COMMUNICATION SYSTEM EECB353Chapter 5 Part 1

DIGITAL TRANSMISSION

Intan Shafinaz MustafaDept of Electrical Engineering

Universiti Tenaga Nasionalhttp://metalab.uniten.edu.my/~shafinaz

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Introduction

Defined as the transmittal of digital signals between two or more points in a communications system.

The signal can be binary or any other form of discrete-level digital pulses.

Original source information may be in digital or analog (i.e have been converted to digital pulses prior to transmission and converted back to analog signal in Rx).

Medium – pair of wires, coaxial cable, optical fiber cable. For analog data, (eg. speech) a digitization process is

needed.

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The world we sense is full of analog signals: Electrical sensors convert the medium they sense

into electrical signals– E.g. transducers, thermocouples, microphones.– (usually) continuous signals

Analog: continuous signals must be converted or digitised for computer processing.

Digital: discrete digital signals that computer can readily deal with.

Analog and Digital Signal Conversion

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Analog and Digital Signal Conversion (ADC)

Take analog signals from analog sensor (e.g. microphone) and digitally sample data

Special hardware devices : Analog-to-Digital converters

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Digital to Analog Converter (DAC)

Takes digital signal, possible after modification by computer(e.g. volume change, equalisation)

Outputs an analog signal that may be played by analog outputdevice (e.g. loudspeaker)

Playback – a converse operation to Analog-to-Digital

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Advantages of Digital Transmission Easier to store as a pattern of I’s and 0’s. Easier to process in computer and digital signal

processor Noise immunity - digital data can be recovered without any

error as long as the distortion and noise are within limits. On the other hand, for an analog message, even a slight distortion or interference in the waveform will cause an error in the received signal.

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Advantages of Digital Transmission Regeneration – data integrity, use repeaters instead of amplifiers to

completely restore signal fidelity.Based on this “noise immunity”, when transporting a bit stream over a long distance, regenerative repeaters or repeater stations are placed along the path of a digital system at distances short enough to ensure that noise and distortion remain within a limit. The property of regenerative repeaters is the main reason for the superiority of digital systems over analog ones.

Cost - Fast development of digital technology prompted a significant drop in cost. Digital circuits are relatively inexpensive and immune to drift.

Multiplexing – capacity utilization i.e several digital signals can easily be multiplexed and transmitted on one channel.

Security and privacy – encryption. Simpler to measure and evaluate - error detection and recovery more

easily and accurate.

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Disadvantages of Digital Transmission Requires more bandwidth – digital Tx system require up to

16 times the bandwidth of an analog system. Cost incurred for analog to digital to analog (A/D/A)

conversion – need additional encoding and decoding circuitry. Require precise synchronization between the clocks in the

Tx and Rx. Compatibility - Incompatible with older version of analog Tx

systems.

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Pulse Modulation

It consists essentially of sampling analog info signals and then converting those samples into discrete pulses and transporting the pulses from a source to a destination over a physical transmission medium.

Pulse Modulation is a process of using some characteristic of a pulse (amplitude, width, position) to carry an analog signal.

Difference between pulse modulation and normal AM and FM is that:

In AM or FM, some parameter of the modulated wave varies continuously with the message

Whereas in pulse modulation some parameter of a sample pulse is varied by each sample value of the message.

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Pulse Modulation

The pulses are usually of very short duration so that pulse modulated wave is “off” most of the time.

Main reason of using pulse modulation because it allows Tx to operate on a very low duty cycle (“off” more

than “on”), as is desirable for certain microwave devices and lasers

The time intervals between pulses to be filled with samples of other messages.

The latter reason conveniently allows several different message to be transmitted on the same channel (TDM) – analogous to computer time sharing, where several users utilize a computer simultaneously.

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Pulse Modulation 4 predominant methods:

Pulse Width Modulation (PWM) - width (active portion of duty cycle) of a constant amplitude pulse is varied proportional to the amplitude of the analog signal at the time the signal is sampled.

