted during the transmission of carrier signal.
More chances of getting different signal interfaces and adding of noise is
more when compared to frequency modulation. Noise addition and signal
interferences are less for frequency modulation. That is why Amplitude
modulation is not used for broadcasting songs or music.
Applications of Amplitude Modulation:

Used to carry message signals in early telephone lines.

Used to transmit Morse code using radio and other communication
systems.
Used in Navy and Aviation for communications as AM signals can travel
longer distances.
Widely used in amateur radio.
All the above modulation will come under continuous wave modulation, where
we will use a sine wave as carrier signal. Continuous wave modulation can be
used in for both digital and analog communications.
When we take the pulse digital modulation, we use a periodic sequence of
rectangular pulses as the carrier signal. Pulse digital modulation is also used in
both analog and digital communications.

Pulse Modulation:
In pulse width modulation, there are different types of modulation for analog
and digital as shown below:
PCM: Pulse Code Modulation for Analog Modulation.
PPM: Pulse Position Modulation for Digital Modulation
PDM: Pulse Duration Modulation for Digital Modulation.
PAM: Pulse Amplitude Modulation for Digital Modulation.

16

1. Pulse Code Modulation (PCM):

Pulse Code Modulation is

first
introduced by Alec Reeves
in the year
1937. In the pulse code
modulation, Analog Signal is
reconstructed to digital
signal for
ease of transmission by
using the
analog signal samples. In
technical
terms, PCM will transmit the
analog in a
digital from, whose signal is
sampled at
regular intervals of time and
quantized
at same quantum levels to
digital
code. We know that digital
code is
nothing but binary code
which
consists of 1s and 0s that
is logic1
and logic0. So we will
transmit
the digital data in the form
of 1s and
0s. When the signal is
received by
the receiver, demodulator in the receiver will demodulate the binary signal
back into pulses with same quantum levels like in modulator and these pulses
are again used for regenerating the required analog signal.
Advantages of Pulse Code Modulation:

Low Noise Susceptibility

The PCM signal is a digital waveform. Digital waveforms are less susceptible to
interference and noise than analog signals. This is because a digital waveform
does not have to reproduce the exact data being transmitted. A transmitted
pulse that is close enough to the expected value of a binary one can be reliably
reproduced into a binary one. This low noise susceptibility allows PCM signals to
transmit farther than analog signals without signal degradation, information
loss, and distortion.
Repeatability
A PCM signal can be received by a repeater device that decodes the data and
retransmits it. This allows PCM signals to be sent very long distances without
data corruption. Repeaters must be placed close enough to the signal source so
that extreme noise does not corrupt the signal. Noise does not accumulate
even after many passes through multiple repeaters. This is because the signal
17

is completely regenerated by each repeater, making it noise-free at the start of

each repeated transmission.
Storage
A PCM waveform may be saved for later recreation or playback. Since PCM data
is digital in origin, it can be stored using a computer or similar device. An
example of a consumer device that stores PCM data is the Digital Versatile Disc
(DVD) technology. The audio portion of a DVD movie is encoded using PCM with
a sampling rate as high as 192 thousand samples per second. This PCM stream
can be piped directly to an amplifier using a digital audio cable, where it is then
decoded into an audible signal.
Encoded Signal
A PCM signal can be modulated in such a way that only a specific decoder can
make sense of the underlying data. This is useful when the data being sent
requires a level of security. The transmitter and receiver each have circuitry
that is analogous to a dictionary. This circuit maps the binary pulse-codes to
their definitions. When a pulse-code is received, the receiver looks up the
meaning in the dictionary. Anyone who intercepted the PCM signal would be left
with meaningless binary data.
Disadvantages of Pulse Code Modulation:

