Sampling Theorem
We will show that a band limited signal
( ) | |
Recall: That a time sampled signal is like taking a snap shot or picture of signal periodically.
( ) ( ) ( ) ( ) ( )
( )
Where
( )
( )
so
( ) ( ) ( ) ( )
{ ( )} ( ) ( )
{ ( )} ( )
Using the phase shift property of the Fourier transform we find that
{ ( )} ( ) ( )
Where
What we also need to take note of is that if is less than the signal spectrum will
have the copies overlapping or aliasing each other. In order to have no overlap we must
have or which is referred to as the Nyquist rate.
Answer: The bandwidth of this signal is 5 kHz and therefore the sampling rate 10 kHz.
Signal Reconstruction
We will now show that a signal ( ) can in fact be reconstructed in its entirety simply by a well-placed
low-pass filter.
For a band limited signal with bandwidth we find the transfer function of the filter for reconstruction
to be
( ) ( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( ) ( ( ))
( ) ( ) ( ( ))
( ) ( ( ))
What we showed above is that the reconstruction of the signal ( ) from ( ) is an interpolation
formula which yields ( ) as a weight sum of all the samples.
For practicality ( ) should be easy to generate. For ( ) let analyze the accuracy of the reconstructed
signal. Lets say that
( ) ( ) ( )
( ) () ( ) ( )
( ) ( )( ) ( ) ( )
This is an interesting result because what it says is that rather than having to walk through a barrage of
mathematically equations to determine how to reconstruct ( ), we can simply find a transfer function
that will essentially undue the results of sampling a signal with a non-ideal sampling function in this
case ( ). We will refer to this transfer function as an equalizer. The end result of apply an equalizer is
that afterwards the signal simply becomes ( ).
Recall: We showed earlier that ( ) can be extracted by a simple, well placed, low-pass
filter.
We have
( ) ( )
( ) ( ) ( )
Quantization
L =Levels
Figure 6: Quantization
What it means to quantize a signal is to limit the possible values that a signal can take on. For instants,
a discrete signal can have any range of amplitudes at a discrete time interval. Well in order to
represent such a signal digitally, lets say, we would have to have an infinite amount of bits to do this.
Since no computer exist that can handle an infinite bit data sample we quantize the data to fit within a
range that we are comfortable with and fits our requirements for the system.
If we decided to represent this signal digitally we could say we need or 3 bits and
now any incoming signal will take on values between and in steps. Another way
to say it is that we can only detect a change in the incoming signals voltage. For an 8
bit binary number we have
Decimal 0 1 2 3 4 5 6 7
( ) ( ) ( )
( ) ( ) ( )
And for the quantized signal let ( ) be the sample of the signal ( ) then
( ) ( ) ( )
( ) [ ( ) ( )] ( )
( ) ( )
The signal ( ) is an undesired and acts as noise which leads to the term quantization noise.
To calculate the power of the quantization noise we do the following
( ) ( )
[ ( ) ( )]
Where
( ) denotes the mean square power. Since the a since function
( ) and ( ) are orthogonal ever where but at we find that
( ) ( ) ( )
( ) ( )
Assuming that the error is equally likely be anywhere between ( ) we will find that the
quantization noise is
( )
Answer: When you see average power think mean square power or ( ) unless
otherwise noted. With this we have
()
( )
( )
Where a the decibel value is calculated as
()
( )
Question: How many bits would be required to represent one sample from this audio
signal?
Answer:
( ) ( )
( )
( ) ( )
Well there is no such thing as 15.61 bits so we must round up to 16 bits in order to
represent the largest possible quantization of the signal.
Although the method of creating and representing a quantized signal via a manipulated pulse train may
vary the data being transmitted does not. See the image below.