Pulse Position Modulation (PPM) - position of a constant-width pulse within a prescribed time slot is varied according to the amplitude of the sample of the analog signal

Pulse Amplitude Modulation (PAM) - amplitude of a constant width, constant-position pulse is varied according to the amplitude of the sample of the analog signal

Pulse Code Modulation (PCM) – the analog signal is sampled and then converted to a serial n-bit binary code for transmission. Each code has the same number of bits and requires the same length of time transmission.

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Modulating signal

PAM

PWM

PPM

PCM

Sample pulse

Notice that the pulse amplitude in PAM and pulse width in PWM are not zero when the signal is minimum.

This done to allow a constant pulse rate and is important in maintaining synchronization in TDM systems.

With PCM, the pulses are fixed of length and fixed amplitude

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Pulse Amplitude Modulation (PAM) The first step in analog-to-digital conversion. This technique takes an analog signal, samples it, and

generates a series of pulses based on the results of the sampling.

PAM is used as an intermediate form of modulation with PSK, QAM and PCM. It is seldom used by itself.

Term sampling means measuring the amplitude of the signal at equal interval.

PAM is not useful to data communication, although it translates the original waveform to a series of pulses, these pulses are still of any amplitude i.e still an analog signal, not digital.

To make them digital, PCM is used to modify the pulses created by PAM.

Analog

signalPAM

signal

The amplitude (height) of the pulses is varied in step with the signal (eg. music, voice, data etc) to be transmitted

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Pulse Code Modulation (PCM)

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).

Why PCM? - The stream of pulses and non-pulse streams of 1’s and 0’s are not easily affected by interference and noise.

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

Filtering - This 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.

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Pulse Code Modulation (PCM) Sampling of a waveform means determining instantaneous amplitudes of a signal at

fixed intervals. The term sampling means measuring the amplitude of the signal at equal intervals.

a) The determination of instantaneous amplitude at uniform intervals.

b) The samples corresponding to instantaneous amplitudes of the input signal.

c) The output which represents the reconstructed input signal.

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Pulse Code Modulation (PCM)

Quantization is the process of allocating levels to the infinite range of amplitudes of sample values of the analog signal.

The maximum amplitude value is +8 and the minimum is –8.

The amplitude values are quantized into four levels.

The full range (from –8 to +8) of values is 16.

Therefore the width of each level is 16 divided by 4 which give 4 volts.

Quantization is done such that half the steps are at the top and half at the bottom.

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Pulse Code Modulation (PCM)

Encoding - This will be the last part of the conversion before the PCM signal will be transmitted/sent.

In this process each step level is assigned a number.

The numbers start with zero at the lowest level.

These assigned numbers are then expressed in binary form (in terms of 0’s and 1’s).

One disadvantage of PCM is that the signal accuracy is reduced because of the quantizing of the samples.

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Block Diagram of PCM System

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Simplified PCM Block Diagram

Figure: A digital communication system

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Block Diagram of PCM System Bandpass filter – limits the freq of analog input signal to the standard voice-

band freq range of 300Hz to 3kHz. Sample-and-hold – periodically samples the analog input signal and converts

those samples to a multilevel PAM signal. ADC – converts the PAM samples to parallel PCM codes. Parallel-to-serial converter – convert to serial binary data. Repeater – to regenerate the digital pulse. Here the effect of noise on a PCM

signal can be removed entirely. Transmission media carrying PCM signals employ regenerative repeaters that are spaced sufficiently close to each other to prevent any uncertainty in the recognition of the binary PCM pulses.

Serial-to-parallel converter – converts to parallel PCM codes. DAC – converts the parallel PCM codes to multilevel PAM signals. Hold circuit – i.e a LPF that converts PAM signals to analog form. An integrated circuit that performs the PCM encoding and decoding functions is

called codec (coder/decoder)

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PCM Sampling Function of sampling cct is to periodically sample the continually

changing analog input voltage and convert those samples to a series of constant amplitude pulses (PAM) that can more easily be converted to binary PCM code.

