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Transmission System: Lecture # 1 by Sajjad Karim

Digital transmission has advantages over analog transmission including regenerative repeaters that can detect and retransmit clean signals to prevent noise accumulation. Digital formats also lend themselves to solid state technology. Frequency division multiplexing (FDM) and time division multiplexing (TDM) allow multiple calls to be transmitted simultaneously over one transmission line. In FDM, the frequency band is subdivided into narrower sub-bands, with each sub-band transmitting one call. In TDM, the transmitted signals are separated in time, with one time slot in consecutive periods allocated to each signal. Pulse code modulation converts analog signals to digital by sampling, quantizing, and encoding the signal.

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Mohsin Nisar
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40 views

Transmission System: Lecture # 1 by Sajjad Karim

Digital transmission has advantages over analog transmission including regenerative repeaters that can detect and retransmit clean signals to prevent noise accumulation. Digital formats also lend themselves to solid state technology. Frequency division multiplexing (FDM) and time division multiplexing (TDM) allow multiple calls to be transmitted simultaneously over one transmission line. In FDM, the frequency band is subdivided into narrower sub-bands, with each sub-band transmitting one call. In TDM, the transmitted signals are separated in time, with one time slot in consecutive periods allocated to each signal. Pulse code modulation converts analog signals to digital by sampling, quantizing, and encoding the signal.

Uploaded by

Mohsin Nisar
Copyright
© Attribution Non-Commercial (BY-NC)
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Download as PPT, PDF, TXT or read online on Scribd
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Transmission System

Lecture # 1

By

SAJJAD KARIM
Digital Versus Analog Transmission

Three notable advantages to digital transmission are:

1. -Regenerative repeaters along the transmission path can detect a

digital signal and retransmit a new, clean (noise free) signal


-These repeaters prevent accumulation of noise along the path
-This is not possible with analog communication systems

2. Digital Format lends itself to solid state technology e.g., to


integrated circuits

3.Theoratically compatible with the digital data, telephone signaling


and computers.
 Principle of FDM and TDM
FDM:
-In a telephone network every subscriber is connected to
exchange via its own pair of wire

-At the Beginning there were separate pair of wires for every
Subscriber between the exchanges

-High cost resulted to use these lines in a multiple way


FDM
-Speech transmission requires narrow frequency band
-The frequency band used in telephony is limited to 300 to 3400Hz
-The speech band does not make use of the entire bandwidth of
a transmission line
-To utilize the full bandwidth a wide frequency band is subdivided into
narrower consecutive sub-bands of 4KHz
-Each sub-band transmits one call, so 12 calls can be transmitted at the
same time via the frequency band
-By modulation with the various sinusoidal waves (carriers)
the individual sub-bands are translated into sub-bands and
then transmitted

-After demodulation on receiver side, the individual speech bands


are available in their original form
-This procedure allows the transmission of several speech bands
via one line
 The other way of making the multiple use of lines is TDM.
 The transmitted telephone signals are separated in time
 Fig shows a period containing 32 time slots, This subdivision is
repeated every 125usec in consecutive periods
 One time slot in each of the consecutive periods is allocated to each
telephone signal

1 2 31 32

125usec

Period subdivided into 32 time slots, each approx 3.9usec


TDM
-The principle of time division multiplex is based on the theory that the
complete waveform is not required for transmitting the signals
-Sample the waveform at regular interval and to transmit only these samples

-On the receiver side the original waveform can be restored


from these samples
-Large intervals occur between each sample, these intervals
can be used for transmitting the samples of several different
Signals one after the another in repeated cycles
-The samples are transmitted as pulses with different
amplitudes (analog) .This procedure is known as pulse
amplitude modulation.
Principles of PCM Technology

-Pulse code modulation is a method of modulation in which


A continuous analog wave is transmitted in an equivalent
digital mode

-Functioning of PCM is Nyquist sampling Theorem which states

If a bandlimited signal is sampled at regular intervals of time and at a


rate equal to or higher than twice the highest significant signal
frequency, then the sample contains all the information of the original
signal, the original signal may then be reconstructed by use of a low
pass filter.
Basic components of PCM System

Three steps required to convert an analog telephone


signal into a digital signal
Sampling, Quantization, Encoding
Band limiting

- low pass filter selects the audible frequencies limited to 300 to 3400 Hz

 Sampling
-The electronic switch take the samples from the telephone signals

-The sampling frequency must be higher than twice the highest frequency
in the analog signal
fs ≥ 2fm

-A sampling Frequency of 8000Hz has been specified internationally for


the frequency band 300Hz to 3400Hz

-The interval between the two consecutive samples, sampling interval is


calculated as

Ts=1/fs=125µsec
Sampling
-Every 125µsec(1/8000 )electronic switch is closed for a short period of
time, that generates a sample in each case

-A pulse amplitude modulated signal (PAM) signal is thus obtained at the outlet
of the electronic switch
Quantization
-First stage in the conversion to a digital signal is quantizing the PAM signal.
e.g., assigning a binary sequence to each voltage sample

-The whole range of amplitude values is divided into quantizing intervals in


example 16,generally 256 are used
No of quantization levels=2exp(n)
n is no of bits required to encode one level
-Allocating all amplitudes within a certain quantizing interval to a fixed value
And the values must be discrete.
-Approximation
-Typically, the converter selects the digital value that is closest to the actual sampled
value.

