Beruflich Dokumente
Kultur Dokumente
Faculty of Engineering
Electrical Department
Forth year
Signal Processing
PCM
Worked by :
Majid Mohammed AlZariey
2390/2012
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
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35
37
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Personal Call
Manager (telephones)
Pulse Code
Modulation
Definition
In Brief, What Is PCM?
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.
sampling a
them over a
conveyed
D.
characters
images
kind of
Electric
transmitted
optomachine did
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
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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
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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.
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
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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.
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
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Angle Modulation:
In the angle modulation, again there are two different types of modulations.
Frequency modulation
Phase modulation.
1. Frequency Modulation:
Advantages of Frequency
Modulation:
the carrier
of the
signal by
the amplitude
huge then the
Modulation:
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
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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.
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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):
pulse
based on
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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):
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):
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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.
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Filtering
Sampling
Quantizing
Encoding
1- Filtering :
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25
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2- Sampling:
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.
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value
assigned
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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
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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
properties
fidelity to
sampling
times per
and the bit
number of
be used to
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
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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.
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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.
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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/
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