Digital Communication

Introduction By- D. Sarma Asst. Prof., Dept. Of ECE, DBCET

Why Digital Communication

Easy to regenerate. Less subject to distortion- Since in two operated in either on or off . Digital circuits are more reliable and of lower cost. Provides encryption and privacy. Finally signal processing.

Non Graceful Degradation

Basic Flow of Digital Communication

Some milestones in the history of electrical communications Version

Milestones in the history of electrical communications contd.

Electromagnetic bands with typical applications

Electromagnetic bands contd..

A few important frequency bands

Some Carrier frequency values and nominal bandwidth that may be available at the carrier frequency

Typical power losses during transmission through a few media

The PCM system

Introduction
Two basic operations in the conversion of analog signal into the digital. Time discretization Amplitude discretization. In the context of PCM, The first is accomplished with the sampling operation The second by means of quantization. PCM involves another step, Conversion of quantized amplitudes into a sequence of simpler pulse patterns (usually binary) - code words.

Block Diagram

 m(t ) is the information bearing message signal that is to be transmitted digitally. m(t ) is first sampled and then quantized. The quantizer converts each sample to one of the values that is closest to it from among a pre-selected set of discrete amplitudes.

Sampling
Analog signal is sampled every TS secs. Ts is referred to as the sampling interval. fs = 1/Ts is called the sampling rate or sampling frequency.

According to the Nyquist theorem, the sampling rate must be at least 2 times the highest frequency contained in the signal.

4.15

 Let a signal x (t ) be band limited to W Hz; that is, X (f ) = 0 for f > W . Let X(nTs ) = X(t ) t=nTs , - < n < represent the samples of x (t ) at uniform intervals of Ts seconds. If Ts 1/2W , than it is is possible to reconstruct x (t ) exactly from the set of samples, {X (n Ts )} .

Types of sampling
There are 3 sampling methods:
Ideal - an impulse at each sampling instant Natural - a pulse of short width with varying amplitude Flattop - sample and hold, like natural but with single amplitude value

The process is referred to as pulse amplitude modulation PAM and the outcome is a signal with analog (non integer) values

Ideal impulse sampling

Consider an arbitrary low pass signal x (t ) shown in Fig 3 . Let

 where (t ) is the unit impulse function of Xs (t) , shown in red in Fig 3. consists of a sequence of impulses; the weight of the impulse at t = nTs is equal to x (nTs ) . X s (t ) is zero between two adjacent impulses.

Fig 3:

 It is very easy to show in the frequency domain that Xs (t ) preserves the complete information of x (t ). X s (t ) is the product of x (t ) and (t n Ts) hence the corresponding Fourier relation is convolution.

That is,

 we see that Xs (f ) is a superposition of X (f ) and its shifted versions (shifted by multiples of fs , the sampling frequency) scaled by 1/ Ts Let X (f ) be a triangular spectrum as shown in Fig. 4(a).

Fig 4:

 it is obvious that we can recover x (t ) from Xs(t ) by passing Xs (t ) through an ideal low pass filter with gain Ts and bandwidth W as shown bellow.

 Of course, with respect to Fig. 4(b), which represents the over-sampled case, reconstruction filter can have some transition band which can fit into the gap between f = W and f = (fs W) . when fs < 2W , (under-sampled case) we see that spectral lobes overlap resulting in signal distortion, called aliasing distortion. In this case, exact signal recovery is not possible

Sampling with a rectangular pulse train

Let yp (t ) represent the periodic rectangular pulse train as shown in Fig.5(b). Let Xs (t ) =[x (t )] yp (t ) Then, Xs (f ) = X (f ) Yp (f ) Therefore

Figure :

Three different sampling methods for PCM

4.30

Quantization
Sampling results in a series of pulses of varying amplitude values ranging between two limits: a min and a max. The amplitude values are infinite between the two limits. We need to map the infinite amplitude values onto a finite set of known values. This is achieved by dividing the distance between min and max into L zones, each of height = (max - min)/L
4.31

Quantization Levels
The midpoint of each zone is assigned a value from 0 to L-1 (resulting in L values) Each sample falling in a zone is then approximated to the value of the midpoint.

4.32

Quantization Error
When a signal is quantized, we introduce an error - the coded signal is an approximation of the actual amplitude value. The difference between actual and coded value (midpoint) is referred to as the quantization error. The more zones, the smaller which results in smaller errors. BUT, the more zones the more bits required to encode the samples -> higher bit rate

4.33

Quantization Error and SNQR

Signals with lower amplitude values will suffer more from quantization error as the error range: /2, is fixed for all signal levels. Non linear quantization is used to alleviate this problem. Goal is to keep SNQR fixed for all sample values. Two approaches:
The quantization levels follow a logarithmic curve. Smaller s at lower amplitudes and larger s at higher amplitudes. Companding: The sample values are compressed at the sender into logarithmic zones, and then expanded at the receiver. The zones are fixed in height.
4.34

Bit rate and bandwidth requirements of PCM

The bit rate of a PCM signal can be calculated form the number of bits per sample x the sampling rate Bit rate = nb x fs The bandwidth required to transmit this signal depends on the type of line encoding used. Refer to previous section for discussion and formulas. A digitized signal will always need more bandwidth than the original analog signal. Price we pay for robustness and other features of digital transmission.

4.35

We want to digitize the human voice. What is the bit rate, assuming 8 bits per sample?

Solution The human voice normally contains frequencies from 0 to 4000 Hz. So the sampling rate and bit rate are calculated as follows:

4.36

PCM Decoder
To recover an analog signal from a digitized signal we follow the following steps:
We use a hold circuit that holds the amplitude value of a pulse till the next pulse arrives. We pass this signal through a low pass filter with a cutoff frequency that is equal to the highest frequency in the pre-sampled signal.

The higher the value of L, the less distorted a signal is recovered.

4.37

We have a low-pass analog signal of 4 kHz. If we send the analog signal, we need a channel with a minimum bandwidth of 4 kHz. If we digitize the signal and send 8 bits per sample, we need a channel with a minimum bandwidth of 8 4 kHz = 32 kHz.

4.38

A brief aside about ADCs

ADCs are used to convert an analogue input voltage into a number that can be interpreted as a physical parameter by a computer.
0111 0110 0100 0101 0011 0010 0001 0000 1111 1110 1100 1010 1101 1011 1001 0000 0110 0111 0011 1100 1001 1011

Resolution= 1 part in 2n

39 Numbers passed from ADC to computer to represent analogue voltage

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