Specialized circuitry is required for transmitting and also for quantizing the
samples at same quantized levels. We can do encoding using pulse code
modulation but we need to have complex and special circuitry.
Pulse code modulation receivers are cost effective when we compared to
other modulation receivers.
Developing pulse code modulation is bit complicated and checking the
transmission quality is also difficult and takes more time.
Large bandwidth is required for pulse code modulation when compared to
bandwidth used by the normal analog signals to transmit message.
Channel bandwidth should be more for digital encoding.
PCM systems are complicated when compared to analog modulation
methods and other systems.
Decoding also needs special equipments and they are also too complex.
2. Pulse Amplitude Modulation (PAM):

In pulse amplitude modulation, the amplitude of regular interval of periodic

pulses or electromagnetic pulses is varied in proposition to the sample of
18

modulating signal or message signal. This is an analog type of modulation. In

the pulse amplitude modulation, the message signal is sampled at regular
periodic or time intervals and this each sample is made proportional to the
magnitude of the message signal. These sample pulses can be transmitted
directly using wired media or we can use a carrier signal for transmitting
through wireless. There are two types of sampling techniques for transmitting
messages using pulse amplitude modulation, they are
FLAT TOP PAM: The amplitude of each pulse is directly proportional to
instantaneous modulating signal amplitude at the time of pulse occurrence and
then keeps the amplitude of the pulse for the rest of the half cycle.
Natural PAM: The amplitude of each pulse is directly proportional to the
instantaneous modulating signal amplitude at the time of pulse occurrence and
then follows the amplitude of the modulating signal for the rest of the half
cycle.
Flat top PAM is the best for transmission because we can easily remove the
noise and we can also easily recognize the noise. When we compare the
difference between the flat
top PAM and
natural PAM, flat top PAM
principle of
sampling uses sample and
hold circuit.
In natural principle of
sampling,
noise interference is
minimum.
But in flat top PAM noise
interference
maximum. Flat top PAM
and natural
PAM are practical and
sampling
rate satisfies the sampling
criteria.
There are two types of
amplitude modulation
signal polarity

pulse
based on

Single polarity pulse amplitude modulation

Double polarity pulse amplitude modulation
In single polarity pulse amplitude modulation, there is fixed level of DC bias
added to the message signal or modulating signal, so the output of modulating
signal is always positive. In the double polarity pulse amplitude modulation, the
output of modulating signal will have both positive and negative ends.
Advantages of Pulse Amplitude Modulation (PAM):

19

 It is the base for all digital modulation techniques and it is simple process
for both modulation and demodulation technique.
No complex circuitry is required for both transmission and reception.
Transmitter and receiver circuitry is simple and easy to construct.
PAM can generate other pulse modulation signals and can carry the
message or information at same time.
Disadvantages of Pulse Amplitude Modulation (PAM):

Bandwidth should be large for transmitting the pulse amplitude

modulation signal. Due to Nyquist criteria also high bandwidth is required.
The frequency varies according to the modulating signal or message
signal. Due to these variations in the signal frequency, interferences will
be there. So noise will be great. For PAM, noise immunity is less when
compared to other modulation techniques. It is almost equal to amplitude
modulation.
Pulse amplitude signal varies, so power required for transmission will be
more, peak power is also, even at receiving more power is required to
receive the pulse amplitude signal.
3. Pulse Position Modulation (PPM):

In the pulse position modulation, the position of each pulse in a signal by taking
the reference signal is varied according to the sample value of message or
modulating signal instantaneously. In the pulse position modulation, width and
amplitude is kept constant. It is a technique that uses pulses of the same
breath and height but is displaced in time from some base position according to
the amplitude of the signal at the time of sampling. The position of the pulse is
1:1 which is propositional to the width of the pulse and also propositional to the
instantaneous amplitude of sampled modulating signal. The position of pulse
position modulation is easy when compared to other modulation. It requires
pulse width generator and monostable multivibrator.
Pulse width generator is used for generating pulse width modulation signal
which will help to trigger the monostable multivibrator, here trial edge of the
PWM signal is used for triggering the monostable multivibrator. After triggering
the monostable multivibrator, PWM signal is converted into pulse position
modulation signal. For demodulation, it requires reference pulse generator, flipflop and pulse width modulation demodulator.
Advantages of Pulse Position Modulation (PPM):

