SAMPLING

Sampling is the fundamental result in the field of information theory, in particular

telecommunications and signal processing. Sampling is the process of converting a signal (for
example, a function of continuous time or space) into a numeric sequence (a function of discrete
time or space). Shannon's version of the theorem states:[1]

A signal consists of different frequency components and as such a frequency spectrum forms.Let us
consider that the highest frequency among the frequency components is B Hz.Now this can be said
that actually the bandwidth of the signal is B Hz. And not the frequency.

Now the signal is multiplied by an impulse at a regular interval Ts.Ts=1/fs,fs=2B Hz.So the signal gets
discrete in nature.

Samples of any message signal are actually impulses as because the signal is multiplied by impulses
to get the samples.This is also a way of discretalizing an analog signal.

Thus if g(t) is the message signal then say the sample at 0 position is G(t)=g(t)xd(t),d(t)=impulse
function which is equal to 1 at t=0 and is equal to 0 at other places.Thus it can be said that,

G(t)=summation[g(nT)xd(t-nT)]

The signal(band-limited) can be reconstructed back using the samples taken at the rate of 1/2B s.

Reconstruction of signal is done with the help of a LPF with BW 2B Hz and h(t) is a sinc function.The
result is also a sinc function where each and every impulses’ amplitude.In other words sinc pulses
are generated for every impulse samples with the amplitude of the sinc pulses being equal to the
corresponding impulses.

Again if the sampling rate is less than the Nyquist rate then proper recovery of the signal is not
possible and in that case aliasing will occur(overlapping of the samples).But if the rate is more or
equal to the Nyquist rate then aliasing will not occur.

Again in this context it is to be noted that a signal is either band limited or time limited but never
together.Yet in general it is time limited & so also band non-limited or has infinite BW.

PCM
Pulse code modulation is a process by which an analog signal is made discrete(A/D conversion
step).In this process the analog signal is made discrete in time from continuous time.

Here the signal is divided by L segments and width of each segment is 2mp/L=V where mp is the
amplitude of the signal.

Now the signal is dicretalized by pointing the mid-point of every segment.

Quantization
Quantization is the process by which an analog signal is made discrete in terms of amplitude.It is a
part of the process PCM.

Now in case of the errors there can be two types of errors:-1)Quantization error,2)Pulse detection
error.

Now Quantization noise=Nq=mean square value of q(t)=mp^2/3(L^2). Where q(t) = distortion in the
reconstructed signal=m(t)’-m(t),m(t)’ is the quantized sample and m(t) is the signal sample(discrete
sample of the signal).

Again power of the message signal=S=mean square value of m(t).

Thus SNR=MSV[m(t)]/mp^/3L^2=3L^2(MSV[m(t)])/L^2.

MSV=Mean Square Value.

Now the value of mp is constant(amplitude) and also 3L^2 and so the SNR depends on the value of
the signal power only.Again as the signal power of the message may vary so the SNR will also vary.
[i.e say in case if the signal is voice signal then people may speak with different power and mostly
softly so SNR willbe poor most of the time).As such non-uniform quantization should occur.

Nq=mp^2/3L^2=(VxL)/2/3L^2=v^2/12.

Thus the noise is proportional to the width of the segments.

As such the following should occur:-

 If the signal power is small then the step size should be reduced/decreased then Nq will
decrease & SNR will increase.

Now for this a compressor is used with two different types of parameters-1)A,2)miu.

Again expander is used at the receiver.So they together are called compander.

According to miu law compander,

SNR=S/N=3L^2/[ln(1+miu)]^2 and miu>>mp^2/MSV[m(t)].