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You are on page 1of 23

Chapter 4

By

Donavon M. Norwood

CS286

Dr. Moh

07/21/2009

Introduction - 4.1

The equipment that is used to convert the information bearing signal into a suitable form

for transmission over the communication channel and then back into a form that is

comprehensible to the end user is called the transmitter and receiver. The transmitter of

a radio system consists of:

Source Encoder – converts the data stream into a form that is more resistant to

degradations in the communication channel.

Modulator – takes a sine wave at a required frequency and modifies the signal

characteristics.

RF Section – generates a signal of sufficient power at a required frequency. It

contains a power amplifier, a local oscillator and a up converter.

Antenna – converts the electrical signal into a wave propagating in free space.

We can look at the encoder and modulator as one subsystem that maps data presented to

it by the user interface onto the RF carrier for processing, amplification, and transmission

by the RF section. The demodulator and decoder does the inverse by taking the received

RF signal and inverse mapping the signal back to the data stream for transmission. 2

Figure 4.1 - Overview of a communication system

3

Figure 4.2 - Structure of the transmitter for a radio system

4

Baseband Systems - 4.2

Formatting involves sampling, equalization, and encoding which makes the

message compatible with digital processing. Transmit formatting transforms

source information into digital symbols. When data compression is used in

addition to formatting, the process is called source coding.

5

Messages, Characters and Symbols - 4.3

In the process of digital transmission the characters are encoded first into a

sequence of bits which is called a bit stream or baseband signal. Groups of

b

b bits form a finite symbol set or word M =2 of such symbols. This is also

known as a M-ary system. The value of b and M is important for the initial design

of any digital communication system. When b = 1, the system is called a binary

system, the size of the symbol set M is 2, and the modulator uses two different

waveforms to represent the binary '1' and the binary '0' (see Figure 4.4 Binary

and quatenary systems). The symbol rate and the bit rate in this case are the

same. When b = 2, the system is called quantentary or 4-ary (M=4) system. At

each symbol time, the modulator uses one of the four different waveforms that

represent the symbol.

Figure 4.4 Binary and quatenary systems

6

Sampling Process - 4.4

analog information into a digital format which is called a discrete pulse-

amplitude-modulated waveform. The sampling process is usually

described in time domain, which is the operation that is basic to digital signal

processing and digital communication. The sampling process can be

implemented in several ways with the most popular being the sample-and-

hold operation. In this operation a switch and storage mechanism form a

sequence of samples of the continuous input waveform, and the output of

this is called pulse amplitude modulation (PAM), because the the

successive output intervals are described as a successive sequence of

pulses with amplitudes derived from the input waveform samples. The wave

of a PAM waveform can be retrieved from a low pass filter if the sampling

rate is chosen properly. The ideal form of sampling is called instantaneous

sampling.

7

Figure 4.5 Sampling Process

8

Aliasing - 4.4.1

the signal seemingly taking the identity of a lower frequency in the spectrum of its

sampled version. In figure 4.6, it shows the part of the spectrum that is aliased due to

under-sampling. The aliased spectral components represent ambigous data that can

be retrieved only under special conditions. In general the ambiguity is not resolved

and ambigous data appears in the frequency band between f s− f m and f m ,

where f s is the maximum frequency and f m is the sampling rate.

9

Aliasing – 4.4.1 (Continued)

'

f

In figure 4.7 we use the higher sampling rate s to eliminate the aliasing by

separating the spectral replicas.

10

Aliasing – 4.4.1 (Continued)

Figure 4.8 and 4.9 show two ways to eliminate aliasing using anti-aliasing filters.

The analog signal is prefiltered so that the new maximum frequency f mis less

than or equal to f s / 2. Thus there are no aliasing components seen in figure 4.8

'

since f s2 f m .

11

Aliasing – 4.4.1 (Continued)

12

Quantization - 4.4.2

predetermined levels; each level can be represented by a symbol from a finite

alphabet. The pulses in figure 4.10 are called quantized samples. When

sample values are quantized to a finite set, this format can interface with a

digital system. After quantization the analog waveform can still be recovered,

but not precisely; improved reconstruction fidelity of the analog waveform can be

achieved by increasing the number of quantization levels.