20

 Pulse position modulation has low noise interference when compared to

PAM because amplitude and width of the pulses are made constant during
modulation.
Noise removal and separation is very easy in pulse position modulation.
Power usage is also very low when compared to other modulations due to
constant pulse amplitude and width.
Disadvantages of Pulse Position Modulation (PPM):

The synchronization between transmitter and receiver is required, which is

not possible for every time and we need dedicated channel for it.
Large bandwidth is required for transmission same as pulse amplitude
modulation.
Special equipments are required in this type of modulations.
4. Pulse Duration Modulation (PDM) or Pulse Width Modulation (PWM):

It is a type of analog modulation. In pulse width modulation or pulse duration

modulation, the width of the pulse carrier is varied in accordance with the
sample values of message signal or modulating signal or modulating voltage. In
pulse width modulation, the amplitude is made constant and width of pulse and
position of pulse is made proportional to the amplitude of the signal. We can
vary the pulse width in three ways
By keeping the leading edge constant and vary the pulse width with respect to
leading edge
By keeping the tailing constant.
By keeping the center of the pulse constant.
We can generate pulse width using different circuitry. In practical, we use 555
Timer which is the best way for generating the pulse width modulation signals.
By configuring the 555 timer as monostable or astable multivibrator, we can
generate the PWM signals. We can use PIC, 8051, AVR, ARM, etc.
microcontrollers to generate the PWM signals. PWM signal generation has n
number of ways. In demodulation, we need PWM detector and its related
circuitry for demodulating the PWM signal.
Advantages of Pulse Width Modulation (PWM):

As like pulse position modulation, noise interference is less due to

amplitude has been made constant.
Signal can be separated very easily at demodulation and noise can also be
separated easily.

21

 Synchronization between transmitter and receiver is not required unlike

pulse position modulation.
Disadvantages of Pulse Width Modulation (PWM):

Power will be variable because of varying in width of pulse. Transmitter

can handle the power even for maximum width of the pulse.
Bandwidth should be large to use in communication, should be huge even
when compared to the pulse amplitude modulation.

22

Demodulation
To recover the original signal from the sampled data, a "demodulator" can
apply the procedure of modulation in reverse. After each sampling period, the
demodulator reads the next value and shifts the output signal to the new value.
As a result of these transitions, the signal has a significant amount of highfrequency energy caused by aliasing. To remove these undesirable frequencies
and leave the original signal, the demodulator passes the signal through analog
filters that suppress energy outside the expected frequency range (greater than
the Nyquist frequency
).The sampling theorem shows PCM devices can
operate without introducing distortions within their designed frequency bands if
they provide a sampling frequency twice that of the input signal. For example,
in telephony, the usable voice frequency band ranges from approximately
300 Hz to 3400 Hz. Therefore, per the NyquistShannon sampling theorem, the
sampling frequency (8 kHz) must be at least twice the voice frequency (4 kHz)
for effective reconstruction of the voice signal.
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 Digital-to-analog converters (DACs). They produce a voltage or
current (depending on type) that represents the value presented on their digital
inputs. This output would then generally be filtered and amplified for use.

23

Analog to Digital Conversion:

The A\D conversion is

generated by carrying out three basic operations:

Filtering
Sampling
Quantizing
Encoding

1- Filtering :

The first step to convert the signal from

analog to digital is to filter out the
higher frequency component of the
signal. This make things easier
downstream to convert this signal.
Most of the energy of spoken language
is
somewhere between 200 or 300 hertz
and
about 2700 or 2800 hertz. Roughly
3000-hertz bandwidth for standard
speech and standard voice
communication is established. Therefore, they do not have to have precise
filters (it is very expensive). A bandwidth of 4000 hertz is made from an
equipment point if view.