There are two basic techniques to perform sampling function:a) Natural Sampling – the tops of the sampled waveforms retain

their natural shapeb) Flat top Sampling – the signal voltage is held constant during

samples, creating a staircase that tracks the changing input waveform

Natural sampling Flat-top sampling

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PCM Samplinga) Natural sampling – tops of the samples pulses retain their natural

shape during the sample interval, thus difficult for an ADC to convert the sample to a PCM code.

b) Flat-top sampling – carried out in sample-and-hold cct. Sample-and-hold cct periodically sample the continually changing

analog input voltage and convert those samples to a series of constant-amplitude PAM voltage levels.

With flat-top sampling, the input voltage is sampled with a narrow pulse and then relatively held const until next sample is taken.

Natural sampling

Flat-top sampling

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Nyquist’s Sampling Theorem

Sampling Frequency is very Important in order to accurately reproduce a digital version of an Analog Waveform

Nyquist’s Theorem:The Sampling frequency for a signal must be at least twice the highest frequency component in the signal.

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Nyquist’s Sampling Theorem

Suppose we are sampling a sine wave. How often do we need

to sample it to figure out its frequency?

A Sine Wave

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Nyquist’s Sampling Theorem

If we sample at 1 time per cycle, we can think it's a

constant

Sampling at 1 time per cycle

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Nyquist’s Sampling Theorem

If we sample at 1.5 times per cycle, we can think it's a

lower frequency sine wave

Sampling at 1.5 times per cycle

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Nyquist’s Sampling TheoremNow if we sample at twice the sample frequency, i.e Nyquist Rate, we start to make some progress. An alternative way of viewing the waveform (re)genereation is to think of straight lines joining up the peaks of the samples. In this case (at these sample points) we see we get a sawtooth wave that begins to start crudely approximating a sine wave

Sampling at 2 times per cycle

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Nyquist’s Sampling TheoremNyquist rate -- For lossless digitization, the sampling rate should be at least twice the maximum frequency responses. Indeed

many times more the better.

Sampling at many times per cycle

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PCM – The Sample Frequency

One of the most critical specification in a PCM system is the selection of sample frequency – govern by the Nyquist Rate.

Nyquist sampling theorem states that for a sample to be reproduced accurately, minimum sampling rate,fs must be twice the higher input frequency, fa,max.

Mathematically, the minimum Nyquist sampling rate, fs is

fs ≥ 2 fa,max

If fs is less than twice fa i.e fs ≤ 2 fa, aliasing or foldover distortion occurs.

This can be overcome by using anti-aliasing filter before sampling to suppress the component before sampling.

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PCM Sampling

A sample-and-hold circuit is a non-linear device (mixer) with 2 inputs which are the sampling pulse and the analog input signal.

The output of sample-and-hold cct are:

2 original inputs (fs and fa)

Sum and difference frequencies (fs fa)

All harmonics of fs and fa (2 fs, 3 fs, 2 fa, 3 fa , …)

Their associated cross products (2fs fa,3 fs fa, …)

Each sum and difference freq generated is separated from its respective centre freq by fa.

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PCM Sampling – Aliasing / Antialiasing As long as fs ≥ 2 fa, none of the side freq from one harmonic will

spill into the SBs of another harmonic, i.e no aliasing occurs.

When fs ≤ 2 fa, the side freq from one harmonic fold over into the SB of another harmonic. The freq that folds over is an alias of the input signal.

An aliasing of foldover distortion cannot be removed through filtering or any other techniques.

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Example 1 - A CD audio laser-disk system has a frequency bandwidth of 20 Hz to 20 Khz. What is the minimum sample rate required to satisfy the Nyquist sampling rate?

Example 2 - For a PCM system with a maximum audio input frequency of 4kHz, determine the minimum sample rate and the alias frequency produced if a 5kHz audio signal were allowed to enter the sample-and-hold circuit.