13

Uniform Quantization – 4.4.4

input having zero mean, and the quantizer being symmetric, the quantizer ouput

and quantization error will have a zero mean.

14

Voice Communication - 4.5

Voice communication has very low speech volumes that predominate: 50% of

the time, the voltage characterizing detected speech energy is less than ¼ of

the rms value of the voltage. Large amplitude are rare, in which only 15% of the

time does the voltage exceed the rms value. Uniform quantization would be

wasteful for speech signals. In a system that uses equally spaced quantization

levels, the quantization noise is the same for all signal magnitudes because the

noise depends on the step size of quantization. Nonuniform quantization can

provide better quantization of the weak signals versus uniform quantization, and

also coarse quantization of the strong signal. The nonuniform quantization is

used to make SNR a constant for all signals within the input range Nonuniform

quantization is achieved by first disorting the original signal with a logarithmic

compression, and then using a uniform quantizer. A device called a expander

at the receiver to for compression. The whole process of compression is called

companding. There are two compression algorithms used today: −law

and A-law.

15

−law Compression characteristic used in North America

[1∣input∣]

∣output∣=log log {1}

16

Pulse Amplitude Modulation (PAM) - 4.6

analog signal with a series of discrete analog pulses in which amplitude of the

information signal at a given time is encoded as a binary number. Pulse

Amplitude Modulation (PAM) is now rarely used and has been replaced by

Pulse Code Modulation (PCM). Two operations involved in the Pulse

Amplitude Modulation (PAM) signal are:

s s s

according to the sampling theorem.

These operations are referred together as sample and hold. The reason for

increasing the duration of each sample is to avoid the use of excessive

bandwidth, since bandwith is proportional to pulse duration.

17

Figure 4.12 Rectangular pulse and its spectrum

18

Pulse Amplitude Modulation (PAM) – 4.6 (Continued)

that the sample has the same amplitude for its whole duration introduces

amplitude distortion as well as delay. The disortion caused by PAM to

transmit a signal called the aperture effect. The disortion can be corrected by

using a equalizer.

19

Pulse Code Modulation (PCM) - 4.7

Pulse Code Modulation (PCM) is a digital scheme for transmitting analog data.

It converts an analog signal into digital form. Using PCM it is possible to digitize

all forms of analog data, including full motion video, voice, music, telemetry, etc.

To obtain a PCM signal from an analog signal at the source (transmitter) of a

communication circuit, the analog signal is sampled at regular time intervals.

The sample rate is several times the maximum frequency of the analog signal.

The amplitude of the analog signal is rounded off to the nearest of several

specific, predetermined levels (quantization). The number of levels is always a

power of 2. The output of PCM is a series of binary numbers, each represented

by some power of 2 bits. At the destination of the communication circuit, the

PCM converts the binary numbers back into pulses having the same quantum

levels as those in the modulator. These pulses are then further processed to

restore the original analog waveform. When PCM is applied to a binary symbol,

the resulting binary wave form what is known as pulse code modulation

waveform. When PCM is applied to a non binary number, the resulting wave is

called M-ary pulse modulation waveform.

20

Modulation - 4.9

Baseband signals are generated at low rates, therefore these signals are

modulated onto a radio frequency carrier for transmission. Baseband signal

s(t) is complex and can be represented mathematically as st =a t e j t ,

where a(t) is the sampling rate and t is the phase. The modulation can be

classified as linear modulation or nonlinear modulation. A modulation

process is linear when a t cos t and a t sin t are linearly related to the

message information signal. Examples of linear modulation are amplitude

modulation in which the modulating signal signal affects only the amplitude

of the modulated signal, and phase modulation is where the modulated

signal affects only the phase of the modulated signal.

21

Figure 4.16 Functional block diagram of a generic modulator

22

Thank You

References:

23

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