24

This band-limiting filter is used to

prevent aliasing (antialiasing). This
happens when the input analog voice
signal
is undersampled, defined by the
Nyquist criterion as Fs < 2(BW). The
sampling frequency is less than the
highest frequency of the input analog
signal. This creates an overlap between
the
frequency spectrum of the samples and
the
input analog
signal. The
lowpass output
filter,
used to
(Basic circuit of filters)
reconstruct the
original input
signal,
is not smart
enough to detect
this
overlap.
Therefore, it
creates a new signal that does not originate from the
source. This creation of a false signal when sampling is called aliasing.
(Filters in reality)

25

NyquistShannon sampling theorem

Fig: Fourier transform of

bandlimited function (amplitude vs frequency)

In the field of digital signal processing, the sampling theorem is a fundamental

bridge between continuous signals (analog domain) and discrete signals (digital
domain). Strictly speaking, it only applies to a class of mathematical functions
whose Fourier transforms are zero outside of a finite region of frequencies (see
Fig ). The analytical extension to actual signals, which can only approximate
that condition, is provided by the discrete-time Fourier transform, a version of
the Poisson summation formula. Intuitively we expect that when one reduces a
continuous function to a discrete sequence (called samples) and interpolates
back to a continuous function, the fidelity of the result depends on the density
(or sample-rate) of the original samples. The sampling theorem introduces the
concept of a sample-rate that is sufficient for perfect fidelity for the class of
bandlimited functions; no actual "information" is lost during the sampling
process. It expresses the sample-rate in terms of the function's bandwidth. The
theorem also leads to a formula for the mathematically ideal interpolation
algorithm.
The theorem does not preclude the possibility of perfect reconstruction under
special circumstances that do not satisfy the sample-rate criterion. (See
Sampling of non-baseband signals below, and compressed sensing.)
The name NyquistShannon sampling theorem honors Harry Nyquist and
Claude Shannon. But in fact the theorem was first discovered by Vladimir
Kotelnikov in 1933. So it is also known by the names NyquistShannon
Kotelnikov, WhittakerShannonKotelnikov, WhittakerNyquistKotelnikov
Shannon, and cardinal theorem of interpolation.

26

2- Sampling:

The second step to convert an

analog voice
signal to a digital voice signal is
to sample
the Filtered input signal at a
constant
sampling frequency. It is
accomplished by using a process
called pulse
amplitude modulation (PAM). This
step uses
the original analog signal to
modulate
the amplitude of a pulse train that has a constant amplitude and frequency.
The pulse train moves at a constant
frequency, called the sampling
frequency. The analog voice signal
sampled at a million times per
or at two to three times per second.
the sampling frequency
determined? A scientist by the
Harry Nyquist discovered that the
analog signal can be reconstructed
enough samples are taken. He
determined that if the sampling
frequency is at least twice the
frequency of the original input
voice signal, this signal can be
reconstructed by a low-pass filter at
destination. The Nyquist criterion is stated like this:

can be
second
How is
name of
original
if

highest
analog
the

Fs > 2(BW)
Fs = Sampling frequency
BW = Bandwidth of original analog voice signal
Digitize Voice :

After you filter and sample (using PAM) an input analog voice signal, the next
step is to digitize these samples in preparation for transmission over a
Telephony network. The process of digitizing analog voice signals is called PCM.
The only difference between PAM and PCM is that PCM takes the process one
step further. PCM decodes each analog sample using binary code words. PCM
has an analog-to-digital converter on the source side and a digital-to-analog
converter on the destination side. PCM uses a technique called quantization to
encode these samples.