-5.3 it will be rounded off to the value of 5.0 and 4.9 it will also be rounded off to
5.0.

The distortion from the exact signal which may arise is called the quantizing error
The difference between the actual sample value and the quantized sample value.

The level of quantizing noise is dependent on the particular quantization process used
and the characteristics of the quantized signal.

Signal to Quantization noise ratio is given by

SNRq=3L2

S/N improves as a function of quantization levels squared


If L approaches to infinity the signal approaches PAM format with no
quantization noise, and signal to quantization noise ratio is infinite.
Uniform Quantization
 Equally large Quantizing intervals used over the whole
amplitude range
 Quantization noise is same for all the signal magnitudes whether

they are high or low.


 Not preferable for those systems where low magnitudes are

more frequent like speech


Non Uniform Quantization
 Unequal Quantizing intervals are used over the whole amplitude

range
 Small quantization intervals for lower signal values

 Larger quantization intervals for higher signal values.

 CCITT recommends two Xtics for non uniform quantization

 A-law for PCM 30 Transmission System

 µ-law for PCM 24 transmission System.


Encoding
 The PCM signal to be transmitted is obtained by encoding
quantizing intervals

 Electronic Encoder allocates an 8- bit PCM word to each sample

 An 8 digit binary code is used for 128 +ve and 128 –ve quantizing
intervals.
Multiplexing
The Transmission of 8 bit binary code requires only part of sampling interval
-In example the transmission of 8 bit binary code requires only one third of a
sampling interval, in remaining time the binary codes of two further signals can
be transmitted
Multiplexing

-8 bit binary code word of several telephone signals are transmitted


consecutively in repeated cycles
-Between two binary codes of the same telephone signal,the binary codes
of the other telephone signals are transmitted
-This creates a PCM time division multiplex signal
-In Eample pulse frame consists of 4 consecutive PCM words.
Multiplexing / Demultiplexing
-Multiplexing and Demultiplexing process can function only if the
transmitter side and receiver side run synchronously

-Separate channels are provided in PCM transmission system for each


direction of speech In a connection

-subscriber A----subscriber B, subscriber B-------- subscriber A


each pair of identically-numbered channel time slots in the pulse
frame of the two transmission directions form one voice channel

-so transmit section is provided with a timing signal generator and


receive section with a timing signal detector.
Multiplexing/Demultiplexing

-so for synchronization additional time slot is used for information


synchronization in PCM 30 system
Pulse Frame
-A bit Sequence containing one binary code word from every input
signal is known as pulse frame

-In PCM 30 Transmission system the pulse frame consist of 32 PCM


words. and 32x8=256 bits.

-8000 samples per second are transmitted as 8-bit PCM words, this
means within a period of 125usec 30 PCM words are transmitted.

-In addition to 30 PCM word 2x8 bits are also transmitted,8 bit for
signalling and 8 bit for frame alignment signal
PCM Transmission Systems
The transmission systems recommended by CCITT are

 PCM 30 systems (Euorope and many non European countries)


 PCM 24 system (Mainly in USA, Canada and Japan)

They combine 30 and 24 telephone channels per transmission


direction
To form a time division multiplex signal

PCM 30 and PCM 24 are also known as ”Primary Transmission


Systems” or Basic Systems. Their most important features are
PCM Waveform types:
 Binary digits are just abstractions----a way to describe the message
information
 We represent the binary digits with electrical pulses in order to
transmit them through base band channel
 When pulse modulation is applied to a binary symbol the resulting
binary waveform is called a pulse-code modulation waveform
 In telephony applications these waveforms are called line codes
 The transmit section of the multiplex unit converts the PCM signal
into pseudoternary signal e.g. an AMI alternate mark inversion, with
no dc component
 Often use a variant of the pseudoternary AMI code the HDB3 code
(Third order high density bipolar), limits consecutive 0’s
 Pseudoternary signals are better suited for line transmission than
are binary signals.
 The alteration between the positive and negative allows the
regenerative repeaters to recover the timing signal required for their
synchronization.

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