27

3- Quantization and Coding:

Quantization : is the process of

converting each analog sample
into a discrete value that can be
a unique digital code word.

value
assigned

As the input signal samples

enter the
quantization phase, they are
assigned
to a quantization interval. All
quantization intervals are
equally
spaced (uniform quantization) throughout the dynamic range of the input
analog signal. Each quantization interval is assigned a discrete value in the
form of a binary code word. The standard word size used is eight bits. If an
input analog signal is sampled 8000 times per second and each sample is given
a code word that is eight bits long, then the maximum transmission bit rate for
Telephony systems using PCM is 64,000 bits per second. Figure 2 illustrates how
bit rate is derived for a PCM system.
Each input sample is assigned a quantization interval that is closest to its
amplitude height. If an input sample is not assigned a quantization interval that
matches its actual height, then an error is introduced into the PCM process. This
error is called quantization noise. Quantization noise is equivalent to the
random noise that impacts the signal-to-noise ratio (SNR) of a voice signal. SNR
is a measure of signal strength relative to background noise. The ratio is usually
measured in decibels (dB). If the incoming signal strength in microvolts is Vs,
and the noise level, also in microvolts, is Vn, then the signal-to-noise ratio, S/N,
in decibels is given by the formula S/N = 20 log10(Vs/Vn). SNR is measured in
decibels (dB). The higher the SNR, the better the voice quality. Quantization
noise reduces the SNR of a signal. Therefore, an increase in quantization noise
degrades the quality of a voice signal. Figure 3 shows how quantization noise is
generated. For coding purpose, an N bit word yields 2N quantization labels.
4- Encoding:

Encoding is the process of representing each quantized sample by an bit code

word.
The mapping is one-to-one so there is no distortion introduced by encoding.
Some mappings are better than others:
A Gray code gives the best end-to-end performance.

28

The weakness of Gray codes is poor performance when the sign bit (MSB) is
received in error.
At the destination (receiver end) of the communications circuit, a pulse code
demodulator converts the binary numbers back into pulses having the same
quantum levels as those in the modulator. These pulses are further processed
to restore the original analog waveform.

PCM Parameters
PCM audio is coded using a combination of various parameters.
Resolution/Sample Size

This parameter specifies the amount of data used to represent each discrete
amplitude sample. The most common values are 8 bits (1 byte), which gives a
range of 256 amplitude steps, or 16 bits (2 bytes), which gives a range of
65536 amplitude steps. Other sizes, such as 12, 20, and 24 bits, are
occasionally seen. Some king-sized formats even opt for 32 and 64 bits per
sample.
Byte Order

When more than one byte is used to represent a PCM sample, the byte order
(big endian vs. little endian) must be known. Due to the widespread use of
29

little-endian Intel CPUs, little-endian PCM tends to be the most common byte
orientation.
Sign

It is not enough to know that a PCM sample is, for example, 8 bits wide.
Whether the sample is signed or unsigned is needed to understand the range. If
the sample is unsigned, the sample range is 0..255 with a centerpoint of 128. If
the sample is signed, the sample range is -128..127 with a centerpoint of 0. If a
PCM type is signed, the sign encoding is almost always 2's complement. In very
rare cases, signed PCM audio is represented as a series of sign/magnitude
coded numbers.
Channels and Interleaving

If the PCM type is monaural, each sample will belong to that one channel. If
there is more than one channel, the channels will almost always be interleaved:
Left sample, right sample, left, right, etc., in the case of stereo interleaved data.
In some rare cases, usually when optimized for special playback hardware,
chunks of audio destined for different channels will not be interleaved.
Frequency and Sample Rate

This parameter measures how many samples/channel are played each second.
Frequency is measured in samples/second (Hz). Common frequency values
include 8000, 11025, 16000, 22050, 32000, 44100, and 48000 Hz.
Integer or Floating Point

Most PCM formats encode samples using integers. However, some applications
which demand higher precision will store and process PCM samples using
floating point numbers.
Floating-point PCM samples (32- or 64-bit in size) are zero-centred and varies in
the interval [-1.0, 1.0], thus signed values.

PCM Types
Linear PCM

The most common PCM type.

Linear pulse-code modulation (LPCM) is a specific type of PCM where the
quantization levels are linearly uniform.[5] This is in contrast to PCM encodings
where quantization levels vary as a function of amplitude (as with the A-law
algorithm or the -law algorithm). Though PCM is a more general term, it is
often used to describe data encoded as LPCM.
30

A PCM stream has two basic

that determine the stream's
the original analog signal: the
rate, which is the number of
second that samples are taken;
depth, which determines the
possible digital values that can
represent each sample.

properties
fidelity to
sampling
times per
and the bit
number of
be used to

Standard sampling resolutions

and rates

Common sample resolutions for LPCM are 8, 16, 20 or 24 bits per sample.
LPCM encodes a single sound channel. Support for multichannel audio depends
on file format and relies on interweaving or synchronization of LPCM streams.
While two channels (stereo) is the most common format, some can support up
to 8 audio channels (7.1 surround).
Common sampling frequencies are 48 kHz as used with DVD format videos, or
44.1 kHz as used in Compact discs. Sampling frequencies of 96 kHz or 192 kHz
can be used on some newer equipment, with the higher value equating to
6.144 megabit per second for two channels at 16-bit per sample value, but the
benefits have been debated.[26] The bitrate limit for LPCM audio on DVD-Video
is also 6.144 Mbit/s, allowing 8 channels (7.1 surround) 48 kHz 16-bit per
sample = 6,144 kbit/s.
There is a L32 bit PCM, and there are many sound cards that support it
Logarithmic PCM

Rather than representing sample amplitudes on a linear scale as linear PCM

coding does, logarithmic PCM coding plots the amplitudes on a logarithmic
scale. Log PCM is more often used in telephony and communications
applications than in entertainment multimedia applications.
There are two major variants of log PCM: mu-law (u-law) and A-law. Mu-law
coding uses the format number 0x07 in Microsoft multimedia files
(WAV/AVI/ASF) and the fourcc 'ulaw' in Apple Quicktime files. A-law coding uses
the format number 0x06 is Microsoft multimedia files and the fourcc 'alaw' in
Apple Quicktime files.
Every byte of a log PCM data chunk maps to a signed 16-bit linear PCM sample.

31

Differential PCM

Values are encoded as differences between the current and the previous value.
This reduces the number of bits required per audio sample by about 25%
compared to PCM.
Adaptive DPCM

The size of the quantization step is varied to allow further reduction of the
required bandwidth for a given signal-to-noise ratio.

The WAVE File Format

The WAVE File Format supports a variety of bit resolutions, sample rates, and
channels of audio. I would say that this is the most popular format for storing
PCM audio on the PC and has become synonymous with the term "raw digital
audio."
Pulse-code modulation (PCM) is a method used to digitally represent sampled
analog signals. It is the standard form of digital audio in computers, Compact
Discs, digital telephony and other digital audio applications. In a PCM stream,
the amplitude of the analog signal is sampled regularly at uniform intervals,
and each sample is quantized to the nearest value within a range of digital
steps.

32

Applications
PCM is used with T-1 and T-3 carrier systems. These carrier systems
combine the PCM signals from many lines and transmit them over a single
cable or other medium.
PCM is also the usual digital method used for music audio playback of
music CDs. While supported by DVDs, DVDs have a greater volume so
they use Linear PCM, which has a higher sampling rate - up to 24-bit at a
sampling rate of 96 kHz.
Pulse code modulation is used in telecommunication systems, air traffic
control systems etc.
Pulse code modulation is used in compressing the data that is why it is
used in storing data in optical disks like DVD, CDs etc. PCM is even used in
the database management systems.
Pulse code modulation is used in mobile phones, normal telephones etc.
Remote controlled cars, planes, trains use pulse code modulations.

33

References
Websites

http://arabia.ni.com/
http://ohda.matrix.msu.edu/
http://www.ieee.org/
http://www.electronicshub.org/
http://en.wikipedia.org/
http://blog.acronymfinder.com/
http://wiki.multimedia.cx/
http://www.codeguru.com/
http://www.cheers4all.com/
http://www.tech-faq.com/
http://www.music.helsinki.fi/
http://www.ehow.com/
http://www.webopedia.com/
http://www.ieee.org/

34